| 00:02.29 | [TK]D-Fender | kripton1x: Go read the guide : |
| 00:02.30 | [TK]D-Fender | ~sipnat |
| 00:02.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:03.00 | kripton1x | hey thanks! |
| 00:04.47 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 00:05.13 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
| 00:08.58 | drmessano | So apparently TCP is listening on my 1.6 box |
| 00:09.11 | drmessano | Just no transforming from TCP <> UDP in calls |
| 00:09.30 | frieze | can a bad sip.conf cause a segfault on startup? |
| 00:11.30 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 00:17.49 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:20.24 | *** join/#asterisk ManxPower (n=manxpowe@17.sub-70-221-193.myvzw.com) |
| 00:21.11 | *** part/#asterisk ManxPower (n=manxpowe@17.sub-70-221-193.myvzw.com) |
| 00:27.34 | [TK]D-Fender | drmessano: You need to specify the transport in your peer entry |
| 00:28.19 | *** join/#asterisk ManxPower (n=manxpowe@17.sub-70-221-193.myvzw.com) |
| 00:28.21 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 00:28.30 | *** part/#asterisk ManxPower (n=manxpowe@17.sub-70-221-193.myvzw.com) |
| 00:28.57 | drmessano | I have |
| 00:29.18 | drmessano | Still digging.. Got debug on now |
| 00:29.43 | drmessano | Ahh.. SIP/2.0 405 Method Not Allowed |
| 00:30.46 | [TK]D-Fender | drmessano: On which? |
| 00:31.08 | drmessano | From the far end box to > Asterisk |
| 00:31.53 | [TK]D-Fender | drmessano: I meant in RESPONSE to which mothod? What is it rejecting? |
| 00:33.21 | *** part/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 00:34.35 | drmessano | Hang on... This fucking terminal window is IMPOSSIBLE to work with.. Too much other crap for the moment |
| 00:34.54 | drmessano | SecureCRT: "Oh, you mean holding a scroll is useful?" |
| 00:37.37 | jaytee | what's SecureCRT? a Putty wannabe? |
| 00:37.48 | drmessano | Other way around |
| 00:37.54 | drmessano | Its about $120 |
| 00:38.00 | drmessano | or something like that |
| 00:38.05 | drmessano | Its pretty decent |
| 00:38.30 | jaytee | ah, so it costs money and it sucks instead of just plain sucks like Putty |
| 00:38.34 | drmessano | lol |
| 00:38.39 | Cutlass | anyone have any idea about that call Park question? |
| 00:38.47 | drmessano | Im trying to get something I can actually look at here |
| 00:38.52 | drmessano | Fucking cant hold the mouse |
| 00:39.01 | jaytee | Cutlass, Colonel Mustard, drawing room, candlestick. Best of luck |
| 00:39.02 | drmessano | Guess I can capture, exit asterisk |
| 00:39.28 | Cutlass | I don't understand your response... |
| 00:40.06 | jaytee | Cutlass, I wasn't here when you asked your question so I just threw out 3 separate items from the board game Clue :-) |
| 00:40.23 | Cutlass | >I want to define an extension in the dial plan which, when called directly, will park the caller and will NOT play back the orbit that (s)he was parked on...I'm using the Park() application, but from what I can tell, there is now way to make it suppress the orbit announcement. Does anyone know how to accomplish this? |
| 00:40.42 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ba09e80bf5a60e4b) |
| 00:41.17 | jaytee | because I'm like that. bombastic, self-righteous, over-opinionated, arrogant and those are just some of my Good traits. |
| 00:41.58 | jaytee | what the hell is an orbit? we got VOIP on the ISS or something? |
| 00:42.43 | [TK]D-Fender | Cutlass: What happens when the caller calls Park? |
| 00:42.59 | Cutlass | it announces the orbit and the user is parked |
| 00:43.11 | [TK]D-Fender | Cutlass: To the caller? |
| 00:43.47 | [TK]D-Fender | Cutlass: Try "ParkAndAnnouce" instead |
| 00:44.02 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 00:44.14 | Cutlass | yes....so the use case would be for a consultative transfer, you would call, hear the orbit and then compelete the transfer... |
| 00:44.25 | Cutlass | ...I understand... |
| 00:44.31 | puppet | anyone know any good program/script to monitor CPU/Memory usage HDD temp / CPU temp? |
| 00:44.40 | Cutlass | ParkAndAnnounce but have it call a dummy extension? |
| 00:44.55 | Cutlass | so the announce is effectively eliminated... |
| 00:45.02 | Cutlass | ...that's a good thought! |
| 00:46.09 | [TK]D-Fender | Cutlass: You're welcome |
| 00:46.40 | Cutlass | thanks again [TK]D-Fender...I'll give it a shot... |
| 00:51.45 | drmessano | AH HA! |
| 00:54.51 | tzanger | puppet: sounds like you want lmsensors |
| 00:54.56 | tzanger | vmstat can do cpu/mem/io pressure |
| 00:55.05 | tzanger | cpu temp/fans/voltages is through lmsensors |
| 00:55.10 | tzanger | hdd stuff is through smartmontools |
| 00:56.33 | drmessano | jaytee: I am getting a BUSY signal from exchange now |
| 00:56.37 | *** join/#asterisk viator (n=matt@cpe-74-70-220-145.nycap.res.rr.com) |
| 00:56.38 | drmessano | Not a fast busy LOL!!!! |
| 00:56.39 | drmessano | Yay |
| 00:57.05 | jaytee | well, that's progress :-) |
| 00:57.49 | drmessano | Yes |
| 00:57.56 | drmessano | Was getting a fast busy previously |
| 00:58.12 | viator | does asterisknow allow forwarding of extensions ie. instead of voicemail can goto cell? or should i goto switchvox? |
| 00:58.23 | jaytee | wonder how many simultaneous sip calls the UM gateway can handle. haven't found anything in the docs on it. |
| 00:59.45 | drmessano | Its set in the dialplan |
| 00:59.50 | drmessano | 100 I think by default |
| 01:01.05 | drmessano | You know.. |
| 01:01.11 | jaytee | if 100 is the default I wonder what you could push it to assuming you had unlimited bandwidth |
| 01:01.29 | jaytee | before it went bat shit crazy and bluescreened |
| 01:01.51 | drmessano | A real coup d'tat would be using AsteriskWin32 1.6 on an Exchange box |
| 01:02.28 | drmessano | You wouldn't have the SIP messaging.. but |
| 01:02.43 | jaytee | about as revolutionary as the Sandanistas overthrowing Somosa. How'd that turn out again? |
| 01:06.31 | drmessano | lol |
| 01:07.28 | drmessano | It would be like driving on the right side of the road in France.. and if someone stops you, you ask them "can you say that again please, en deutsch? What, dont speak german? Youre welcome.. " |
| 01:13.03 | *** join/#asterisk sasargen_ (n=chatzill@173.100.22.119) |
| 01:13.41 | [TK]D-Fender | viator: Probably possible. What options does it show you? What do you see being executed in CLI when you call? |
| 01:17.17 | joako | Has anyone gotten IMAP voicemail working with Exchange 2003? |
| 01:18.22 | drmessano | Ouch |
| 01:18.39 | drmessano | I wouldnt just SMTP it, TBH |
| 01:18.50 | drmessano | errr |
| 01:18.54 | drmessano | I would just SMTP it, TBH |
| 01:19.32 | joako | That's no fun |
| 01:19.57 | joako | Problem with SMTP is when you delete the email the message is still in the voicemail box.... |
| 01:20.37 | drmessano | You can tell Asterisk to delete after sending |
| 01:20.47 | drmessano | Exchange 2003 IMAP is horribly broken.. You'll probably add latency to your Asterisk box |
| 01:20.57 | drmessano | Just from the persistent connections |
| 01:21.09 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 01:21.39 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:22.51 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 01:24.13 | drmessano | OMG |
| 01:24.49 | drmessano | I just saw the Beyonce' DirecTV commercial and bought 3 years of Satellite TV for 5 TVs |
| 01:25.01 | drmessano | I dont have a place for the dish, or 5 TVs |
| 01:25.02 | jblack | OMG WTF O RLY NO WAI! |
| 01:25.21 | drmessano | But she wanted to upgrade me |
| 01:25.57 | jblack | I've been a dtv customer since the late 90s. I like their customer service. |
| 01:28.51 | *** join/#asterisk Meaw (n=dino@213.244.81.144) |
| 01:35.49 | Carlos_PHX | Even she couldn't make be be willing to put up with their DVR. |
| 01:38.14 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:44.00 | drmessano | If had Beyonce, I wouldnt need to watch TV.. I'd *make* TV |
| 02:31.56 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 02:32.14 | *** join/#asterisk pawpro (n=Miranda@host86-147-119-69.range86-147.btcentralplus.com) |
| 02:35.22 | pawpro | Hi all. Can someone tell me why when I make the call in LAN from Asterisk 1 (using autodialout) to Asterisk 2 (Answer(),MusicOnHold()) using SIP ulaw there is never RTP session established but when i call from SIP client registered to Asterisk 1 who does dial(x@asterisk1) everything works? |
| 02:36.21 | [TK]D-Fender | pawpro: pastebin is your friend... |
| 02:36.27 | [TK]D-Fender | ~pb |
| 02:36.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 02:36.37 | pawpro | i know it why? |
| 02:36.57 | [TK]D-Fender | pawpro: Show us the debug of the call the works, and the one that fiails |
| 02:37.07 | pawpro | ok |
| 02:39.42 | *** join/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 02:40.02 | wwalker | it there documentation anywhere about the return codes of app_dial? There's lots of docs on what variable may be set by app_dial before it exits, but nothing that I can find on what it's return codes are. |
| 02:40.20 | wwalker | s/it/is/ |
| 02:48.12 | pawpro | Successfull call is here http://pastebin.com/d3e83e12f please mind that there is xlite client from 172.x.x.x connecting to Asterisk 1 |
| 02:48.29 | pawpro | and unsuccessful call between asterisks only http://pastebin.com/d3e83e12f |
| 02:48.32 | pawpro | ups |
| 02:48.44 | pawpro | http://pastebin.com/m14036b5d |
| 02:50.20 | pawpro | i set rtp timeout to 3 sec and so in second case you will find that Asterisk 2 (the one that answers the call and plays MOH) will hungup because of rtptimeout Nov 24 03:30:20] NOTICE[2648]: chan_sip.c:15732 do_monitor: Disconnecting call 'SIP/10.10.10.98-b50fcee8' for lack of RTP activity in 4 seconds |
| 02:51.45 | [TK]D-Fender | pawpro: checked your firewalls? |
| 02:51.47 | pawpro | Just for clarification Asterisk1 10.10.10.98, asterisk 2 10.10.10.99, xlite 172.x.x.x, (SER 10.10.10.83). Thank you very much for taking your time to look into my problem |
| 02:51.54 | pawpro | service iptebales STOP |
| 02:52.25 | pawpro | <PROTECTED> |
| 02:52.36 | [TK]D-Fender | Via: SIP/2.0/UDP 10.10.10.83;branch=z9hG4bK1638.c435c61.2 |
| 02:52.38 | [TK]D-Fender | Via: SIP/2.0/UDP 10.10.10.98:5060;received=10.10.10.98;branch=z9hG4bK5e825216;rport=5060 |
| 02:52.43 | [TK]D-Fender | 2 via's is also interesting |
| 02:52.53 | [TK]D-Fender | SER in the middle perhaps an issue |
| 02:53.10 | pawpro | only in second case? |
| 02:53.24 | [TK]D-Fender | pawpro: Try by removing hops. |
| 02:53.51 | [TK]D-Fender | pawpro: make things as simple as you can |
| 02:54.22 | [TK]D-Fender | pawpro: And when in doubt always disable reinvites, globally, and per peer |
| 02:56.21 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 02:58.24 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 02:58.40 | pawpro | I called straight to ASTERISK 2 and the call gets rtptimed out |
| 02:59.47 | pawpro | I mean instead of doing dial sip/x@ser i dialed sip/x@10.10.10.99 . SIP messages reflect that but there is no RTP |
| 03:00.25 | pawpro | it has to be something with the call being initiaded by autodialout |
| 03:10.46 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 03:11.22 | *** join/#asterisk ratmandu (n=ratmandu@173-24-253-160.client.mchsi.com) |
| 03:12.47 | ratmandu | I've got a TDM800p and im using asterisk 1.6+dahdi, and cannot seem to get it to dial out on an FXO line |
| 03:13.02 | ratmandu | I get [Nov 23 21:04:25] WARNING[12962]: app_dial.c:1450 dial_exec_full: Unable to create channel of type 'dahdi' (cause 0 - Unknown) |
| 03:13.21 | pawpro | everything works after playing sound after seting autodialout call |
| 03:13.29 | pawpro | thanks for your time |
| 03:15.57 | *** join/#asterisk fusss (n=chatzill@ip70-187-234-43.dc.dc.cox.net) |
| 03:17.03 | [TK]D-Fender | ratmandu: pastebi your failed attempt along with the output of "dahdi show status" and "dahdi show channels" |
| 03:17.05 | [TK]D-Fender | ~pb |
| 03:17.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 03:17.51 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1176162578.dsl.bell.ca) |
| 03:21.09 | ratmandu | [TK]D-Fender, http://pastebin.com/d695fc807 |
| 03:22.20 | [TK]D-Fender | ratmandu: please show that actual complete failed call... |
| 03:22.33 | [TK]D-Fender | ratmandu: verbose 10 |
| 03:22.58 | [TK]D-Fender | ratmandu: and your chan_dahdi.con |
| 03:24.53 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 03:27.14 | ratmandu | http://pastebin.com/d3f11f0e |
| 03:27.18 | ratmandu | added at the bottoms |
| 03:27.21 | ratmandu | -s |
| 03:29.06 | Carlos_PHX | [TK]D-Fender: Am I remembering correctly that you've used text to speech? |
| 03:29.35 | [TK]D-Fender | Carlos_PHX: Nope |
| 03:30.52 | [TK]D-Fender | ratmandu: Ok, please trash all comments from both of thsose files and repastebin and link in channel |
| 03:34.48 | yidiyuehan | Hi,everybody, |
| 03:35.04 | yidiyuehan | I found an interesting point, not sure related to asterisk or not. |
| 03:35.07 | yidiyuehan | <PROTECTED> |
