| 00:00.06 | rue_mohr | CIF is just res. 252x388 I think |
| 00:00.19 | [TK]D-Fender | rue_mohr: And that is how much bw? |
| 00:00.37 | rue_mohr | they say its 24mbytes |
| 00:00.44 | rue_mohr | /sec |
| 00:01.11 | Gugge | so you need a connection faster than 200Mbit ? |
| 00:01.20 | [TK]D-Fender | rue_mohr: "they say" sounds like "I'm not looking and they are calling me on it and ar fucking with me". |
| 00:01.22 | rue_mohr | thats what they tell me |
| 00:01.38 | rue_mohr | no I fought with them over it for 2.5 weeks |
| 00:02.18 | rue_mohr | according to them, it takes 200mbits and a 2Ghz+ processor |
| 00:02.22 | rue_mohr | to stream a webcam |
| 00:02.25 | [TK]D-Fender | rue_mohr: Fought with people? Pardon. |
| 00:02.33 | [TK]D-Fender | *cough* |
| 00:02.33 | rue_mohr | <rue_mohr> #videolan #mplayer, and #gstreamer |
| 00:02.39 | [TK]D-Fender | rue_mohr: FUCK THEM |
| 00:02.52 | [TK]D-Fender | rue_mohr: YOU aren't loking and they are fucking with you |
| 00:03.00 | rue_mohr | unfortunatly, their the only video codec people on irc |
| 00:03.06 | Gugge | rue_mohr: yep, it takes 200Mbit and a 2Ghz processor to stream a webcam .. thats why you can buy cheap wireless webcams :P |
| 00:03.13 | rue_mohr | :) |
| 00:03.25 | [TK]D-Fender | rue_mohr: We do video in here too you know... |
| 00:03.39 | [TK]D-Fender | rue_mohr: Asterisk has CODECS for that shit. |
| 00:03.47 | rue_mohr | yes, but you do all hate me now |
| 00:03.49 | [TK]D-Fender | rue_mohr: Perhaps you should have read some of the big print. |
| 00:04.10 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 00:04.14 | [TK]D-Fender | rue_mohr: http://www.google.ca/#hl=en&source=hp&q=h.263+bandwidth&aq=0&aqi=g1&aql=&oq=H.263+ban&gs_rfai=&fp=5b1fceef9c80000d |
| 00:04.15 | rue_mohr | well, I was searching for it from terms of video, not asterisk |
| 00:04.24 | rue_mohr | voip protocols are good at realtime |
| 00:04.40 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 00:04.52 | [TK]D-Fender | rue_mohr: and guess what , CODECS are VIDEO TERMS. The fact asterisk can USE them is no excusse not to use it as a REFERENCE for things to LOOK FOR |
| 00:05.27 | rue_mohr | you can talk softly I shouted to 3 channels about it for 3 weeks |
| 00:05.42 | [TK]D-Fender | rue_mohr: You have communication issues. I'm quite aware. |
| 00:05.49 | [TK]D-Fender | rue_mohr: "H.263 lets users scale bandwidth usage and can achieve full-motion video (30 frames per second) at speeds as low as 128K bit/sec. With its flexibility and bandwidth and storage savings, H.263 has a low total cost of ownership and provides a quick return on investment." |
| 00:06.24 | rue_mohr | I better tell them |
| 00:06.37 | [TK]D-Fender | rue_mohr: Quote I got in a 10 SECOND search. Sure as shit doesn't sound like I need 5 MEGABIT |
| 00:06.52 | [TK]D-Fender | rue_mohr: The fact you couldn't pull this up yourself is disturbing. |
| 00:07.13 | rue_mohr | I didn't have the term H.263 to search for |
| 00:07.14 | [TK]D-Fender | rue_mohr: Pick a codec. "+ bandwidth" in a Google search |
| 00:07.35 | [TK]D-Fender | rue_mohr: "streaming video" is generic shit. It can mean ANYTHING |
| 00:07.46 | rue_mohr | I was looking for things like mjpeg, mpeg, rtsp, webcam stream |
| 00:08.10 | [TK]D-Fender | rue_mohr: None of those imply a fixed amount |
