| 00:04.04 | *** join/#asterisk snadge (n=snadge@starbug.ugh.net.au) |
| 00:04.12 | snadge | "all circuits are busy now, please try again later" |
| 00:04.37 | snadge | whenever i try to dial out.. the solution so far has been to just keep pressing hangup then redial.. until it eventually works |
| 00:07.15 | doolittlework | Kobaz: i get 2 erroes can you tell me if they are mission critical |
| 00:07.21 | doolittlework | make[2]: *** [/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/wctdm.o] Error 1 |
| 00:07.38 | doolittlework | make[1]: *** [_module_/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi] Error 2 |
| 00:08.49 | snadge | im not a complete nob.. i've logged into the asterisk console.. enabled sip show debug.. placed a call.. and have a crapload of debugging message, but dont know what im looking for |
| 00:10.19 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:10.22 | doolittlework | snadge: are u using free pbx? |
| 00:11.03 | snadge | ok maybe i am a complete nob then.. i am using freepbx |
| 00:11.14 | doolittlework | lol |
| 00:11.24 | doolittlework | are u dualing of sip trunks? |
| 00:11.49 | snadge | theres only one sip drunk for outgoing |
| 00:12.01 | snadge | its passed through to our isps sip server |
| 00:12.18 | doolittlework | if your trunk fails for some reason you will get that message |
| 00:12.22 | snadge | it can place and receive calls normally.. just i keep getting busy out messages for some reason |
| 00:12.43 | snadge | normally if i hang up and try a few times.. it will go through (which is annoying) |
| 00:13.26 | snadge | sigh.. i'll just use skype for now i need to make a call lol |
| 00:13.33 | doolittlework | past your sip_addiional.conf xxx out passwords |
| 00:13.47 | *** join/#asterisk rbd_ (n=rbd@rrcs-98-101-33-14.midsouth.biz.rr.com) |
| 00:14.18 | rbd_ | anyone tried asterisk on Amazon's EC2? do you use ztdummy on it? how's the voice quality? does meetme work? |
| 00:14.46 | *** part/#asterisk johnyjj2 (n=mainacco@p9g117.traco.pl) |
| 00:20.08 | bcrisp | yikes call transfering issues |
| 00:20.45 | bcrisp | Hear the "transfer" announcement but when attempting to enter an extension it warns in cli: "Did not read data" |
| 00:20.58 | *** join/#asterisk jtodd (i=chk13q9q@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 00:20.58 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 00:23.21 | doolittlework | what channels are u using bcrisp? |
| 00:23.30 | bcrisp | SIP |
| 00:24.24 | bcrisp | internal SIP -> outgoing sip provider -> mobile |
| 00:25.14 | etfonhomey1 | What dtmfmode are you using? |
| 00:25.20 | bcrisp | default |
| 00:25.30 | etfonhomey1 | Which is? |
| 00:25.35 | bcrisp | looking |
| 00:25.56 | bcrisp | ok dtmf mode for outbound SIP is rfc2833 |
| 00:27.03 | etfonhomey1 | And is that the dtmfmode your SIP provider specifies (if they do)? |
| 00:27.24 | bcrisp | yes |
| 00:27.38 | etfonhomey1 | Who is your provider? (just curios) |
| 00:27.47 | bcrisp | callcentric |
| 00:27.55 | bcrisp | thats the outbound provider |
| 00:28.36 | bcrisp | it reads the # for transfer ok (dial plan with tk options), just doesn't appear to read the ext digits |
| 00:29.11 | etfonhomey1 | You should be good with RFC2833: http://www.callcentric.com/faq/8#174 |
| 00:29.18 | bcrisp | i.e. Dial(SIP/number@callcentric,10,tk) |
| 00:30.01 | etfonhomey1 | bcrisp, are you having any packet loss issues? |
| 00:30.08 | etfonhomey1 | or jitter? |
| 00:30.24 | bcrisp | not that i can see |
| 00:30.35 | bcrisp | calls are fine - no jitter |
| 00:30.57 | bcrisp | i think it may be a context issue... i wonder |
| 00:31.24 | bcrisp | i can test that pretty quick |
| 00:31.26 | etfonhomey1 | And just to be sure that you're using rfc2833, do a "sip show peer xxxx" where xxx is your CallCentric entry in sip.conf |
| 00:31.32 | *** join/#asterisk Sabrina-- (n=chatzill@modemcable091.107-22-96.mc.videotron.ca) |
| 00:32.22 | LemensTS | When I do; sip show peer exten, it says..: Expire: 3473 I see it decrements 3473, 3472, etc every second. I set qualify=5000, is this where it is getting that from? |
| 00:33.04 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 00:33.12 | bcrisp | etfonhomey1, yes it shows rfc2833 |
| 00:33.12 | Sabrina-- | hi guys! please can someone tell me what the "out pulse number" is ? we're talking to a provider, and they are asking us what number we will out pulse and that we need to give them a valid CPN. |
| 00:33.57 | Sabrina-- | (I know this is not related to Asterisk, however I didn't know where else to ask..) |
| 00:34.15 | etfonhomey1 | bcrisp, I could see it being a context issue. pastebin your dial plan. |
| 00:34.30 | bcrisp | etfonhomey1, im almost certain it is now that i look |
| 00:34.34 | doolittlework | etfonhomey1: he is using freepbx |
| 00:34.55 | etfonhomey1 | I thought Manx beat that out of him yesterday! |
| 00:35.02 | bcrisp | id like to be able to set up a forward to cell# that has ability to transfer |
| 00:35.05 | bcrisp | thats not usual i think |
| 00:35.07 | doolittlework | lol |
| 00:35.29 | bcrisp | ? |
| 00:35.33 | bcrisp | doolittlework, who is |
| 00:36.03 | doolittlework | bcrisp: who is what/ |
| 00:36.15 | bcrisp | is not using freepbx |
| 00:36.48 | doolittlework | "all circuits are busy now, please try again later" ?????????? |
| 00:37.03 | bcrisp | ? |
| 00:37.41 | bcrisp | im confused now |
| 00:37.58 | doolittlework | we can not help you if you do not share info |
| 00:38.11 | doolittlework | paste your sip_additional.conf |
| 00:38.21 | bcrisp | doolittlework, who are you talking to? |
| 00:38.25 | doolittlework | you |
| 00:38.40 | bcrisp | uh |
| 00:38.44 | etfonhomey1 | Anyone getting DSCP set on your signaling traffic? I have "tos_sip=cs3" and a tcpdump of my outgoing sip traffic is still marked 0. |
| 00:38.52 | doolittlework | we can not help u buddy if you dont give us more to go on |
| 00:38.56 | bcrisp | doolittlework, im not using freepbx |
| 00:39.11 | bcrisp | doolittlework, you quoted a different users chat i believe |
| 00:39.16 | doolittlework | paste your sip.conf |
| 00:39.23 | etfonhomey1 | And dialplan? |
| 00:39.30 | doolittlework | sry dude |
| 00:39.35 | bcrisp | ya... wtf lol |
| 00:39.40 | doolittlework | lol |
| 00:39.59 | bcrisp | ~punish doolittlework |
| 00:40.00 | infobot | ACTION forces doolittlework to install and use Windows |
| 00:40.11 | doolittlework | lol |
| 00:40.38 | bcrisp | i think its time for another coffee break |
| 00:40.44 | doolittlework | true |
| 00:41.35 | doolittlework | when do you wanna forward on busy or no answer? |
| 00:42.55 | bcrisp | we having someone screening calls thats on the road |
| 00:43.02 | bcrisp | he then needs to transfer to whoever he wants |
| 00:43.25 | etfonhomey1 | bcrisp, using a cell phone? |
| 00:43.30 | bcrisp | righto |
| 00:43.43 | bcrisp | brb |
| 00:44.52 | Sabrina-- | guys please can someone just tell me what an out pulse number and CPN are ? |
| 00:45.34 | LemensTS | is there any way to override the registration seconds that is in the sip phone? |
| 00:47.10 | bcrisp | Sabrina - did you google? |
| 00:47.31 | doolittlework | bcrisp: so you phone him of a trunk of * and then he wants to transfer the call to someone else over the * server?? |
| 00:47.42 | bcrisp | right |
| 00:48.08 | bcrisp | ~CPN |
| 00:48.27 | bcrisp | doolittlework, yes |
| 00:48.36 | bcrisp | Sabrina--, CPN = calling party number |
| 00:48.49 | Sabrina-- | bcrisp, did, but all i found was pulse dialing (rotary phones) |
| 00:49.16 | bcrisp | Sabrina--, they are asking you for the number that will show in callerID |
| 00:49.24 | bcrisp | DID # / callerid |
| 00:50.59 | Sabrina-- | bcrisp, but they're a wholesale provider, shouldn't they just forward the caller ID ? |
| 00:51.21 | bcrisp | Sabrina--, who is "they" ? |
| 00:51.59 | Corydon76-dig | That's a question for your provider |
| 00:54.40 | Sabrina-- | we're applying for an account with Powernet Global, and one question which we hadn't encountered before is: What number will we outpulse? and we need to provide them with a valid CPN. |
| 00:55.01 | bcrisp | Sabrina--, did you ask them "what do you mean?" |
| 00:57.03 | citywok | Sabrina--: either you outpulse it, or they set it for you. |
| 00:57.14 | citywok | they wouldnt ask what you are going to outpulse... that seems dumb |
| 00:57.52 | Sabrina-- | not yet, I was planning on asking them tomorrow, just wanted to see if I could get a clear understanding first as in all honesty I am puzzled |
| 00:58.11 | citywok | typically you either set it to your main phone number, or they set it for you if you dont have any custom requirements |
| 00:58.24 | citywok | we set it becasue we send out any of a hundred different numbers for our callerid |
| 00:58.49 | bcrisp | and we don't because callcentric is a pain |
| 00:59.18 | citywok | sucky |
| 00:59.22 | *** join/#asterisk chendy (n=chatzill@58.251.100.196) |
| 01:00.04 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 01:00.49 | Sabrina-- | ok i see, but do all providers deal with outpulse numbers ? because we never were asked such a question or had even encountered the term out pulse |
| 01:01.24 | bcrisp | citywok, is it possible for a call forwarded from * to a cell be able to transfer to another ext inside * ? |
| 01:01.29 | citywok | thats only used in not sip settings. T1's, FXO/S |
| 01:01.30 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 01:01.51 | citywok | bcrisp: check out the dial commands t & T functions |
| 01:01.57 | bcrisp | citywok ya i have |
| 01:02.01 | citywok | one of them allow the called party to transfer the call |
| 01:02.20 | citywok | if neither of those do it, then i wouldnt know of any other way than to dial back in to the system. |
| 01:02.23 | bcrisp | yeah.. my problem is that when the cell user types # it does play "Transfer" but then it wont accept any data |
| 01:02.32 | citywok | DTMF? |
| 01:02.38 | bcrisp | checked and correct |
| 01:02.51 | citywok | is there a valid dialplan? |
| 01:02.53 | bcrisp | im gonna try something dif now.. lets see |
| 01:02.56 | citywok | i'm not sure what context the t/T works out of |
| 01:03.02 | Sabrina-- | bcrisp, citywok thanks a lot for your help :) I appreciate it |
| 01:03.22 | bcrisp | ya... i think once its bridged the cell # is in a context that cant see any other extensions :/ |
| 01:04.31 | citywok | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 01:04.41 | citywok | it says set GOTO_ON_BLINDXFR |
| 01:05.32 | citywok | anyways i'm taking off, that should work for you, good luck with it! |
| 01:05.57 | bcrisp | citywok you da man |
| 01:12.25 | bcrisp | hrm |
| 01:13.47 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:13.56 | etfonhomey1 | bcrisp, did that work? |
| 01:14.02 | bcrisp | nah |
| 01:14.12 | bcrisp | im looking in core show application dial |
| 01:14.31 | bcrisp | the problem is the called party's context |
| 01:15.05 | etfonhomey1 | Can you pastebin the snippet of your dialplan that covers the Press # to Transfer part? |
| 01:15.44 | bcrisp | its not part of a dialplan |
| 01:15.48 | bcrisp | features.cofn |
| 01:15.49 | bcrisp | conf |
| 01:15.54 | doolittlework | bcrisp |
| 01:16.00 | etfonhomey1 | Ah. |
| 01:16.09 | doolittlework | is your call to mobile sip or zap? |
| 01:16.15 | bcrisp | im using Dial(SIP/<number>@provider,,tk) |
| 01:16.16 | etfonhomey1 | sip |
| 01:17.00 | bcrisp | when its answered, the callee is part of a context that can't see other extensions |
| 01:17.11 | bcrisp | (that would be a default behavior if i were calling a customer etc) |
| 01:17.46 | bcrisp | id like to be able to say "once answered, put callee in the foo context |
| 01:19.39 | *** join/#asterisk Iamnacho (n=Iamnacho@ip174-70-129-163.ks.ks.cox.net) |
| 01:19.50 | *** join/#asterisk mr_ian_ (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 01:19.55 | etfonhomey1 | bcrisp, does the CLI show you which context each leg is in during the call? |
| 01:20.03 | bcrisp | yep |
| 01:20.25 | bcrisp | i may be able to use the U option in dial |
| 01:20.34 | etfonhomey1 | bcrisp, pastebin your CLI output for a test call. I just want to see what you're seeing. |
| 01:21.19 | bcrisp | ok one sec |
| 01:24.45 | bcrisp | i cut out the first portion.. |
| 01:24.47 | bcrisp | http://pastebin.ca/1718975 |
| 01:24.52 | bcrisp | but this shows what i suspected |
| 01:25.12 | etfonhomey1 | bcrisp, I've got time to kill. So, we'll figure this out. I have a spare DID I could try to configure on my * box to do what you're doing. I'm restoring an NTFS partition after a failed GParted shrink. |
| 01:25.46 | bcrisp | ya.. im sure it is because the callee's default context doesnt see the internal extensions |
| 01:26.02 | bcrisp | just not sure how to set callee context when its answered |
| 01:27.08 | etfonhomey1 | "Unable to find extension '3' in context 'from-callcentric'" looks like the key line to me. |
| 01:27.17 | bcrisp | well ya lol |
| 01:27.23 | bcrisp | because i cant set the context |
| 01:27.38 | bcrisp | (it stops on first DTMF digit because it has no extensions in it :) |
| 01:28.07 | bcrisp | there is a TRANSFER_CONTEXT variable that i will try now |
| 01:28.20 | etfonhomey1 | You could put some pattern matching in you from-callcentric context + a GoTo to get you somewhere else. |
| 01:28.35 | bcrisp | ya.. |
| 01:28.41 | etfonhomey1 | But, I'm sure that's not best practice... |
| 01:28.55 | bcrisp | ya i dont want my customers transferring calls due to my sloppy code hehe |
| 01:29.10 | etfonhomey1 | Although, the only thing coming in that context are incoming SIP calls from your SIP provider... |
| 01:30.14 | etfonhomey1 | How'd the TRANSFER_CONTEXT variable work? |
| 01:30.51 | bcrisp | it worked |
| 01:30.57 | bcrisp | :) |
| 01:31.12 | etfonhomey1 | Sweet! |
| 01:31.15 | bcrisp | that is sweet! |
| 01:31.18 | etfonhomey1 | On to the next thing! |
| 01:31.29 | bcrisp | id like to try an attended transfer also |
| 01:32.12 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 01:32.21 | etfonhomey1 | Does the other party hear the spoken "Transfer" on the blind xfer? |
| 01:32.35 | bcrisp | no |
| 01:32.42 | *** join/#asterisk jtodd (i=x41xok44@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 01:32.42 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:32.54 | bcrisp | they hear moh |
| 01:33.10 | etfonhomey1 | Ah. Sweet. |
| 01:33.21 | bcrisp | although i could be wrong .. im sitting right next to the phone and have a headset on |
| 01:34.48 | bcrisp | thats pretty sweet tho |
| 01:35.29 | etfonhomey1 | Get the attended xfer? |
| 01:35.35 | bcrisp | yep |
| 01:35.56 | etfonhomey1 | Good stuff. |
| 01:36.36 | etfonhomey1 | Did you have to use two underscores in the variable name __TRANSFER_CONTEXT? |
| 01:38.46 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 01:38.51 | bcrisp | no |
| 01:38.58 | bcrisp | just TRANSFER_CONTEXT=contextName |
| 01:39.30 | bcrisp | brb |
| 01:40.28 | rbd_ | anyone have any luck getting asterisk working with clean voice quality in a virtualized environment (esxi or KVM, specifically?) |
| 01:41.08 | rbd_ | and specifically with using ztdummy/dahdi-dummy and meetme...I'm wondering if I'd run into choppy voice issues |
| 01:44.11 | heliosj | rbd: I can't say that it's impossible.. but most of the VMs I've seen do not support access to the RTC, which would mean MeetMe wouldn't work right. |
| 01:44.37 | etfonhomey1 | rbd_, I have not, but it wouldn't take very long to setup a test environment in ESXi. |
| 01:44.52 | heliosj | I think Amazon has a kernel patch that might emulate it.. but I'm not sure how well that would work. |
| 01:46.03 | nny | hmm |
| 01:46.18 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 01:46.21 | nny | can't seem to get a script to connect to localhost asterisk manager, it is certainly an issue with iptables |
| 01:46.28 | nny | this is what i have for the port |
| 01:46.29 | nny | http://pastebin.com/m2e76d700 |
| 01:46.36 | nny | am i being stupid here? |
| 01:46.58 | nny | if i stop iptables the script works fine :\ |
| 01:47.56 | rbd_ | ok...that's what I've been seeing too, thanks |
| 01:48.54 | bcrisp | nny i dont think its your iptables |
| 01:49.08 | bcrisp | but you could try disabling anyway |
| 01:49.30 | bcrisp | what does your manager.conf look like? (srub your secret of course) |
| 01:49.43 | nny | bcrisp: blow open for testing |
| 01:49.57 | nny | bcrisp: the minute iptables stops it works heh |
| 01:50.02 | bcrisp | oh hah |
| 01:50.09 | bcrisp | i disabled mine |
| 01:50.23 | bcrisp | manager.conf has ip sec in it too |
| 01:58.45 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 01:59.04 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 01:59.16 | nny | hmm yeah but only iptables state changing causes the issue, so I doubt it's an asterisk config error |
| 01:59.21 | nny | i'll delve further |
| 01:59.52 | bcrisp | no i understand |
| 02:00.03 | bcrisp | i have an ext firewall configured and ipsec on manager.conf |
| 02:00.45 | bcrisp | etfonhomey1, i had voicemail.conf configured to send voicemail files in email |
| 02:00.54 | bcrisp | all of a sudden that stopped happening and i dont know why :) |
| 02:02.05 | etfonhomey1 | Getting anything on the root's email? |
| 02:02.11 | etfonhomey1 | in root's email* |
| 02:02.22 | bcrisp | not that i can see |
| 02:02.55 | etfonhomey1 | Where is your * box sending the email? |
| 02:03.12 | etfonhomey1 | To an Exchange server or Linux smtp or a hosted email account? |
| 02:03.22 | bcrisp | imap server |
| 02:03.25 | bcrisp | and exchange |
| 02:03.28 | bcrisp | neither recieve now |
| 02:03.41 | bcrisp | s/recieve/receive/ |
| 02:04.24 | etfonhomey1 | Well, IMAP servers receive the email via SMTP, no? It's the clients that sync their mailboxes via IMAP, right? |
| 02:04.46 | bcrisp | ya |
| 02:04.50 | bcrisp | sorry yes |
| 02:05.25 | dlynes | etfonhomey1: not quite...smtp servers receive it via smtp, and imap is a message retrieval protocol for clients....the imap message store and the smtp server are not necessarily on the same machine |
| 02:05.59 | etfonhomey1 | Fire up Wireshark on the Exchange server and see if a connection attempt is being made from your * box to your SMTP server immediately after a VM is left. |
| 02:06.00 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:06.13 | etfonhomey1 | Might also check your sendmail relay settings. |
| 02:06.26 | *** join/#asterisk lcuevas (n=lcuevas@190.142.104.254) |
| 02:06.28 | dlynes | bcrisp: you're running asterisk on windows? |
| 02:06.32 | bcrisp | no |
| 02:06.43 | dlynes | bcrisp: then what's all the talk about exchange server? |
| 02:06.48 | etfonhomey1 | Thanks dlynes, I don't use IMAP much, I just know that when I've configured it, it always wants an SMTP server as well. |
| 02:06.50 | etfonhomey1 | (on clients) |
| 02:07.43 | dlynes | etfonhomey1: yeah...if you're using exchange, you typically don't need an smtp server (exchange has its own proprietary mechanism)....however, exchange still offers smtp, imap and pop3 connectors |
| 02:08.06 | dlynes | etfonhomey1: you don't typically need imap or pop3 on exchange either (unless you've got some non-windows clients) |
| 02:08.43 | etfonhomey1 | If you're sending email to Exchange from *, you would configure sendmail to relay through the Exchange SMTP connector. Just need to add the access rules on the Exchange SMTP part and the correct relay config in your sendmail. |
| 02:09.15 | etfonhomey1 | dlynes, I'm very familiar with Exchange. Just never use IMAP that's all. |
| 02:10.07 | etfonhomey1 | bcrisp, let's see you sendmail relay line. |
| 02:10.15 | bcrisp | :) |
| 02:10.20 | bcrisp | <- linux newb |