| 03:35.33 | ratmandu | [TK]D-Fender, dahdi-channels.conf http://rafb.net/p/dXRdQv64.html |
| 03:35.46 | ratmandu | chan_dahdi http://rafb.net/p/uKnh5n93.html |
| 03:35.54 | *** join/#asterisk RobH (n=RobH@130-227.127-70.tampabay.res.rr.com) |
| 03:37.02 | [TK]D-Fender | ratmandu: I would be very sure first if your ports are configured as they should on the card itself. |
| 03:37.22 | ratmandu | yeah, they are |
| 03:38.05 | [TK]D-Fender | yidiyuehan: Why would you plug 1 line split into 2 ports? |
| 03:38.35 | [TK]D-Fender | ratmandu: try another port to dial out of. Perhaps that one is defective |
| 03:38.52 | yidiyuehan | D-Fender, ok, what I want is this: call in, ring one phone, upon talking to this phone, I want to press some DTMF tones like 888 to ring another phones, and three party will be in a conference. |
| 03:39.09 | ratmandu | i've tried 5, 6, and 7... not 8 yet... got kinda tired of climbing in the system closet |
| 03:39.18 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 03:39.45 | [TK]D-Fender | ratmandu: Tried swapping the modules? |
| 03:40.19 | [TK]D-Fender | yidiyuehan: What phone is doing the answering? |
| 03:40.31 | [TK]D-Fender | yidiyuehan: because this is not a multi-port thing.. |
| 03:41.08 | [TK]D-Fender | yidiyuehan: You never plug 1 line into 2 ports on a card like that |
| 03:41.46 | yidiyuehan | hi, D-Fender, I use a normal IP Phone for the answering part, and use an analog phone to dial to two FXO ports. |
| 03:42.24 | [TK]D-Fender | yidiyuehan: Why is an analog phone involved in this? |
| 03:42.45 | ratmandu | [TK]D-Fender, noI only have the 4 port FXS and the 4 port FXO in the card |
| 03:43.10 | ratmandu | I have tried moving them between areas, if that is what you mean |
| 03:43.12 | yidiyuehan | D-Fender, analog phone is the calling part, |
| 03:43.15 | [TK]D-Fender | ratmandu: Yes, you need to think a little outside the box... swap the ORDER of the modules.. maybe the BASE card has issues... |
| 03:43.24 | [TK]D-Fender | ratmandu: try to confirm by swapping your ports |
| 03:43.29 | ratmandu | yeah, I did that |
| 03:43.50 | yidiyuehan | D-Fender, analog phone ==> pstn line ==> splitter ==> 2 FXO ports ==> IP phone and IVR |
| 03:43.57 | [TK]D-Fender | ratmandu: Ok, then I'm a little short on ideas.... have you physically tested the lines going to your card with an analog phone? |
| 03:44.11 | ratmandu | yes |
| 03:44.12 | *** join/#asterisk moy (n=moy@187.133.5.154) |
| 03:44.18 | [TK]D-Fender | yidiyuehan: you do NOT plug 1 phone into 2 ports! |
| 03:44.33 | [TK]D-Fender | yidiyuehan: this idea makes no sense |
| 03:45.03 | [TK]D-Fender | yidiyuehan: And there is no "transfer" going on here either |
| 03:45.14 | [TK]D-Fender | yidiyuehan: Your concept seems quite broken. |
| 03:45.19 | yidiyuehan | D-Fender, as I don't know how to achieve this with a single port, call in, one phone answer and talk a while, remote party press 8888, second party will be auto involved and three parties are in a conference. |
| 03:45.34 | [TK]D-Fender | yidiyuehan: You need to start what it is you want to do, from the beginning. |
| 03:46.01 | [TK]D-Fender | yidiyuehan: this is a 3-way call.. it is not Raw-Cat Science. |
| 03:46.23 | [TK]D-Fender | yidiyuehan: You know that basically every kind of phone interface for * is natively capable of this, right? |
| 03:46.30 | [TK]D-Fender | yidiyuehan: Zaptel FXS, SIP phones, etc... |
| 03:46.43 | [TK]D-Fender | yidiyuehan: Phones don't get split between 2 ports for it |
| 03:48.01 | yidiyuehan | D-Fender, yes 3-way call I do know, however, what I need to do is: talk to first party, must press dtmf tone like 8888 to call for second party, and put all in a conference |
| 03:48.47 | jblack | Transfer party A to conference, call party B, transfer them to conference. |
| 03:48.55 | jblack | then, call the conference yourself. |
| 03:49.15 | [TK]D-Fender | yidiyuehan: Yes, a 3-way call is VERY basic. |
| 03:49.25 | [TK]D-Fender | yidiyuehan: so why the crazy cross-wiring? |
| 03:49.26 | yidiyuehan | jblack, then we need to use channelredirect am i right? |
| 03:49.32 | [TK]D-Fender | jblack: NO. |
| 03:49.45 | [TK]D-Fender | yidiyuehan: What is your misunderstanding here. |
| 03:49.51 | [TK]D-Fender | yidiyuehan: this is a SIMPLE feature. |
| 03:50.05 | [TK]D-Fender | yidiyuehan: No AMI, no AGI, no multiple ports, NOTHING. |
| 03:50.14 | [TK]D-Fender | yidiyuehan: Why are you complicating this? |
| 03:51.31 | yidiyuehan | D-Fender, maybe I do have a misunderstanding, but again what I want is: remote party (A) calls in, IP phone B ansers the call, after a while, A press digit (8888), here A only can press digit, no conference and any other features available, Party C will be called and A,B,C will be in a conference |
| 03:52.16 | jql | it's kinda like a magic trick, except more awesome |
| 03:52.29 | [TK]D-Fender | yidiyuehan: please clarify "party A" as being an analog phone on one of your Zap ports... |
| 03:52.55 | yidiyuehan | D-Fender, Party A is a remote phone calling from another side, |
| 03:53.04 | yidiyuehan | B & C are sitting behind * |
| 03:53.09 | [TK]D-Fender | yidiyuehan: that is an empty answer. FILL IT |
| 03:53.17 | [TK]D-Fender | "other side" <- |
| 03:53.34 | [TK]D-Fender | yidiyuehan: Please be spcecific, because solutions need to be. |
| 03:53.47 | yidiyuehan | D-Fender, analog phone is using a separate analog line which is not plugged into the * |
| 03:54.19 | [TK]D-Fender | yidiyuehan: Please try your description of the chain of this call in COMPLETE form from the beginning and we'll try this again. |
| 03:55.18 | [TK]D-Fender | yidiyuehan: yidiyuehan And syour last description of "analog phone is using a separate analog line which is not plugged into the *" tells me it must be using MAGIC to talk to * |
| 03:55.42 | [TK]D-Fender | yidiyuehan: So please understand the scope of the word "complete". |
| 03:55.47 | yidiyuehan | D-Fender, ok, the call flow is analog phone => PSTN Line A => Telecom => PSTN Line B => * => IP Phone B. And after that analog phone can only press dtmf digit 888, IP phone C will be invoked and A,B,and C will be in a conference. |
| 03:56.32 | yidiyuehan | D-Fender, i hope it's clear for you to understand? |
| 03:57.39 | yidiyuehan | D-Fender, analog phone is just a testing phone simply used to dial into the * from another telephone line. |
| 03:57.55 | [TK]D-Fender | yidiyuehan: Yes, this would have been must more simply said as "PSTN caller coming in on my Zap FXO hits *, navigates IVR and Dials a SIP device. I want the Zap FXO channel to be able to do a 3-way call." |
| 03:58.23 | [TK]D-Fender | yidiyuehan: the difference between FXO & FXS is huge here. |
| 03:59.45 | [TK]D-Fender | yidiyuehan: there is no features.conf option for 3-way call, only transfer. this makes things extremely difficult |
| 03:59.58 | [TK]D-Fender | yidiyuehan: I do not believe there is a clean way to do this. |
| 04:00.03 | yidiyuehan | D-Fender,sorry for my poor description. the key point here is, only analog phone can press digits to invites another party C. |
| 04:00.26 | [TK]D-Fender | yidiyuehan: Why would the PSTN caller be given this ability? Why wouldn't the SIP user on the insdie do it? |
| 04:01.02 | yidiyuehan | D-Fender, what I have tried is, inside features.conf, do a channelredirect to meetme, and call party C to this meetme as well. |
| 04:01.41 | [TK]D-Fender | yidiyuehan: Redirect will trash the other side of the call though... thats the problem. |
| 04:01.52 | [TK]D-Fender | yidiyuehan: And you can't really pass on other steps for it to do |
| 04:02.36 | yidiyuehan | D-Fender, this application is a kind of paging system for my HQ and branch office for emergency purpose, and that's the manual I read through.... |
| 04:02.45 | [TK]D-Fender | yidiyuehan: I do not see a way to make this work unfortunately. |
| 04:03.39 | [TK]D-Fender | yidiyuehan: But why is the PSTN interface given the power to cruise around like its a local pohne? Why would it have to transfer itself off of the call it got bridged to? |
| 04:03.44 | yidiyuehan | D-Fender, that's why I think of an idea to use a splitter, which is a virtual conference point which makes it much easier to achieve. unfortunately with one line plugging to two FXO ports, the IVR couldn't detect any DTMF tone... |
| 04:04.03 | [TK]D-Fender | yidiyuehan: If they want to use this 3rd party to do a "page", why involve that SIP phone in the first place? |
| 04:04.50 | [TK]D-Fender | yidiyuehan: You don't plug 1 physical line into 2 ports. That will screw stuff up and risk damaging equipment. |
| 04:04.59 | [TK]D-Fender | yidiyuehan: this is NOT a solution. |
| 04:07.02 | yidiyuehan | D-Fender, because they want to talk to somebody first, and then press some digit to activate the paging system.... |
| 04:07.38 | tzanger | [TK]D-Fender: it's amazing how people try to route around the solution |
| 04:07.44 | [TK]D-Fender | yidiyuehan: And the person they are talking to can't transfer them? |
| 04:07.48 | yidiyuehan | D-Fender, yes it may not make sense, the idea coming from a few phones with one PSTN Line for residential place. |
| 04:08.12 | yidiyuehan | D-Fender, unluckily the activation must be done from remote side.... |
| 04:08.29 | [TK]D-Fender | yidiyuehan: yidiyuehan because...? |
| 04:09.18 | yidiyuehan | D-Fender, actually it must be possible to activate from both sides, and i know from local side is not a problem, only the remote side part. |
| 04:09.51 | [TK]D-Fender | yidiyuehan: Just about the only option you have is a Redirect.... but that CANNOT 3-way this call. What you want cannot be done. |
| 04:10.16 | [TK]D-Fender | yidiyuehan: there is no mechanism that can keep that existing call up & bridged |
| 04:11.27 | yidiyuehan | D-fender, maybe I need to think about another way to achieve this, thank you for your clarification~~:_) |
| 04:12.56 | [TK]D-Fender | yidiyuehan: well we've just covered all there is. there is no Features.conf option for it, you can use features.conf to transfer the SIP device, but that drops the PSTN caller. You can use a dynamic feature to force-Redirect the Zaptel channel, but then you lose the SIP caller. No optiosn exist there... |
| 04:13.35 | [TK]D-Fender | tzanger: 2 words.... Marie Antoinette :) |
| 04:17.19 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:22.00 | tzanger | heh |
| 04:23.03 | [TK]D-Fender | scores another 3-pointer on his "historical drive-by" score-card |
| 04:23.18 | [TK]D-Fender | loads in another clip and chamers a round... |
| 04:23.23 | [TK]D-Fender | chambers* |
| 04:25.23 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:35.58 | Carlos_PHX | Hot tip for the night, if you ever have the choice between watching "Pan's Labyrinth" or dragging your nuts through salty broken glass, the choice is obvious. |
| 04:38.59 | jaytee | rofl |
| 04:39.07 | jaytee | that bad huh? |
| 04:43.14 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:46.54 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 04:51.12 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-13-32.phlapa.east.verizon.net) |
| 04:52.10 | Carlos_PHX | Worse. |
| 04:52.50 | jameswf | taking donations http://dontcallmyboss.blogspot.com/2008/11/wish-list.html |
| 05:06.54 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 05:17.21 | Carlos_PHX | Damn, that is a sweet gun. |
| 05:18.34 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 05:19.24 | drmessano | has an original taurus millenium |
| 05:23.06 | *** join/#asterisk chendy (n=chatzill@59.40.223.115) |
| 05:39.53 | jameswf | Carlos_PHX: have you been to the bass pro shop outdoor world in mesa... |
| 05:39.54 | *** join/#asterisk demonist (n=hi@jstickland.ca) |
| 05:40.02 | jameswf | flippin unreal hugr |
| 05:40.07 | jameswf | *huge |
| 05:42.16 | demonist | hello, is there an FXS card for the asterisk which will provide line and ring current for attached stations (for example: allow two modems , each attached to an FXS port to get dialtone and dial eachother through the asterisk pbx) |
| 05:42.25 | demonist | would a digium FXO/FXS card be good for this |
| 05:43.27 | demonist | i need to have dialup support in this lab for ccnp studies, and i dont want to spend money on an adtran. |
| 05:43.51 | drmessano | That would be FXS |
| 05:43.57 | drmessano | Errr |
| 05:44.04 | drmessano | You said that.. |
| 05:44.14 | drmessano | You _can_ but it wont work |
| 05:45.11 | drmessano | Generally a newer modem wont negotiate low enough to complete a connection over an FXS device |
| 05:45.36 | [TK]D-Fender | demonist: Now WHY would you want to do this? |
| 05:46.02 | drmessano | He needs to dial a modem connected to a router for CCNP I would guess |
| 05:46.22 | [TK]D-Fender | drmessano: When people seem crazy, I try not to guess... |
| 05:46.43 | Carlos_PHX | jameswf: Yes, it's pretty impressive. |
| 05:48.23 | demonist | drmessano, yes. |
| 05:49.00 | demonist | can you think of any other way to connect two modems togehter, without going over the pstn |
| 05:49.52 | demonist | or two routers ISDN interface without having an actual line on the ISDN net |