| 00:08.27 | [TK]D-Fender | rue_mohr: Webcams come in a different specs and have nothing to do with the FORMAT. |
| 00:08.29 | rue_mohr | I dont know if its the way I read you, but over the past 3 years you sure sound angry |
| 00:08.52 | [TK]D-Fender | rue_mohr: And those formats you mentioned aren't fixed by size or even necessarily relevant for your use |
| 00:08.55 | timeshell | rue_mohr : NO, that's just how he is |
| 00:09.06 | rue_mohr | he wasn't always |
| 00:09.08 | [TK]D-Fender | timeshell: indeed :) |
| 00:09.11 | timeshell | Just a very rude and inconsiderate person |
| 00:09.57 | rue_mohr | used to be pretty patient, when you get to know a LOT I think you start to get tired of repeating the same answers, and get kinda snappy with people who ask them |
| 00:10.25 | rue_mohr | I tried to start a bot to look for keywords and automatically answer the top 80% repeated quesions |
| 00:10.35 | rue_mohr | so far (the last 6 years) it just logs channels... |
| 00:11.00 | timeshell | The thing is, people who ask the questions are usually newbs who haven't had enough exposure to even know how to look for the answers to the questions they' |
| 00:11.02 | timeshell | re asking |
| 00:11.16 | timeshell | And in fact don't even know what question to ask. |
| 00:11.34 | [TK]D-Fender | rue_mohr: http://www.google.ca/#hl=en&source=hp&q=video+streaming+formats+and+bandwidth&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=5b1fceef9c80000d |
| 00:11.41 | rue_mohr | yes, not knowing the right terms to search for will usually not lead to search results that point to what you want |
| 00:11.53 | [TK]D-Fender | rue_mohr: Seriously. You fought with IRC people and got nowhere. How hard would this google search have been? |
| 00:12.05 | rue_mohr | yes, I'm just looking for a H.263 library for linux |
| 00:12.17 | [TK]D-Fender | rue_mohr: "One hour of video encoded at 300 kbit/s (this is a typical broadband video in 2005 and it is usually encoded in a 320Ã240 pixels window size) will be:" <--- answer from the FIRST LINK |
| 00:12.28 | [TK]D-Fender | rue_mohr: 300kbps != 5 MEGAbits |
| 00:12.32 | rue_mohr | [TK]D-Fender, well most of my google searches were trying to work out how streaming mpeg worked |
| 00:12.48 | [TK]D-Fender | rue_mohr: You haven't even firmly picked a format yet. |
| 00:12.48 | rue_mohr | cause I'm not a video streaming guru |
| 00:12.52 | rue_mohr | no |
| 00:13.12 | [TK]D-Fender | rue_mohr: You don't need to be a guru. You just need to actually pick a FORMAT to compare and a reason to use it. |
| 00:13.23 | rue_mohr | I determined that none of the existing video software (vlc, mplayer, ffmpeg) can do realtime streaming of a v4l device |
| 00:13.40 | rue_mohr | [TK]D-Fender, and I had no pallette of formats |
| 00:13.52 | [TK]D-Fender | rue_mohr: You jsut made your first V4L reference |
| 00:13.55 | rue_mohr | from what I understood, everything is encapsulated in mpeg |
| 00:14.00 | [TK]D-Fender | rue_mohr: Finally some details |
| 00:14.15 | [TK]D-Fender | rue_mohr: Anything else you care to add so we don't ahve to get them at large interfvals? |
| 00:14.17 | *** join/#asterisk nightwind (~nightwind@daimon.vixel.org) |
| 00:14.34 | mmlj4 | rue_mohr: looking at your wish list earlier... the problem to me is finding FX* devices cheap... which means chinese or something like that |
| 00:14.38 | rue_mohr | usb webcam (cheapo) via v4l laptop computer via wireless to workstation |