| 02:11.09 | bcrisp | dlynes: the mails are going TO exchange |
| 02:11.09 | bcrisp | dlynes: not via |
| 02:12.14 | etfonhomey1 | bcrisp, define(`SMART_HOST',[192.168.250.11])dnl in your sendmail.mc replace 192.168.250.11 with the IP of your Exchange server. |
| 02:12.48 | etfonhomey1 | bcrisp, which version of Exchange? |
| 02:13.00 | carrar | mmm m4 |
| 02:14.14 | bcrisp | ... |
| 02:14.19 | bcrisp | me no have exchange server |
| 02:14.45 | bcrisp | im on a 1and1 linux server with centos5 |
| 02:15.35 | bcrisp | its ok;.. ill need to troubleshoot tomorrow thanks for your help tho |
| 02:15.46 | bcrisp | nite |
| 02:19.29 | dlynes | etfonhomey1: btw..a better solution probably would've been to get him to tail his sendmail log, if the exchange box is having issues receiving from the linux box |
| 02:19.51 | dlynes | etfonhomey1: but if he's using centos, he's probably using postfix, not sendmail |
| 02:20.23 | LemensTS | whatss the point of qualify=5000 if the sip phones override it? |
| 02:20.46 | LemensTS | i mean qualify=2000 |
| 02:20.59 | dlynes | LemensTS: qualify=2000 => asterisk->phones, not phones->asterisk |
| 02:21.17 | etfonhomey1 | dlynes, sendmail is what I use by default on all the CentOS 5.3/5.4 installs I've done in the last few weeks/ |
| 02:22.14 | etfonhomey1 | I can't say that I am 100% sure, but I'm making my config changes to sendmail.mc at least. |
| 02:22.41 | LemensTS | dylnes: the qualify=2000 helps with nat keeping the port open right? |
| 02:23.53 | etfonhomey1 | LemensTS, yes |
| 02:24.29 | etfonhomey1 | LemensTS, by sending a SIP OPTIONS message I believe. |
| 02:24.49 | LemensTS | etfonhomey1: thx. So what is Expire: 3600 good for? (when i do sip show peer exten) |
| 02:24.57 | nny | any iptables peeps here? |
| 02:25.05 | nny | I swear I have the manager rule right, but it doesn't work |
| 02:25.08 | LemensTS | I know its how often to reregister, but what is that useful for |
| 02:25.12 | nny | http://pastebin.com/m2e76d700 |
| 02:26.02 | LemensTS | etfonhomey1: 3600seconds is 1 hour til reregister on all my phones...that seems odd |
| 02:26.16 | nny | the sad thing is, the script and the asterisk install are on the same server :\ |
| 02:27.17 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 02:27.37 | nny | i could just disable iptables but that seems kinda silly |
| 02:28.21 | nny | http://pastebin.com/m42ac8055 is full output of iptables |
| 02:28.57 | nny | am I missing something that allows the loopback connector to accept connections? I don't think i have a mysql rule, but local db connections work fine |
| 02:38.38 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 02:48.21 | LemensTS | How often does a nat port need to have traffic to stay open...on most routers...? |
| 02:51.03 | *** join/#asterisk lowtek (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 02:51.15 | lowtek | Hey guys, are release candidates considered stable? i.e., 1.4.28rc1? |
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| 03:11.40 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 03:13.27 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
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| 03:15.48 | odie_flocon | Hey |
| 03:17.23 | LemensTS | exten => _8500,n,Gotoif($["${CALLERID(number)}" = "12223334444"]?dial) |
| 03:17.29 | LemensTS | is something wrong with this in 1.4? |
| 03:17.43 | LemensTS | think ive tried every combination |
| 03:18.00 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:19.07 | voipmonk | how do you know you're getting the 1 when the number comes in |
| 03:20.26 | *** part/#asterisk odie_flocon (n=Odie@S010600219113b6e8.cg.shawcable.net) |
| 03:21.50 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:24.09 | LemensTS | exten => _8500,n,NoOp(${CALLERID(num)}) |
| 03:24.20 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 03:24.37 | LemensTS | <PROTECTED> |
| 03:25.06 | LemensTS | I gotta be doing something dumb |
| 03:26.21 | *** join/#asterisk jrlenz (i=ralf@mako.jrlenz.com) |
| 03:26.52 | LemensTS | hmmm now its working... |
| 03:28.28 | jrlenz | has anyone tried using a sierra aircard as a gsm bridge, or is it even possible ? |
| 03:30.05 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
| 03:31.35 | x86 | free toll-free termination over IAX2... anyone have suggestions? |
| 03:32.05 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
| 03:32.46 | voipmonk | muahha |
| 03:32.49 | voipmonk | you're welcome |
| 03:33.04 | voipmonk | scrapes the rest of the pixie dust and puts it back in his satchel |
| 03:34.53 | LemensTS | lol im ready for 1.6.2, they fixed it where these dumb linksys/sipura keep-alives dont get bad-events |
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| 04:03.04 | p3nguin | lemensts: To match callerID number, you may as well do it the other way: exten => 8500/12223334444,1,Verbose(1,do stuff). |
| 04:03.53 | p3nguin | x86: I don't know about IAX2, but there are a couple SIP ones I use. |
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| 04:42.36 | Cynagen | I've got a question for you guys. |
| 04:44.29 | Cynagen | I want to set up a custom outbound trunk that runs a bash script, waits for an incoming call, and then answers and connects the SIP device initiating the call to that new incoming line |
| 04:53.10 | dlynes | Cynagen: Thought you said you had a question? |
| 04:54.44 | dlynes | lowtek: no...that's why they're release candidates, and not releases |
| 05:06.01 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:0:0:0:2) |
| 05:06.06 | Cynagen | I want to set up a custom outbound trunk that runs a bash script, waits for an incoming call, and then answers and connects the SIP device initiating the call to that new incoming line |
| 05:06.12 | Cynagen | How can I do this? |
| 05:06.56 | Cynagen | Like how do I run an external script/application, and then, how do I wait for the incoming, and connect that to the line that the SIP device opened. |
| 05:07.40 | Cynagen | basically I want to enable outward dialing on my google voice account using Asterisk through my linksys SIP device |
| 05:10.42 | ChannelZ | I don't get what the bash script is for, but I also don't use google voice and thus understand it's intricacies |
| 05:11.52 | Cynagen | the bash script is to call wget a few times, to 1, log into google voice service, save session cookies, and then 2, using saved cookies, dial a call via POST on wget, already have the script figured out, I just need an automated function to accomplish this |
| 05:12.21 | Cynagen | then wait for the incoming call from google on the gizmo SIP account, and connect that to the open channel from the linksys device placing the call |
| 05:13.53 | Cynagen | here's the layout, i've got planned: |
| 05:14.33 | Cynagen | POTS phone <-> linksys SIP router <-> asterisk <-> gizmo5 <-> google voice |
| 05:14.51 | Cynagen | now, i've got the circuit working already, just no outbound dialing |
| 05:15.11 | ChannelZ | well there's no real 'waiting for the call' in an interactive sense.. you just setup your dialplan to answer the gizmo call and dial out to the other. |
| 05:15.12 | Cynagen | somebody calls my google voice number, it goes through that chain, without asterisk |
| 05:15.24 | Cynagen | no |
| 05:15.27 | Cynagen | sighs |
| 05:15.29 | Cynagen | you misunderstand me |
| 05:15.34 | Cynagen | i'm not redirecting calls |
| 05:15.59 | Cynagen | i've got to do this roundabout crack ass way to initiate an outbound call on my google voice number |
| 05:17.39 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 05:18.46 | Cynagen | basically, to make an outgoing call, I have to go to the google voice page, and punch in the phone number I wanna call, and tell it to call my gizmo SIP account |
| 05:19.13 | Cynagen | then my phone on my linksys SIP device rings, I answer, and it calls the other number, and connects me to that outbound line |
| 05:19.23 | Cynagen | i hear dead air for a short period of time, then ringing |
| 05:20.21 | Cynagen | i wanna take having to get on my computer to make the outbound call out of the equation and have asterisk do it for me |
| 05:24.44 | Cynagen | i can't find any solid documentation on anything remotely close to this |
| 05:28.29 | ChannelZ | well this is thinking out loud, I have no idea if this would actually work. Your "dial out" extension calls System and runs your script that does whatever. Next it puts you on hold or maybe throws you into a conference to keep your channel opens. The incoming call from Google comes in on whatever it comes in on, and it's dialplan uses Bridge() to connect the channel to your channel |
| 05:28.57 | p3nguin | Try googling for "orgasmatron gizmo google" |
| 05:29.06 | Cynagen | wha? |
| 05:29.17 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 05:29.33 | p3nguin | and/or pygooglevoice |
| 05:29.41 | Cynagen | i'd say, put me on hold, with ringing as the hold tone |
| 05:29.46 | Cynagen | so i hear ringing the entire way |
| 05:29.50 | x86 | anyone use sipbroker? |
| 05:29.58 | Cynagen | and anyone else using the phone at my house will hear normal ringing |
| 05:30.10 | x86 | I'd like someone to test call me at *97687601 |
| 05:30.36 | p3nguin | not enough numbers |
| 05:31.09 | ChannelZ | http://geeklad.com/free-phone-calls-with-google-voice-asterisk-and-gizmo |
| 05:31.32 | ChannelZ | never underestimate the amount of jurry rigging someone will do to get something for nothing |
| 05:31.46 | Cynagen | that's it right there |
| 05:32.00 | Cynagen | and yeah, i figured somebody else may have found some info and done it before me |
| 05:32.03 | Cynagen | i figured as much |
| 05:32.09 | drmessano | Considering Google bought Gizmo5 and it work without scripts soon enough... |
| 05:32.22 | Cynagen | i would hope so |
| 05:32.32 | Cynagen | but i might keep asterisk in the mix for later |
| 05:32.32 | p3nguin | It's just a bother to have to get a web browser to make calls. |
| 05:32.39 | Cynagen | yup |
| 05:32.51 | Cynagen | when you don't have a web browser, it blows |
| 05:33.05 | Cynagen | also, side note, you can't call google voice numbers from payphones |
| 05:33.06 | p3nguin | The pygooglevoice script should be helpful. |
| 05:33.17 | p3nguin | Uh, why not? |
| 05:33.21 | Cynagen | i tried it |
| 05:33.32 | Cynagen | the payphone refused |
| 05:33.47 | Cynagen | i don't know why, but it refused |
| 05:33.56 | p3nguin | I don't have a payphone handy, but I would expect it to work the same as any other number. You deposit 35 cents and dial the number. |
| 05:34.04 | Cynagen | it's 50 cents now |
| 05:34.10 | p3nguin | inflation |
| 05:34.12 | drmessano | WTF is a payphone |
| 05:34.16 | Cynagen | it didn't work for me |
| 05:34.17 | drmessano | Its 2009, fuckers |
| 05:34.36 | drmessano | Is that like a cellphone connected to a wall you put quarters in? |
| 05:35.13 | ChannelZ | payphones are cell phones for prostitutes |
| 05:35.20 | Cynagen | lmfao |
| 05:36.28 | drmessano | i thought cellphones were cellphones for prostitutes :) |
| 05:37.45 | ChannelZ | I suppose they are now with all the disposables |
| 05:45.23 | Cynagen | p3nguin: can i use orgasmatron with AsteriskNOW? |
| 05:48.35 | [TK]D-Fender | Cynagen: Unless you're jane Fonda at which point you'll probably short it out... |
| 05:48.48 | Cynagen | it won't install |
| 05:49.04 | Cynagen | WARNING: The Orgasmatron installer only works with PBX in a Flash. |
| 05:49.04 | Cynagen | You appear to be running an incompatible version. DO NOT PROCEED. |
| 05:49.13 | Cynagen | man i love being locked out |
| 05:50.37 | [TK]D-Fender | Cynagen: You've got teh source. Get busy |
| 05:51.11 | drmessano | hahah |
| 05:51.18 | drmessano | Barbarella, is that you? |
| 05:51.26 | [TK]D-Fender | drmessano: pwned |
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| 06:01.06 | Cynagen | no source code |
| 06:01.12 | Cynagen | it's an ELF |
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| 06:01.22 | drmessano | Not a DWARF? |
| 06:01.48 | drmessano | Maybe it's a gnome and looks like an Elf |
| 06:01.54 | Cynagen | sighs |
| 06:03.57 | Cynagen | well i can look at it from a brighter light |
| 06:04.09 | [TK]D-Fender | sets his laser printer on "KILL" |
| 06:04.13 | Cynagen | once i have this running, i can set my SIP device to use the external address for my computer |
| 06:04.40 | Cynagen | asterisknow running, i can take my SIP gateway with me, and plug it into any internet connection, and viola |
| 06:07.50 | [TK]D-Fender | O.o |
| 06:08.10 | Cynagen | hmm... |
| 06:08.18 | Cynagen | naw, that wouldn't work... |
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| 06:11.41 | JohnTeddy | What is a good simple quality headset with headphone/mic. To use with a laptop. Something that isn't too annoying or wrapping around my head etc. |
| 06:12.14 | JohnTeddy | This isn't asterisk related. But I figure you guys would probably give the best recommendation since you've seen these products more than me. |
| 06:12.19 | JohnTeddy | quality mic/headphones |
| 06:16.32 | JohnTeddy | something that a telemarketer would use, that is my litmus test. If a telemarketer would like it. |
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| 06:27.42 | ChannelZ | hmm twitter.com hacked? |
| 06:29.20 | hitesh | hello ever one I am new to asterisk. I installed asterisk on ubuntu 8.04 and asterisk gui tools. I configured the required settings for the gui tool. I did a configcheck. For gui tools I send the request to localhost. but when I put the username and password to it and click on login the cli shows me warning for channel.c that no channel type registered for local and for util.c ast_carefulwrite: write() returned error: Broken pipe. It keeps on going. am |
| 06:29.21 | hitesh | I doing something wrong? |
| 06:31.09 | [TK]D-Fender | hitesh: GUI's are not supported here. Go look at whatever resource that "tool"'s developers offer |
| 06:31.47 | hitesh | ok thank you [TK]D-Fender |
| 06:33.25 | MaliutaLap | hehe dev, tool. hehe |
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| 09:02.51 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 09:16.16 | siera04 | hi, i use asterisk 1.6.0.19 now. |
| 09:16.52 | siera04 | when i run command 'conference list 9999' through AMI, the response is |
| 09:17.00 | siera04 | Response: Follows |
| 09:17.01 | siera04 | Privilege: Command |
| 09:17.03 | siera04 | No active conferences. |
| 09:17.05 | siera04 | --END COMMAND-- |
| 09:17.27 | siera04 | But when i run this command in CLI, the user is in conference now. |
| 09:17.31 | siera04 | what's the problem? |
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| 09:56.06 | *** join/#asterisk Han (n=han@unaffiliated/han) |
| 09:56.27 | Han | Which table in the asterisk database holds the inbound routes? |
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| 10:14.38 | TSM2 | its all in the extentions.conf i think |
| 10:14.48 | TSM2 | routes are not stored in the astdb |
| 10:15.03 | TSM2 | unless ofcourse you have programmed it specialy that way |
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| 10:19.16 | *** join/#asterisk fabsoft (n=fabsoft@host177-255-static.28-79-b.business.telecomitalia.it) |
| 10:19.26 | fabsoft | hi all |
| 10:20.26 | Gido-E | fabsoft do you live in catalonia? |
| 10:21.14 | fabsoft | anyone can help me with asterisknow? i've a tdm400 with a fxo port and another one fxs. after o configured a trunk for fxo and a simple extension for fxs, i cant get nothing to work |
| 10:21.45 | fabsoft | the console message always are chan unavailable.. |
| 10:24.54 | Gido-E | fabsoft i can tel you, the chan is unavailable. But i think you had that figured out. |
| 10:31.22 | *** join/#asterisk fabsoft (n=fabsoft@host177-255-static.28-79-b.business.telecomitalia.it) |
| 10:32.32 | fabsoft | anyone can help me to setting up working a tdm400 with 1 fxs + 1fxo with asterisknow ? just a couple of tips |
| 10:32.59 | Gido-E | fabsoft doe you have a /proc/dahdi or /proc/zap ? |
| 10:34.08 | fabsoft | Gido-E i've the dahdi |
| 10:35.25 | fabsoft | the /proc/dahdi/1 reports on chan 0 the fxs+, and on chan 2 the fxo |
| 10:39.33 | Gido-E | fabsoft, ok. That is the Kernel part. |
| 10:39.48 | fabsoft | Gido-E, i notice that the chan which asterisk uses is ZAP, and not DAHDI... can this be the problem ? |
| 10:40.01 | Gido-E | yep! |
| 10:41.30 | fabsoft | thi sound strange to me |
| 10:42.24 | TSM2 | fabsoft: dahdi_genconf |
| 10:42.25 | fabsoft | how the fuck last asterisknow release doesnt support dahdi with simple analog ports? |
| 10:42.35 | fabsoft | TSM2i did it already |
| 10:42.44 | TSM2 | dahdi show channels |
| 10:42.50 | TSM2 | dahdi_scan |
| 10:42.55 | fabsoft | TSM2 report nothing |
| 10:43.07 | TSM2 | have you got dahdi? |
| 10:43.34 | TSM2 | did you run dahdi_genconf after dahdi was started? |
| 10:43.47 | Gido-E | TSM2 he has /proc/dahdi/1 |
| 10:43.55 | Gido-E | with his hardware. |
| 10:44.21 | fabsoft | did dahdi_genconf at first run |
| 10:44.33 | Gido-E | fabsoft you can edit, the dialplan :-). |
| 10:44.46 | fabsoft | also the kernel reports the 2 chans are loaded correctly |
| 10:44.51 | TSM2 | fabsoft: im bored, if you have screen ile have a look |
| 10:45.09 | Gido-E | TSM2 pre holliday mood? |
| 10:45.32 | TSM2 | Gido: fucked off mood and cant be botherd to do anything, as long as its working, alls good |
| 10:47.24 | fabsoft | TSM2: in the control panel, i noticed the dahdi compatibility mode, this cause the diaplan to use the old name ? ie: ZAP/1 ZAP/3 ? |
| 10:47.44 | Gido-E | fabsoft nope, not for me. :-) |
| 10:47.53 | Gido-E | I don't know the official statement. |
| 10:48.15 | fabsoft | Gido-E, what do you mean, do you have the ZAP or DAHDI notation? |
| 10:48.31 | Gido-E | fabsoft with dadhi, you sould use dahdi notation. |
| 10:48.44 | fabsoft | ook |
| 10:48.56 | TSM2 | i use ZAP notation |
| 10:49.05 | fabsoft | Gido-E, TSM2, can i use the console to test what get wrong ? |
| 10:49.15 | TSM2 | you use FREEPBX thats where that comes from |
| 10:49.29 | fabsoft | from asterisknow |
| 10:49.33 | fabsoft | last release |
| 10:50.05 | TSM2 | just leave it as it is |
| 10:50.22 | fabsoft | ook,. doesn works NOTHING |
| 10:50.23 | TSM2 | its fine, most peeps use ZAP compatabilty as it just makes some things work better |
| 10:50.37 | TSM2 | still no idea what you want to do |
| 10:50.52 | fabsoft | TSM2, is the only trunk i've seen |
| 10:51.36 | fabsoft | TSM2, i need to configure a fxo and a fxs port |
| 10:51.38 | fabsoft | stop |
| 10:52.32 | TSM2 | if its not showing in your asterisk CLI dahdi show channels, you need to get that sorted first, its nothing to do with dialplans |
| 10:52.39 | fabsoft | TSM2, after configured via control panel, a trunk for fxo, and a extension for fxs, i cant call fxs, i cant call via fxo, neither receive |
| 10:54.13 | *** join/#asterisk Han (n=han@unaffiliated/han) |
| 10:54.44 | Han | Can I update the asterisk database somehow when updating to a new release? |
| 10:56.43 | TSM2 | Han, dont worry |
| 11:02.19 | *** join/#asterisk s0lid (n=s0lid@124.104.45.64) |
| 11:04.05 | Han | I do worry since we did a new install a new machine and want to import the old databases. |
| 11:07.57 | TSM2 | why, just copy it |
| 11:08.06 | TSM2 | its not as if it stores anything about routing |
| 11:08.13 | TSM2 | its just a BDB format db |
| 11:09.06 | TSM2 | its in /var/lib/asterisk/astdb |
| 11:10.33 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
| 11:14.06 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
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| 11:23.01 | *** join/#asterisk morex (n=m@5ace0728.bb.sky.com) |