| 05:50.17 | drmessano | Faking ISDN will be even more fun |
| 05:51.29 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 05:54.41 | [TK]D-Fender | demonist: Again... WHY? |
| 05:57.11 | demonist | making a cisco lab here |
| 05:57.19 | demonist | people need to do their WAN labs |
| 05:57.29 | [TK]D-Fender | demonist: Well it might work, might not. |
| 05:58.26 | demonist | i know it can be done with adtran equipment |
| 05:58.33 | demonist | but that stuff is damn expensive |
| 05:59.12 | florz | demonist: I suppose a small second-hand analog pbx would be cheaper, easier, less power-consuming, more reliable? |
| 05:59.32 | demonist | can you recommend anything? |
| 05:59.36 | demonist | meridian pbx? |
| 05:59.39 | demonist | key system... |
| 05:59.46 | florz | demonist: I think it's called ebay |
| 05:59.58 | demonist | im banned from there |
| 06:00.07 | drmessano | .... |
| 06:00.09 | demonist | =( |
| 06:00.29 | drmessano | Considering how hard it is to get banned from eBay, that scares me |
| 06:00.44 | demonist | Nortel Meridian - $1,000 |
| 06:00.47 | demonist | but, thats a big one |
| 06:01.17 | drmessano | Surely you can find some black box PBX on there |
| 06:01.19 | demonist | that thing is an oven |
| 06:01.39 | florz | I mean, I don't know how many thousands of simultaneous connections you want to have, but ... |
| 06:02.51 | demonist | http://cgi.ebay.ca/Nortel-Meridian-Norstar-Telephone-System-w-5-Telephones_W0QQitemZ390004755259QQcmdZViewItemQQptZLH_DefaultDomain_2?hash=item390004755259&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318 |
| 06:03.14 | drmessano | http://cgi.ebay.com/Data-Labs-USA-DL-424-832-PBX_W0QQitemZ110312470742QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item110312470742&_trksid=p3286.c0.m14&_trkparms=72%3A1234|66%3A2|65%3A12|39%3A1|240%3A1318 |
| 06:03.18 | drmessano | There you go |
| 06:04.01 | demonist | why pay so much for that |
| 06:04.03 | drmessano | That one I posted uses POTS phones |
| 06:04.15 | *** join/#asterisk chendy (n=chatzill@59.40.223.115) |
| 06:04.26 | demonist | i dont think this meridian is isdn... |
| 06:04.33 | demonist | maybe it is |
| 06:04.33 | drmessano | A second Hand PBX will likely use some proprietary signalling |
| 06:04.36 | drmessano | and not POTS |
| 06:04.40 | demonist | ah |
| 06:04.48 | drmessano | Which is useless for your modem deal |
| 06:07.05 | drmessano | I wonder if a couple USR modems would work over a dedicated loop |
| 06:07.41 | drmessano | You could always get (4) 12v lantern batteries and make your own 48v DC circuit.. throw those bad boys on there |
| 06:07.43 | drmessano | heh |
| 06:07.56 | demonist | how about faking dsl |
| 06:08.04 | demonist | with a mini ip dslam |
| 06:08.17 | drmessano | Telco doesnt even fake DSL that well, what makes you think you can? |
| 06:08.36 | demonist | hah |
| 06:08.42 | demonist | with a mini ip dslam =) |
| 06:09.03 | demonist | and some dsl modems |
| 06:10.15 | drmessano | http://cgi.ebay.ca/Coastcom-Loop-IP-DSLAM-24-Port-1U-RackMount-Line-Access_W0QQitemZ150308015276QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item150308015276&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318 |
| 06:10.19 | drmessano | There you go |
| 06:10.26 | drmessano | Perfect for the home user needing a DSLAM |
| 06:10.27 | drmessano | Next? |
| 06:11.24 | demonist | thats nice |
| 06:11.38 | demonist | but what do the pairs connect with |
| 06:11.43 | demonist | one of those octopussy cables |
| 06:12.07 | [TK]D-Fender | demonist: Just asking makes us wonder if you're qualified :) |
| 06:12.29 | drmessano | RJ-21 |
| 06:12.58 | demonist | isnt it amphenol |
| 06:13.18 | demonist | like all the pairs would go to a punch down block and onto some amphenol connector |
| 06:13.31 | drmessano | Yes, an RJ21 |
| 06:13.36 | demonist | oh |
| 06:13.46 | demonist | i did not know of that jack registration =) |
| 06:14.00 | [TK]D-Fender | demonist: just keep digging... |
| 06:14.13 | drmessano | I may get that DSLAM to put in my rack |
| 06:14.22 | drmessano | Thats kinda hardcore |
| 06:14.27 | drmessano | No, not really |
| 06:15.02 | demonist | what, i never came in here claiming to be a telephony expert |
| 06:15.17 | drmessano | Hang on |
| 06:15.20 | demonist | here, i will admit "hi im a fucking lamer who knows nothing of telephony" |
| 06:15.24 | drmessano | This is a telelphony channel? |
| 06:15.27 | drmessano | err |
| 06:15.31 | drmessano | telephony |
| 06:15.32 | drmessano | Oh crap |
| 06:15.38 | demonist | well, its telephony related |
| 06:15.40 | [TK]D-Fender | OMGZ! |
| 06:15.40 | drmessano | I thought this was apache.. Damn names with A's |
| 06:15.52 | drmessano | Screw you guys, I get a web server to build |
| 06:15.55 | *** part/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 06:16.01 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 06:16.13 | drmessano | <--- LOL'ed, to himself. Quietly. |
| 06:16.53 | demonist | i wonder what dsl standard that dslam offers |
| 06:17.18 | demonist | time to get the fucking manual |
| 06:17.50 | demonist | if im going to get a dslam, might as be one i can use for something either than a cisco lab |
| 06:18.24 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 06:19.12 | demonist | well, forget this telephony stuff, im just going to hire some monkeys to close and open switches all day |
| 06:19.50 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 06:20.23 | demonist | well, with this dslam i would have to get a media converter, because its uplink is fibre |
| 06:20.38 | demonist | fibre uplink port, and its not weather hardened. |
| 06:28.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:30.35 | [TK]D-Fender | alrighty... checkout time, later all |
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| 06:58.26 | drmessano | (. ) (. ) |
| 06:58.32 | drmessano | ( .) ( .) |
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| 07:16.39 | loompek | i have a question.. in case i use Dial(SIP/1&SIP/2&SIP/3), in case of an incomming call when someone in the group answers, all the rest get a missed call.. is there any way to fix that? |
| 07:18.51 | drmessano | By deisgn |
| 07:18.53 | drmessano | design |
| 07:19.10 | drmessano | You dialed 3 extensions, 1 answered, so the other 2 *missed* it |
| 07:20.05 | ratmandu | "It's not a bug... It's a feature!" |
| 07:20.48 | drmessano | More like "It's doing exactly what you told it to do. Want it to stop? Tell it to do something else." |
| 07:22.27 | drmessano | I wonder if there is a SIP verb for "There, there.. it's ok, someone else got it. You didn't miss it, pookie.." |
| 07:27.56 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
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| 07:39.58 | hi365 | [groosly off topic]im trying to repin an iso, and having problems with comps.xml/metadata stuff. anyone ehre good with that? |
| 07:40.14 | baliktad | yes, it's called CANCEL |
| 07:40.22 | hi365 | hu? |
| 07:40.37 | baliktad | that was for drmessano |
| 07:41.07 | hi365 | :) |
| 07:41.21 | hi365 | cd .. |
| 07:41.29 | baliktad | ...wrong window? |
| 07:43.37 | drmessano | lol |
| 07:47.26 | *** join/#asterisk sack (n=sack@159.Red-83-46-156.dynamicIP.rima-tde.net) |
| 07:48.00 | loompek | well... |
| 07:48.06 | loompek | can it be done? |
| 07:49.39 | baliktad | loompek what phones are you using |
| 07:50.14 | loompek | umm... some lg nortel, some koncept ip phones, ... |
| 07:50.55 | baliktad | I had the same problem as you, using linksys SPA-942's |
| 07:51.21 | baliktad | linksys recently made available a new firmware version that allows you specify no missed call notifications |
| 07:53.59 | baliktad | also, asterisk 1.6 has this note: |
| 07:54.04 | baliktad | A new option to Dial() for telling IP phones not to count the call as "missed" when dial times out and cancels. |
| 07:54.11 | drmessano | I think 5.1.7 had that |
| 07:54.57 | baliktad | no, 5.1.7 only had the ability to enable/disable the missed call shortcut |
| 07:55.10 | drmessano | ok |
| 07:55.23 | baliktad | 6.1.3 is the first release to allow you to totally ignore missed calls |
| 07:55.54 | baliktad | it can be defined on a per-line basis as well, in case you have one private extension and one shared extension |
| 07:56.06 | drmessano | 6.1.3? |
| 07:56.22 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:57.28 | baliktad | http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US%2FLayout&cid=1169083356524&packedargs=sku%3D1138743806996&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=5652495017B03&displaypage=download#versiondetail |
| 07:57.40 | baliktad | blarg @ linksys's crappy url's |
| 07:57.59 | baliktad | 6.1.3(a) released for spa-942 30 sep 2008 |
| 07:58.18 | drmessano | Oh shit... |
| 07:58.44 | drmessano | Where were these firmwares 2 months ago |
| 07:58.58 | drmessano | 5.2 hadn't even been listed for the 941 |
| 07:59.03 | baliktad | it adds native SLA functionality and LDAP directory support as well |
| 07:59.05 | drmessano | Now they show them going back |
| 08:00.27 | drmessano | Hmm |
| 08:02.53 | baliktad | hmm, maybe asterisk 1.6 will solve loompek's problem without any changes |
| 08:03.02 | baliktad | SIP now adds a header to the CANCEL if the call was answered by another phone in the same dial command, or if the new c option in dial() is used. |
| 08:03.43 | trogs | the LDAP doesn't really work properly. |
| 08:04.03 | trogs | but the BLF/speeddials work great |
| 08:04.13 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 08:04.20 | baliktad | I haven't tried LDAP yet, saw reports of other people unable to get it to work |
| 08:04.48 | *** join/#asterisk km2 (n=x@mobile-166-217-228-154.mycingular.net) |
| 08:05.10 | baliktad | I tried to get the SLA support going, but the release notes are so sparse in documenting how it's supposed to work |
| 08:05.23 | drmessano | Yeah |
| 08:05.25 | trogs | ah |
| 08:05.26 | drmessano | I noticed |
| 08:05.33 | trogs | well if you search for spa932 configs |
| 08:05.36 | trogs | for the spa962 |
| 08:05.39 | trogs | it's exactly the same as that |
| 08:05.48 | trogs | looks like they just ripped the code from the 962 and threw it on the 942 |
| 08:06.22 | baliktad | trogs did you get it working with multiple 942's? |
| 08:06.23 | trogs | http://www.yourexodus.com/index.php?option=com_content&view=article&id=80:spa962-spa942-and-blf-busy-lamp-field&catid=45:articles&Itemid=88 |
| 08:06.54 | trogs | that details it. |
| 08:07.02 | baliktad | I read that page too :S |
| 08:07.25 | trogs | hmm, just works for me. |
| 08:07.30 | trogs | oh |
| 08:07.30 | baliktad | but I use regular *, no trixbox |
| 08:08.04 | trogs | you need the usr= and sub= parameters |
| 08:08.07 | trogs | to get it going. |
| 08:08.14 | trogs | and make sure you set the phone's mode to asterisk. |
| 08:09.00 | trogs | set the extension you're using for it to disabled, and share call appearance to shared |
| 08:09.19 | trogs | then your extended function should show something like |
| 08:09.21 | trogs | fnc=sd+blf+cp;usr=567@yourpbxip;sub=567@yourpbxip;nme=Daniel |
| 08:10.27 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:10.55 | trogs | i am using trix, but its like 2.0 or something (yeah, real old pbx, gotta upgrade that some time) .. maybe it does some magic though, dunno. |
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| 08:14.16 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 08:14.18 | joelsolanki | hi room |
| 08:14.26 | *** join/#asterisk getsmart (n=getsmart@host1-65-dynamic.55-82-r.retail.telecomitalia.it) |
| 08:15.12 | joelsolanki | i have extension registered to asterisk 4028539999 |
| 08:15.28 | joelsolanki | my extensions.conf looks like this. |
| 08:15.36 | joelsolanki | [incoming] |
| 08:15.37 | joelsolanki | exten => _4028539999X.,1,noop |
| 08:15.37 | joelsolanki | exten => _4028539999X.,n,Playback(hello-world) |
| 08:15.37 | joelsolanki | exten => _4028539999X.,n,Hangup |
| 08:15.49 | baliktad | please use a pastebin |
| 08:16.00 | loompek | there should be answer() |
| 08:16.04 | loompek | before playback |
| 08:16.04 | *** join/#asterisk YoShiKi_99 (n=yoshiki_@LAubervilliers-151-11-33-91.w193-251.abo.wanadoo.fr) |
| 08:16.07 | joelsolanki | oh sorry. |
| 08:16.08 | YoShiKi_99 | Hello :) |
| 08:16.25 | loompek | or i'm mistaken |
| 08:16.54 | joelsolanki | when i make call from 4028539999 to 4028539999 it says person you are dialing is not avialable. |
| 08:17.09 | baliktad | you know that the pattern _4028539999X. will only match extensions 12 digits or longer |
| 08:17.27 | joelsolanki | but when i change _4028539999X. to just _X. then it works |
| 08:17.37 | joelsolanki | yes i know baliktatd |
| 08:17.39 | baliktad | read up on dialplan matching |
| 08:17.53 | loompek | _ means the beginning, something like ^ in regexp matching.. rightr? |
| 08:17.59 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 08:18.03 | baliktad | _ means it's a pattern |
| 08:18.12 | baliktad | X means 0-9 |
| 08:18.17 | joelsolanki | yes |