| 00:14.47 | [TK]D-Fender | rue_mohr: Because you made a nasty wide-open blanket statement on the bandwidth requirements stating NO details. |
| 00:15.04 | rue_mohr | mmlj4, look for old T1 channelbanks |
| 00:15.19 | mmlj4 | oh! yeah, that would work |
| 00:15.21 | rue_mohr | [TK]D-Fender, sorry, wish I had logs |
| 00:15.39 | rue_mohr | mmlj4, remember to get a T1 card WITH a good echo canceler |
| 00:15.48 | rue_mohr | I use a mainstreet |
| 00:16.00 | [TK]D-Fender | rue_mohr: Well you came in HERE asking that way. How you fought with those other channels isn't something we're expected to be psychic about. |
| 00:16.25 | rue_mohr | newbridge mainstreet 3624 |
| 00:16.28 | [TK]D-Fender | rue_mohr: Describe the end-point requirements a bit more... |
| 00:17.52 | [TK]D-Fender | rue_mohr: how do you expect to hook into the "stream"? How many clients? |
| 00:18.16 | rue_mohr | [TK]D-Fender, ok, the lawn mowing robot has a 333Mhz laptop on it, with a usb webcam and a 802.11b wireless card, the only other decive ont eh wireless is a WRT54G thats hooked to a dual P3-1.2Ghz machine to monitor/control the robot |
| 00:19.15 | rue_mohr | this isn't an asterisk thing and I'm not sure how it came up |
| 00:19.27 | [TK]D-Fender | rue_mohr: If you're dealing with laptops you could set Ekiga up with video on it and just DIAL it from another soft-phone via *... or DIRECT. Jsut set Ekiga to "auto-answer" |
| 00:19.42 | rue_mohr | hmm |
| 00:19.57 | [TK]D-Fender | rue_mohr: Well * CAN be a part of the solution. You just want to "get video" on demand. A million tools to do that. |
| 00:20.15 | [TK]D-Fender | rue_mohr: How do you think those cheap-shit D-Link Wi-Fi cameras work? |
| 00:20.23 | [TK]D-Fender | rue_mohr: Its all the same reall. |
| 00:20.24 | [TK]D-Fender | y |
| 00:20.42 | rue_mohr | no linux video software that exists right now seems to be abel to push more than 1fps live |
| 00:21.16 | rue_mohr | I honestly didn't think to look to voip software |
| 00:21.19 | [TK]D-Fender | rue_mohr: Pardon? Ekiga (previously GnomeMEtteing) has done this for I dunno... over a DECADE... |
| 00:22.49 | rue_mohr | hehe, those people in the video channels are really behind the times |
| 00:24.13 | [TK]D-Fender | rue_mohr: I won't vouch for how it is you came by the answer you got. |
| 00:25.17 | [TK]D-Fender | rue_mohr: Windows 95 came with NetMeeting which did this 15 years ago. Seriously.... that was in the age of 100 Mhz computers. |
| 00:39.40 | *** join/#asterisk unspin (~unspin@S01060026f2f3042d.vc.shawcable.net) |
| 00:42.43 | rue_mohr | thankyou for the peptalk, I wont give up on my project! |
| 00:45.06 | [TK]D-Fender | \o/ |
| 00:45.33 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 00:52.53 | xheliox | man, the pain.. |
| 00:53.13 | xheliox | I can't bare to be on the same planet as these people |
| 00:53.14 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:53.15 | xheliox | I really can't |
| 00:57.22 | dlynes | Any recommendations on an iax or sip softphone for android? |
| 01:00.16 | [TK]D-Fender | dlynes: sipdroid |
| 01:02.29 | Gugge | sipdrod works "okay" ... needs srv support, and for some reason it turns on the screen whenever it reregisters on my desire |
| 01:16.40 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