| 11:23.29 | morex | Hey what's the proper way to set the device state for a local channel? I don't want my queue to ring an agent who's already on the phone. |
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| 11:37.47 | DennisG | has someone experience with the rtptimeout function ? |
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| 11:46.25 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 11:46.35 | shamelessn00b | Hi guys |
| 11:46.47 | shamelessn00b | I want to do something similar to http://blog.mserver.nl/2008/07/09/asterisk-playing-audio-streams/ but using VLAN player |
| 11:46.51 | shamelessn00b | is it possible |
| 11:50.59 | *** join/#asterisk sdr_ (n=sdr@82.103.69.146) |
| 11:53.15 | sdr_ | hi all my asterisk is taking too long to answer. I'm using sip, and the INVITE packets are delivered to the machine according to tcpdump but asterisk does not respond until some time |
| 11:54.09 | sdr_ | then wen i start dialing out the signalization is broken again |
| 11:54.38 | sdr_ | any ideas? |
| 11:58.15 | sdr_ | anyone? |
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| 12:20.49 | gavimobile | is it a bad idea to test asterisk on a windows vbox |
| 12:21.02 | TSM2 | its glitchy somtimes |
| 12:21.38 | gavimobile | tsm2, I dnot have another spare computer to test on, do you know of a better solution for me |
| 12:21.43 | DennisG | well i have no problems with a virtual asterisk in vmware workstation (windows 7 host) |
| 12:21.49 | DennisG | just to mention.. |
| 12:22.04 | gavimobile | DennisG: that's goood news |
| 12:22.33 | gavimobile | do you think it'll make a difference on xp pro 86x |
| 12:22.37 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:22.38 | gavimobile | x86* |
| 12:22.52 | DennisG | but i have a vanilla asterisk 1.6.1 with Ubuntu 9.10 server edition =) |
| 12:23.10 | DennisG | i don't think so gavimobile |
| 12:23.16 | DennisG | because it's the emulation |
| 12:23.25 | gavimobile | DennisG: so on you are windows pc you have vbox with ubuntu and on ubuntu you installed asterisk |
| 12:23.40 | DennisG | yes with aptitude install asterisk =) |
| 12:23.54 | DennisG | compiled with the addons and the mysql module =) |
| 12:24.13 | DennisG | i don't use a bri/pri card |
| 12:24.18 | DennisG | just SIP -> SIP |
| 12:25.04 | gavimobile | DennisG: thanks for the feedback... I never installed asterisk before, but I would like to setup a auto call center at my office.i downloaded this iso AsteriskNOW-1.5.0-i386-1of1.iso cause as the asterisk website this was recommended for begenniers |
| 12:25.12 | gavimobile | what cha think? |
| 12:25.18 | gavimobile | have you heard of this version? |
| 12:25.42 | DennisG | yeah i've heard about it. but i have never used it |
| 12:25.47 | DennisG | i use the normal asterisk version |
| 12:26.11 | DennisG | but i have experience with the voiceroute druid, freepbx and that sort of pbx's |
| 12:27.07 | DennisG | just try asteriskNOW, there is no problem if you happy with it =) |
| 12:28.06 | gavimobile | ok... I think im gona do vbox with centos4 or 5 |
| 12:28.36 | gavimobile | cause I saw a screenshot on asterisk website with centos |
| 12:28.46 | gavimobile | so im guessing its recommended |
| 12:28.54 | gavimobile | not that ubuntu is no good |
| 12:29.05 | gavimobile | thanks for you are advice DennisG |
| 12:29.33 | DennisG | no problem! you can always use a GUI for configuring |
| 12:29.43 | DennisG | good luck with your asterisk PBX ! |
| 12:30.47 | gavimobile | thanks dood |
| 12:31.11 | DennisG | =) |
| 12:31.46 | Akiraa | 1.What's a softphone app you would or actually do use? 2.Do you have a test infrastructure in place for VOIP networks? 3.Before deploying an IP network, do you do some testing with virtual devices rather than actual hardware? |
| 12:33.13 | _Raptor_ | Akiraa: zoiper is quite ok |
| 12:33.30 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:33.30 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 12:33.39 | shamelessn00b | I want to do something similar to http://blog.mserver.nl/2008/07/09/asterisk-playing-audio-streams/ but using VLAN player |
| 12:36.30 | *** join/#asterisk danj1980 (n=dan@91.108.6.182) |
| 12:38.18 | *** join/#asterisk lukhas (n=lucas@bearstech/lukhas) |
| 12:38.21 | lukhas | hello |
| 12:38.53 | danj1980 | Hi everyone |
| 12:39.31 | danj1980 | Does anyone have a cheap provider for UK Mobile calls? Low volumes at the moment. |
| 12:41.29 | shamelessn00b | app_rtsp does it play only video streams?? |
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| 13:11.43 | reptiles | howdy |
| 13:12.04 | reptiles | when i dial in on PSTN, and out on SIP, i can not hear the rings of the phone on the end of the SIP. |
| 13:12.16 | reptiles | is this normal? and how would i do things if i want to hear the rigs? |
| 13:19.46 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
| 13:23.49 | gaetronik | answer |
| 13:24.05 | gaetronik | or another command i don't remember |
| 13:25.04 | voipmonk | ? |
| 13:25.10 | *** part/#asterisk lukhas (n=lucas@bearstech/lukhas) |
| 13:25.35 | jblack | So, the at&t data revolt is at 3 eastern today, eh? |
| 13:25.52 | gaetronik | maybe i didn't understand the question |
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| 13:30.13 | reptiles | nope |
| 13:30.42 | reptiles | i already have answer() at the beginning of the context, and that sems to actually be what's breaking it. |
| 13:31.05 | reptiles | if i just jam the call straight into a Dial(SIP/sss), i hear the rings. |
| 13:31.09 | reptiles | i think. |
| 13:31.10 | reptiles | hrm. |
| 13:31.14 | reptiles | sec, lemme test that again. |
| 13:32.30 | reptiles | yeah, if the answering context starts with Dial(SIP/sss), i hear the sipphone ringing. |
| 13:32.40 | reptiles | if i do Answer, then Dial, i don't get ringing. |
| 13:33.18 | reptiles | i've tried setting progressinband=(yes/no/never) in the sip profile. |
| 13:33.59 | reptiles | but, basically, if the line is answered using answer(), and then passed to a Dial(), it doesn't seem to pass the ring tones back to the caller. |
| 13:34.02 | reptiles | what am i missing here? |
| 13:34.18 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) [NETSPLIT VICTIM] |
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| 13:38.53 | SuPrSluG | reptiles: if you answer the phone and then put someone on hold and dial out should the person on hold hear the ringing? |
| 13:39.39 | SuPrSluG | reptiles: essentially, that is what you are doing |
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| 13:53.14 | reptiles | supr, ok, that's sounds about right. however, what if i want the caller to hear the ringing? |
| 13:53.41 | reptiles | or, would that be more of a "transfer"? |
| 13:54.45 | SuPrSluG | you could use the r option at the end of the dial command. that might work |
| 13:55.23 | SuPrSluG | <PROTECTED> |
| 13:55.27 | reptiles | yeah, but if the final destingation is busy, it does ring ring bussy busy |
| 13:55.56 | reptiles | bascially, what i want is to enable in-band progress |
| 13:56.27 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 13:57.01 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
| 13:57.50 | CGMChris | I am having problems with dead sip channels hanging and staying in 'sip show channels' and 'core show channels' until I restart asterisk. I already have rtptimeout=60 and rtpholdtimeout=61 in sip.conf. Anything else I can do to fix this problem? |
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| 14:12.00 | x86 | `/3 |
| 14:13.04 | CGMChris | must be too early in the morning to get help...everyone is still coffee deprived at this point I suppose. |
| 14:14.38 | danj1980 | Does anyone have a cheap provider for UK Mobile calls? Low volumes at the moment. |
| 14:15.17 | x86 | heh... cheap + (mobile anywhere in europe) doesn't compute |
| 14:15.26 | x86 | from what I've seen |
| 14:15.44 | *** join/#asterisk devafree (i=kannan@122.165.250.150) |
| 14:16.41 | *** join/#asterisk grandpapadot (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 14:16.47 | grandpapadot | Hey guys, are release candidates considered stable? i.e., 1.4.28rc1? |
| 14:17.14 | CGMChris | no. |
| 14:17.26 | reptiles | well, more stable than beta, i suppose. |
| 14:17.34 | reptiles | but there is a reason its called a candidate. |
| 14:18.09 | CGMChris | release candidate = "we think it's good, but we're not really sure, so you test it and see if it works" |
| 14:20.19 | x86 | I thought that's the way "stable" releases were too? ;) |
| 14:20.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:20.42 | grandpapadot | x86 lol yep |
| 14:21.43 | x86 | although I've been quite fond of 1.6.1.11 |
| 14:21.52 | x86 | no headaches yet |
| 14:22.18 | grandpapadot | That's something else I was trying to figure out ... what's the difference between the two 1.6 branches? |
| 14:22.36 | voipmonk | read the changelog |
| 14:22.39 | x86 | I'm not sure |
| 14:22.41 | voipmonk | thats why its there |
| 14:22.53 | voipmonk | it shows a log of changes for the release |
| 14:23.04 | voipmonk | changes in the release, rather |
| 14:23.12 | grandpapadot | rather .. the philosophy behind the split .. usually there's a good reason |
| 14:23.18 | x86 | ...but probably wont show you differences in branches ;) |
| 14:23.21 | Katty | peeks in |
| 14:23.25 | x86 | heya Katty |
| 14:23.33 | x86 | anyone use SipBroker? |
| 14:24.29 | x86 | any time I try to send a call to them, I get: Got SIP response 513 "Message too big" back from 64.34.162.221 |
| 14:24.38 | x86 | no matter what number I dial and how I dial it |
| 14:24.44 | Katty | is working on the grocery list |
| 14:25.33 | Katty | suggestions? |
| 14:25.40 | grandpapadot | Also, anyone successfully went from zaptel/1.4 to dahdi/1.6 on a server? Is there a clean upgrade path? |
| 14:25.45 | Katty | x86: i've never worked with sipbroker before |
| 14:26.17 | *** join/#asterisk Strykar (n=wakka@122.170.17.158) |
| 14:30.52 | Katty | hmm :< a friend had to put his kitty down, she had feline leukemia :< |
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| 14:36.33 | creativx | bg |
| 14:36.36 | creativx | i just got a kitty |
| 14:38.36 | jaytee | Anyone know if there are any plans to support Fax for Asterisk on 64 Asterisk platforms in the future? |
| 14:40.06 | eppigy | Katty: i have to go grocery shopping too |
| 14:42.05 | jaytee | Dave tiene que ir al supermercado y carneceria |
| 14:43.42 | *** join/#asterisk qu1ckkkk (i=c419ffd6@gateway/web/freenode/x-fnsruriyzzsfjzbp) |
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| 14:44.34 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:45.24 | Katty | jaytee: even if it did support faxing, it wouldn't be all that reliable |
| 14:45.28 | Katty | jaytee: especially outbound :/ |
| 14:46.12 | Katty | jaytee: might have better luck with a 3rd party setup where your clients can print and recieve faxes via email |
| 14:46.26 | Katty | jaytee: but you already knew all this, of course. |
| 14:46.32 | Katty | eppigy: what's you on your list so far? |
| 14:46.55 | Katty | eppigy: i have juice, bagels, cream cheese, soup, ferret food, cheese, conditioner, and bread |
| 14:47.43 | jaytee | Katty, I want to take incoming faxes only and route them to my Exchange server which can terminate t.38 fax but balks at answering a call from a standard fax machine. I need to have * act as a t.38 gateway or relay. |
| 14:48.12 | Katty | jaytee: inbound faxing isn't so bad. |
| 14:49.24 | jaytee | Katty, I've gotten it to work internall by faxing from an fax machine hooked to a Linksys 2102 with t38 passthrough enabled but I can't get Exchange to answer when I try to route a fax call directly from the PSTN to Exchange. |
| 14:49.37 | Katty | nods |
| 14:49.40 | jaytee | s/internall/internally |
| 14:49.56 | *** join/#asterisk pyite_hsv (n=pyite_hs@unaffiliated/pyite) |
| 14:49.58 | Katty | i've never tried to do that...i've always just used mutt to forward it to a distrobution group on the exchange box. |
| 14:50.21 | Katty | or individual accounts if using dids |
| 14:50.22 | jaytee | using Fax for Asterisk or some other solution? |
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| 14:50.54 | Katty | res_fax |
| 14:51.02 | Katty | the one you register with digium for for 1 free license key |
| 14:51.21 | *** part/#asterisk qu1ckkkk (i=c419ffd6@gateway/web/freenode/x-fnsruriyzzsfjzbp) |
| 14:51.41 | snadge | 8==D ---. |
| 14:51.52 | Katty | hi snadge |
| 14:52.01 | *** mode/#asterisk [+b %snadge!*@*] by russellb |
| 14:52.09 | russellb | snadge: don't do that crap in here, wtf? |
| 14:52.15 | Katty | jaytee: i take the uniqueid and dump it into a folder, then email that file, then delete that file |
| 14:52.32 | Katty | russellb: i must have missed something |
| 14:53.20 | Katty | jaytee: course i don't use a fax machine |
| 14:54.16 | russellb | jaytee: What version of OCS do you use? |
| 14:54.18 | Katty | jaytee: and it comes off an IVR not a pots line |
| 14:56.04 | *** mode/#asterisk [-b %snadge!*@*] by russellb |
| 14:56.23 | snadge | ok no more ascii penii |
| 14:56.34 | russellb | :-p |
| 14:57.00 | snadge | katty interpreted that as hello though.. thats interesting :D |
| 14:57.13 | Katty | i'm a very interesting person. |
| 14:57.15 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:57.21 | snadge | perhaps in some ancient cultures that was a method of greeting |
| 14:57.39 | Katty | hi Naikrovek |
| 14:57.51 | Naikrovek | greetings and salutations dearest katty |
| 14:58.08 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 14:58.46 | [TK]D-Fender | snadge: My yogurt has more culture than you... even if its just bacterial. |
| 14:59.14 | Katty | i've not heard that line before, but it's cute! |
| 15:02.00 | Naikrovek | trying to convince my supervisor that consuming 100% of the bandwidth of our ISP is not a good idea |
| 15:02.08 | Naikrovek | "oh it'll only be for about 10 days" he says |
| 15:02.54 | voipmonk | and they'll blame you when stuff no worky |
| 15:02.54 | Naikrovek | he doesn't understand that email will slow to a crawl, web surfing will slow to a crawl, and phones will stop working. damned ISP says we're not a big enough customer to implement QoS |
| 15:03.00 | Naikrovek | voipmonk: ohyes |
| 15:03.07 | voipmonk | thats how it works |
| 15:03.20 | voipmonk | document your protests |
| 15:03.26 | Naikrovek | I have |
| 15:03.30 | voipmonk | and go over your supers head |
| 15:03.47 | Naikrovek | Owner/CEO is with me |
| 15:03.49 | voipmonk | then slash his tires |
| 15:04.07 | voipmonk | so he can experience the lag on the 'net while he waits for a tow |
| 15:04.24 | Naikrovek | he wasn't here when we tried this the first time |
| 15:04.29 | ManxPower-work | voipmonk: You've been reading BOFH again |
| 15:04.35 | voipmonk | :) |
| 15:04.36 | Naikrovek | has no idea wtf he's talking about |
| 15:04.42 | voipmonk | LOL |
| 15:04.54 | voipmonk | BOFH should be mandatory reading for this field |
| 15:05.21 | ManxPower-work | *nod* |
| 15:07.51 | Naikrovek | BOFH is a little too deliberate for me. I'm a more subtle guy, usually |
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| 15:10.56 | eppigy | Katty: oatmeal, tuna, smoked salmon, red meat, milk, dried fruit, cashews |
| 15:12.43 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
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| 15:21.15 | Katty | oh, i need milk. |
| 15:21.38 | Katty | eppigy: are you going to make any dishes? |
| 15:22.18 | snadge | hows this for awesome.. telstra technician comes to hook up my adsl2 today |
| 15:22.37 | voipmonk | I like salmon and tuna with my mercury? |
| 15:23.02 | snadge | he doesnt have any crimps for a phone socket.. he doesn't have a test adsl modem.. so he leaves me with a wire dangling underneath the house, with stripped ends on it |
| 15:23.25 | snadge | tells me that i should plug an adsl modem into it, and it should have sync.. if it doesnt, and i havn't any problems.. let him know |
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| 15:23.38 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.30 | Katty | voipmonk: fish is very healthy for you |
| 15:26.07 | [TK]D-Fender | Katty: http://tinyurl.com/2p2k2u <- sometimes less so |
| 15:26.29 | eppigy | Katty: i guess so :< |
| 15:26.40 | Katty | eppigy: you guess so? |
| 15:26.45 | Katty | eppigy: you're going to eat tuna cold from a can? :P |
| 15:26.50 | eppigy | YES |
| 15:26.52 | eppigy | no :< |
| 15:27.05 | eppigy | I use paper plates |
| 15:27.09 | eppigy | and eat tuna sammich |
| 15:27.16 | Katty | so cold tuna, bread, paperplate |
| 15:27.21 | eppigy | yesh |
| 15:27.24 | eppigy | :D |
| 15:27.28 | Katty | you poor thing |
| 15:27.34 | eppigy | no man that that is good |
| 15:27.40 | eppigy | that iz just one of six meals |
| 15:27.42 | eppigy | is |
| 15:27.49 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:27.54 | Katty | elaborate on other 5 |
| 15:27.56 | snadge | so i have a mate coming over to drill a hole in the floor, and attach a phone socket to the skirting board.. this is apparently the norm |
| 15:28.54 | Katty | hi Chainsaw |
| 15:29.44 | Chainsaw | Hi there Katty :) |
| 15:30.00 | *** join/#asterisk keyp (n=keyp@66.184.128.98) |
| 15:30.51 | eppigy | Katty: well morning will be protein shake with complexe carb powder |
| 15:31.28 | eppigy | 10am is oatmeal or peanutbutter sammich |
| 15:31.42 | eppigy | lunch is some restraunt |
| 15:31.58 | eppigy | 2pm is tuna sammich |
| 15:32.27 | eppigy | dinner is red meat and goof carb source |
| 15:32.30 | eppigy | *good |
| 15:32.56 | eppigy | after gym another protein shake with carb powder |
| 15:33.01 | eppigy | if it is a workout day |
| 15:33.21 | eppigy | if not maybe yogurt and peanut butter sammich |
| 15:33.21 | DocAwesome | mmm... protein shake is a good idea right about now |
| 15:33.24 | DocAwesome | it's morning snack time |
| 15:33.46 | Katty | checks eppigy's list for vegetables |
| 15:33.53 | eppigy | then usually around 11pm I am hungry again |
| 15:33.55 | Naikrovek | protein shake = snack? |
| 15:33.56 | eppigy | and make something random |
| 15:34.07 | DocAwesome | Naikrovek: it's not a huge shake, so yes |
| 15:34.14 | DocAwesome | only about 300mL |
| 15:34.15 | eppigy | Katty: veggies are worked in at lunch |
| 15:34.20 | Katty | k |
| 15:34.21 | eppigy | you need very little |
| 15:34.29 | Naikrovek | i remember when i used to work out, i'd drink those protein shakes 3x daily and i am so sick of them now |
| 15:34.30 | Katty | i usually get my veggies in afternoon snack |
| 15:34.33 | DocAwesome | only about 10mL of protein powder |
| 15:34.37 | Katty | cause i usually have 2 cups of soup for snack |
| 15:34.39 | eppigy | i am going for more carbs, protein, and calories |
| 15:35.03 | eppigy | 3720 calories per day |
| 15:35.21 | Katty | lots of peanut butter and banana! |
| 15:35.42 | eppigy | yesh |
| 15:35.52 | eppigy | oh yeah I forgot dried fruit and nuts |
| 15:37.01 | Naikrovek | anyone know a good AMI extension status viewing application BESIDES iSymphony |
| 15:37.13 | Naikrovek | we've outgrown FOP |
| 15:41.41 | Katty | there's HUD |
| 15:41.45 | Katty | but i like isymphony better. |
| 15:41.52 | Naikrovek | i think everyone does except me |
| 15:42.01 | *** join/#asterisk simplydrew (n=simplydr@19.sub-75-251-149.myvzw.com) |
| 15:42.05 | Naikrovek | it's even based on eclipse, which i love |
| 15:42.09 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 15:42.12 | Katty | just curious, but are you using isymphony free version or paid version? |
| 15:42.24 | Naikrovek | was using free, and i realize that paid would be better |
| 15:42.34 | Katty | k |