| 08:18.20 | baliktad | . means the last character repeated 1 or more times |
| 08:18.24 | loompek | . means 0-infinity numbers |
| 08:18.28 | loompek | oo.. |
| 08:18.28 | baliktad | no |
| 08:18.29 | loompek | 1 or more |
| 08:18.31 | baliktad | . does not mean 0 |
| 08:18.44 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 08:18.54 | loompek | so the correct and prolly most accurate matching would be |
| 08:18.58 | loompek | _4028539999 |
| 08:19.12 | loompek | or just |
| 08:19.13 | loompek | 4028539999 |
| 08:19.27 | baliktad | yes |
| 08:19.45 | baliktad | no _, no X, no . |
| 08:20.01 | joelsolanki | so should i just keeep 4028539999 ?? |
| 08:20.04 | joelsolanki | and remove _ ? |
| 08:20.13 | baliktad | it depends on what you want to do |
| 08:20.29 | baliktad | if you want to match on that exact number, then you should use just that exact number |
| 08:20.39 | *** join/#asterisk LND (n=LND@fazer1-adsl.demon.co.uk) |
| 08:21.24 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:21.25 | joelsolanki | got it. |
| 08:21.25 | loompek | so why should i use _ in any case? |
| 08:21.27 | joelsolanki | let me try |
| 08:21.41 | baliktad | now would be a great time to refer to chapter 5 of TfoT "Dialplan Basics" |
| 08:21.49 | joelsolanki | cool it worked :) |
| 08:22.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:22.28 | joelsolanki | yes agree |
| 08:24.08 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 08:41.51 | *** join/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 08:42.34 | farah | hi all |
| 08:42.57 | farah | anyone knows how to run automatically and periodically a CLI command? |
| 08:43.24 | farah | i want to run the command "iax2 show netstats" every 1 minute for example |
| 08:43.29 | angryuser | farah: external script with cron |
| 08:43.43 | farah | what is cron? |
| 08:43.52 | angryuser | farah: what is internet ? |
| 08:43.57 | farah | lol |
| 08:44.25 | angryuser | farah: search google "linux cron" |
| 08:44.34 | farah | ok thank you |
| 08:44.50 | *** join/#asterisk mvanbaak (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 08:46.23 | farah | angryuser: got it..thank you very much |
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| 09:11.30 | joelsolanki | does any company provide custom sounds for asterisk as per our requirement ? |
| 09:13.19 | angryuser | joelsolanki: digium does |
| 09:14.14 | joelsolanki | ok good. i will talk to them. |
| 09:14.19 | joelsolanki | :) |
| 09:16.34 | angryuser | Let's say i will write a agi script in php and forward all incoming to that script, i have 1-n call's, the script is started separately for each call or i need to do something for multitasking ? |
| 09:17.37 | angryuser | i will have something like ->> read dtmf >> soome sql request's > play some audio > go to the queue |
| 09:17.39 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:18.32 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 09:36.30 | mvanbaak | angryuser: if you use normal Agi() calls then yes, the script will be started for every time you use Agi() |
| 09:36.38 | mvanbaak | if you want to scale, have a look at fastagi |
| 09:37.41 | angryuser | mvanbaak: thank's |
| 09:44.52 | *** join/#asterisk ElDios (n=ElDios@85-18-35-21.ip.fastwebnet.it) |
| 09:44.56 | ElDios | EHLO |
| 09:47.11 | loompek | 200 HI |
| 09:48.38 | ElDios | ^_^ |
| 09:49.13 | angryuser | mvanbaak: all this agi stuff is new to me, how should i configure interpreter of php ? add some alias in apache conf ? all this interaction between asterisk and php is someway not clear ;) |
| 09:49.50 | yang | hello loompek faxman :) |
| 09:50.01 | loompek | ssup |
| 09:50.18 | yang | look outside its SNOWING ! :) |
| 09:50.44 | ElDios | here too |
| 09:50.47 | ElDios | (near Milan) |
| 09:51.01 | yang | first snow |
| 09:51.11 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 09:51.35 | mvanbaak | angryuser: what I would do is create the daemon in pure PHP |
| 09:52.11 | SwK | angryuser, agi's in PHP dont run under apache they run standalone either as a daemon for FastAGI or as little more then a shell script if using normal AGI... |
| 09:52.54 | mvanbaak | well, actually what I would do is rewrite the agi in python or C, but that's just me ;) |
| 09:54.21 | mvanbaak | brb |
| 09:55.05 | *** join/#asterisk mvanbaak (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 09:55.44 | angryuser | mvanbaak: ah nice |
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| 10:08.59 | mark_csi | hi all, I'm hoping someone can help. I'm using asterisk 1.4.22 with a tdm815p. In my dialplan any incoming pstn just calls reception, but sometimes the phones are not ringing - in the cdr they are described as answered!?! |
| 10:11.06 | Maliuta | I put it to you that they are, infact, ringing. You just can't hear them :P |
| 10:12.24 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 10:12.40 | mark_csi | Maliuta - I think you're probably right ;-) |
| 10:14.12 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
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| 10:34.29 | ratmandu | what is the equivalent to using insecure=very in asterisk 1.6? |
| 10:35.40 | mvanbaak | insecure=invite,port |
| 10:36.42 | ratmandu | thanks |
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| 10:47.02 | mort_gib | I have to experiment with ISDN Bri cards... Is Digium cards any good?? |
| 10:47.27 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 10:54.01 | yang | mort_gib: sangoma, digium, xorcom are good, or any cheap HFC card would work to some extent |
| 10:55.22 | yang | mort_gib: those are being sold in stores as ISDN modems for very cheap |
| 11:26.08 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:27.17 | mort_gib | yang: I'm using Sangomas A500 cards, but they are causing issues... |
| 11:27.49 | mort_gib | Yang: -Or rather, I'm having issues in one country with the A500 cards |
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| 11:38.56 | mark_csi | Maliuta - I think you're probably right ;-) |
| 11:39.12 | mark_csi | oops wrong window |
| 11:40.00 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
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| 12:02.49 | Assid | heya |
| 12:12.18 | *** part/#asterisk iobug (n=gustav@93.123.102.99) |
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| 12:48.00 | *** part/#asterisk cvs (n=cvs@unaffiliated/cvs) |
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| 12:50.15 | hi365 | anyone using a snom m3? |
| 12:50.26 | *** part/#asterisk Flawless (n=chrivers@zeus.sikkerhed.org) |
| 12:52.55 | mort_gib | hi365: Yes |
| 12:53.19 | hi365 | mort_gib: did you get paging (via sip headers) working with it? |
| 12:53.32 | mort_gib | hi365: Paging?? |
| 12:53.44 | hi365 | agins/intercom/auto answer |
| 12:53.51 | hi365 | *paging |
| 12:53.56 | mort_gib | :-) No, haven't played with that |
| 12:54.16 | mort_gib | Had BIG issues before I upgraded to latest FW though |
| 12:54.30 | puppet | I got an idea, just want to check if you think it is correc in here. My phone dies pretty quick when I use SIP, i have had qualify on now, turn that off should increase it right? |
| 12:54.35 | WimpMan | Soso |
| 12:54.35 | hi365 | mort_gib: what version are you on? |
| 12:54.42 | WimpMan | Oops |
| 12:56.05 | mort_gib | snom-m3-SIP/01.16//03-Jul-08 13:43 |
| 12:56.25 | hi365 | same here - yet no paging |
| 12:57.24 | mort_gib | To be honest I haven't had the need... What is paging supposed to give you?? |
| 12:58.08 | *** join/#asterisk ccesario_ (n=ccesario@189.20.219.10) |
| 12:58.15 | hi365 | the ability to have the phone auto-answered when you call it. think: "<beep> Jane, please bring me a coffee <hangup>" |
| 12:58.38 | hi365 | instead of actualy wating for the dumb secretary to figure you that the phoe is ringing and answer it |
| 12:58.58 | hi365 | (a small price to pay for having blond secretaries) |
| 12:59.02 | *** join/#asterisk sasargen_ (n=chatzill@68-244-127-44.pools.spcsdns.net) |
| 12:59.22 | *** join/#asterisk tmjb (n=tane@fit-211.yubc.net) |
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| 13:00.11 | shazaum | hi guys |
| 13:04.50 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:05.00 | tmjb | hello I have one way audio situation here is the setup ipphone1 ---nat ---internet ---asterisk and firewall server --- ipphone2. What happens when i call ipphone2 i can hear them but they can not here me. on the asterisk server i open all ports 5060 udp/tcp and 10000-20000. Any ideas how to solve this .Thank you |
| 13:05.32 | Maliuta | ~sipnat |
| 13:05.33 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:05.54 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 13:08.12 | mort_gib | hi365: :-) Yeah, telephony for blondes... |
| 13:09.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:10.37 | *** join/#asterisk plasmid (n=noway@c-71-225-10-10.hsd1.pa.comcast.net) |
| 13:10.47 | plasmid | I get this on my e-mail now: |
| 13:10.47 | plasmid | Dial *98 to access your voicemail by phone. |
| 13:10.47 | plasmid | Visit http://AMPWEBADDRESS/recordings/index.php to check your voicemail with a web browser. |
| 13:10.58 | puppet | plasmid: www.google.com |
| 13:11.00 | plasmid | how do I access my voicemail anywhere ? |
| 13:11.08 | coppice | blondes don't need special telephony, but idiots might |
| 13:11.09 | tmjb | jbot, tnx |
| 13:11.14 | plasmid | yes i know.. google is my best friend. |
| 13:11.24 | puppet | plasmid: go there then? |
| 13:11.38 | puppet | plasmid: I did and foud it on 20 sec |
| 13:11.47 | plasmid | that's commendable puppet. |
| 13:12.45 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 13:12.55 | plasmid | i did 2, but that's internal ip address. I need to access it worldwide. |
| 13:12.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:12.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:13.24 | puppet | plasmid: well then you google on howto portforward on your router |
| 13:14.50 | plasmid | ok... if I type: http://192.168.1.120/recordings/index.php i get this: The requested URL /recordings/index.php was not found on this server. |
| 13:14.54 | plasmid | so it's not working internally. |
| 13:15.06 | plasmid | heh.. at least I am getting the e-mail to my gmail acct. |
| 13:15.30 | puppet | plasmid: installed it your self or using trixbox or something like that? |
| 13:15.49 | plasmid | no.. i didn't use trixbox or something like that. |
| 13:17.06 | puppet | then u need to install ampportal |
| 13:20.17 | plasmid | oh. darn.. Lol. Maybe I didn't install ampportal.. googling it now. |
| 13:20.56 | mark_csi | hi all, I've added a few options to etc/modprobe.conf for asterisk - does anyone know how to reload this without rebooting the server? |
| 13:25.05 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 13:25.25 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.199.65) |
| 13:27.47 | plasmid | hmm.... amportal is already in my system: amportal start -> asterisk already running. |
| 13:28.26 | plasmid | yet I can't access 192.168.1.120/recordings/index.php |
| 13:31.32 | feeds | mark_csi: reload? |
| 13:31.42 | feeds | mark_csi: I mean try reload |
| 13:33.38 | mark_csi | feeds: thx |
| 13:34.33 | feeds | mark_csi: No problem ;) |
| 13:34.58 | *** join/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 13:35.36 | HeMan | Hi! We get "socket_read: Out of idle IAX2 threads for I/O, pausing!" and then a lot of "channel.c:1068 in channel_find_locked: Avoiding initial deadlock for channel '0x618050'" |
| 13:35.50 | HeMan | after that our asterisk-machine dies |
| 13:36.00 | HeMan | any ideas what this could be? |
| 13:39.20 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.90) |
| 13:41.01 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:41.12 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 13:42.29 | mark_csi | HeMan: sounds like it could be a bug, what version? |
| 13:43.13 | HeMan | 1.4.18 |
| 13:44.09 | HeMan | I know that 1.4.22 is out but I haven't had the time to update |
| 13:45.03 | mark_csi | HeMan: I'm using IAX on 1.4.22 without issue |
| 13:45.33 | HeMan | mark_csi: we only use IAX to IAXMODEM and hylafax there |
| 13:46.26 | HeMan | but we have problem with hylafax to so I think I'll just disable our fax capabilities for now |
| 13:46.47 | mark_csi | HeMan: ya got me there, could it be that your iax channels in the dialplan have no hangup() |
| 13:48.08 | HeMan | mark_csi: we never call out with the fax, just inbound |
| 13:48.22 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 13:49.11 | *** join/#asterisk protocols (n=protocol@p5791FEDD.dip.t-dialin.net) |
| 13:49.24 | HeMan | mark_csi: and I was hoping that iaxmodem handled the hangup |
| 13:51.23 | mark_csi | HeMan: I'm not sure, I always end a dialplan with a hangup() anyway - not sure if it's good practice or not |
| 13:52.18 | HeMan | I try that and install 1.4.22 on our testserver |
| 13:52.38 | mark_csi | HeMan: sorry I couldn't be more helpful |
| 13:52.56 | HeMan | mark_csi: np |