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| 01:35.05 | rue_mohr | xheliox, me!? please by all means make a wireless rov to show me how its done |
| 01:37.28 | *** join/#asterisk paranox (paranox@anomaly.shits.me) |
| 01:38.16 | paranox | has anyone ever seen this? |
| 01:38.17 | paranox | kernel: dahdi: HDLC Receiver overrun on channel WCT1/0/16 (master=WCT1/0/16) |
| 01:38.34 | paranox | it started happening at random on my asterisk server last thursday |
| 01:38.51 | paranox | and the only way to get our lines back up is to power cycle the isdn ntu |
| 01:39.09 | rue_mohr | hmm |
| 01:39.18 | rue_mohr | how long did the system work for? |
| 01:39.52 | *** join/#asterisk diegomad (~mad@186.86.2.62) |
| 01:40.14 | WIMPy | paranox: Sounds like an interrupt problem. |
| 01:41.53 | paranox | rue_mohr: since i've worked here (~10 months) |
| 01:42.35 | rue_mohr | ah, well |
| 01:42.43 | paranox | WIMPy: as in a pci interrupt? |
| 01:42.52 | rue_mohr | open er up and see if the motherboards caps are ok or if the tops are blown open |
| 01:43.05 | paranox | did that |
| 01:43.08 | paranox | everything looks fine |
| 01:43.14 | rue_mohr | ok |
| 01:43.35 | *** join/#asterisk diegomad (~mad@186.86.2.62) |
| 01:43.40 | paranox | in my pri debug, the issue starts with: -- Timeout occured, restarting PRI |
| 01:43.41 | rue_mohr | I'm out of lucky-shot ideas |
| 01:43.49 | paranox | then scrolls: Sending Set Asynchronous Balanced Mode Extended |
| 01:43.52 | rue_mohr | what the card? |
| 01:43.52 | paranox | until i reset the ntu |
| 01:44.04 | paranox | digium te122 |
| 01:44.28 | paranox | could it be the telco's ntu acting up? |
| 01:44.45 | rue_mohr | (this is where I try to get details needed for one of you people who know how to answer the question) |
| 01:44.52 | paranox | they have reset the circuit, though i have not made them drag out an engineer to test their equipment yet |
| 01:45.26 | rue_mohr | hmm, you should see alarms on the lights on the card if thats the case (I think) |
| 01:45.41 | rue_mohr | paranox, how long can you down the system to try things? |
| 01:45.51 | rue_mohr | do you know how to make a T1 loopback connector? |
| 01:46.01 | paranox | at the moment, not very long |
| 01:46.07 | rue_mohr | hmm |
| 01:46.17 | paranox | i've seen mention of a loopback connector so i could make one |
| 01:46.29 | rue_mohr | 1 sec |
| 01:46.41 | paranox | the annoying thing is there doesn't seem to be a patern to it |
| 01:47.01 | paranox | ie. sometimes it will stay up for 30 minutes, sometimes it will fail after 10 minutes |
| 01:47.17 | rue_mohr | ouch |
| 01:47.24 | paranox | at the moment it has been up for ~50 minutes so i am doing well ;) |
| 01:47.33 | rue_mohr | http://eds.dyndns.org/~ircjunk/images/dscn9333_T1loopback.jpg |
| 01:48.05 | rue_mohr | so they are having to reset their system every ~30 mins? |
| 01:48.19 | *** join/#asterisk Kyosh (whoa@96.246.232.130) |
| 01:48.48 | paranox | i am having to reset the telco's ntu |
| 01:49.00 | paranox | the asterisk box doesn't need a reset |
| 01:49.10 | rue_mohr | oh |
| 01:49.17 | rue_mohr | oh I see |
| 01:49.18 | rue_mohr | sorry |
| 01:49.35 | paranox | which is why i think it might be their equipment playing up |
| 01:50.30 | paranox | also, unplugging and replugging the e1 cable doesn't work either to bring it back up :S |
| 01:52.36 | rue_mohr | have you asked them to run a test-set on the ntu? |