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| 15:42.40 | Katty | what part about it have you 'outgrown'? |
| 15:42.51 | Naikrovek | but all i want is something to show the status of all the extensions that are currently registered |
| 15:42.59 | Naikrovek | i have far too many extensinos for FOP to show easily |
| 15:43.14 | Naikrovek | i could spend a day tweaking it to get them all to show or i could spend an hour finding a windows gui app that does what i want |
| 15:43.30 | Katty | you know isymphony has a list view, right? |
| 15:43.31 | hluesea | hello channel, i have problem. I bought and installed g729 codec then my working server have fault and now it is not working |
| 15:43.46 | Naikrovek | Katty: yes, but i want something that is AMI only, and no server components |
| 15:43.52 | Katty | k |
| 15:44.09 | Naikrovek | i've found some C# manager API client code I may be able to adapt |
| 15:44.16 | keyp | Naikrovek, have you seen this one? http://www.asteriskguru.com/tools/switchboard.php |
| 15:44.20 | Naikrovek | has lots and lots and lots of development experience. |
| 15:44.46 | hluesea | are anyone know the segmentation fault or it is recovery ? |
| 15:45.09 | Katty | hmm, that looks interesting |
| 15:45.15 | Naikrovek | hluesea: if you bought those from Digium you may be able to just contact them |
| 15:45.17 | Katty | i think it would confuse our receptionist here |
| 15:45.34 | Naikrovek | yes, i'm trying to find something for our receptionist as well |
| 15:45.54 | Naikrovek | i dont' want her to be able to click around and hang up anyone's call or anything, just use it to see who is on the phone and who is not |
| 15:45.58 | hluesea | ok i am on it, thanks Naikrovek |
| 15:45.59 | Naikrovek | or who has voicemail maybe |
| 15:46.07 | Katty | nods |
| 15:46.20 | Katty | we had issues with our previous receiptionist using call barge |
| 15:46.36 | Naikrovek | yeah i don't think ours will do that, but you didn't either i'm sure |
| 15:46.47 | Katty | no, not in a million years |
| 15:47.03 | Katty | isymphony doesn't show a barged caller |
| 15:47.06 | Katty | but FoP does. |
| 15:47.23 | *** join/#asterisk etfonhomey (n=etfonhom@office.archvision.com) |
| 15:47.27 | Katty | i saw something supsicious so went back to check the call logs..that's when i found out |
| 15:51.55 | Naikrovek | i used to have an admin working for me who would use dameware to snoop silently on all manner of people |
| 15:52.13 | Katty | boredom? |
| 15:52.13 | Naikrovek | he was drop kick fired |
| 15:52.44 | Naikrovek | he said he wanted to read other people's email, so he woudl snoop on their screens |
| 15:53.32 | voipmonk | wow |
| 15:54.02 | Katty | yes but /why/? |
| 15:54.09 | Katty | did he want to make sure no one was talking about him? |
| 15:54.42 | Naikrovek | i think he enjoyed freaking people out |
| 15:55.11 | Naikrovek | found out later that he would mysteriously know what people were talking about before he would approach them, then act as if he was part of the conversation the whole time |
| 15:55.22 | Katty | that's just....weird. |
| 15:56.01 | Naikrovek | he would read private email between marketing manager and ceo, then act all buddy buddy with them when he'd see them speaking together. he would regurgitate points made in the emails he read |
| 15:56.02 | Katty | ah well |
| 15:56.21 | Naikrovek | as well as the IM conversations he watched |
| 15:56.37 | Naikrovek | dude wasn't obviously creepy, but his actions were despicable |
| 15:57.11 | Katty | probably just really wanted to belong |
| 15:58.20 | p3nguin | Sounds like he had too much time on his hands. |
| 15:58.26 | Naikrovek | if so, his radar dish was bent so bad he couldn't see that people liked him |
| 15:58.42 | Naikrovek | he did belong, more than i ever did actually |
| 15:59.25 | voipmonk | he needs to apply for a CIA job |
| 15:59.34 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 15:59.34 | Katty | there you go |
| 15:59.43 | Katty | he'd probably be very good at that (= |
| 16:00.21 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 16:01.00 | Naikrovek | everyone is really good at something |
| 16:01.04 | Naikrovek | ... or so they say |
| 16:01.17 | Katty | is really good at stuffing her face |
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| 16:02.59 | p3nguin | Do FoP and iSymphony both do pretty much the same things? |
| 16:03.13 | Naikrovek | p3nguin: vaguely |
| 16:03.34 | p3nguin | They are the same type of software, aren't they? |
| 16:03.55 | *** join/#asterisk Pe_Elle (n=jeremy@69-29-65-119.dyn.centurytel.net) |
| 16:04.26 | p3nguin | I'm barely familiar with FoP and I've never even seen iSymphony. |
| 16:04.34 | p3nguin | heard of it, but never used it. |
| 16:04.41 | Pe_Elle | I have a configuration question is the right channel to ask it in? |
| 16:04.57 | p3nguin | pe_elle: If it's about configuration of Asterisk, then probably. |
| 16:05.16 | ManxPower-work | Pe_Elle: If it's about configuration of a GUI for Asterisk, then probably not. |
| 16:05.53 | p3nguin | Ah yes, nice addendum. |
| 16:06.11 | *** join/#asterisk tgunr (n=tgunr@polymicro.net) |
| 16:06.31 | Katty | i would consider isymphony much more user friendly |
| 16:06.41 | Katty | and not nearly as buggy as FoP |
| 16:06.43 | Pe_Elle | sweet. I am getting an error. I searched google, but didn't come up with anything. I am trying to load chan_dahdi. The error I get is: "Signalling requested on channel 3 is FXO Kewlstart but line is in ISDN PRI signalling" |
| 16:07.17 | voipmonk | what sort of lines are you tring to use, Pe_Elle ? |
| 16:08.15 | Pe_Elle | it's a T1 card connected to a adit 600 |
| 16:09.00 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 16:09.49 | *** join/#asterisk holister (n=ryan@static-151-204-189-39.pskn.east.verizon.net) |
| 16:10.07 | holister | my meetme conferences suddenly stopped working...I can't figure out why |
| 16:10.10 | voipmonk | paste your conf file for dahdi at pastie.org or pastebin.ca |
| 16:10.19 | holister | the config file has conf => 900, |
| 16:10.24 | p3nguin | Does iSymphony client have any problems running on 64-bit Linux? |
| 16:10.38 | holister | it was generated by pbx....don't know what it's supposed to look like |
| 16:10.43 | holister | freepbx rather |
| 16:12.13 | LemensTS | shhhh dont tell them that |
| 16:14.22 | Katty | p3nguin: dunno, haven't tried. occasionally there's a guy in here by the name of seanmh |
| 16:14.28 | Katty | p3nguin: ask him. |
| 16:15.07 | eppigy | 8[] |
| 16:15.31 | Katty | eppigy: i just can't decide on lunch |
| 16:16.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.45 | diatonic_ | Naikrovek, I'm using FOP2 and it works pretty well, but web based. It does have user based security that can restrict things like barging calls and seeing voicemail. Not free, but cheap, and works pretty well. |
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| 16:16.46 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 16:20.30 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 16:24.53 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
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| 16:26.12 | diatonic | Eerily quiet in here |
| 16:26.14 | Geminizer | Hello all. Is there a way to pass an extension between contexts? For example, I would like to have a context which prompts the user to enter a number, then goto another contect and repeat the entered number back to the caller |
| 16:26.27 | dlynes | diatonic: even diatonic |
| 16:26.32 | Geminizer | s/contect/context |
| 16:26.32 | dlynes | diatonic: erm catatonic |
| 16:27.13 | dlynes | Geminizer: astdb? |
| 16:27.28 | dlynes | Geminizer: global vars? |
| 16:27.39 | dlynes | Geminizer: sql database calls? |
| 16:27.57 | dlynes | Geminizer: macros? |
| 16:28.07 | eppigy | Katty: our company is providing lunch :D |
| 16:28.10 | dlynes | Geminizer: put on your thinking cap, man :) |
| 16:28.31 | Geminizer | dlynes: thanks.. |
| 16:28.49 | dlynes | Geminizer: many ways to skin a cat in asterisk |
| 16:28.52 | [TK]D-Fender | Geminizer: Why do you need a whole new context jsut to read a number back? |
| 16:29.31 | [TK]D-Fender | Geminizer: thats a single SayDigits() command |
| 16:30.30 | bcrisp | my annual car premium went up to 19,000 |
| 16:30.34 | Geminizer | b/c I am intitially asking a user to enter a number, then press 1 to do one thing with it, or press 2 to do something else with it. I don't want the user to immediately press 1 or 2 before entering the number |
| 16:30.35 | bcrisp | can you believe that? |
| 16:30.43 | Katty | eppigy: yum. save me some. |
| 16:31.13 | bcrisp | is having a panic attack |
| 16:32.03 | *** join/#asterisk inkubus08 (n=inkubus0@174.54.207.45) |
| 16:32.22 | voipmonk | Geminizer: then use "Plaback" which doesnt allow the user to enter a number until the prompt has finished |
| 16:32.27 | voipmonk | Playback, rather |
| 16:32.44 | Katty | bcrisp: breathe deeply |
| 16:33.29 | bcrisp | ty |
| 16:33.41 | Geminizer | voipmonk: That I agree with, and Playback is being used. However, the options for pressing 1 or 2 shouldn't be made available under after the user has entered a valid number |
| 16:34.00 | Katty | bcrisp: what causes your attacks? |
| 16:34.16 | dlynes | bcrisp: 19K? holy shizzit |
| 16:34.17 | Geminizer | I shouldn't be putting extension entries for 1,2, and _NXXNXXXX all in the same context |
| 16:34.32 | dlynes | bcrisp: How many accidents, speeding tickets, and DUIs do you have? |
| 16:34.34 | Geminizer | because the caller could immediately press 1 without entering a number |
| 16:34.34 | bcrisp | katty: a letter from my car insurance saying that my annual premium was shooting up 400% |
| 16:34.47 | Geminizer | a valid phone number that is |
| 16:34.50 | bcrisp | dlynes: 0,0, and 0 |
| 16:34.50 | Katty | bcrisp: sounds like it's time to visit a new insurance agency |
| 16:35.09 | dlynes | bcrisp: so why is it so high? Did the insurance company give you a reason? |
| 16:35.11 | bcrisp | the other person on my policy had some major traffic issues that i was unaware of... |
| 16:35.14 | coppice | if they're shooting up I guess they'll need the cash |
| 16:35.28 | dlynes | bcrisp: so take them off the policy, then |
| 16:35.30 | Katty | bcrisp: oh. well that's easy to fix. take them off your policy ;) |
| 16:35.34 | bcrisp | dlynes: yep |
| 16:35.44 | dlynes | bcrisp: that's what my wife did with me |
| 16:35.52 | Katty | bcrisp: and if they ask why, just tell them you still love them but you can't afford it! |
| 16:35.57 | dlynes | bcrisp: I had two accidents and three speeding tickets |
| 16:36.01 | bcrisp | also found out that my prior mortgage co (that went bankrupt) didn't collect sufficient escrow funds, now i owe several thousand |
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| 16:36.16 | Katty | bcrisp: these things happen. |
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| 16:36.28 | Katty | bcrisp: you will get there, it will just take some time. |
| 16:36.33 | dlynes | bcrisp: wtf? How does that have anything to do with you? |
| 16:36.55 | bcrisp | dlynes: they said that the prior company paid but wasnt collecting escrow from me.. |
| 16:37.00 | *** join/#asterisk etfonhomey1 (n=etfonhom@office.archvision.com) |
| 16:37.14 | bcrisp | (i was paying.. but they were shutdown by the FBI due to bad book keeping, etc) |
| 16:37.15 | dlynes | bcrisp: dood...the US is so severely fubar |
| 16:37.46 | dlynes | bcrisp: I'd be taking the bastards to court if that happened here |
| 16:37.50 | dlynes | bcrisp: and i'd probably win |
| 16:37.51 | bcrisp | im not gonna whine any more in here |
| 16:37.58 | Katty | bcrisp: oh that's okay. |
| 16:38.02 | Katty | bcrisp: we all whine (= |
| 16:38.04 | bcrisp | my head is going to explode |
| 16:38.06 | Katty | bcrisp: especially me ;P |
| 16:38.17 | Katty | bcrisp: oh it's not going to explode.... |
| 16:38.29 | Katty | bcrisp: one day at a time deary, one day at a time. |
| 16:38.34 | bcrisp | was also robbed and vandalized |
| 16:38.50 | bcrisp | is full of complaints, i apologize |
| 16:38.53 | dlynes | bcrisp: dood...do you live in the ghetto, or something? |
| 16:39.03 | bcrisp | dlynes: not until people stopped paying their mortgages |
| 16:39.13 | Katty | that's our problem too |
| 16:39.15 | bcrisp | dlynes: its my reward for paying |
| 16:39.22 | Katty | all the houses around us went up for sale, but they didn't sale, so they went up for rent |
| 16:39.32 | Katty | err didnt' sell |
| 16:39.37 | dlynes | bcrisp: ah...one of those things |
| 16:39.42 | dlynes | bcrisp: that's gotta suck |
| 16:39.45 | bcrisp | now they are section 8-ing houses that were selling for 300,000 in '05 |
| 16:39.51 | Katty | hopefully we'll find a house and get out of city limits soon (= |
| 16:39.58 | bcrisp | so mine dropped by 70% |
| 16:40.13 | dlynes | bcrisp: and we thought our 15% drop was bad |
| 16:40.23 | dlynes | bcrisp: mind you, most of that 15% drop has been recouped now |
| 16:40.26 | bcrisp | im now upside down 65% lol |
| 16:40.28 | dlynes | bcrisp: everyone's condo crazy again |
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| 16:40.39 | ManxPower-work | I paid off my cabin in 2 yrs. Don't hate me. |
| 16:40.56 | dlynes | bcrisp: our condo's in Vancouver, though...2010 olympics is making all the speculators go nuts |
| 16:41.20 | bcrisp | we had the superbowl 2 miles from my house last year |
| 16:41.31 | dlynes | bcrisp: and it still didn't help? |
| 16:41.38 | bcrisp | not with all the foreclosures |
| 16:41.45 | bcrisp | there are over 100,000 vacant houses in our city |
| 16:41.46 | dlynes | bcrisp: we were kinda hoping the superbowl coming to indianapolis was going to help our real estate there |
| 16:41.53 | dlynes | bcrisp: so I'm guessing it probably won't |
| 16:42.20 | dlynes | bcrisp: Our house went from about $90K in value to $40K in value almost overnight |
| 16:42.30 | dlynes | bcrisp: luckily we didn't have a mortgage |
| 16:42.53 | bcrisp | dlynes: i suppose if i can survive without going crazy now then ill be ok |
| 16:43.12 | dlynes | bcrisp: what city are you in that's like that, if you don't mind? |
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| 16:43.20 | bcrisp | phoenix, az |
| 16:43.23 | dlynes | ah |
| 16:43.30 | dlynes | yeah...phoenix is quite famous nowadays |
| 16:43.42 | bcrisp | all the damn californian investors |
| 16:43.47 | bcrisp | drove prices up then went bankrupt |
| 16:43.48 | dlynes | Canadians are going down there like crazy right now |
| 16:43.59 | dlynes | Picking up all the condos for pennies on the dollar |
| 16:44.19 | bcrisp | you know my property taxes only dropped by $1.02 |
| 16:44.34 | p3nguin | If only I had a few billion pennies... |
| 16:45.02 | dlynes | bcrisp: yeah...I feel for you...with all the shit that went down in Indianapolis (major foreclosures...still a huge tide going through), our taxes went up, even though our value went down by 60% |
| 16:45.18 | dlynes | bcrisp: Major corruption in Marion county government, I think |
| 16:45.47 | bcrisp | its sickening when you think of all the people charging crap like new cars, TVs etc, then just not paying mortgage and getting bailed out heh |
| 16:45.49 | dlynes | Tons of Canadians going down to Vegas right now, buying up condos as well |
| 16:46.09 | dlynes | Major rental market in both Phoenix and Vegas |
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| 16:55.28 | Katty | goes to the store |
| 16:57.19 | telnettech | i need a 12pk and bread please :) |
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| 16:59.00 | mcrownover | hi everyone |
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| 17:00.04 | Geminizer | if using astdb, are stored variables saved anywhere such that I can access them outside of the dialplan? |
| 17:01.21 | ManxPower-work | Geminizer: AstDB is really a BerkleyDB |
| 17:01.51 | ManxPower-work | I don't know anything about locking/etc for multiple processes accessing the same DB |
| 17:03.09 | *** join/#asterisk fainsys (n=fainsys@74.7.247.152) |
| 17:06.39 | TSM2 | If the PSTN is digital ISDN, why would i hear hiss even when doing a loopback call, is this some sort of VAD or comfort noise they add |
| 17:07.44 | voipmonk | Geminizer: just us mysql |
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| 17:09.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 17:10.01 | Geminizer | voipmonk: can asterisk dialplans interface with mysql, or can astdb somehow be managed via mysql, or both? |
| 17:11.57 | p3nguin | astdb data can be read in the asterisk CLI, so surely there is some way for you to get data outside of a dialplan. |
| 17:12.46 | [TK]D-Fender | Geminizer: Before you ask too many details about a tool, perhaps it'd be a good idea what you expect to USE IT FOR. |
| 17:12.56 | *** join/#asterisk d00gster (n=doughant@94.96.110.232) |
| 17:13.43 | Geminizer | I'm just interested in reading from it... |
| 17:14.04 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 17:14.29 | voipmonk | Geminizer you can interface with mysql via the asterisk dialplan to input and retrieve information , what are you looking to DO , spell it all out - give me the 50,000 ft view. :) |
| 17:14.54 | voipmonk | give US the view, cuz I may need to leave here shortly.... I smell cookies |
| 17:15.19 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 17:15.22 | *** join/#asterisk pacmanfan (n=andrewk@75-121-116-246.dyn.centurytel.net) |
| 17:16.04 | Geminizer | in simple terms... the dialplan I want to use asks a caller for a number. The number the caller specifies gets assigned to a global variable. I would like to use ASTDB to get the value of this variable |
| 17:16.17 | TSM2 | any ideas how on the VoiceMail app, how i could have a little script that stops users logging in if they still have their password set to default, would that have to be done within the aplication itself, as it seems everything is done though VoiceMailMain |
| 17:16.45 | [TK]D-Fender | Geminizer: Continue... so far nothing happens with this input.. |
| 17:17.33 | [TK]D-Fender | TSM2: * knows of no such"default" |
| 17:17.38 | Geminizer | all I really need is the value assigned to the global variable (work with it outside of asterisk) |
| 17:18.07 | Geminizer | from that point on, I could to whatever with that content |
| 17:18.13 | Geminizer | "could do" |
| 17:18.30 | TSM2 | [TK]D-Fender: ok if i knew what the default was an i set it, but im thinking that all of the VM auth etc is done within the VM app and not within dialplan |
| 17:19.05 | Geminizer | or is there a way to invoke applications from a diaplan (e.g. exten => 1,n,invoke(mysql, "insert into....")) |
| 17:19.59 | Geminizer | my main concern comes down to: how can asterisk take what the caller specifies and dump that information to something that can be accessed externally? |
| 17:21.00 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 17:21.18 | Gido-E | Geminizer you can excute bash scripts :-) |
| 17:21.37 | Gido-E | Geminizer login to the CLI, core show application system |
| 17:22.00 | bcrisp | ~book |
| 17:22.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:22.43 | bcrisp | Geminizer you can use AGI scripts or execute bash |
| 17:23.00 | bcrisp | AGI scripts allow you to gather info from asterisk in the lang you prefer |
| 17:23.05 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:23.26 | bcrisp | you can also use the asterisk manager interface (AMI) to remotely connect in and interact with * |
| 17:23.40 | Gido-E | bcrisp also a good evening. It is already dark here, and the time is ~ 18:30 |
| 17:24.05 | bcrisp | :/ |