| 13:54.28 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 14:04.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:07.27 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:09.20 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:12.17 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 14:17.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.30 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 14:33.15 | tzanger | hmm, isn't 'F' a hookflash in a Dial() command? |
| 14:33.27 | tzanger | e.g. Dial(Zap/1/F*69) type thing? |
| 14:34.09 | lmadsen | I've never heard of that... |
| 14:34.18 | lmadsen | not that it doesn't exist :) |
| 14:34.52 | *** join/#asterisk stencil_ (i=asr33@unaffiliated/stencil) |
| 14:37.16 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 14:37.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:37.47 | *** join/#asterisk joelsolanki (i=joelsola@124.125.150.158) |
| 14:42.18 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 14:42.34 | Dovid | I know this is OT but does anyone here know vicidial ? |
| 14:42.50 | patrick-- | hey all, im trying to build agx-asterisk-addons , but keep getting this error: http://pastebin.com/ma49ba30 anyone got an idea on that? |
| 14:43.00 | tzanger | lmadsen: it doesn't exist |
| 14:43.21 | tzanger | Dial(Zap/1,,M(foo)) and [macro-foo] with Flash and SendDTMF() works nicely though |
| 14:46.15 | Carlos_PHX | Brrr...had to fire up the heater for the first time this year. Got up and it was down to 72 in the house. The humanity. |
| 14:46.38 | lmadsen | tzanger: ok, that's what I was thinking :) |
| 14:48.55 | neurosys | Is it usual that some switches will change the default way they accept peer auths, then try to charge you 150 an hr to get it "working" ? |
| 14:49.52 | tzanger | neurosys: sounds like a valid business model |
| 14:50.16 | neurosys | tzanger: hmm :( |
| 14:50.17 | [TK]D-Fender | patrick--: What ver of * are you running? |
| 14:50.42 | Assid | anyone by chance know anyone in asterlink? |
| 14:50.45 | patrick-- | Asterisk 1.4.18, |
| 14:51.06 | [TK]D-Fender | patrick--: http://sourceforge.net/project/showfiles.php?group_id=209138 <- say .13 & .17 supported |
| 14:51.09 | Assid | their registration server is down.. and i cant get through their toll free either.. email is pretty much useless |
| 14:51.12 | [TK]D-Fender | patrick--: Try one of those |
| 14:51.21 | lmadsen | Assid: find bkw_ I think |
| 14:51.21 | patrick-- | right, thanks |
| 14:51.29 | patrick-- | mhh |
| 14:51.33 | patrick-- | ill have to downgrade? :D |
| 14:51.35 | Assid | lmadsen: as i am told.. he no longer is with them |
| 14:51.42 | lmadsen | ahhh, then I have no idea |
| 14:52.11 | patrick-- | [TK]D-Fender: is a downgrade necessary? |
| 14:53.01 | [TK]D-Fender | patrick--: Well they say very specific versions that it is for. Would you ignore something that is written that clearly? |
| 14:53.42 | patrick-- | no, actually not :D |
| 14:54.56 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:55.24 | feeds | In sip.conf, what do I need to enter to subscribe client 1000 to mailbox 12345@employees? |
| 14:56.41 | feeds | I mean is it mailbox= or vmexten= |
| 14:56.42 | feeds | ? |
| 14:56.57 | [TK]D-Fender | feeds: mailbxo- |
| 14:57.04 | [TK]D-Fender | feeds: mailbox= |
| 14:57.21 | feeds | [TK]D-Fender: thanks. |
| 15:04.48 | *** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk) |
| 15:06.55 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:07.56 | stmaher | hello everyone.. I have a new asterisk install .. but monkeys doesnt work.. I have my entry and the phone picks up |
| 15:08.22 | guax | load chan_banana.so |
| 15:09.28 | stmaher | load chan_banana.so ??? |
| 15:09.29 | [TK]D-Fender | stmaher: pastebin is your friend... show us.. |
| 15:09.31 | [TK]D-Fender | ~pb |
| 15:09.31 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:09.33 | [TK]D-Fender | ^^^ |
| 15:09.53 | guax | stmaher: to the monkeys =P |
| 15:11.17 | stmaher | [TK]D-Fender heya.. There isnt much to paste.. Logs dont show anything.. and all i have is exten => 704,1,BackGround(tt-monkeys) |
| 15:11.58 | [TK]D-Fender | stmaher: Screw logs, you should be looking at CLI. set verbose 10 , and enable SIP debug if thats the protocol your phone is using |
| 15:12.35 | guax | or watch for the full log if its enabled |
| 15:13.11 | stmaher | Ok.. fyi.. a tcpdump shows only rtp going to the asterisk box.. and not to the phone.. |
| 15:13.14 | stmaher | There is no firewall |
| 15:13.40 | stmaher | <PROTECTED> |
| 15:13.40 | stmaher | <PROTECTED> |
| 15:13.51 | *** join/#asterisk RobH (n=RobH@130-227.127-70.tampabay.res.rr.com) |
| 15:14.42 | stmaher | invite/200 OK shows they both agree on g711 Ulaw |
| 15:15.33 | [TK]D-Fender | stmaher: PASTEBIN the actual call with SIP debug as requested |
| 15:15.38 | stmaher | ok thanks |
| 15:16.47 | stmaher | [TK]D-Fender http://www.pastebin.ca/1265842 thanks |
| 15:17.57 | [TK]D-Fender | stmaher: Now change your first priotiry to Answer. |
| 15:18.04 | [TK]D-Fender | priority* |
| 15:19.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:20.19 | stmaher | [TK]D-Fender Same result http://www.pastebin.ca/1265844 |
| 15:25.14 | [TK]D-Fender | stmaher: There is no ANSWER in there |
| 15:25.46 | [TK]D-Fender | stmaher: And what are you doing on that ancient version? |
| 15:25.49 | stmaher | [TK]D-Fender eh.. there should be.. |
| 15:25.59 | stmaher | [TK]D-Fender debians apt-get |
| 15:26.20 | stmaher | [TK]D-Fender reason.. Hoping I wouldnt have to do any zaptel driver install.. |
| 15:26.35 | stmaher | [TK]D-Fender or the newer / more confuzing dahdi? |
| 15:26.45 | [TK]D-Fender | stmaher: Invalid approach to an unnecessary process |
| 15:27.20 | [TK]D-Fender | stmaher: Zaptel/DAHDI is *option* Always has been. They are SEPARATE packages |
| 15:27.33 | stmaher | [TK]D-Fender yeah I need to get a digium card working with it too |
| 15:27.38 | guax | stmaher: try this: exten => 704,1,Answer exten => 704,n,BackGround(tt-monkeys) |
| 15:27.50 | stmaher | guax thats what I have.. |
| 15:27.52 | [TK]D-Fender | stmaher: HUH!? |
| 15:28.24 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
| 15:28.34 | stmaher | [TK]D-Fender Never mind thats another days problem |
| 15:28.38 | guax | update your log, its not what he is telling us. (remember dialplan reload(i do forget sometimes)) |
| 15:28.52 | stmaher | guax thanks ill try that |
| 15:29.14 | [TK]D-Fender | stmaher: If you have a card, then you need one or the other. If you ahve no card, you only need them if you need a timing source for * apps |
| 15:29.17 | guax | needs english classes |
| 15:29.56 | andrebarbosa | anyone has a tdm400 detecting verizon's callerid? |
| 15:30.20 | [TK]D-Fender | andrebarbosa: Should work fine.. |
| 15:30.29 | andrebarbosa | not working for me |
| 15:30.42 | [TK]D-Fender | andrebarbosa: PASTEBIN is your friend, show us your configs. |
| 15:30.43 | [TK]D-Fender | ~pb |
| 15:30.44 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:30.47 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:31.38 | andrebarbosa | I've cidstart=ring and cidsignalling=bell |
| 15:31.42 | andrebarbosa | and usecallerid=yes |
| 15:32.00 | andrebarbosa | but already tried, dtmf, v23 |
| 15:32.18 | [TK]D-Fender | andrebarbosa: Also need "callerid=asreceived" <--- |
| 15:32.32 | andrebarbosa | let me check that one |
| 15:32.38 | andrebarbosa | oh |
| 15:32.58 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:33.06 | andrebarbosa | I'm using 1.2.29 and latest zaptel 1.2 |
| 15:33.28 | [TK]D-Fender | andrebarbosa: This parm predates 1.2 |
| 15:34.39 | stmaher | [TK]D-Fender guax http://www.pastebin.ca/1265851 thanks |
| 15:34.49 | andrebarbosa | i'll pastif |
| 15:34.52 | andrebarbosa | just a second |
| 15:35.00 | feeds|Busy | Is there any way to change the vm-intro path in the config? |
| 15:35.28 | feeds|Busy | because I know what my issue is I just don't know how to solve it ^^ |
| 15:35.32 | [TK]D-Fender | stmaher: Where is the CALL!? |
| 15:35.58 | [TK]D-Fender | feeds|Busy: it looks in the lib/sounds folder.... |
| 15:35.58 | andrebarbosa | http://pastebin.com/m1a2540c3 |
| 15:36.05 | [TK]D-Fender | feeds|Busy: as specified in asterisk.conf. |
| 15:36.19 | stmaher | [TK]D-Fender its at verbose 10 and sip debug.. |
| 15:36.26 | stmaher | [TK]D-Fender is there anything else I could give you ? |
| 15:36.27 | [TK]D-Fender | stmaher: there is NO call in there. |
| 15:36.41 | stmaher | [TK]D-Fender Yes there is.. SIP signally? |
| 15:36.55 | stmaher | *signalling |
| 15:37.16 | [TK]D-Fender | stmaher: NVM. I see no dialplan apps being called. Why? |
| 15:37.23 | guax | stmaher: open your log, and run a core set verbose 15, then make the call and paste the log, |
| 15:37.58 | *** join/#asterisk kc2tnk (n=fskrotzk@host.textwise.com) |
| 15:38.20 | feeds|Busy | [TK]D-Fender: I have these directories: http://asterisk.pastebin.com/m562f3048 |
| 15:38.48 | [TK]D-Fender | andrebarbosa: cidsignalling=bell cidstart=ring <- remove. And you need to restart * for changes to take effect |
| 15:39.01 | andrebarbosa | ok |
| 15:39.04 | andrebarbosa | remove ring |
| 15:39.09 | andrebarbosa | what is the one by default? |
| 15:39.21 | [TK]D-Fender | feeds|Busy: astvarlibdir => /var/lib/asterisk <- under "sounds" in there |
| 15:39.24 | stmaher | [TK]D-Fender guax http://www.pastebin.ca/1265857 |
| 15:39.28 | [TK]D-Fender | andrebarbosa: Just do it |
| 15:39.36 | andrebarbosa | :p |
| 15:39.36 | [TK]D-Fender | andrebarbosa: BOTH |
| 15:39.41 | andrebarbosa | ah ok |
| 15:39.45 | andrebarbosa | both |
| 15:40.19 | feeds|Busy | [TK]D-Fender: so I should do Playback(sounds/vm-intro) for example, if I want to play vm=intro? |
| 15:40.53 | [TK]D-Fender | feeds|Busy: no, its in the base folder. remove your path call. And why are you looking to play it directly? |
| 15:42.33 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:42.48 | andrebarbosa | [TK]D-Fender, didn't work |
| 15:42.58 | andrebarbosa | Caller ID: (N/A) |
| 15:43.04 | feeds|Busy | [TK]D-Fender: I want to let after 15 secs play the VoiceMail() app. And it prints: http://asterisk.pastebin.com/fe809712 |
| 15:43.15 | [TK]D-Fender | andrebarbosa: Go plug an analog phon on your line and prove it even has it functional |
| 15:43.32 | andrebarbosa | already did |
| 15:43.34 | andrebarbosa | it works |
| 15:43.50 | [TK]D-Fender | feeds|Busy: You are missing the basic sound files that should come with *. |
| 15:43.57 | [TK]D-Fender | feeds|Busy: Go reinstall |
| 15:45.09 | [TK]D-Fender | feeds|Busy: Or go to asterisk.org and check the HTTP Download server and manually grab the tarball for the codec version you want and install it yourself |
| 15:45.45 | feeds|Busy | I have the sounds: http://asterisk.pastebin.com/f3cc7616c |
| 15:45.52 | feeds|Busy | bu tonly .gsm |
| 15:45.58 | feeds|Busy | * but only |
| 15:46.53 | [TK]D-Fender | feeds: Go check your permissions, and never show a folder like that without the proof of the precise PATH it represents |
| 15:46.53 | stmaher | [TK]D-Fender I also tried chaging the localnet.. but no difference.. |
| 15:47.03 | stmaher | [TK]D-Fender asterisk is not sending any rtp to the phoen |
| 15:47.09 | [TK]D-Fender | stmaher: I am still NEVER seeing a good sample. |
| 15:47.24 | [TK]D-Fender | stmaher: Where's the good apstbin witht he matching dialplan output? |
| 15:47.35 | stmaher | [TK]D-Fender what verbosity would you like? |
| 15:47.39 | stmaher | ive done 10 and 15? |
| 15:47.50 | stmaher | http://www.pastebin.ca/1265857 |
| 15:47.54 | [TK]D-Fender | stmaher: You have clearly done something wrong. 10 shoud do. |
| 15:47.56 | *** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 15:48.14 | stmaher | [TK]D-Fender Doubt it.. its a fresh install.. |
| 15:48.25 | [TK]D-Fender | stmaher: do YOU see the "answer" being called... or ANY dialplan app? I know I don't |
| 15:49.02 | stmaher | [TK]D-Fender Ok.. I have it at verbosity 15.. what LEVEL would you like it at.. if it doesnt show what you need.. can you please suggest another command |
| 15:49.32 | [TK]D-Fender | stmaher: "set verbose 10" |
| 15:49.41 | feeds | [TK]D-Fender: I know the problem. Normally, when I want to Playback a file in any exten, I have to write not Playback(xyz) but Playback(/var/lib/asterisk/sounds/xyz). Can't I change that in voicemail.conf and everything will be ok.... I just need to know how... |
| 15:49.50 | [TK]D-Fender | stmaher: Something is horribly wrong if its at 10 and you don't see dialplan execution. |
| 15:50.05 | [TK]D-Fender | feeds: You have bad paths <- |
| 15:50.23 | [TK]D-Fender | feeds: Now do as I said and go prove the exact path of that folder dump you just handed me. |
| 15:51.06 | feeds | [root@gandalf sounds]# pwd: /var/lib/asterisk/sounds |
| 15:51.37 | [TK]D-Fender | feeds: and PERMISSIONS. Do not make this a blow-by-blow process... |
| 15:52.22 | feeds | chown is asterisk and chgrp is asterisk too |
| 15:55.23 | *** join/#asterisk cianmaher (n=Cian@84.203.233.126) |
| 15:55.44 | andrebarbosa | [TK]D-Fender, i've look at chan_zap.c and bell and ring are the defaults values |
| 15:55.55 | andrebarbosa | so have them or not is the same |