| 01:52.54 | rue_mohr | they usually have a few layers of loopback tests they can do |
| 01:53.04 | rue_mohr | tough when its an intermittent problem tho |
| 01:53.09 | paranox | not yet |
| 01:53.16 | paranox | yeah not sure how that will go |
| 01:53.31 | paranox | it will probably work when they test it and they'll say there is nothing wrong with it |
| 01:54.20 | rue_mohr | like trying to prove the rtp levels of a polycom phone are hosed, I dont know what you can do |
| 01:54.22 | WIMPy | Ok, if un- and replugging the E1 doesn;t help but powercycling the NT does, the problem seems to lie outside ouf your responsibility. |
| 01:54.46 | rue_mohr | trick is otherwise proving it |
| 01:54.55 | paranox | heh, exactly |
| 01:55.02 | WIMPy | What kind of line is used to carry the E1? |
| 01:55.02 | rue_mohr | maybe ask for a test next time it goes belly up |
| 01:55.21 | paranox | rue_mohr: i think that's my plan |
| 01:55.26 | paranox | WIMPy: what do you mean? |
| 01:55.40 | WIMPy | The physical line. |
| 01:55.44 | rue_mohr | cause, come to think of it, WIMPy has a point, if repluging it dosn't work, its still in a failure mode |
| 01:56.13 | paranox | WIMPy: it's a cat5e patch cable between the ntu and pbx |
| 01:56.20 | rue_mohr | before the ntu |
| 01:56.23 | rue_mohr | fiber? |
| 01:56.28 | rue_mohr | sdl? |
| 01:56.33 | WIMPy | The other side. |
| 01:56.44 | paranox | oh, sorry |
| 01:56.50 | WIMPy | From CO to the NT. |
| 01:57.20 | rue_mohr | string... |
| 01:57.46 | paranox | it's over twisted pair |
| 01:57.48 | rue_mohr | * still has no tin-can interface cards... |
| 01:58.49 | WIMPy | Yes, but what kind of line? Uk2? DSL? The NT should give a clue about what it's converting from/to. |
| 01:59.05 | rue_mohr | tin-can has its problems, people can only have two calls on hold at once |
| 01:59.20 | rue_mohr | single pair must be dsl |
| 01:59.22 | rue_mohr | ? |
| 01:59.26 | paranox | sorry, no idea |
| 02:00.03 | *** join/#asterisk diegomad (~mad@186.86.2.62) |
| 02:00.03 | WIMPy | Probably, yes. |
| 02:00.16 | WIMPy | And that's known to have issues. |
| 02:02.00 | paranox | heh, no reset for an hour now |
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| 02:06.02 | thehar | ping? |
| 02:06.13 | *** join/#asterisk diegomad (~mad@186.86.2.62) |
| 02:06.26 | WIMPy | pong! |
| 02:06.30 | thehar | hehe |
| 02:06.45 | thehar | typical PRI cable can just be a passthru, right? |
| 02:06.49 | thehar | i never remember |
| 02:07.12 | WIMPy | passthru? |
| 02:07.22 | thehar | patch |
| 02:07.39 | thehar | straight-thru |
| 02:07.48 | WIMPy | It's a straight 1:1, yes. |
| 02:08.16 | thehar | thought so |
| 02:08.31 | thehar | (*&*(&@#$# piece of (*&$ panasonis d500 vs adtran 924e are fighting |
| 02:09.04 | paranox | strange |
| 02:09.20 | paranox | link just went down but reset itself without any intervention this time |
| 02:09.45 | WIMPy | That's the normal issues you get with DSL. |
| 02:09.55 | WIMPy | Quite annoying :-( |
| 02:10.00 | thehar | urrrrrrrlll |
| 02:51.59 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 02:54.47 | dlynes | [TK]D-Fender, thanks |
| 02:55.21 | dlynes | [TK]D-Fender, do you happen to know if sipdroid works well when the asterisk server is behind a router? |
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| 02:56.13 | [TK]D-Fender | dlynes: Why would any client give a crap about that? |