| 17:24.14 | Geminizer | bcrisp -- exactly the info I needed. thanks |
| 17:24.27 | bcrisp | Geminizer, np |
| 17:26.16 | *** join/#asterisk RypPn2 (i=TuMbL@rosscom.co.uk) |
| 17:26.58 | *** join/#asterisk odie_flocon (n=Odie@dsl-vlan426-66-18-230-102.nucleus.com) |
| 17:27.30 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 17:28.16 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
| 17:29.07 | CGMChris | Can anyone confirm that running DeadAGI on a live channel *can potentially* result in a defunct/zombie process and a hung SIP channel? |
| 17:31.52 | odie_flocon | how would you recreate that cgmchris |
| 17:32.41 | CGMChris | odie_flocon: Reproduce a SIP channel hitting RTPTimeout, at which point the "live" channel is disconnected..and have DeadAGI run as the next step in the dialplan. |
| 17:32.43 | p3nguin | tsm2: The forcename and forcegreeting voicemail options check to see if the user's password is the same as the mailbox number... perhaps you can read the code that makes those work, and you can create something of your own based on it. |
| 17:33.43 | p3nguin | forcegreetings, actually. |
| 17:34.26 | odie_flocon | so what do you need confirmed? can you not reproduce it? |
| 17:34.34 | CGMChris | odie_flocon: For me, the sip channel hangs indefinately and loses communication with the child AGI process, which becomes defunct/zombized. Obviously I shouldnt be running DeadAGI, and was only doing so by mistake, so the fix for me is to use AGI. |
| 17:34.45 | *** join/#asterisk axelilly (n=jfenner@66.181.75.57) |
| 17:34.56 | TSM2 | p3nguin: ile have a look thanks |
| 17:35.14 | CGMChris | odie_flocon: I was just looking for verification, if someone else had run into the same issue. |
| 17:35.47 | odie_flocon | ok. I will probably be able to try it in a week or so. |
| 17:36.06 | axelilly | I am trying to set up a call queue and my phones are connected to asterisk using the chan_sccp-b module. The agent phones can log into the queue, but they are always marked as invalid. Any one have a hint for me as to how to get them valid? |
| 17:36.59 | bcrisp | axelilly, you can pastebin your CLI output |
| 17:37.13 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 17:39.07 | TSM2 | p3nguin: ive just looked, does not seem to force user to change password, only if password matches mailbox then it checks to see if name has been recorded and if not asks user to record it |
| 17:39.18 | TSM2 | p3nguin: looks like it would need a code change |
| 17:39.20 | p3nguin | That's what I said. |
| 17:39.23 | axelilly | bcrisp: here is the postebin of the CLI output http://fpaste.org/IZej/ |
| 17:39.36 | p3nguin | tsm2: err, wait... |
| 17:40.25 | p3nguin | tsm2: I screwed up. Those settings force the user to do something if the password is the same as the mailbox number. Perhaps you can use that code to create something of your own. |
| 17:40.43 | p3nguin | tsm2: I mistyped what I meant originally. |
| 17:41.06 | p3nguin | or did I? Hell, I don't know. |
| 17:41.08 | TSM2 | p3nguin: thats what i thought, ile have to modify the app and compile it, a job for next week |
| 17:41.10 | grandpapadot | Hey guys, I'm trying to find some info on the reason there are two 1.6 branches, any pointers would be great! Thanks... |
| 17:41.53 | p3nguin | Yeah, I think that's what I said the first time, but now you've made me think I said something else. |
| 17:42.26 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:42.32 | bcrisp | axelilly: try looking at the queues.conf.sample in your source |
| 17:42.44 | *** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath) |
| 17:42.49 | bcrisp | axelilly: im not positive, but i think you may need to preload some modules |
| 17:43.01 | p3nguin | tsm2: Anyway, if you are good with source code, you might be able to develop something based on the idea of those two options. |
| 17:43.30 | axelilly | bcrisp: oh man, I just figured it out I think |
| 17:43.37 | sbrath | Any tips on asterisk voice prompts? When I call into a Bridge on my Asterisk or the Directory, the voice is a bit crackley? |
| 17:44.00 | bcrisp | sbrath: what codecs? |
| 17:44.10 | sbrath | Where do I look? |
| 17:44.24 | bcrisp | start CLI, then call into the menu |
| 17:44.31 | bcrisp | or trigger some sound prompt |
| 17:45.09 | bcrisp | also, what type of channels are you using for the calls? |
| 17:45.27 | grandpapadot | Also, if you were going to deploy a new asterisk server, and reliablity is most important, which branch/version would you deploy? Any feedback would be welcome. Thanks! |
| 17:45.41 | sbrath | I have a merlin legend -> asterisk via a T1/PRI flip cable |
| 17:45.54 | p3nguin | grandpapadot: I'm satisfied with 1.4 |
| 17:46.09 | grandpapadot | p3nguin: Which revision? (thanks) |
| 17:46.20 | p3nguin | I use 1.4.24.1 |
| 17:46.39 | bcrisp | sbrath, are the prompts always crackily regardless? have you tried SIP internal calls? |
| 17:46.47 | p3nguin | grandpapadot: I would go up to the current 1.4 release if I were you, though. |
| 17:46.56 | p3nguin | I'll get around to upgrading my systems eventually. |
| 17:47.02 | *** join/#asterisk mcrownover (n=markcrow@remote.gawest.com) |
| 17:47.39 | sbrath | Even from sip, the Voice sounds "muffled" and "quiet" |
| 17:47.43 | grandpapadot | Yea, that's what were running now, and it's pretty stable, but running on debian 4. We're deploying a bunch of new servers, debian 5, and I was hoping that I could use this opportunity to move to 1.6.? but wanted to solicit feedback. |
| 17:47.56 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 17:48.18 | bcrisp | im on 1.6.1.11 and its working great .. but my setup is not very complicated |
| 17:49.05 | bcrisp | sbrath, is your CPU getting swamped? do you have other software besides OS / * installed? |
| 17:49.13 | p3nguin | grandpapadot: What do you expect to gain from going to 1.6? Did you read the changelogs? |
| 17:49.17 | grandpapadot | bcrisp: Thanks. |
| 17:49.48 | grandpapadot | p3nguin: Just browsed, haven't dug that deep yet. I was just expecting to get current, actually, if it was stable. Seems the biggest change will be zaptel->dahdi. |
| 17:50.18 | voipmonk | if it aint broke |
| 17:50.25 | voipmonk | ur about to break it |
| 17:50.29 | sbrath | bcrisp: not really, 2G ram, and load average is 0.00 |
| 17:50.30 | bcrisp | if it aint broke, tax it more |
| 17:50.55 | bcrisp | sbrath: were you able to tell what format of audio is playing for the sound prompts? |
| 17:51.00 | *** join/#asterisk DennisG (n=Gebruike@84.30.136.208) |
| 17:51.19 | bcrisp | also.. are they standard prompts, or you recorded them yourself? |
| 17:51.21 | sbrath | bcrisp: no, I didn't see anything when I was in the cli, do I need to turn on more debugging? |
| 17:51.29 | sbrath | These are the standard prompts. |
| 17:51.42 | bcrisp | sbrath: i think so, maybe up the verbosity |
| 17:51.59 | bcrisp | asterisk -rvvvvvvvv |
| 17:52.09 | p3nguin | core set verbose 10 |
| 17:52.16 | bcrisp | ty p3nguin |
| 17:52.45 | p3nguin | If you already have CLI open, there's no reason to stop it and run it again just to add a bunch of Vs. |
| 17:52.53 | bcrisp | sbrath: you might also nav to /var/lib/asterisk/sounds/en (if thats where the sound files are) and try playing them outside of * itself to judge the quality |
| 17:53.44 | bcrisp | the only other thing i can think of is bandwidth problems |
| 17:53.48 | bcrisp | but im a total newb |
| 17:53.54 | p3nguin | "core set verbose 2147483647" if you want to max it out. :P |
| 17:54.11 | bcrisp | CLI: [NOTICE]: User is going crazy with verbosity |
| 17:54.15 | voipmonk | ... |
| 17:54.28 | grandpapadot | The new 1.6 faxdetect, is it based on nv_detect or something new? Anyone know? |
| 17:54.32 | sbrath | crisp: <DAHDI/23-1> Playing 'dir-usingkeypad.gsm' (language 'en') |
| 17:54.36 | sbrath | That's all I get. |
| 17:54.56 | bcrisp | ok sbrath, go play that sound file manually and see how it sounds to u |
| 17:57.37 | axelilly | when using the Queue app, when all agents are busy, incoming calls should get MOH instead of ringing right? |
| 17:57.53 | [TK]D-Fender | axelilly: Depends what you configured |
| 17:58.13 | axelilly | [TK]D-Fender: is that configured in queues.conf? |
| 17:58.32 | [TK]D-Fender | axelilly: And in your dialplan |
| 17:58.52 | LemensTS | If you turn on qualify= in sip.conf for a device, Asterisk will |
| 17:58.52 | LemensTS | send an OPTIONS request every minute to the device and check |
| 17:58.52 | LemensTS | if it replies. Each OPTIONS request is retransmitted a number |
| 17:58.52 | LemensTS | of times (to handle packet loss) and if we get no reply, the |
| 17:58.52 | LemensTS | device is considered unreachable. From that moment, we will |
| 17:58.53 | LemensTS | send a new OPTIONS request (with retransmits) every tenth |
| 17:58.55 | LemensTS | second. |
| 17:58.56 | Katty | one of my polycoms says 18/12/2009 rather than 12/18/2009... anyone know where the setting to change that is? |
| 17:58.57 | LemensTS | whoops sorry.. |
| 17:59.10 | axelilly | [TK]D-Fender: to play MOH, in queues.conf, do I just need to define musicclass? |
| 17:59.20 | Katty | nevermindi found it |
| 17:59.37 | [TK]D-Fender | axelilly: You need to not be telling it to ring, etc |
| 18:00.07 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:00.19 | axelilly | [TK]D-Fender: do you know what directive in queue.conf tells it to ring? |
| 18:00.39 | [TK]D-Fender | axelilly: Extensions.conf <- |
| 18:00.57 | [TK]D-Fender | axelilly: Start by reading the apps INSTRUCTIONS, and the sample configs |
| 18:01.08 | [TK]D-Fender | axelilly: "core show application queue" |
| 18:01.20 | *** join/#asterisk roguefalcon (n=rogue@75-151-54-65-Nashville.hfc.comcastbusiness.net) |
| 18:01.52 | *** join/#asterisk wolfeyes (i=29117b99@gateway/web/freenode/x-lyyknxcvbinqbafw) |
| 18:02.20 | *** part/#asterisk wolfeyes (i=29117b99@gateway/web/freenode/x-lyyknxcvbinqbafw) |
| 18:03.11 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 18:03.32 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 18:04.01 | Naikrovek | anyone in here have an IBM hardware fail under support contract? |
| 18:04.10 | Naikrovek | if so, how long did it take for ibm to repair |
| 18:04.32 | [TK]D-Fender | Naikrovek: No, they stopped printing it on paper so heavy as to risk damaging the unit should you place it on top. |
| 18:04.40 | [TK]D-Fender | Naikrovek: Far more environmentally friendly as well. |
| 18:04.42 | Naikrovek | yuk yuk |
| 18:04.44 | Naikrovek | hehe |
| 18:05.04 | grandpapadot | lol (took me a minute) |
| 18:05.26 | bcrisp | axelilly, the r option in Queue() will ring instead of moh |
| 18:05.35 | grandpapadot | Hey TK, still running 1.4? Moved to 1.6 yet? |
| 18:06.34 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 18:06.42 | angryuser | Naikrovek, it depends on your support contract |
| 18:06.53 | Naikrovek | well i'm talking 24x7 |
| 18:07.04 | Naikrovek | 4 hour response |
| 18:07.05 | Naikrovek | max |
| 18:07.05 | angryuser | Naikrovek, 24/7 + what ? |
| 18:07.12 | angryuser | so it will 4 hours |
| 18:07.18 | angryuser | be* |
| 18:07.31 | coppice | they call it a support contract because it contracts when you need it most |
| 18:07.35 | Naikrovek | 4 hours to response though, right? not 4 hours to the fix |
| 18:07.44 | Naikrovek | or do they just replace what's busted |
| 18:08.09 | angryuser | Naikrovek, 4 hours to fix hardware, if all your hdd are dead, they are replaced, the date is your problem |
| 18:08.10 | *** join/#asterisk BrianR___ (i=brianr@osiris.978.org) |
| 18:08.17 | sbrath | bcrisp: I played the gsm files on windows and they sound quite good. |
| 18:08.19 | angryuser | data* |
| 18:08.28 | ruben23 | hi guys, anyone have idea on this error log on my asterisk server---->http://pastebin.com/m166e85c4 |
| 18:08.43 | Naikrovek | nice |
| 18:08.44 | Naikrovek | okay |
| 18:08.48 | Naikrovek | angryuser: tyvm |
| 18:08.57 | bcrisp | sbrath: are you using a softphone to test sip internal calls? |
| 18:09.17 | Katty | eppigy: what did you have for lunchles. |
| 18:09.24 | sbrath | I was using a softphone to test internal, I don't have a hard sip phone configured yet... Configuring the 7960 is next. |
| 18:09.28 | [TK]D-Fender | grandpapadot: FFS get off 1.4 |
| 18:09.37 | bcrisp | sbrath: which softphone? |
| 18:09.39 | hardwire | rawr? |
| 18:09.50 | sbrath | Zopier communicator |
| 18:10.01 | bcrisp | ok, in zoiper, look at the configured codecs |
| 18:10.13 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 18:10.18 | bcrisp | diable all but GSM to test |
| 18:10.23 | ruben23 | http://pastebin.com/m166e85c4 this keeps getting out on my asterisk CLI ogs.. |
| 18:10.24 | bcrisp | s/diable/disable/ |
| 18:11.43 | sbrath | did it, still sounds all nazel and crackley |
| 18:11.43 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 18:11.43 | bcrisp | p3nguin: any ideas for sbrath? |
| 18:11.58 | Katty | recommend musinex |
| 18:12.01 | p3nguin | I wasn't paying attention, so give me a minute to read it. |
| 18:12.06 | dlynes | is with Katty. |
| 18:12.13 | ManxPower-work | sbrath: try using ulaw. In some situations on some versoions of Asterisk with some compilers, GSM sounds likke crap |
| 18:12.20 | dlynes | Or even Nasonex works, too :0 |
| 18:12.37 | Katty | dlynes: yeah but that can be damaging to the nasal cavity |
| 18:12.58 | dlynes | Katty: it can? |
| 18:13.06 | ruben23 | :-( |
| 18:13.11 | dlynes | Katty: wonder why my doctor prescribed it for hay fever, then |
| 18:13.15 | sbrath | for the record, a few days ago I was trying to get my TE110P to talk to a Merlin using AsteriskNOW... And it wouldn't bring the channels up.... I figured out the problem... AsteriskNow.. I installed all the latest modules and it started talking to Merlin from the get go :) |
| 18:13.33 | sbrath | How do I change the default playbak codec in the config to be ulaw? |
| 18:13.40 | Katty | dlynes: it can cause nose bleeds, amongst other things |
| 18:13.41 | bcrisp | this is strange, im testing the ability to forward SIP calls to cell #s with the options in place for the callee to transfer calls. it works in some cellphones and not others |
| 18:13.46 | dlynes | Katty: oh |
| 18:13.48 | dlynes | Katty: nasty |
| 18:14.06 | dlynes | Katty: and musinex does the same thing, but without the side effects? |
| 18:14.07 | Katty | dlynes: along with sinus infections... |
| 18:14.14 | Katty | dlynes: musinex isn't a nasal spray |
| 18:14.25 | Katty | dlynes: your nose isn't designed to have things sprayed into it |
| 18:14.28 | dlynes | Katty: but it clears up hay fever congestion? |
| 18:14.44 | Katty | dlynes: nasonex is an allergy medication |
| 18:14.52 | Katty | dlynes: so if you have allergies which cause a hay fever, then sure |
| 18:15.10 | ruben23 | <PROTECTED> |
| 18:15.16 | dlynes | Katty: so what's musinex then? I thought because of the similarity in names, it was for the same contraindication |
| 18:15.51 | Katty | dlynes: mucinex is an expectorant |
| 18:16.02 | dlynes | ruben23: perhaps you were writing to a network drive that has aborted the mount? |
| 18:16.13 | sbrath | Even changing the Zopier to u-law is still nazel and crackley.... ? |
| 18:16.14 | Katty | dlynes: it breaks up mucus and makes you cough it all up >.< |
| 18:16.22 | dlynes | Katty: ah...like Dr Buckley's |
| 18:16.25 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.25 | Katty | dlynes: including nasal congestion |
| 18:16.40 | Katty | dlynes: but it's not meant to be For Allergies |
| 18:16.46 | dlynes | Katty: tastes like shit, but works great |
| 18:16.57 | Katty | dlynes: if you have allergy problems you should probably find an antihistamine |
| 18:17.21 | dlynes | Katty: yeah...I don't like antihistamines, because of all the steroids in them |
| 18:17.30 | Katty | nods |
| 18:17.35 | Katty | i can certainly understand that |
| 18:17.35 | ManxPower-work | sbrath: With softphones you depend on the OS for way too much with regards to audio. |
| 18:17.43 | Katty | but people take psuedophedrine for allergies |
| 18:17.44 | p3nguin | Just because a phone/device only uses ulaw, does that make asterisk play only .ulaw files to the phone/device? I expect it would transcode. |
| 18:17.46 | ruben23 | dlynes: thanks for the reply..im using lvm on my disk..what might i do to trace and correct.. |
| 18:17.47 | Katty | and it's not for allergies.... |
| 18:17.53 | Katty | it just...dries you up (= |
| 18:18.12 | ManxPower-work | p3nguin: yes it will (normally) transcode |
| 18:18.43 | [TK]D-Fender | [13:17]<Katty>but people take psuedophedrine for allergies <- BOOYEAH |
| 18:18.44 | dlynes | Katty: yeah...and all the meth freaks use it to make meth |
| 18:18.45 | sbrath | But even if I call from a hardphone going over the PRI into the asterisk, it still sounds about the same as the soft phone, maybe just a bit quieter. |
| 18:18.54 | [TK]D-Fender | Katty: 1 of the 2 only drugs I take. |
| 18:19.12 | Katty | [TK]D-Fender: affedrin(sp)? is very handy for a lot of things |
| 18:19.21 | [TK]D-Fender | Katty: Don't fall for that phenylephidrine crap |
| 18:19.27 | p3nguin | manxpower-work: Your answer was contradicting. |
| 18:19.36 | Katty | [TK]D-Fender: it's in the same family, but not very potent |
| 18:19.54 | odie_flocon | later all |
| 18:19.55 | ManxPower-work | p3nguin: how so? |
| 18:19.57 | Katty | course neither is psuedophedrine compared to affedra |
| 18:20.11 | Katty | s/affedra/ephedra/ |
| 18:20.12 | [TK]D-Fender | Katty: s/not very potent/dismissed as placebo by medical community/ |
| 18:20.28 | p3nguin | manxpower-work: I expect * to use a gsm file and transcode it when playing to a ulaw device. The question was whether * would play a .ulaw file when playing to a ulaw-only device. |
| 18:20.56 | Katty | dlynes: but regardless, if you start getting nose bleeds, or if your nose starts itching a lot, i would dicontinue immediately. |
| 18:21.00 | ManxPower-work | p3nguin: Asterisk will not transcode if it does not need to. |
| 18:21.10 | p3nguin | tests |
| 18:21.28 | p3nguin | I have a ulaw-only phone, so I'll see if it plays .ulaw files. |
| 18:21.32 | mcrownover | would anyone have a good resource to point me to for setting up MOH? I |
| 18:21.39 | Katty | i think ephedra was commonly used in weight loss pills |
| 18:21.47 | mcrownover | i'm taking over our asterisk box and am relatively new at this |
| 18:22.04 | ManxPower-work | p3nguin: "core show translations" will give you a general idea of the CPU needs for the various transcoding |
| 18:22.09 | Katty | ephedra is a stimulant so it probably increases metabolism or something |
| 18:22.36 | p3nguin | When a sound file is played, the verbosity does not include the file's extension, so I'm not sure how I'm going to determine. |
| 18:22.52 | bcrisp | hmm mine does |
| 18:22.52 | dlynes | mcrownover: google: "site:voip-info.org moh" |
| 18:23.06 | p3nguin | "Playing 'transfer' (language 'en')" |
| 18:23.15 | p3nguin | This doesn't say .gsm, .ulaw, or anything. |
| 18:23.21 | bcrisp | Playing 'pbx-transfer.ulaw' (language 'en') |
| 18:23.24 | dlynes | mcrownover: keep in mind some of that might be out of date, but it's a good starting point |
| 18:23.57 | dlynes | mcrownover: there might also be something in your asterisk source distribution in the directory 'doc' |
| 18:24.14 | mcrownover | we do have mpg123 on the box but the other admin had it working with an internet music stream that has since died |
| 18:24.31 | dlynes | mcrownover: anything in the 'doc' directory should be up-to-date |
| 18:24.34 | mcrownover | so i need to replace it and would prefer to just do mp3's on our box |
| 18:24.51 | x86 | anyone make calls out to sipbroker? |