| 15:56.48 | andrebarbosa | this config should be working.. maybe i've a problem with something else.. :( |
| 15:57.37 | cianmaher | Hi all, I dont think the following is possible but here it goes... I need callers to be able to navigate through available agents who are assigned to a queue but not on a call. Does anyone know of anyway to do this? Or if anyone would be interested in doing development for me? |
| 16:01.59 | mort_gib | cianmaher: Let the caller select the queue they want using ivr |
| 16:02.05 | mort_gib | Easy |
| 16:02.37 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-46e65ec17e34f6c8) |
| 16:02.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:05.29 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:07.13 | *** join/#asterisk justin_work (n=justin@74.7.161.194) |
| 16:07.21 | justin_work | Morning, folks |
| 16:07.38 | *** join/#asterisk funxion (n=funxion@63.214.236.169) |
| 16:08.21 | justin_work | Trying to figure something out - if I set an extension to "ring all" multiple outside lines, users calling that extension get dead air until someone answers - no ring sound. Is there a way to remedy this? |
| 16:09.49 | beek | justin_work: You have the 'r' parameter in the dial statement? |
| 16:10.08 | [TK]D-Fender | andrebarbosa: Prove it with a real phone |
| 16:10.20 | justin_work | Not sure - it's a Switchvox, so I don't have as direct access to the settings as I'd like |
| 16:11.40 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:12.00 | beek | justin_work: Sorry -- I haven't worked with a Switchvox so I can't offer anything else. |
| 16:13.17 | justin_work | Thanks, beek |
| 16:14.20 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 16:16.26 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 16:20.31 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 16:26.01 | *** part/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:26.11 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:26.36 | [TK]D-Fender | cianmaher: you want "agents" without using normal Queues? |
| 16:26.38 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:27.09 | anonymouz666 | I wonder why someone would use chan_agent |
| 16:28.18 | [TK]D-Fender | anonymouz666: more loggable and hot-deskable... |
| 16:29.20 | anonymouz666 | more loggable? |
| 16:29.31 | *** join/#asterisk ecrist (n=ecrist@chunk.ip6.secure-computing.net) |
| 16:29.36 | ecrist | anyone have a hosted voip provider they recommend? |
| 16:29.38 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:29.43 | anonymouz666 | there's nothing you can't do using read() and some DB queries :) |
| 16:29.48 | anonymouz666 | using the addqueuemember |
| 16:30.13 | ecrist | currently with ironvoice (formerly heavylogic), but their service is less than dependable |
| 16:30.36 | [TK]D-Fender | anonymouz666: No argument here...jsut another way I guess. Still waiting on a solid answer for his goals of course |
| 16:31.11 | [TK]D-Fender | ecrist: What is it that you mean by "hosted voip provider" exactly? |
| 16:31.13 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.2.87) |
| 16:31.28 | ecrist | our phones connect to their server, via the net |
| 16:32.09 | [TK]D-Fender | ecrist: This is #asterisk you know... we tend to run our OWN servers here... |
| 16:32.21 | ecrist | [TK]D-Fender: I'm fully aware. |
| 16:32.39 | ecrist | but, I also know many folks in here help *run* such systems for these providers |
| 16:32.55 | jpcansa | hi, how can i restrict one extension or a group of extensions from making outgoing calls via certain trunks, while allowing unrestricted access to other extensions? |
| 16:33.28 | mort_gib | jpcansa: Context |
| 16:34.30 | jaytee | ~book |
| 16:34.31 | jbot | rumour has it, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 16:34.52 | *** join/#asterisk abalashov (n=sasha@97.81.69.51) |
| 16:35.30 | jaytee | jpcansa, ^^^^^^^^^ pg 119 Chapter 5 |
| 16:35.46 | jpcansa | thks |
| 16:35.50 | jaytee | yw |
| 16:37.06 | jameswf | omfg |
| 16:37.32 | jameswf | bbq idk my bff jill |
| 16:38.29 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:38.56 | mark_csi | hi all, I've an issue with incoming zap calls not ringing my reception phones. no errors logged and it's not even consistent |
| 16:39.10 | mark_csi | only highlighted in cdr database |
| 16:39.32 | etfonhomey | mark_csi, sounds like my first experience with analog FXO cards. |
| 16:40.23 | mark_csi | etfonhomey: what did you do in the end? |
| 16:41.38 | tzafrir_laptop | mark_csi, err... what phones are those? |
| 16:41.50 | tzafrir_laptop | What do you see in the CLI trace? |
| 16:41.50 | etfonhomey | etfonhomey, I bought a different brand of card. But you mention that it's not ringing your reception phones, there are a ton of variables between the call coming in a zap channel on your * box to correctly routing it to your internal extensions. |
| 16:42.35 | tzafrir_laptop | etfonhomey, don't jump to conclusions. At the moment it's ENODATA |
| 16:43.22 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 16:43.41 | mark_csi | etfonhomey: it's a real simple dialplan - exten => s,1,Dial(SIP/801&SIP/802&SIP/803,90) |
| 16:43.57 | mark_csi | after that it's just hangup() |
| 16:44.37 | mark_csi | tzafrir_laptop: they are 3 types (we're experimenting) Polycom, Snom and Linksys |
| 16:44.54 | etfonhomey | mark_csi, stick with the Polycom. |
| 16:45.16 | mark_csi | etfonhomey: I think I prefer it to the others anyway |
| 16:45.28 | etfonhomey | mark_csi, simply your dialplan even more and just dial one extension. |
| 16:45.49 | tzafrir_laptop | mark_csi, again, what do you see in the CLI trace? |
| 16:45.56 | tzafrir_laptop | core set verbose 3 |
| 16:45.58 | tzafrir_laptop | (or more) |
| 16:46.01 | etfonhomey | mark_csi, if that still doesn't work, "core set verbose 10", call it and then pastebin the results. |
| 16:46.07 | Qwell | mark_csi: beep |
| 16:46.16 | putnopvut | Qwell: you jerk |
| 16:46.20 | Qwell | putnopvut: <3 |
| 16:46.31 | *** part/#asterisk abalashov (n=sasha@97.81.69.51) |
| 16:46.54 | mark_csi | etfonhomey/tzafrir: will do |
| 16:47.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:48.29 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:49.49 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:52.39 | *** join/#asterisk johann8384 (n=johann83@intra.netlogic.net) |
| 16:53.35 | mark_csi | guys - it's difficult to check this it doesn't do it all the time. |
| 16:54.55 | etfonhomey | mark_csi, Add some NoOps to show the Caller ID information. I'm just curious. When I was having intermittent problems like this, the times when the call would fail, there would be no CallerID information detected (or sent) from the incoming call. |
| 16:55.40 | *** part/#asterisk hummb (i=anon@theos.org) |
| 16:55.55 | mort_gib | I have one installation where I don't get ANY CallerID from incoming calls |
| 16:56.01 | mort_gib | -Very strange |
| 16:57.00 | mark_csi | etfonhomey: I think you are onto something as it seems to be callerid related |
| 16:57.38 | mark_csi | etfonhomey: do I just put NoOps() and position 1 then everything else after? |
| 16:58.59 | etfonhomey | mark_csi, I never did figure out the issue. I eventually had a port go bad on my card. |
| 17:05.14 | *** join/#asterisk hi365_m (n=hi365@213.151.43.64) |
| 17:05.54 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 17:07.46 | *** join/#asterisk khronos (i=tom@sip02.kineticvoip.com) |
| 17:08.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:08.28 | Micc | Where do I setup the authentication for subscribe? |
| 17:08.40 | mark_csi | etfonhomey: this is what I get in the cdr: http://www.pastebin.ca/1265902 |
| 17:09.24 | *** join/#asterisk synchris (n=synchris@athedsl-100148.home.otenet.gr) |
| 17:09.38 | [TK]D-Fender | Micc: what "subscribe"? |
| 17:10.42 | etfonhomey | mark_csi, what are the 2nd and 4th columns? |
| 17:10.50 | Micc | TKD-Fender, I"m trying to setup BLF support, but i keep getting an authentication error for subscribe. |
| 17:10.57 | khronos | <PROTECTED> |
| 17:11.14 | mark_csi | etfonhomey: sorry they are src and billsec |
| 17:12.27 | etfonhomey | mark_csi, src = channel id? |
| 17:12.36 | [TK]D-Fender | Micc: Go read about the "hint" priority" on the WIKI under "presence" |
| 17:12.44 | [TK]D-Fender | khronos: You don't say! |
| 17:13.52 | mark_csi | etfonhomey: src is populated with the callerid when possible. not sure of the mappings |
| 17:13.55 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:14.18 | Micc | TKD-Fender, I already read that but I'll go over it again just in case I missed something. |
| 17:14.21 | etfonhomey | mark_csi, guess you're not in the US? |
| 17:15.12 | mark_csi | etfonhomey: 'fraid not irelad |
| 17:15.16 | mark_csi | etfonhomey: 'fraid not ireland |
| 17:22.01 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 17:23.11 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:24.16 | Peaceful | I upgrade from asterisk 1.2.24 to 1.4.24 and then dtmf only works for asterisk itself (dtmfmode= rfc2833 or inband or auto) or only works over the PSTN over our digium cards (dtmfmode=info). What am I missing??? |
| 17:24.39 | Peaceful | I'm going to spend all day googling and hanging around here and trying stuff until I get this fixed! |
| 17:24.55 | Peaceful | so far, two hours of googling haven't turned anything up :-( |
| 17:26.05 | Peaceful | I tried adding relaxdtmf=yes to [general] in sip.conf, but no effect |
| 17:27.42 | etfonhomey | mark_csi, love Ireland. you symptoms sound exactly like what I was having. I had the telco come out and check the lines twice and both times they said everything was fine. If I hooked up an analog phone it always worked. |
| 17:28.02 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 17:28.49 | etfonhomey | mark_csi, it was either something I did when building the zaptel stuff or something with the card. |
| 17:29.40 | Micc | http://pastebin.com/m1e2fd478 |
| 17:30.09 | Micc | Anyone know how to fix these subscribe errors? |
| 17:30.12 | mark_csi | etfonhomey, thx. I just can't get it to give me the problem again. I know as soon as I leave it they'll call me. |
| 17:30.39 | etfonhomey | mark_csi, that was my experience. I was using a TDM400 w/o echo cancellation and 4 FXO modules. |
| 17:31.55 | mark_csi | etfonhomey: I've a tdm815p with echo cancellation, I've placed 10 test calls all of which worked |
| 17:32.18 | *** join/#asterisk betamax (i=betamax@gateway/gpg-tor/key-0x2D032A3C) |
| 17:32.33 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:33.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:43.34 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 17:43.54 | etfonhomey | mark_csi, Have you tried Digium's support? |
| 17:45.17 | jameswf | some people are to stubborn to call tech support.... |
| 17:45.34 | mark_csi | etfonhomey, no not yet, only discovered the problem this morning. (and I'm stubborn :P) |
| 17:46.52 | etfonhomey | mark_csi, I had 2 POTS lines and 4 FXO ports (until one went dead). I tried moving lines around and the symptoms were the same regardless. |
| 17:50.36 | mark_csi | etfonhomey, I think I'll give Digium a call on Wed, I'll be onsite then, I'm heading home now. Thx for your help |
| 17:50.58 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 17:51.05 | etfonhomey | mark_csi, can't say I was much help, just that I've been there... Good luck! |
| 17:52.35 | [TK]D-Fender | Micc: What is doing the subscribe? |
| 17:52.43 | mark_csi | etfonhomey, thx anyways, talk to you again sometime |
| 17:53.43 | [TK]D-Fender | BBIAb |
| 17:58.24 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 17:58.34 | CrashHD | Hello |
| 17:58.53 | CrashHD | any documented issues doing sip to sip traffic between two local instances of asterisk on a single machine? |
| 17:59.07 | CrashHD | I'm seeing some reports of lost packets |
| 17:59.19 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:59.39 | CrashHD | and a couple of rtcp debugs show really skewed numbers |
| 18:01.01 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 18:01.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 18:07.36 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 18:13.11 | jameswf | ~book |
| 18:13.12 | jbot | somebody said book was probably 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://tfot.leifmadsen.com or see ~buybook |
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| 18:56.26 | grandpapadot | This is kind of cool: http://udigits.com |
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| 19:09.16 | jameswf | anyone know the AT&T Edge network subnets? |
| 19:12.36 | grandpapadot | jameswf: lol, you give AT&T too much credit. AT&T's networks are a hodge podge of purchased entities, mergers, etc. |
| 19:13.20 | jameswf | my blackberry pulls a different subnet every connection... makes security difficult |
| 19:13.34 | jameswf | jerks |
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| 19:13.51 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 19:14.02 | grandpapadot | Yea, them jerks! lol |
| 19:14.05 | grandpapadot | How dare they. |
| 19:19.17 | neurosys | [TK]D-Fender: Thanks for the peer lead last friday. You were correct (never doubted you for a sec.) |
| 19:19.56 | *** join/#asterisk gcbirzan (n=gcbirzan@pida/gcbirzan) |
| 19:20.54 | [TK]D-Fender | neurosys: You're welcome |
| 19:22.09 | gcbirzan | I'm looking at the output from iax set debug and it says "FORMAT : 2", is that the codec it's using? And, if so, how can I tellw hat codec that is? |
| 19:23.27 | gcbirzan | -- Call accepted by 192.168.2.200 (format gsm), or that |
| 19:23.47 | gcbirzan | goes use wall on head. |
| 19:25.01 | [TK]D-Fender | BBIAB |
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| 19:58.17 | bradleyprice86 | I am having a problem configuring my trunk between an asterisk and cisco call manager system. I can dial out from asterisk to the cisco call manager, but when I dial from the cisco call manager I get the "ss-noservice" message. |
| 19:58.23 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 19:58.54 | bradleyprice86 | I think it is a problem with my incoming . |
| 19:59.18 | bradleyprice86 | Anyone think that they might be able to help me? |
| 19:59.20 | funxion | check ur codec and protocol |
| 20:01.52 | bradleyprice86 | I know I have the protocol and codec setup correctly. |
| 20:01.58 | bradleyprice86 | That was my last obstacle. |
| 20:02.21 | bradleyprice86 | It seems that no matter what I change my context to, it uses from-sip-external |
| 20:03.13 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 20:10.55 | farkus_ | I've been using "conspy 9" to look at the asterisk output. Is there a better method, like getting the output to a log, etc.? |
| 20:11.34 | neurosys | My 6 year old in 1st grade made the Honor Roll! Im so proud. Yes, I'm aware that this is not the place. But im too happy :P |
| 20:14.07 | n3hxs | Ahh, that is what the Asterisk by his/her name means :) |
| 20:16.02 | funxion | anyone in here from the south florida area and looking for a job? |
| 20:16.24 | neurosys | Um. I live in south florida :P |
| 20:16.35 | funxion | are you looking for a job? |
| 20:16.51 | neurosys | Not exactly heh |
| 20:17.12 | neurosys | Do you need a voip setup or something? |
| 20:17.19 | funxion | well |
| 20:17.24 | funxion | www.seamobile.com |
| 20:17.26 | tzanger | it's funny |
| 20:17.36 | tzanger | I have two Nortel ATAs, the internal ATA and a quad T1 card and dual T1 channel bank surrouding this cheap-ass Nortel CICS |
| 20:17.42 | tzanger | so I can covert this company over to Asterisk without distrupting their current phones |
| 20:18.18 | funxion | and.... |
| 20:18.25 | funxion | is it an old meridian? |
| 20:18.40 | tzanger | it's an old Nortel CICS (think MICS but smaller, or think nano-option11) |
| 20:19.52 | funxion | o |
| 20:21.07 | funxion | whats the problem |
| 20:24.34 | farkus_ | What's the best way to view the console output of * if you're not at the console? |
| 20:26.25 | funxion | ssh to the box |
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| 20:44.39 | StephenF | are there any problems with installing ztdummy after asterisk has already been installed? |
| 20:44.54 | StephenF | I didnt need it before,, but now I need MeetMe so i need a time source |
| 20:45.58 | abalashov | You may need to recompile asterisk, because asterisk will detect whether to build app_meetme on compile based on its detection of the presence of zaptel headers. |
| 20:46.08 | abalashov | So if zaptel wasn't there before, app_meetme was probably not built. |
| 20:46.28 | StephenF | right its not |
| 20:46.46 | StephenF | so just need to re-run make install after install zaptel |
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| 20:47.19 | [TK]D-Fender | StephenF: * must be recompiled if Zapte was installed after it |
| 20:47.39 | [TK]D-Fender | StephenF: You should trash your source folder, reextract from scratch and go from there. |
| 20:47.48 | StephenF | hmm |
| 20:47.52 | [TK]D-Fender | StephenF: Everything except "make samples" (if you know whats good for you) |
| 20:47.59 | StephenF | lol right |
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| 21:04.26 | *** mode/#asterisk [+o denon] by ChanServ |
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| 21:06.35 | *** join/#asterisk EI5GTB (n=Paul@78.16.153.30) |
| 21:06.45 | EI5GTB | evening guys |
| 21:06.57 | EI5GTB | anyone reccomend the best sounding (free) voice synth? |
| 21:07.23 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-2fe04edb81a5fcea) |
| 21:07.29 | jblack | EI5GTB: no such animal. |
| 21:07.34 | neurosys | heh |
| 21:07.51 | EI5GTB | is there not voice synths out there that work with asterisk? |
| 21:07.52 | jblack | Your choices range from "useless" to "horrid" |
| 21:08.04 | EI5GTB | iv seen dial plans with like "talk(bla ets hello bla) |
| 21:08.09 | EI5GTB | i see =/ |
| 21:08.11 | neurosys | EI5GTB: so far, ive found festival+OGI to work "best" :-P |
| 21:08.20 | jblack | bah. least terrible. |
| 21:08.30 | neurosys | jblack: true true.. |
| 21:10.07 | jblack | The world needs an IBM or a sun to buy some neat proprietary project, and make it free software to piss off microsoft. |
| 21:10.50 | jblack | since the problem is to big/hard for conventional free software, we have to wait for dirty pool. |
| 21:10.52 | [TK]D-Fender | jblack: Same thing :) |
| 21:11.05 | [TK]D-Fender | jblack: "fest free" != necessarily acceptable. |
| 21:11.11 | [TK]D-Fender | best* |
| 21:11.28 | jblack | I never used the b word. |
| 21:11.30 | [TK]D-Fender | EI5GTB: How dynamic is your content? |
| 21:11.43 | [TK]D-Fender | EI5GTB: And how many maximum simultaneous channels? |
| 21:11.51 | EI5GTB | oh, 3 max i could imagine |
| 21:12.04 | EI5GTB | as for dynamic......not very |
| 21:12.19 | EI5GTB | pretty static actually |
| 21:12.23 | EI5GTB | ivm menu.. |
| 21:12.28 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-74d29d7cc9804e11) |
| 21:12.36 | EI5GTB | its not a task critical operation, its for a bit of fun really |
| 21:12.44 | [TK]D-Fender | EI5GTB: then pay for the 1 channel Cepstral license and generate static filess off of it. You could optionally create a cahce-driver setup for this. |
| 21:12.50 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 21:12.54 | [TK]D-Fender | cache* |
| 21:13.08 | EI5GTB | i see... how much can i expect top pay? |
| 21:13.56 | jblack | If I were a quazillionaire, I think that's what I'd donate. Development for tts and speech recog |
| 21:14.15 | jblack | Just throw a big pile of money at a big pile of phd students. |
| 21:14.22 | EI5GTB | asterisk is just a hobby for me (strange, yes) so any outlays are band considering i wont get an income from it |
| 21:14.28 | [TK]D-Fender | EI5GTB: 10-20$ IIRC |
| 21:14.32 | EI5GTB | oh, i see |
| 21:15.29 | Ritzerisk | im checking to see if its possible (its really complicated to explain but) build a Time entry system first where you put your employee number , then control number , start time , finish time , and then export it to a txt file with multiple entries im assuming its alot of dtmf input and txt output with ???? |
| 21:15.45 | `Sauron | ~trixbox |
| 21:15.45 | jbot | from memory, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 21:15.46 | [TK]D-Fender | EI5GTB: confirmed @ $29.99 USD |
| 21:15.56 | EI5GTB | i see |
| 21:16.01 | EI5GTB | tnx |
| 21:16.07 | `Sauron | ~openpbx |
| 21:16.07 | jbot | extra, extra, read all about it, openpbx is a free software PBX written in PERL. Written by Voicetronix. Maybe you meant callweaver, which was once caller openpbx. |
| 21:16.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:16.20 | `Sauron | ~freepbx |
| 21:16.20 | jbot | [~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 |
| 21:16.32 | [TK]D-Fender | Ritzerisk: Yes possible, not necessarily very difficult and possibly entirely doable by dialplan. |
| 21:16.54 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 21:19.00 | Ritzerisk | where you think would i start to look for docs on to learn how it works or do i have to do some type of scripting |
| 21:19.05 | Ritzerisk | like perl |
| 21:19.12 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:20.04 | [TK]D-Fender | Ritzerisk: basic IVR's (BOOK), and "core show application system" |
| 21:20.32 | [TK]D-Fender | Ritzerisk: You should read the complete function & application list to learn what *'s dialplan has to ofer. |
| 21:20.36 | [TK]D-Fender | offer* |
| 21:21.04 | [TK]D-Fender | Ritzerisk: And I already answered your qeustion. I said nothing of external scripting. May not be needed. |
| 21:21.19 | Ritzerisk | so theres like if then statements to program and such .... i have the asterisk Book rel 2 |
| 21:21.29 | Ritzerisk | ohhh snazzy |
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| 21:26.24 | [TK]D-Fender | checkout time, heading home, later all. |
| 21:26.45 | *** part/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212) |
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| 21:48.52 | cianmaher | Hi all, is there any way to get an agents status in a queue as variable in 1.2 to use in dialplan? |
| 21:49.23 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
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| 21:56.47 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:59.50 | goobsoft | I'm having a NAT problem that I can't figure out. My phone is behind a nat and asterisk is not. I have nat=yes for that sip user and qualify=5000. I'm not getting sound either way. If I run tcpdump on the asterisk server, it shows the packets from the phone are getting to the server, but they don't show when rtp debug is enabled. Can someone help me with this here? |
| 22:00.43 | harry_v | port forwarding in the fw. open up ports 10k-20k in the wirewall |
| 22:01.20 | goobsoft | I tried that with no luck... let me double check again |
| 22:01.24 | [TK]D-Fender | Phone-side needs nothing. Its all * config |
| 22:01.25 | harry_v | dont forget to config rtp.conf |
| 22:01.30 | [TK]D-Fender | ~sipnat |
| 22:01.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:01.34 | [TK]D-Fender | ^^^^ |
| 22:01.48 | *** join/#asterisk DeVilSoulBlacK (n=aandaluz@200.93.197.157) |
| 22:02.26 | cianmaher | perhaps firewall on phone side is blocking rtp traffic? |
| 22:02.40 | goobsoft | In my case asterisk is on an openWRT router, so it is not behind the nat, though it is behind a firewall that has 5060 and 10000-60000 UDP open. |
| 22:03.23 | goobsoft | The phone works when I'm on the local net, but I'm at my parents house and they have a linksys router. I opened up 10000-20000 (what's in my rtp.conf) and that didn't help |
| 22:03.49 | [TK]D-Fender | goobsoft: Go read the guide. |
| 22:04.32 | goobsoft | Well, let me ask two specific question that should help me a lot... |
| 22:05.04 | goobsoft | If * runs the play command, shouldn't rtp debug show outgoing packets? |
| 22:05.32 | [TK]D-Fender | goobsoft: Depends if RTP can be set up |
| 22:05.41 | harry_v | TK, what percentage of people usaully get the RTP configurations wrong the first time? |
| 22:05.50 | [TK]D-Fender | goobsoft: pastebin your sip.conf and maybe we can see if you made a mistake. Mask only passwords |
| 22:05.52 | [TK]D-Fender | ~pb |
| 22:05.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:05.54 | harry_v | Goob, common problem is misconfiguration |
| 22:06.08 | [TK]D-Fender | harry_v: what RTP configurations? |
| 22:06.12 | harry_v | or no configuration |
| 22:06.21 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:06.32 | harry_v | from what goob is describing. |
| 22:07.33 | [TK]D-Fender | harry_v: This should not be a guessing game. We should have seen the sip.conf a while agi, no forwarding is required and he's confirmed the ports he has opened. |
| 22:07.37 | [TK]D-Fender | ago* |
| 22:07.43 | goobsoft | http://pastebin.com/d7f9e66e4 |
| 22:07.53 | *** part/#asterisk xeno42 (n=nxeno42@r.omnipotent.net) |
| 22:08.20 | harry_v | right now, im focusing on why patch -p1 <path/to/patchfile/ is being excepted in contrib but after rebooting the server Festival still refuses to work in asterisk. |
| 22:08.29 | harry_v | so your saying he has done everything right then |
| 22:08.35 | [TK]D-Fender | goobsoft: Now pastebin a failed call with SIP debug enabled and verbose 10 |
| 22:08.52 | [TK]D-Fender | harry_v: No, I'm saying he hadn't shown us his config so I naturally trust NOTHING. |
| 22:09.11 | [TK]D-Fender | harry_v: I jsut know that no forwarding is required |
| 22:10.54 | goobsoft | http://pastebin.com/d69c04039 |
| 22:12.16 | goodjoke | is there any way to change message playback order? |
| 22:12.30 | goodjoke | so when people dial in, they get the newest message first |
| 22:12.34 | *** part/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 22:12.53 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.186) |
| 22:12.59 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 22:14.12 | [TK]D-Fender | goobsoft: Ok, so you go from "Answer" right into "Echo", and get nothing? do a PLAYBACK in between... just anything really. Something to ensure the RTP stream is up. |