| 18:24.53 | p3nguin | bcrisp: Are you simply using Playback(pbx-transfer) in the dialplan to make that play? |
| 18:24.54 | dlynes | mcrownover: you don't need mpg123....just install asterisk add-ons |
| 18:25.02 | bcrisp | p3nguin no |
| 18:25.03 | dlynes | mcrownover: it has res_mp3.so |
| 18:25.17 | dlynes | mcrownover: erm format_mp3.so I mean |
| 18:25.19 | grandpapadot | Is dahdi required in 1.6 for MeetMe? |
| 18:25.20 | mcrownover | do the addons work with freepbx? |
| 18:25.21 | bcrisp | i was testing transfers and saw that in my CLI |
| 18:25.24 | x86 | I'm getting "513" SIP error codes when I Dial(SIP/anything@sipbroker.com) |
| 18:25.28 | [TK]D-Fender | grandpapadot: Yes |
| 18:25.35 | grandpapadot | Thanks, TK. |
| 18:25.35 | dlynes | mcrownover: also, voip-info.org has plenty of info on how to set up streaming moh |
| 18:25.49 | p3nguin | bcrisp: But transferring doesn't cause a sound to play, so something in the dialplan make it play -- probably Playback(). |
| 18:25.56 | sbrath | ManxPower-work: my times for translations are 2ms and less |
| 18:26.02 | mcrownover | dylnes: thanks for the help |
| 18:26.20 | bcrisp | p3nguin: oh... im testing by dialing # from the callee |
| 18:26.36 | bcrisp | and it says "Transfer" |
| 18:26.49 | p3nguin | Oh, I hate DTMF transfers. |
| 18:26.55 | bcrisp | hehe |
| 18:27.04 | [TK]D-Fender | SIP bcrisp What phones? |
| 18:27.09 | [TK]D-Fender | bcrisp: What phones? |
| 18:27.16 | hardwire | p3nguin: if i could be without inband signalling.. life would be amazing |
| 18:27.24 | bcrisp | [TK]D-Fender, im sorry you've lost me |
| 18:27.55 | bcrisp | [TK]D-Fender, im attempting dtmf transfers from a cell phone callee |
| 18:27.59 | [TK]D-Fender | bcrisp: What model of phones are you testing transfers with? |
| 18:28.13 | bcrisp | oh.. iphone, HTC, droid |
| 18:28.25 | [TK]D-Fender | bcrisp: Softphone clients then... |
| 18:28.30 | bcrisp | zoiper |
| 18:28.32 | bcrisp | xlite |
| 18:28.37 | [TK]D-Fender | bcrisp: Ok... |
| 18:28.52 | bcrisp | the transfer works ok from the iphone but not from the droid phone apparently |
| 18:28.56 | [TK]D-Fender | bcrisp: And phone that offers native transfer is preferrable to use their native functionality |
| 18:29.11 | bcrisp | i see |
| 18:29.12 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 18:30.17 | bcrisp | softphone reaches outside cell # (screening agent on the road). From there we attempt an attended transfer to an internal sip device |
| 18:30.29 | bcrisp | first by setting the TRANSFER_CONTEXT variable to the proper context |
| 18:30.57 | bcrisp | and use the ,kt options on the Dial() |
| 18:31.19 | bcrisp | default # for blind transfer as per features.conf |
| 18:31.35 | *** join/#asterisk sebbl (n=Momofu@109.192.162.148) |
| 18:33.30 | p3nguin | bcrisp: I tried it the way you were doing it... Playing 'pbx-transfer' (language 'en') |
| 18:33.43 | bcrisp | you mean via dtmf? |
| 18:33.44 | p3nguin | It just does not show the filename extension. |
| 18:33.47 | p3nguin | yes |
| 18:34.00 | bcrisp | maybe its a change in 1.6? |
| 18:34.25 | sbrath | p3nguin: What version of Asterisk do you not see the extension on? |
| 18:34.34 | p3nguin | My cordless phone doesn't have a transfer button, so I called it and used # to make it say "transfer," which is what I think you were doing. |
| 18:34.43 | p3nguin | sbrath: 1.4.24.1 |
| 18:34.44 | bcrisp | p3nguin, yep exactly |
| 18:35.28 | sbrath | and I'm doing it on 1.6 and seeing it, probably a version thing. Try removing the directory for the files you don't want to play and see ifyou get an error :) |
| 18:36.21 | p3nguin | There's no directory of files I don't want to play. |
| 18:36.38 | p3nguin | All of the sound files are in the asterisk/sounds directory. |
| 18:37.25 | p3nguin | I was just trying to determine if, as a fact, asterisk would play file.ulaw to a ulaw-only phone. |
| 18:38.17 | bcrisp | p3nguin, can you rename your non-ulaw sounds? |
| 18:38.34 | ManxPower-work | p3nguin: Transfer = Flash / Recall on Analog pgones |
| 18:41.09 | p3nguin | bcrisp: I could move them, but that sould skew the results, since the non-ulaw files wouldn't even be present. |
| 18:41.12 | *** join/#asterisk joako_ (n=ston3d@opensuse/member/joak0) |
| 18:41.28 | bcrisp | right |
| 18:41.52 | bcrisp | you could see if it was trying to play other formats first tho right? |
| 18:42.05 | bcrisp | thinks it defaults to highest allowed |
| 18:43.15 | p3nguin | Heck, I don't know... I can't do too much work right now, as I am elbow-deep in BBQ ribs. |
| 18:43.16 | *** join/#asterisk stegbth (n=stegbth@p4FD0967B.dip0.t-ipconnect.de) |
| 18:43.24 | stegbth | hello everybody |
| 18:44.01 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 18:44.03 | stegbth | since about two weeks i have the problem, one of my extension, doesn't count to 20s |
| 18:44.06 | *** join/#asterisk ayrjola (n=ayrjola@a88-115-162-170.elisa-laajakaista.fi) |
| 18:44.06 | stegbth | it times out |
| 18:44.21 | stegbth | i set already autofallthrough=no |
| 18:44.31 | stegbth | then i get Timeout, but no rule 't' in context 'capi-in' |
| 18:46.35 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 18:46.59 | dlynes | stegbth: exten => t,1,Hangup |
| 18:48.07 | dlynes | stegbth: what do you mean it doesn't count to 20s? you mean exten => 1,1,Dial(SIP/peername,20)? |
| 18:48.24 | dlynes | stegbth: or do you mean exten => 1,1,Set(TIMEOUT(response)=30)? |
| 18:48.35 | bcrisp | i think he set a timeout but not a timeout handler |
| 18:49.35 | stegbth | ah, sometime i have to speak with somebody ;) |
| 18:49.50 | bcrisp | huh? |
| 18:49.52 | stegbth | found the error, was typo in the extension a 8 instead of 5 |
| 18:49.59 | bcrisp | heh |
| 18:50.05 | stegbth | so my extension stopped working |
| 18:50.08 | bcrisp | 5 != 8??? |
| 18:50.22 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
| 18:50.32 | stegbth | yes, i know 5 != 8 ;) that was the error |
| 18:50.49 | grandpapadot | So dahdi-dummy is the new loopback driver replacing ztdummy? |
| 18:51.01 | bcrisp | :) |
| 18:51.20 | axelilly | Any ideas on why when a called is put into a queue it only hears ringing and no MOH? And no, I did not call Queue with the r argument. |
| 18:52.21 | bcrisp | did you set music class? |
| 18:52.44 | axelilly | bcrisp: yea. And I know that MOH is working since I can hear it when I put a call on hold. |
| 18:53.30 | bcrisp | axelilly, did you try Set(CHANNEL( |
| 18:53.40 | bcrisp | <musicclass>)=whatever) prior to Queue() ? |
| 18:53.53 | axelilly | bcrisp: no, I guess I'll try that next. |
| 18:54.03 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 18:54.07 | bcrisp | sry.... Set(CHANNEL(musicclass)=yourclass) |
| 18:54.16 | bcrisp | that is supposed to override the queue moh - just to test it out |
| 18:54.29 | ruben23 | hi, how do i correct this error problem on my asterisk server--suing versionAsterisk 1.4.27 |
| 18:55.06 | bcrisp | reads ruben23's mind |
| 18:55.43 | jaytee | grandpapadot, yep and /etc/zaptel.conf is now /etc/dahdi/system.conf and /etc/asterisk/zapata.conf is now /etc/asterisk/chan_dahdi.conf |
| 18:56.29 | *** join/#asterisk correcaminos (n=laguilar@201.166.73.171.cable.dyn.cableonline.com.mx) |
| 18:57.07 | bcrisp | axelilly, you can also try the StartMusicOnHold(class) function |
| 18:57.13 | [TK]D-Fender | GAH |
| 18:57.13 | axelilly | bcrisp: check this CLI output. It looks like it's trying to start the MOH http://pastebin.com/d7b8d1d69 |
| 18:57.23 | jaytee | GAH? |
| 18:57.29 | ruben23 | sorry-->http://pastebin.com/m166e85c4 |
| 18:58.44 | bcrisp | hmm axelilly maybe a phone issue |
| 18:59.01 | axelilly | bcrisp: phone issue eh? |
| 18:59.10 | bcrisp | axelilly, im not entirely sure.. |
| 18:59.20 | p3nguin | bcrisp: How about SetMusicOnHold() instead of starting it? |
| 18:59.24 | bcrisp | axelilly, maybe try it with a softphone |
| 18:59.40 | bcrisp | it looks like it played the default moh |
| 18:59.49 | axelilly | bcrisp: yea, I only have default |
| 18:59.55 | p3nguin | and they are both applications, not functions. |
| 19:00.32 | bcrisp | p3nguin.. sorry bad choice of words on my part |
| 19:00.38 | bcrisp | aren't all functions in upper case? |
| 19:00.48 | jaytee | thinks that millenia from now alien archaeologists will be puzzling over [TK]D-Fender's cryptic "GAH" entry in chat. |
| 19:00.50 | p3nguin | core show functions |
| 19:01.09 | axelilly | bcrisp: seems more like Queue is not answering the call fo rsome reason |
| 19:01.17 | Katty | peeks in |
| 19:01.41 | Katty | HELLO THAR |
| 19:02.14 | *** join/#asterisk blebleble (n=godie@s7.n194.n222.n216.static.myhostcenter.com) |
| 19:02.30 | blebleble | I am having the same issue outlined here http://www.freepbx.org/forum/freepbx/users/phones-remain-in-use-when-they-transfer-a-call (no resolution), basically if users in our queue transfer a call to another department they do not get another queue call until the call they transfered is finished, it keeps them in use the entire time, anyone know of a fix for this |
| 19:02.49 | Katty | blebleble: you should probably be asking that in #freepbx |
| 19:03.15 | blebleble | Katty: did, just no response everyone seems to be idling thus thought i would try here |
| 19:03.17 | axelilly | bcrisp: I solved it. |
| 19:03.30 | Katty | blebleble: we don't support freepbx here |
| 19:03.35 | axelilly | bcrisp: Queue() wasn't answering the call. You need to proceed it with: Answer() |
| 19:03.42 | angler | blebleble, does the call happen to go over an IAX trunk to another Asterisk system when it's transfered? |
| 19:03.53 | blebleble | angler: does not just internal |
| 19:03.56 | blebleble | Katty: thanks |
| 19:06.03 | bcrisp | axelilly, ya that makes sense |
| 19:06.55 | axelilly | bcrisp: yea, silly mistake |
| 19:07.59 | p3nguin | What if you preceed Queue() with a Playback()? |
| 19:08.13 | p3nguin | Would you still need to Answer() it first? |
| 19:08.46 | axelilly | p3nguin: yes...you need to answer it in order for the channel to hear the Queue app |
| 19:09.05 | p3nguin | But Playback() can answer a channel... |
| 19:09.11 | p3nguin | See my reason for asking? |
| 19:10.21 | p3nguin | I can't imagine you would ever want a caller to go directly to a queue, i.e. exten => yourexten,1,Queue(queue1). |
| 19:10.25 | axelilly | p3nguin: yes, very observant...I would think that it'd work |
| 19:10.27 | *** join/#asterisk Obeliks_ (i=obeliks@gentoo/contributor/Obeliks) |
| 19:10.36 | axelilly | p3nguin: I do |
| 19:10.47 | p3nguin | I can't imagine it. |
| 19:11.06 | axelilly | p3nguin: the call is originally handled by IPCC on CallManager....which then punts the call to Asterisk for the call center. |
| 19:11.15 | p3nguin | If I called you, how would I know who I reached and what is going on with the call? |
| 19:11.26 | p3nguin | hmm |
| 19:11.41 | axelilly | p3nguin: because call manager played a menu for you. |
| 19:11.48 | p3nguin | right, I see that now. |
| 19:11.56 | p3nguin | I was thinking asterisk was out front. |
| 19:12.57 | p3nguin | So if there is no need for another Playback, then your finding of Answer() is certainly the way to go. |
| 19:13.56 | axelilly | p3nguin: yea, should be good. |
| 19:14.32 | *** join/#asterisk jkroon (n=jkroon@dsl-244-46-242.telkomadsl.co.za) |
| 19:16.15 | axelilly | p3nguin: now I just need to figure out how to get it to announce hold times. |
| 19:16.26 | bcrisp | axelilly... very easy in queues.conf |
| 19:16.43 | axelilly | bcrisp: yea, not working...it's only saying the position in line. |
| 19:16.53 | bcrisp | pastebin your queues.conf |
| 19:17.00 | bcrisp | i have announce holdtimes so i can reference it |
| 19:17.27 | p3nguin | Didn't we determine that some calls would need to be answered before it could estimate a hold time? |
| 19:21.03 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) [NETSPLIT VICTIM] |
| 19:21.03 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 19:21.04 | axelilly | bcrisp: here that pastebin http://pastebin.com/mef8e1d1 |
| 19:22.38 | axelilly | p3nguin: I didn't realize that. |
| 19:22.45 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
| 19:23.32 | axelilly | p3nguin: do you know how many? |
| 19:24.26 | p3nguin | No idea, really. |
| 19:24.35 | Katty | I"M IN THE MOOOD FOR LOVEEEE!!!! |
| 19:24.42 | Katty | eppigy: SIMPLY BECAUSE YOU"RE NEARRR MEEEEEEEEE |
| 19:28.31 | grandpapadot | Hey guys, in 1.6 menuselect, what does module embedded do? Basically link the modules to the binary (like including drivers in your linux kernel)? |
| 19:28.42 | grandpapadot | ... vs load and unload at runtime ... |
| 19:28.50 | *** join/#asterisk DennisG (n=Gebruike@84.30.136.208) |
| 19:29.44 | axelilly | Is there a way to turn on the MWI other then voice mail? I'd like to turn on the WMI if the phone is logged into a queue and turn it off when logged out. |
| 19:30.53 | axelilly | unless anyone has a better method of indicating if a phone is logged into a queue. |
| 19:31.36 | *** join/#asterisk Da-Geek (n=Da-Geek@87-194-2-213.bethere.co.uk) |
| 19:31.46 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 19:32.07 | DennisG | axelilly, take a look at the BLF section =) |
| 19:32.08 | [TK]D-Fender | axelilly: ----> |
| 19:32.10 | [TK]D-Fender | ~devstate |
| 19:32.10 | infobot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/community/russell/asterisk-1.4/func_devstate-1.4 , or http://www.asterisk.org/node/48325 |
| 19:32.29 | DennisG | haha 3 times the same answer |
| 19:32.32 | Qwell | MWI light is controlled differently |
| 19:33.54 | axelilly | yea, and I'm using Cisco 7941s...no custom light |
| 19:34.22 | LemensTS | if i do sip set debug peer 1000 at the cli, and i want it to log it to a file in logger.conf, what do i need to set in logger.conf? I tried just filename => debug that didnt put it in it |
| 19:35.24 | [TK]D-Fender | axelilly: Push an XLM browser page to them |
| 19:35.54 | p3nguin | or an XML page, even. |
| 19:35.58 | axelilly | [TK]D-Fender: know of any docs that explain how to do that? |
| 19:36.00 | DennisG | LemensTS, open logger.conf, look at the messages rule.. the messages file is stored in /var/log/asterisk/ =) |
| 19:36.18 | [TK]D-Fender | axelilly: Yeah, they have "Cisco" stamped in the corner. |
| 19:36.35 | [TK]D-Fender | axelilly: I'm sure you could go and try to look a little |
| 19:36.35 | axelilly | [TK]D-Fender: man, cisco's site is always such a mess |
| 19:37.43 | axelilly | [TK]D-Fender: yep, I'm on it |
| 19:38.30 | p3nguin | axelilly: Are you using SIP or SCCP images on your phones? |
| 19:38.37 | axelilly | p3nguin: SCP |
| 19:38.40 | axelilly | SCCP |
| 19:39.16 | p3nguin | Is there any type of "presense" indicator using that image? There sure doesn't seem to be in the SIP image. |
| 19:39.58 | *** join/#asterisk sniperjazz (n=sniper_j@203.175.76.218) |
| 19:41.18 | bcrisp | would Dial() options be lost if the callee has call forwarding set up? (i.e. * => cell (with transfer options) => cell forwarding to other # |
| 19:41.28 | *** join/#asterisk dkirker-openmobl (n=dkirker@pcp065241pcs.wireless.calpoly.edu) |
| 19:41.39 | sniperjazz | hi |
| 19:41.49 | sniperjazz | any buddy expert in asterisk? |
| 19:41.59 | sniperjazz | please contact me |
| 19:42.03 | sniperjazz | i need some helppppppppp |
| 19:42.07 | p3nguin | Is that really all you came here for? |
| 19:42.22 | p3nguin | You wanted to know if anyone is an expert, then you planned to leave? |
| 19:42.46 | bcrisp | p3nguin: but he stayed in a holiday inn last night... |
| 19:43.07 | Katty | i need some helpppppppppppppppppp too |
| 19:44.03 | dlynes | sniperjazz: i ain't your buddy...and putting more p's on the end of help ain't gonna get you help any faster...say what it is you need help with, and why you need help |
| 19:44.20 | dlynes | sniperjazz: AND DON'T MSG Me |
| 19:45.29 | p3nguin | http://catb.org/~esr/faqs/smart-questions.html |
| 19:46.34 | Katty | "Try to find an answer by asking a skilled friend" |
| 19:46.40 | Katty | DEAR SKILLED FRIENDS. |
| 19:46.51 | Katty | oh dangit, now i fogot my question |
| 19:47.53 | p3nguin | has a tooth ache |
| 19:48.07 | Katty | :< |
| 19:48.16 | Katty | do you have a cavity |
| 19:48.35 | p3nguin | I think my carnivorous activities earlier may have lodged something between a couple back teeth. |
| 19:48.42 | Katty | ohisee. |
| 19:48.54 | [TK]D-Fender | OU812? |
| 19:49.03 | bcrisp | help ME help Youuuuuu! |
| 19:49.44 | Katty | i'll help /you/ in a minute |
| 19:50.40 | Katty | i'm bored. |
| 19:50.57 | Katty | fridays are so slowwwwwwwwwwwwwwww |
| 19:51.26 | bcrisp | ill forward my collectors' calls to you Katty :) |
| 19:51.55 | Katty | kay |
| 19:51.58 | Katty | i'll sing to them |
| 19:52.27 | bcrisp | i should just forward their calls to a callcenter |
| 19:52.41 | bcrisp | torture script :) |
| 19:52.44 | p3nguin | debt collections? |
| 19:52.53 | bcrisp | ya |
| 19:53.21 | bcrisp | gah.. all the secure IM software out there looks amateurish |
| 19:53.23 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 19:53.30 | p3nguin | Answer the calls, once you find out who they are looking for, "Please hold; I'll transfer you," and then transfer them to endless MoH. |
| 19:53.45 | bcrisp | is there a built-in torture script? |
| 19:53.52 | p3nguin | It's all you. |
| 19:53.52 | bcrisp | press 1 if you are a member of the govt.... |
| 19:53.57 | p3nguin | dialplan. |
| 19:54.09 | bcrisp | some of the conf file samples refer to torture script |
| 19:54.17 | p3nguin | I've noticed that. |
| 19:54.23 | Corydon76-dig | codefreeze was the master of that |
| 19:54.29 | *** join/#asterisk trumee (n=rs4@cpc2-cmbg15-0-0-cust1000.5-4.cable.virginmedia.com) |
| 19:54.44 | bcrisp | wish i could find a tar with the dialplan + sounds :) |
| 19:54.46 | Corydon76-dig | He had an IVR with 100 different political parties, and the major ones were all at the end |
| 19:55.40 | bcrisp | that is great |
| 19:55.42 | Corydon76-dig | and at the end, a message would play that an error occurred, and the whole IVR would restart |
| 19:55.51 | bcrisp | haha |
| 19:56.18 | bcrisp | "Please dial your full name, followed by your street address, then press # |
| 19:56.40 | bcrisp | then just randomly error out and say "im sorry, that is invalid input" |
| 19:56.44 | grandpapadot | I swear I saw somewhere that you could store the host name in realtime config db tables, anyone seen this? I've been digging through the docs and can't find ... |
| 19:57.47 | Corydon76-dig | grandpapadot: No, hostname is in asterisk.conf and you cannot store that file in realtime |
| 19:58.16 | Corydon76-dig | Nor can you store any of the files that configure realtime in realtime (chicken-and-egg problem) |
| 19:59.08 | grandpapadot | Corydon76-dig: Thanks, but I mean I *think* I read somewhere I could "add" the hostname to the table so I could store multiple asterisk CDR logs in the same db/table ... Is this possible? |
| 19:59.17 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:59.40 | *** join/#asterisk gheegh (n=chatzill@pool-173-79-210-223.washdc.fios.verizon.net) |
| 19:59.46 | [TK]D-Fender | grandpapadot: thats adding a field IIRC, not a "table |
| 19:59.57 | Corydon76-dig | grandpapadot: oh, for CDR. Yes, you're looking for cdr_adaptive_odbc |
| 20:00.00 | bcrisp | i thought you could create user fields |
| 20:00.11 | grandpapadot | Sorry, wasn't clear. |
| 20:00.14 | grandpapadot | .. and thanks |
| 20:00.23 | gheegh | Hey all.. wondering if anyone has seen a good example of a simple "click here to call" button for my website. doesn't need to be sophisticated or handle presence.. etc. |
| 20:00.34 | gheegh | anyone have any thoughts or directions you can point me? |
| 20:00.43 | bcrisp | um |
| 20:01.08 | [TK]D-Fender | gheegh: Go lookup "call files", "AMI Originate" ont he WIKI |
| 20:01.11 | [TK]D-Fender | ~wikis |
| 20:01.11 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 20:01.12 | bcrisp | you can originate a call to their number, once they answer bridge |
| 20:01.12 | [TK]D-Fender | ~bok |
| 20:01.13 | Corydon76-dig | Adaptive CDR is pretty neat, if I do say so myself |
| 20:01.17 | [TK]D-Fender | gheegh: And the BOOK... |
| 20:01.22 | [TK]D-Fender | ~book |
| 20:01.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:01.27 | Corydon76-dig | but then again, I wrote it, so I'm biased. ;-) |
| 20:01.52 | bcrisp | ~Adaptive CDR |
| 20:01.54 | gheegh | Okie dokie.. clearly i deserved it.. my google-fu failed.. so I came here.. i'll go check the wiki |
| 20:04.05 | grandpapadot | So basically, with 1.6, you can store any channel variable in your CDR table with the [alias] section? |
| 20:04.41 | grandpapadot | That's pretty cool. |
| 20:05.40 | *** join/#asterisk CLeron (i=CLeron@d206-116-30-244.bchsia.telus.net) |
| 20:05.42 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:06.02 | Corydon76-dig | http://blogs.asterisk.org/2007/01/05/cdr_custom_odbc/ |
| 20:07.07 | grandpapadot | very cool |
| 20:07.22 | grandpapadot | how did I miss this? lol you should see the hacking I have in place to capture call details |
| 20:07.28 | Corydon76-dig | http://svn.digium.com/view/asterisk/branches/1.6.2/configs/cdr_adaptive_odbc.conf.sample?revision=182360&view=markup |
| 20:07.36 | Corydon76-dig | It's in 1.6 only |
| 20:08.22 | roguefalcon | I am sure this is simple but I will ask since I have throughly tried to figure it out on my own |
| 20:08.53 | Corydon76-dig | 1.6.2 is the only one that has the "static" keyword, though |
| 20:08.55 | roguefalcon | I setup Skype for Asterisk and when it calls in it immediately hangs up |
| 20:09.22 | roguefalcon | exten => ...,1,Dial(SIP/12345)) |
| 20:09.24 | Corydon76-dig | roguefalcon: did you install the g729 codec module? |
| 20:09.46 | grandpapadot | Hey anyone run extensions.conf successfully from static-realtime? |
| 20:09.49 | roguefalcon | default install of astriskNOW... so... I don't know if I did or not |
| 20:09.59 | Corydon76-dig | roguefalcon: that would be a no. |
| 20:10.23 | roguefalcon | Corydon76-dig: I will do that now |
| 20:10.37 | Corydon76-dig | SFA comes with g729 licenses but you have to get the codec module yourself |
| 20:10.55 | *** join/#asterisk vader-- (n=me@68.36.9.8) |
| 20:10.57 | vader-- | hello |
| 20:11.42 | bcrisp | hello lord vader |
| 20:12.20 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 20:13.15 | CLeron | can any one help me? Im trying to set up a Asterisk im able to connect the extantion with xlite but i cant dail out threw the trunk. THe trunk is connected and seems fine i think i need to setup calling rules for failing out? |
| 20:13.54 | bcrisp | CLeron, what do you mean by "dialing out through the trunk" |
| 20:14.25 | bcrisp | Cleron, pastebin you extensions.conf and sip.conf |
| 20:15.38 | CLeron | im using a SIP account for outgoing calls |
| 20:16.30 | CLeron | how do i access extensions.conf and sip.conf, sofar i have been using the web based graphical manager to amke the changes |
| 20:17.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.19 | [TK]D-Fender | CLeron: vi extensions.conf |
| 20:17.24 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 20:18.13 | Nugget | if you want a GUI to manage asterisk then use gvim. |
| 20:21.13 | grandpapadot | Nugget: lol |
| 20:24.21 | CLeron | <PROTECTED> |
| 20:24.22 | bcrisp | CLeron, please use main chat, not private message plz |
| 20:24.29 | CLeron | No prob |
| 20:24.32 | bcrisp | k |
| 20:24.38 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 20:24.42 | bcrisp | we would need to see your full dialplan |
| 20:25.04 | bcrisp | you using freepbx? |
| 20:25.24 | CLeron | hmm if i give you my login to the pbx can you check it out that way? |
| 20:25.35 | CLeron | its web based |
| 20:25.48 | bcrisp | CLeron: are you using asterisknow? |
| 20:25.52 | CLeron | yes |
| 20:26.25 | bcrisp | please try #asterisknow |
| 20:26.32 | CLeron | ok |
| 20:26.35 | CLeron | thanx |
| 20:27.17 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:27.33 | pigpen | With asterisk 1.6, with the inbound lines being IAX and the clients being SIP, is it wise to enable/force jitter buffer on both sides of the bridge?? |
| 20:28.00 | pigpen | I have had little reason to use jitter buffer until now, so I have little experience with it. |
| 20:29.05 | vader-- | whats your older asterisk system running? os and asterisk version |
| 20:30.47 | gheegh | sorry to ask newbie questions on Call files.. but if it's not working.. and I've confirmed that modules.conf has autoload=yes, what else could I do to troubleshoot? |
| 20:32.32 | CLeron | anyone know how to configure AsteriskNow to dial out threw SIP account since no one seems to be awake in #asterisknow |
| 20:35.13 | [TK]D-Fender | CLeron: ... |
| 20:35.15 | [TK]D-Fender | ~freepbx |
| 20:35.16 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:35.18 | [TK]D-Fender | ^^^^^^^^^ |
| 20:36.14 | Katty | hi folks |
| 20:36.29 | bcrisp | hi |
| 20:36.44 | Katty | i had some soup. but didn't realize there was spinach in it. eww :< |
| 20:36.52 | p3nguin | bleh |
| 20:37.09 | gheegh | Yummy.. i like spinach.. :-) |
| 20:37.25 | Katty | i like /raw/ spinach, but not soggy slimey spinach in my soup |
| 20:37.28 | p3nguin | I can handle raw spinach in a salad, but once it has been cooked and is floppy, it's out of the question. |
| 20:37.54 | Katty | agrees |
| 20:38.27 | bcrisp | i love it raw or mushy |
| 20:39.09 | Skeeter- | [TK]D-Fender, was that you that told me that IAX2 directed pickup is an impossible task? |
| 20:39.12 | gheegh | me too |
| 20:40.20 | jaytee | cooked spinach makes me gag |
| 20:40.42 | bcrisp | windows 7 sounds make me gag |
| 20:40.47 | gheegh | Amen |
| 20:41.30 | gheegh | So, question.. if i am logged in as root, and I create a callfile.. will asterisk not run it because i'm not logged in as "asterisk"? |
| 20:41.55 | Corydon76-dig | Depends on whether Asterisk can read/modify the file |
| 20:42.32 | Corydon76-dig | You could chown the file, though |
| 20:42.38 | gheegh | yep |
| 20:42.59 | lesouvage | I want to use content of the logging of Asterisk (Registration from 'IMSI) to trigger Asterisk to run a context setting up a sip entry in sip.conf and an extension in extensions.conf. |
| 20:44.36 | lesouvage | What way can I discover the "Registration from 'IMSI" part in the logging and use it as a trigger? For the rest I know what to do but I miss the first step. |
| 20:44.41 | gheegh | hrm.. that's not working.. which is annoying. :-) |
| 20:45.04 | bcrisp | gheegh: did you set the "damnitjustwork = yes" setting? |
| 20:45.37 | gheegh | i thought i did.. but i might have been distracted by the thought of spinach soup |
| 20:47.27 | gheegh | ok |
| 20:47.58 | *** join/#asterisk Chocho (n=chatzill@ip-166-28-71-77.bgwan.com) |
| 20:48.27 | gheegh | so, that's a bit annoying.. :-) you have to create a different directory.. chown it there, before you move it.. which makes scripting file generation difficult.. |
| 20:48.28 | gheegh | :-) |
| 20:49.18 | bcrisp | hm login as asterisk |
| 20:49.33 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:50.09 | p3nguin | If you're using a script, you could be chowning the file easily from within the script. |
| 20:50.58 | [TK]D-Fender | ... mv not chown |
| 20:51.19 | gheegh | right.. ok.. thanks! |
| 20:52.58 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 20:53.42 | Geminizer | Hello all (again). If I have a DID associated with my PBX, how (and where) do I associate that DID with a dialplan context? |
| 20:53.42 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:54.50 | [TK]D-Fender | Geminizer: A "DID" is something that arrives at a TELCO. |
| 20:54.59 | [TK]D-Fender | Geminizer: Maybe you could narrow things down. |
| 20:55.22 | bcrisp | Geminizer, are you using a sip provider? |
| 20:55.43 | Geminizer | someone dials 1-877-123-4567.... that eventually reaches my PBX (yes, I am using a SIP provider)... |
| 20:56.12 | Geminizer | where is the entry point defined for that "channel" (forgive the poor terminology) |
| 20:56.19 | p3nguin | As long as your ITSP is sending calls to 18771234567, you can create an exten for that number. |
| 20:56.22 | [TK]D-Fender | Geminizer: It matches against a peer in sip.conf and goes tot he context it points to based on what they dialed in |
| 20:57.19 | bcrisp | Geminizer: does your sip provider have any * config samples on their site? |
| 20:57.33 | p3nguin | You have to remember that extensions are rules that say what you want to happen when a specific alpha-numeric sequence is called. |
| 20:58.00 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:58.20 | Geminizer | that makes sense |
| 20:58.50 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:59.02 | p3nguin | So, create exten => 8771234567,1,SomeCommand() |
| 20:59.04 | Geminizer | ok... I am using telasip as a sip trunk provider, and it is defined in my main inbound route |
| 20:59.21 | p3nguin | What context do calls from that peer go into? |
| 20:59.26 | bcrisp | gem, can you patebin your sip.conf minus any passwords? |
| 20:59.30 | [TK]D-Fender | Geminizer: And that statement says pretty much nothing |
| 20:59.40 | [TK]D-Fender | "defined in my main inbound route" |
| 20:59.41 | Geminizer | and in the [telasip] context as defined in sip.conf... |
| 20:59.56 | Geminizer | which has a context param of "from-trunk" |
| 21:00.14 | bcrisp | Geminizer: pastebin it :) |
| 21:00.34 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 21:00.35 | p3nguin | Okay, so your [from-trunk] context in extensions.conf is where you need to create the exten for incoming calls. |
| 21:00.45 | [TK]D-Fender | *sigh* |
| 21:01.11 | bcrisp | [TK]D-Fender, can you tell me how to make asterisk work correctly please? thanks :) |
| 21:01.23 | Geminizer | http://pastebin.com/m752087a0 |
| 21:01.41 | [TK]D-Fender | bcrisp: 1st you start sticking forks into electical outlets until you find a live one.... |
| 21:01.54 | bcrisp | [TK]D-Fender, awesome! |
| 21:02.01 | p3nguin | geminizer: My last statment still stands. |
| 21:02.03 | bcrisp | Geminizer: now pastebin your extensions.conf |
| 21:02.13 | bcrisp | you need a [from-trunk] context in it |
| 21:02.18 | [TK]D-Fender | waits for hilarity to ensue |
| 21:02.36 | p3nguin | or since you are using a GUI, put it in one of the included files. |
| 21:02.47 | [TK]D-Fender | p3nguin: Nope |
| 21:02.52 | p3nguin | why not? |
| 21:03.12 | [TK]D-Fender | p3nguin: Seriously.... does it even have to be said? |
| 21:03.16 | Cynagen | what was the alternative GUI for managing Asterisk you guys suggested somebody use? |
| 21:03.22 | p3nguin | apparently |
| 21:03.42 | [TK]D-Fender | p3nguin: If you're using a GUI for this you shouldn't be screwing around in the config files anyway <- |
| 21:03.44 | p3nguin | There needs to be an applicable context somewhere. |
| 21:04.03 | bcrisp | shot i didnt realize he was using a GUI |
| 21:04.12 | *** join/#asterisk steve745 (n=W33DS@pool-96-243-156-36.tampfl.fios.verizon.net) |
| 21:04.15 | bcrisp | s/shot/shoot/ |
| 21:04.19 | p3nguin | All those buzzwords should have given it away. |
| 21:04.30 | p3nguin | "main inbound route" |
| 21:04.31 | [TK]D-Fender | bcrisp: And you do now? What brought this revelation? |
| 21:04.35 | steve745 | anyone know how to use bridge() in the dial plan |
| 21:04.39 | Corydon76-dig | Cynagen: it's called vim |
| 21:04.45 | Cynagen | haha |
| 21:04.49 | Cynagen | i prefer nano |
| 21:04.52 | vader-- | question im trying to play the privacy-phone sound file that tells the person to enter in their phone number and im trying to read the phone number in. I tried the playback command but the read doesn't happen until the sound file is completly done and the background playback treats what they enter in as an extension, any ideas? |
| 21:05.01 | bcrisp | [TK]D-Fender, |
| 21:05.08 | bcrisp | "or since you are using a GUI, put it in one of the included files." |
| 21:05.38 | [TK]D-Fender | bcrisp: What was p3nguin talking... and also assuming. |
| 21:05.39 | Corydon76-dig | Seriously, trying to get people in here to recommend a GUI is like trying to get Larry Ellison to recommend an open source database |
| 21:06.29 | Geminizer | umm, asterisk is open source, is it not? |
| 21:06.44 | p3nguin | It has to be included in a file, regardless of whether you manually enter the info or have your GUI do it. You won't get calls until you have the right peer configuration, an applicable context, and a valid exten that does something useful. |
| 21:06.50 | [TK]D-Fender | Geminizer: Generally, yes |
| 21:07.12 | Cynagen | okay, this officially blows ass |
| 21:07.24 | Cynagen | i'm trying to install asterisk 1.6.1.11 |
| 21:07.43 | Cynagen | and ./configure errors out "termcap support not found" |
| 21:07.47 | Cynagen | i have termcap installed |
| 21:07.59 | [TK]D-Fender | Cynagen: Go install the prereq's * has had for the past DECADE |
| 21:08.04 | [TK]D-Fender | Cynagen: libtermcap <- |
| 21:08.09 | Corydon76-dig | Cynagen: libncurses5-devel |
| 21:08.11 | Cynagen | facepalms |
| 21:08.27 | bcrisp | ya ncurses one got me initially |
| 21:09.17 | eppigy | NEIN |
| 21:09.49 | [TK]D-Fender | DU HAST MICH |
| 21:09.55 | eppigy | DA |
| 21:10.13 | Katty | hi dave. |
| 21:10.41 | eppigy | hiya |
| 21:12.09 | Katty | dirpy.com <- pretty fun. |
| 21:12.21 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
| 21:12.26 | roguefalcon | so if SFA comes with g729 licenses where do I find them? |
| 21:12.53 | Chainsaw | In the cupboard under the stairs. |
| 21:12.57 | roguefalcon | lol |
| 21:12.59 | Cynagen | noooowwww i have SIP |
| 21:13.03 | Cynagen | facepalms |
| 21:13.07 | Defraz | Hey all having an issue with a TE110P. When I call out and the number is busy or disconnected it reports back unavailable. shouldn't tell me if it is busy or disconnected or something. |
| 21:13.15 | roguefalcon | I think I installed the module but it sure doesn't seem to think I did |
| 21:13.16 | Corydon76-dig | roguefalcon: same place the SFA licenses are installed |
| 21:13.43 | tzafrir | Defraz, PRI? |
| 21:13.46 | roguefalcon | so how do I register it? |
| 21:14.03 | Defraz | yes PRI |
| 21:14.07 | eppigy | lays down next to Katty |
| 21:14.09 | Corydon76-dig | roguefalcon: with the register utility, same as the SFA licenses |
| 21:14.16 | tzafrir | If so, the next stop is to look at the pri debug, I guess |
| 21:14.45 | Defraz | so just pri debug? |
| 21:14.56 | roguefalcon | This is my first time to Skype with asterisk... so then I need to purchase a license to G729 or I got one when I signed up for SFA? |
| 21:15.00 | tzafrir | Where do you see it as "unavailable"? How can you tell it is busy? |
| 21:15.14 | Katty | eppigy: is it naptime already?! |
| 21:15.22 | eppigy | i am kind of sleepy |
| 21:15.31 | eppigy | but I think i am just ready to gop |
| 21:15.32 | eppigy | go |
| 21:15.56 | Katty | http://farm3.static.flickr.com/2727/4190811941_98497f9c4d_b.jpg <- Super. |
| 21:16.06 | Katty | eppigy: yes, me too...another 1.5hrs to go here tho :< |
| 21:17.53 | pigpen | Has anyone used the open source g729 with asterisk?? (http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/) |
| 21:18.37 | roguefalcon | Corydon76-dig: so I need to buy a g729 license? |
| 21:18.56 | Katty | http://i.imgur.com/2FOf7.jpg <- wait, what? |
| 21:19.12 | Katty | ^- that's warp 2, isn't it? |
| 21:19.14 | pigpen | roguefalcon, if you are using the "paid" version, then yes. |
| 21:19.30 | pigpen | however I ran across the above. |
| 21:19.54 | Katty | a bit /over/ warp 2, but not nearly warm 3 |
| 21:19.58 | Katty | s/warm/warp/ |
| 21:20.18 | roguefalcon | pigpen, so SFA doesn't come with it? I am trying to read back through the readme |
| 21:20.39 | pigpen | SFA? |
| 21:20.52 | bcrisp | skype for * |
| 21:20.52 | roguefalcon | skype for asterisk |
| 21:20.59 | pigpen | ah. |
| 21:21.14 | pigpen | well at the bottom of that page, it does note a royalty fee may need to be paid. |
| 21:21.15 | jaytee | I saw that CNN blurb on a Demotivational Poster that said "18 Times the Speed of Light? Because Fuck You, Einstein!!" |
| 21:21.17 | bcrisp | roguefalcon, that info should be on digiums site |
| 21:21.20 | pigpen | heh, so probably yes. |
| 21:21.28 | *** join/#asterisk voipmonk_ (n=shido6@69.171.158.152) |
| 21:21.37 | Katty | jaytee: lol |
| 21:21.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:21.39 | Katty | jaytee: nice. |
| 21:21.48 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 21:21.52 | jaytee | it was one of my favs from Demotivational |
| 21:22.01 | bcrisp | thats great |
| 21:22.18 | bcrisp | i like the GFY T-shirt |
| 21:22.42 | jaytee | I like the "I bring nothing to the table" tshirt from tshirthell.com |
| 21:23.25 | jaytee | or the one that on the front of the shirt says, "Rape is no laughing matter" and on the back says, "Unless you're raping a clown" |
| 21:24.11 | Corydon76-dig | roguefalcon: your SFA license should have come with g729 licenses. If not, you need to contact your reseller (or whomever you purchased your SFA licenses from) |
| 21:24.25 | p3nguin | Whoever it was that told me sip definitions for incoming "sip trunks" should be type=user ... that broke my phone calls. Every call that was coming IN was being send to the default context rather than the one defined in sip.conf. |
| 21:24.38 | p3nguin | I changed it to type=peer, and now I can get calls again. |
| 21:24.51 | bcrisp | p3nguin, hunt him down |
| 21:24.52 | *** join/#asterisk StevenR_ (n=foo@wan2.sghs.org.uk) |
| 21:24.57 | Geminizer | there is a table asterisk.incoming which, at first glance, appears to store the extenstion to destination mappings... am I in the right place or no? |
| 21:25.57 | Corydon76-dig | Geminizer: That's not part of Asterisk. That's some other system. |
| 21:26.04 | Geminizer | ahh, ok |
| 21:26.45 | p3nguin | bcrisp: I'm pretty sure I know who it was, but I'm not going to point fingers. |
| 21:26.58 | bcrisp | p3nguin, surely not me |
| 21:27.03 | p3nguin | bcrisp: I would rather get useful feedback about the experience instead. |
| 21:27.06 | p3nguin | no, not you. |
| 21:27.28 | bcrisp | my * roots arent deep yet |
| 21:27.43 | p3nguin | I'm just hoping that I didn't miss any important calls. |
| 21:27.43 | Cynagen | god i hate how the instructions for the orgasmatron and the original setup were only for pbx in a flash |
| 21:27.54 | Cynagen | and rely heavily on you using piaf |
| 21:28.11 | *** join/#asterisk smooth_penguin (n=smoove@59.95.10.233) |
| 21:28.14 | bcrisp | wha? |
| 21:28.16 | Cynagen | i need some generic "for all asterisk setups" gvoice setup |
| 21:29.03 | p3nguin | Did you look into the pygooglevoice script rather than trying to hack orgasmatron? |
| 21:29.09 | Cynagen | i did |
| 21:29.16 | eppigy | Katty: 30 MINUTES |
| 21:29.16 | p3nguin | No luck? |
| 21:29.19 | Cynagen | i don't see instructions on how to integrate it |
| 21:29.24 | p3nguin | oh |
| 21:29.27 | Cynagen | just instructions for piaf |
| 21:29.31 | Cynagen | it's fucking annoying |
| 21:29.35 | p3nguin | aye |
| 21:29.36 | Cynagen | it's all coded for piaf |
| 21:29.42 | Cynagen | all the scripts for install and shit |
| 21:29.45 | Cynagen | sighs |
| 21:29.55 | Cynagen | i like pygooglevoice |
| 21:29.59 | Katty | http://img129.yfrog.com/img129/9176/k8tha.jpg <- this is hilarious |
| 21:30.02 | Cynagen | i'm going to use it for other projects in the future |