| 22:14.25 | [TK]D-Fender | goodjoke: Apparently not |
| 22:14.40 | [TK]D-Fender | goodjoke: Short of modding the source. |
| 22:15.27 | goobsoft | Here are the two extentions I am using for testing. http://pastebin.com/d7fff78cd |
| 22:15.42 | goobsoft | When I dial 601, I hear nothing. Do you want the sip log? |
| 22:18.15 | goobsoft | Wait, I may have figured something out. |
| 22:18.39 | goobsoft | http://pastebin.com/d36b8f5fc |
| 22:19.21 | goobsoft | rtp debug actually shows it sending the packets, but 192.168.1.100 is the private ip address of my phone |
| 22:19.53 | goobsoft | It needs to be sending those packets to 66.25.84.137. I thought nat=yes would make that happen. |
| 22:21.33 | goobsoft | btw, I'm using * version 1.4.21-1. I upgraded from 1.2 something. |
| 22:25.44 | funxion | anyone in here from the south florida area and looking for a job? |
| 22:26.23 | funxion | msg me if ur interested |
| 22:27.09 | harry_v | to bad file was not here he would know a answer to my problem |
| 22:27.37 | harry_v | funxion, for * installs? |
| 22:27.45 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 22:29.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:33.47 | funxion | some but thats not all |
| 22:33.58 | funxion | mainly voip centric |
| 22:34.04 | funxion | cisco and * |
| 22:34.09 | funxion | some quintum |
| 22:34.20 | funxion | www.seamobile.com |
| 22:34.33 | funxion | check out system engineer under employment |
| 22:35.52 | funxion | actually its under about us then career oputunities |
| 22:37.12 | *** join/#asterisk telecos (n=sergio@67.166.219.87.dynamic.jazztel.es) |
| 22:42.00 | goobsoft | Here's the full log of a call to 601, with the sound file replaced with one that works. http://pastebin.com/d5adc5c52 |
| 22:44.45 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 22:45.06 | *** join/#asterisk stephank (n=urk@2002:52c5:cf78:0:21c:c4ff:fece:ea94) |
| 22:45.52 | harry_v | fuxion, a few years ago somone integrated a asterisk box with a att sat dish and some wifi phones for some ships that would communicate in a remote area of alaska. For its purpous, worked well. |
| 22:47.16 | FruitBasket | is there a way to make "sip debug on" log to a specific file? |
| 22:49.53 | FruitBasket | == Spawn extension (macro-dialexten, dial, 3) exited non-zero on 'SIP/accountA-1db44780' \n -- User hung up <-- I almost never see "User hung up". What produces it? how can I get clarity like that on all calls? there was another 26 seconds later where they left voicemail, but it didn't give anything else.. |
| 22:51.39 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:58.23 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:02.35 | FruitBasket | I recently upgraded from asterisk 1.4.18 to .22, but when I do asterisk -r it's still saying .18... I _just_ did a restart through the console, and it still says .18 |
| 23:03.03 | FruitBasket | <PROTECTED> |
| 23:04.06 | FruitBasket | "Asterisk 1.4.22, Copyright (C) 1999 - 2008", then says "Connected to Asterisk 1.4.18" ... |
| 23:12.39 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 23:12.57 | goobsoft | http://pastebin.com/d7856e386 If you look at the last line, it shows the connected peer and lists NAT as N, when in my sip.conf, it has nat=yes for user 101. How is this possible? |
| 23:14.38 | *** join/#asterisk ipguy (n=dkavadas@129.94.190.121) |
| 23:14.44 | ipguy | hi all |
| 23:15.11 | ipguy | can someone explain to me how asterisk can save me money on my phone calls. |
| 23:15.35 | ipguy | do i need to get a voip account with engin or someone else ? |
| 23:15.37 | [TK]D-Fender | ipguy: Who said * had anything to do with that? |
| 23:16.04 | [TK]D-Fender | ipguy: ITSP's may very well cost less than more traditionaly termination depending on what you pay now and your needs... |
| 23:16.27 | [TK]D-Fender | goobsoft: It's showing the WAN IP for your phone, its fine... |
| 23:16.31 | WimpMan | # make money -fast |
| 23:17.17 | ipguy | [TK]D-Fender: so if i setup an asterisk box, can anyone with a phone call me ? |
| 23:17.43 | ipguy | [TK]D-Fender: i just need to get my head around this |
| 23:17.47 | [TK]D-Fender | ipguy: Depends what you connect to your * server |
| 23:18.28 | goobsoft | But If nat is N won't * will adhear to the Contact and Via headers? That means that it will send rtp data to 192.168.1.100 instead of 66.25.84.137, won't it? |
| 23:18.38 | ipguy | [TK]D-Fender: well, let me ask this then, what do i need to setup asterisk so anyone with a phone can call me. |
| 23:18.47 | [TK]D-Fender | ipguy: * is a telephony toolkit that can let you use hardware interfaces to plug in lines, phones, etc. Also use soft-phones (Similar to the Skype client for example), "hard" IP phones, etc, and internet-based IP telephony providers |
| 23:18.53 | Deeewayne | ipguy: I have an IAXy sitting in the Republic of Georgia which registers to an Asterisk box in the US so my wife can speak to her family without buying calling cards. |
| 23:19.12 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 23:20.03 | ipguy | Deeewayne: but she needs that setup, if the IAX box wasn't there, could she just call your * box ? |
| 23:20.47 | [TK]D-Fender | ipguy: If you want a PSTN # she can dial you need an interface to take in a line you already have so that * can use it, or get a new # via an ITSP. |
| 23:20.49 | [TK]D-Fender | ~itsp |
| 23:20.50 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:21.39 | [TK]D-Fender | ipguy: "need" is not the term we should be looking for. "how" is what is flexible. Do you have something specific you are looking to do? |
| 23:22.07 | SQLDarkly | I am having an audio issue. 2 out of 10 Calls will have a very audible buzzing then cease. This issue is reproducible and i was thinking to do a sniffer trace on a good call then replicate the issue so I have both a good trace and a bad trace. The problem is when I finally have both cap files I see no difference. |
| 23:22.28 | ipguy | so for everyone to be avle to contact me they will need a computer and a soft phone ? |
| 23:22.48 | SQLDarkly | I have no telephony hardware installed and am using SIP. I have some cards on order but SIP doesnt use hardware timing if im not mistaken |
| 23:23.03 | *** join/#asterisk Segnale007 (n=Pietro@host86-135-19-235.range86-135.btcentralplus.com) |
| 23:23.13 | ipguy | [TK]D-Fender: specifically, i want to save money on calls and ITSP rental costs |
| 23:23.39 | [TK]D-Fender | ip Call you make? Incoming calls? For very specific people? |
| 23:23.51 | [TK]D-Fender | ipguy: and what "ITSP rental costs" are you referring to? |
| 23:23.58 | ipguy | [TK]D-Fender: calls both ways, for everyone |
| 23:24.09 | SQLDarkly | I was thinking perhaps a firmware issue on the phones.... |
| 23:24.14 | [TK]D-Fender | SQLDarkly: SIP has nothing to do with timing of itself. |
| 23:24.18 | ipguy | [TK]D-Fender: for out primary home phone |
| 23:24.29 | ipguy | out=our |
| 23:25.07 | [TK]D-Fender | ipguy: Where are you located? |
| 23:25.07 | SQLDarkly | Exactly why I do not think timing is the issue. Just stating the timing I am using at current |
| 23:25.18 | [TK]D-Fender | SQLDarkly: Could be bandwidth issues as well. But what phones? |
| 23:25.19 | ipguy | [TK]D-Fender: AUS, Sydney |
| 23:25.32 | SQLDarkly | I changed servers both switch and nic to 1gig full |
| 23:25.34 | [TK]D-Fender | ipguy: yay... Telsta territoy... |
| 23:25.42 | SQLDarkly | cisco 79xx |
| 23:25.47 | [TK]D-Fender | dang typos |
| 23:25.55 | [TK]D-Fender | SQLDarkly: Shouldn't be the phone so much as BW |
| 23:26.02 | [TK]D-Fender | SQLDarkly: "(jitter, etc) |
| 23:26.19 | [TK]D-Fender | ipguy: Well you'll have to shop around to see about providers in your area |
| 23:26.23 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 23:27.26 | ipguy | [TK]D-Fender: so i must have an ITSP account so everyone can call my * box ? |
| 23:27.37 | SQLDarkly | Seeming as though my traffic captures show no difference. How would I go about diagnosing this issue? Is there somewhere I can tweak the jitterbuffer? |
| 23:27.41 | [TK]D-Fender | ipguy: For the "normal world", yes. |
| 23:27.58 | [TK]D-Fender | ipguy: They get the call and send it to you over a VoIP protocol. |
| 23:28.06 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 23:28.10 | [TK]D-Fender | ~itsp |
| 23:28.11 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:28.12 | [TK]D-Fender | ^^^ |
| 23:29.03 | ipguy | [TK]D-Fender: what is FWD ? |
| 23:29.37 | SQLDarkly | I will also state that I am using QoS on my network and yes voice traffic has priority. |
| 23:30.02 | [TK]D-Fender | ipguy: A P2P type service that USEWD to be "free", but is no longer |
| 23:30.11 | [TK]D-Fender | ipguy: Something you can largely forget about. |
| 23:31.03 | ipguy | [TK]D-Fender: and sipborker is ? |
| 23:31.09 | ipguy | sipbroker |
| 23:31.16 | [TK]D-Fender | ipguy: As far as saving money goes, with an * server typocally you can connect with others via the internet for free as well. They only need a client and can connect to your box directly, or though a service like ekiga.net / FWD (Less advised for obvious reasons), etc |
| 23:31.29 | [TK]D-Fender | ipguy: SIPBroker is a straight-up ITSP IIRC |
| 23:31.56 | ipguy | [TK]D-Fender: "Less advised for obvious reasons" ?? why ? |
| 23:32.09 | ipguy | <PROTECTED> |
| 23:32.24 | ipguy | o got a stack of questions |
| 23:32.27 | [TK]D-Fender | If I Recall Correctly. |
| 23:32.43 | [TK]D-Fender | ipguy: and ITSP has been linked up to, TWICE |
| 23:32.53 | [TK]D-Fender | ipguy: pay attention to the JBOT info-lets |
| 23:33.03 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:33.18 | [TK]D-Fender | ipguy: And FWD is a FORMERLY FREE SIP proxy service. Hence no longer FREE. |
| 23:33.28 | ipguy | [TK]D-Fender: so sipbroker is an ITSP, and sipbroder is free. |
| 23:33.43 | outtolunc | haha |
| 23:33.47 | [TK]D-Fender | ipguy: Ekiga.net should do fine if they want to hook up to a generic servive, and you can do the same and then can call each other for free |
| 23:34.16 | [TK]D-Fender | ipguy: No, SIPBroker is a full-on ITSP, not just a proxy service, and they COST. |
| 23:34.22 | ipguy | [TK]D-Fender: so my pots friends can call me also ? |
| 23:34.32 | [TK]D-Fender | ipguy: FWD was just a P2P service |
| 23:34.47 | SQLDarkly | Is there a global jitter configuration as the one in sip.conf |
| 23:34.48 | [TK]D-Fender | ipguy: ITSP is so you can call POTS, and POTS can call you |
| 23:34.55 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 23:34.57 | [TK]D-Fender | SQLDarkly: jsut sip.conf IIRC |
| 23:35.32 | ipguy | off to buy a VOIP book |
| 23:35.37 | *** part/#asterisk ipguy (n=dkavadas@129.94.190.121) |
| 23:35.50 | [TK]D-Fender | ~book |
| 23:35.50 | jbot | somebody said book was probably 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://tfot.leifmadsen.com or see ~buybook |
| 23:39.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:39.52 | tzafrir_laptop | http://julius.sourceforge.jp/en_index.php |
| 23:41.43 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 23:41.46 | *** join/#asterisk Segnale007 (n=Pietro@host86-135-19-235.range86-135.btcentralplus.com) |
| 23:42.51 | beek | [TK]D-Fender: Good evening... I have another T1 question I hope you can answer. I have my * box between the PSTN and the legacy PBX. Card is an A104d. Timing comes from the PSTN (pri_cpe) and gets sent to PBX (pri_net). If I use another port to connect to another PRI, from a different provider, who supplies the timing? |
| 23:43.47 | CrashHD | you specify in your zapata or zaptel.confs the priority order which you get timing from which ports |
| 23:44.05 | [TK]D-Fender | beek: you set primart, secondary, etc.. |
| 23:44.37 | [TK]D-Fender | beek: 1,1,0 -> 2,2,0 -> 3,3,0 and 4,0,0 (saying port for GIVES timing) |
| 23:44.46 | [TK]D-Fender | four* |
| 23:45.02 | beek | Ah... I see. I'm looking at my options for some outbound and Verizon is an option, so I wondered how that would work. Thanks. You too CrashHD . |
| 23:55.40 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 23:56.46 | jasonwoot | is 'remote unix connection' followed by 'remote unix disconnection' |
| 23:57.00 | jasonwoot | * is 'remote unix connection' followed by 'remote unix disconnection' an indication of my sip trunks dropping? |
| 23:57.21 | [TK]D-Fender | jasonwoot: No its another process accessing CLI or AMI |
| 23:57.32 | jasonwoot | uh oh... |
| 23:57.37 | [TK]D-Fender | jasonwoot: Typically a way for us to detect GUI users :) |
| 23:57.40 | drmessano | jasonwoot: FreePBX? |
| 23:57.48 | [TK]D-Fender | high-5's drmessano |
| 23:58.28 | jasonwoot | drmessano: no, but I did start apache back up |
| 23:58.45 | drmessano | apache doing what? |
| 23:59.05 | Daejeo | dance |
| 23:59.10 | drmessano | If its not FreePBX, what is apache doing in the equation? |
| 23:59.20 | jasonwoot | this box previously had a gui, that's been disabled for some time |
| 23:59.29 | jasonwoot | odd that when I turn apache on, something connects to it |
| 23:59.43 | drmessano | Its the web interface.. not odd at all |