| 21:30.03 | bcrisp | offers Cynagen a tissue |
| 21:30.05 | Cynagen | but, goddamn |
| 21:30.13 | bcrisp | :) |
| 21:30.29 | Cynagen | gimme some french fries and i'll be happy to sit and wait for them to pull their heads out of their asses |
| 21:30.50 | roguefalcon | so I skyped the digium pre-sales team and asked and they said "I am not sure... can we call you back" |
| 21:31.00 | roguefalcon | :-( |
| 21:31.03 | bcrisp | rofl |
| 21:31.25 | [TK]D-Fender | roguefalcon: Perhaps you should be talking to TECH SUPPORT for "support" of your install. "Sales" seem to have done their job already. |
| 21:32.12 | roguefalcon | lol |
| 21:32.24 | roguefalcon | [TK]D-Fender: not if I need a license for g729 |
| 21:32.33 | roguefalcon | the FAQ says SFA does come with g729 |
| 21:32.40 | Defraz | so I watched the debug looks like the pri is reporting back the right info but asterisk just plays the all-circuits are busy. |
| 21:32.47 | dlynes | roguefalcon: does or doesn't? |
| 21:32.58 | roguefalcon | http://www.digium.com/en/docs/SFA/sfa_faq.php |
| 21:33.14 | roguefalcon | it technically says 'yes' |
| 21:33.41 | [TK]D-Fender | roguefalcon: So go ask for "Support" in setting it up |
| 21:33.51 | Defraz | http://pastebin.ca/1719811 |
| 21:34.19 | Defraz | so if asterisk isn't seeing it then would it be the zaptel drivers? |
| 21:34.25 | [TK]D-Fender | defENTIRE CALL PLEASE |
| 21:34.43 | Defraz | okay |
| 21:34.49 | *** join/#asterisk voipmonk__ (n=shido6@69.172.69.80) |
| 21:35.06 | roguefalcon | [TK]D-Fender: I am under the impression they would want me to pay for support... is that not the case? |
| 21:35.22 | Katty | oh em gee, this afternoon is /dragging/ |
| 21:35.25 | [TK]D-Fender | roguefalcon: You paid for a product. taht includes installation support. |
| 21:35.37 | roguefalcon | [TK]D-Fender thanks |
| 21:35.39 | [TK]D-Fender | roguefalcon: Please get a clue, and then get off your ass and go ask for it. |
| 21:35.54 | roguefalcon | [TK]D-Fender: I guess I take back my thanks |
| 21:36.21 | Defraz | http://pastebin.ca/1719816 |
| 21:37.24 | [TK]D-Fender | Defraz: Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] <-- invalid PSTN # |
| 21:37.40 | Defraz | yea that number is disconnected |
| 21:37.49 | p3nguin | So what would make type=user ignore the context= and go to default, but setting type=peer it will obey context= and I can get the call? This is from a SIP ITSP, inbound calls only. |
| 21:38.12 | [TK]D-Fender | p3nguin: It wouldn't |
| 21:38.16 | p3nguin | It did. |
| 21:38.20 | [TK]D-Fender | p3nguin: I see nothing |
| 21:38.22 | Defraz | when I call it with a land line it says disconnected but when I call it with the PRI and asterisk it says the all circuits are busy. |
| 21:38.37 | *** join/#asterisk correcaminos (n=laguilar@201.166.73.171.cable.dyn.cableonline.com.mx) |
| 21:38.46 | jaytee | quittin time, back later |
| 21:38.58 | [TK]D-Fender | defNo, the TELCO "says" busy", because that's AUDIO PROGRESS. PRI is OOB |
| 21:39.00 | Defraz | When I call it with a sip provider from the dame box it says disconnected. The PSTN plays the message |
| 21:39.31 | [TK]D-Fender | Defraz: PRI gives you OOB progress. SIP can do whatever the other side wants to |
| 21:40.01 | Defraz | So the OOB is saying the number doesn't exist? |
| 21:40.12 | [TK]D-Fender | [16:37]<[TK]D-Fender>Defraz: Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] <-- invalid PSTN # |
| 21:40.13 | [TK]D-Fender | ^^^^^^ |
| 21:40.38 | Defraz | so then I need to take that and play a message about the number being unavailabe or not in service. |
| 21:41.12 | [TK]D-Fender | Defraz: i think there is another channelvar with q.sig, or other PRI status codes in it you can use |
| 21:41.25 | [TK]D-Fender | defBut "Dialstatus" is lowest level and too generic |
| 21:41.28 | [TK]D-Fender | (for yo) |
| 21:41.34 | Defraz | okay |
| 21:47.36 | p3nguin | [tk]d-fender: Here's something for you to "see" since you feel that I was lying about it: http://pastebin.com/d630a841c |
| 21:47.57 | p3nguin | type=user, context is ignored, it checks for phone number exten in default. |
| 21:48.12 | [TK]D-Fender | p3nguin: Its not the context that's ignored. |
| 21:48.17 | [TK]D-Fender | p3nguin: Found no matching peer or user for '204.155.28.10:5060' |
| 21:48.26 | [TK]D-Fender | p3nguin: it didn't match your entire damn PEER |
| 21:48.30 | p3nguin | If I change to type=peer or type=friend, it works just fine. |
| 21:48.47 | [TK]D-Fender | p3nguin: Because the auth method is DIFFERent between "peer and "user" |
| 21:49.03 | p3nguin | host= isn't good enough for a user? |
| 21:49.48 | [TK]D-Fender | p3nguin: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034907.html |
| 21:50.19 | [TK]D-Fender | p3nguin: Go read up on how these are used |
| 21:50.37 | bcrisp | so much love |
| 21:51.05 | p3nguin | This writing seems to claim that peer and friend should have the same functionality. |
| 21:51.33 | [TK]D-Fender | p3nguin: And user is separate... there are other docs that explain the difference, but I don't have time to hunt them down right now. |
| 21:51.44 | p3nguin | But when I use type=friend, the call is received as I expected. When I use type=peer, the call is received like I expected. |
| 21:51.57 | p3nguin | type=user == fail |
| 21:52.20 | [TK]D-Fender | p3nguin: A little searching will tun them up |
| 21:52.34 | [TK]D-Fender | p3nguin: ;Like I said, the auth process is DEFFERENT. Go read up on how. |
| 21:53.14 | [TK]D-Fender | GAH... can't type... and its checkout time.. |
| 21:53.16 | [TK]D-Fender | BBL |
| 21:53.24 | [TK]D-Fender | p3nguin: I'm sure you'll figure it out... |
| 21:54.07 | bcrisp | is it possible to change a caller's position in line? |
| 21:54.11 | bcrisp | for a queue |
| 21:55.01 | voipmonk__ | bcrisp: like if they want to hang up and call back or - live while in the queue? |
| 21:55.21 | bcrisp | here's what im trying to do: |
| 21:55.43 | bcrisp | caller enters queue, presented with the option while waiting in queue to leave a message (without losing queue position) |
| 21:56.51 | bcrisp | hmm hard to explain... basically the message is playedback to the queue member |
| 21:57.09 | bcrisp | so i thought i might need to generate some kind of call and manually set its position in the queue.. on bridge play the sound file |
| 21:57.47 | bcrisp | does that make sense ? |
| 21:58.01 | voipmonk__ | yep |
| 21:58.23 | voipmonk__ | wow , I just left |
| 21:58.30 | voipmonk__ | who am i |
| 21:58.39 | voipmonk | there we go |
| 21:58.40 | bcrisp | it would be easy enough to just record a sound when the user dials the 1# queue-context extensions |
| 21:58.58 | bcrisp | but the rest is a mystery |
| 22:00.21 | *** part/#asterisk JohnTeddy (i=unstable@tor/regular/sid) |
| 22:02.26 | bcrisp | im studying the following now to get an idea: http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback |
| 22:03.47 | etfonhomey1 | bcrisp, what's the project for today? |
| 22:03.50 | *** join/#asterisk correcaminos (n=laguilar@201.166.73.171.cable.dyn.cableonline.com.mx) |
| 22:04.25 | bcrisp | etfonhomey1: queued caller can leave a message which takes his/her place in the queue line, played back to the agent once their queue position is 1st |
| 22:05.08 | etfonhomey1 | bcrisp, very interesting. I'd be interested to see how you do it when you're done. |
| 22:05.11 | ruben23 | hi |
| 22:05.43 | etfonhomey1 | I use call queues for my help desk. Wouldn't be a bad thing to have if someone wants to leave a message rather than wait in line. |
| 22:06.01 | bcrisp | ya, some airlines do it |
| 22:06.09 | bcrisp | who wants to wait on hold? |
| 22:06.44 | bcrisp | probably have 1 to leave a message (in queue), and 2 for callback |
| 22:07.00 | bcrisp | ah |
| 22:07.12 | bcrisp | i can manipulate QUEUE_PRIO |
| 22:07.38 | *** join/#asterisk etfonhomey (n=etfonhom@32.179.143.44) |
| 22:07.46 | bcrisp | hm |
| 22:09.15 | etfonhomey | Have you heard over others doing it with Asterisk? |
| 22:09.23 | bcrisp | not yet |
| 22:09.26 | p3nguin | diatonic: Can you paste your hylafax config.ttyIAX1 once more for me? |
| 22:10.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:12.33 | *** part/#asterisk etfonhomey (n=etfonhom@32.179.143.44) |
| 22:15.31 | ruben23 | hi nayone have idea on this log ----> http://pastebin.com/m25e35e49 |
| 22:15.55 | *** join/#asterisk bmoraca (n=bmoraca@66-242-174-254.ceres.bvn.net) |
| 22:16.04 | bcrisp | i think i could set some channel variables when user hits the queue context extension, record a message, then write a script to generate a callfile that sets the queue/queue priority. Then the membermacro could be modified to read channelvars and play the recorded msg |
| 22:16.07 | bcrisp | what a mouth ful |
| 22:16.37 | bmoraca | is there a way to debug cdr_mysql to find out why, exactly, it's erroring out? |
| 22:17.28 | bcrisp | (or i might just be crazy) |
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| 22:24.31 | lesouvage | If I copy a callfile using mc it works and if I move the file with the "mv" command an error messgae appears. [Dec 18 23:21:08] WARNING[2303]: pbx_spool.c:482 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/call' |
| 22:25.08 | lesouvage | How can I solve this? |
| 22:25.15 | mcrownover | what is the proper format for MOH files mp3 or wav?? |
| 22:25.59 | diatonic | p3nguin, mine is here: http://pastebin.com/d49144c67 - It was generated by elastix when I added a virtual fax modem. I've also run it on Trixbox by setting up according to post #3 on this page: http://is.gd/5sQwE |
| 22:26.04 | mcrownover | it seems that freepbx is converting mp3 back to wav |
| 22:28.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 22:35.53 | CGMChris | I am still having problems with SIP channels hanging after RTPTimeout occurs. http://pastebin.com/m1425f84a |
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| 22:38.20 | Tech_Travis | p3nguin: Did your asterisk and hylafax stop working? |
| 22:38.20 | *** join/#asterisk DrGeek (n=geek@c-24-21-244-173.hsd1.or.comcast.net) |
| 22:38.20 | DrGeek | g'day all. |
| 22:39.11 | DrGeek | I'm running asterisk 1.6.2-rc2 on ubuntu 9.10. After installing freepbx 2.6.0RC2, asterisk fails to load. |
| 22:39.30 | JunK-Y | which error? |
| 22:39.31 | DrGeek | How do I start to troubleshoot this one? |
| 22:39.53 | JunK-Y | what happen when you do asterisk -cvvv ? |
| 22:39.57 | *** part/#asterisk axelilly (n=jfenner@66.181.75.57) |
| 22:40.10 | DrGeek | The tail of asterisk -cvvv is |
| 22:40.13 | DrGeek | <PROTECTED> |
| 22:40.26 | DrGeek | with exit status 1 |
| 22:40.27 | CGMChris | DrGeek: Pastebin. |
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| 22:40.36 | CGMChris | DrGeek: or check /var/log/asterisk/messages |
| 22:41.32 | DrGeek | http://pastebin.com/m155c8346 |
| 22:42.09 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:42.26 | JunK-Y | and it stops there? what about if you noload => func_curl.so in your modules.conf |
| 22:42.30 | *** join/#asterisk mcrownover (n=markcrow@remote.gawest.com) |
| 22:42.50 | JunK-Y | is it the original * version or you installed the ubuntu package? |
| 22:42.58 | DrGeek | JunK-Y, yeah |
| 22:43.05 | [TK]D-Fender | JunK-Y: B-Ing! |
| 22:43.16 | JunK-Y | [TK]D-Fender: yay! |
| 22:43.18 | JunK-Y | :) |
| 22:43.21 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 22:43.22 | [TK]D-Fender | \o/ |
| 22:43.33 | mcrownover | can someone tell me how to upload a MOH wav larger than 2mb after modifying /etc/php.ini to allow 5mb files? |
| 22:43.40 | JunK-Y | thanks god, its over! hehehe |
| 22:43.48 | mcrownover | mine keeps blowing up telling my file size is too large |
| 22:43.51 | [TK]D-Fender | mcrownover: Completely restart Apache |
| 22:43.57 | mcrownover | aah |
| 22:44.09 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 22:44.15 | JunK-Y | i was suppose to do it in 4 years (12 semesters), i did it in 3.33 years (10 semesters) |
| 22:44.15 | DrGeek | after noload func_curl.so: http://pastebin.com/m46d8d9f3 |
| 22:44.24 | [TK]D-Fender | JunK-Y: 1/2 Evil! |
| 22:44.27 | CGMChris | [TK]D-Fender: Any idea why my sip channels are hanging? http://pastebin.com/m1425f84a |
| 22:44.58 | mcrownover | thanks TK - its what i get for workin late on friday.... |
| 22:45.39 | [TK]D-Fender | CGMChris: No idea |
| 22:46.05 | [TK]D-Fender | CGMChris: Except maybe Swifty is clocked and can't tell the call died |
| 22:46.16 | [TK]D-Fender | -c |
| 22:46.17 | DrGeek | JunK-Y, still does not launch after noload func_curl.so ... ideas? |
| 22:46.20 | JunK-Y | DrGeek: and gdb asterisk then run -cvvvvv ? |
| 22:46.48 | DrGeek | installing gdb... |
| 22:46.52 | JunK-Y | asterisk -V? |
| 22:47.00 | DrGeek | Asterisk 1.6.2.0~rc2-0ubuntu1.1 |
| 22:47.12 | JunK-Y | ha its the ubuntu package. |
| 22:47.16 | *** join/#asterisk Greek-Boy (n=email@41.188.154.137) |
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| 22:47.43 | JunK-Y | wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.0.tar.gz |
| 22:47.52 | JunK-Y | install this and see if you have the same problem. |
| 22:48.05 | JunK-Y | since we dont give a real support for all distros out there. |
| 22:48.36 | Katty | HELLO THAR SWEETPEA |
| 22:48.39 | Katty | hugs JunK-Y |
| 22:48.46 | DrGeek | JunK-Y, ok |
| 22:48.55 | JunK-Y | hugs katty too! |
| 22:48.59 | JunK-Y | whats up? |
| 22:49.09 | DrGeek | JunK-Y, I'll use the src distro... Just in case, do I need to pass any configure options to include debug support for gdb if we need it? |
| 22:49.40 | DrGeek | JunK-Y, this is the gdb output: http://pastebin.com/m286b9bc4 |
| 22:49.42 | JunK-Y | no, its fine, ./configure && make -j2 && make install would be enough for now |
| 22:51.21 | *** join/#asterisk isamar (n=isamar@189.14.112.59) |
| 22:51.24 | isamar | hi folks... |
| 22:51.49 | *** part/#asterisk mcrownover (n=markcrow@remote.gawest.com) |
| 22:51.56 | isamar | anybody using trixbox here ? |
| 22:52.54 | isamar | need to know if trixbox pro version has any evaluation |
| 22:52.55 | isamar | <PROTECTED> |
| 22:53.21 | bcrisp | hm |
| 22:53.35 | Katty | isamar: #trixbox |
| 22:53.40 | isamar | okey |
| 22:53.44 | Katty | isamar: we do not officially support trixbox here |
| 22:53.54 | isamar | asked there and had only echo of my voice :-) |
| 22:54.04 | isamar | Katty: np |
| 22:54.09 | russellb | trixbox eats babies |
| 22:54.16 | isamar | just a shot.. sorry |
| 22:54.20 | isamar | hehehe |
| 22:54.29 | isamar | okey.. thanks |
| 22:54.45 | Katty | almost time for the weekend!!! yay |
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| 22:57.46 | JunK-Y | DrGeek: so? |
| 22:58.03 | bcrisp | cant...figure...this out |
| 22:58.38 | bcrisp | is there any way to gather position in a queue? |
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| 22:58.47 | DrGeek | JunK-Y, just got it compiled... |
| 22:58.49 | DrGeek | starting. |
| 22:59.07 | JunK-Y | so problems fixed? |
| 22:59.10 | DrGeek | JunK-Y I've got a CLI> ! |
| 22:59.27 | JunK-Y | dont forget to remove the noload in ur modules.conf (if you need it) |
| 22:59.30 | Katty | snowman.yahoo.com <- friend of mine made this. |
| 22:59.32 | Katty | ^- enjoy. |
| 23:00.03 | DrGeek | JunK-Y, I'll default configs for now until I get the hang of asterisk. Thank you for your help! |
| 23:00.39 | JunK-Y | DrGeek: enjoy the CLI! |
| 23:00.45 | JunK-Y | ive to jet, see ya. |
| 23:01.17 | DrGeek | :) |
| 23:01.57 | eppigy | NEIN |
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| 23:07.17 | p3nguin | tech_travis: I've only had one successful fax received since I started. |
| 23:07.50 | p3nguin | tech_travis: I have had two partials today, but only about the top one inch of the pages. |
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| 23:16.05 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.0 (2009/12/18), Asterisk 1.6.1.12 (2009/12/18), 1.6.0.20 (2009/12/18), 1.4.28 (2009/12/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bu |
| 23:16.21 | *** join/#asterisk Tech_Travis (n=Tech_Tra@mail.techglia.com) |
| 23:16.39 | bcrisp | ugh... anyone used QUEUE_PRIO ? |
| 23:16.54 | leifmadsen | Asterisk 1.4.28, 1.6.0.20, 1.6.1.12, and 1.6.2.0 (NEW!) are now available! See http://www.asterisk.org for more information. Thanks for your continued support of Asterisk! |
| 23:17.04 | *** part/#asterisk Tech_Travis (n=Tech_Tra@mail.techglia.com) |
| 23:17.49 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 23:18.08 | bmoraca | damnit, i just installed 1.6.1.11 yesterday |
| 23:19.58 | leifmadsen | heh :) |
| 23:20.01 | leifmadsen | it's easy to install 1.6.1.12 |
| 23:20.18 | ruben23 | hi, how do i uninstall asterisk..? i wan to downgrade for some issues i have on my current version.. |
| 23:20.30 | leifmadsen | ruben23: just install over top |
| 23:20.43 | leifmadsen | rm -f /usr/lib/asterisk/modules ; cd /usr/src/asterisk-version ; make install |
| 23:20.55 | ruben23 | leifmadsen: but, im installaing a lower version.. |
| 23:21.06 | leifmadsen | ruben23: yes... see above |
| 23:21.18 | leifmadsen | you don't have to "uninstall" to downgrade |
| 23:21.29 | leifmadsen | just remove current modules, and reinstall |
| 23:21.54 | ruben23 | <PROTECTED> |
| 23:28.27 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
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| 23:42.37 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 23:42.47 | LemensTS | im having one way audio issues am desperate for help here is a sip debug on that peer http://pastebin.ca/1719932 |
| 23:43.23 | LemensTS | i notice line 71 and 72 are duplicated... |
| 23:47.33 | [TK]D-Fender | LemensTS: thats fine and I see you debugging HALF of whats involved in this call, and showing none of the configs |
| 23:48.53 | *** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath) |
| 23:49.05 | sbrath | how do I get rid of this message: Received SIP subscribe for peer without mailbox: (null) |
| 23:49.22 | [TK]D-Fender | sbrath: Make a mailbox for that peer or tell it to stop trying |
| 23:49.34 | *** part/#asterisk stegbth (n=stegbth@p4FD0967B.dip0.t-ipconnect.de) |
| 23:50.03 | sbrath | But if I call the extension it goes to a mailbox, but I don't have anything in voicemail.conf |
| 23:50.23 | LemensTS | TKD: http://pastebin.ca/1719943 here is settings |
| 23:50.34 | LemensTS | and more specific settings on peer |
| 23:52.48 | bcrisp | cd sbrath: tell your phone to stop asking |
| 23:53.03 | sbrath | fair enough :) |
| 23:53.19 | bcrisp | the phone is asking to subscribe to voicemail |
| 23:53.22 | bcrisp | but has no voicemail |
| 23:55.28 | [TK]D-Fender | [18:50]<sbrath>But if I call the extension it goes to a mailbox, but I don't have anything in voicemail.conf <- that is DIALPLAN and has nothing to do with the other |
| 23:56.02 | bcrisp | sbrath: do you WANT voicemail for that SIP device? |
| 23:56.09 | sbrath | yes. |
| 23:56.38 | bcrisp | sbrath: if so, in sip.conf set the mailbox setting |
| 23:56.44 | bcrisp | i.e. Mailbox=123 |
| 23:57.11 | bcrisp | then in voicemail.conf, under [default] |
| 23:57.18 | bcrisp | set up the mailbox: |
| 23:57.30 | bcrisp | 123 => <password>,<name>,<email> ..... |
| 23:57.45 | sbrath | my sip config is in users.conf it seems, and mailbox= is set. |
| 23:57.55 | bcrisp | ok so in extensions.conf |
| 23:57.59 | bcrisp | you can do this |
| 23:58.18 | bcrisp | exten => 123, 1,VoiceMail(123) |
| 23:58.42 | bcrisp | sorry |
| 23:58.49 | bcrisp | exten => 123,1,VoiceMail(SIP/123) |
| 23:59.11 | bcrisp | ugh i think im wrong lol |
| 23:59.15 | bmoraca | as5400s are kind of cool...once you figure them the hell out, anyway |
| 23:59.17 | bcrisp | easier... |
| 23:59.26 | bcrisp | Voicemail(123@default) |