| 02:56.22 | [TK]D-Fender | dlynes: they don't KNOW it is <- |
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| 03:14.27 | dlynes | [TK]D-Fender, i suppose so, if you put your asterisk server on the dmz |
| 03:15.18 | [TK]D-Fender | dlynes: * needs to be correctly configured for itself. This is not the client's problem. |
| 03:15.36 | [TK]D-Fender | dlynes: The road is not responsible for you putting gas in yuor car |
| 03:16.29 | dlynes | yeah yeah...was just hoping somebody had found a way around the router issue by now, but i guess not |
| 03:17.39 | [TK]D-Fender | dlynes: Router issue? |
| 03:17.57 | [TK]D-Fender | dlynes: Holy shit... hasn't theis gone around the block enough timess EVERY week? |
| 03:18.03 | [TK]D-Fender | ~sipnat |
| 03:18.04 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:18.06 | [TK]D-Fender | ^^^ |
| 03:18.44 | [TK]D-Fender | dlynes: How many years have you been here? |
| 03:18.55 | [TK]D-Fender | dlynes: 4 or 5 now? |
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| 07:09.05 | rishikesh | musiconhold extension not working for me with this http://pastebin.com/GZCrE8Xq |
| 07:09.06 | rishikesh | why? |
| 07:10.32 | *** join/#asterisk ox42 (0x42@devio.us) |
| 07:11.47 | ox42 | Does getting a BYE and the CANCEL after INVITE is a normal call flaw (I thought it should be CANCEL and the BYE) |
| 07:16.18 | ChannelZ | rishikesh: answer() first |
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| 07:44.25 | rishikesh | musiconhold extension not working for me with this http://pastebin.com/GZCrE8Xq why? |
| 07:46.45 | J4zen | Goodmorning guys, i'm having some issues i hope you can help me with or point me in the right direction. Our SIP "trunk" provider has changed their infrastructure in such a way that (as they claim) SIP and RTP streams are coming from two different IP-addresses. The result is that outbound calls still work fine, but inbound calls are dropped as Asterisk doesn't recognise the inbound peer's address (as it differs from the registered "t |
| 07:46.49 | J4zen | it drops the call immediatly |
| 07:46.53 | J4zen | http://pastebin.com/zFj4QzsH this is what happends |
| 07:47.27 | J4zen | how can i create a situation where it drops all anonymous inbound calls, except for the inbound calls coming from the two specified ip addresses? |
| 07:48.16 | rishikesh | how do i setup musiconhold extension |
| 07:48.39 | calmh | rishikesh: 09:16:18 <ChannelZ> rishikesh: answer() first |
| 07:48.48 | rishikesh | i did it |
| 07:48.50 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 07:49.17 | rishikesh | whether that 8888 and 121 should be same or different? |
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| 07:57.50 | rishikesh | plz help me |
| 07:58.44 | AliRezaTaleghani | rishikesh: am not so expert, but, what is up with u? |
| 07:58.48 | AliRezaTaleghani | let me know |
| 07:59.17 | rishikesh | i want to setup extension 121 as musiconhold |
| 07:59.35 | rishikesh | means when a user dial 121 it will play musiconhold |
| 08:00.02 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
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| 08:00.25 | AliRezaTaleghani | define a context : [onholdcon] |
| 08:00.51 | AliRezaTaleghani | add this lines: 121,1,Answer() |
| 08:01.01 | AliRezaTaleghani | 121,n,MusicOnHold(default) |
| 08:01.13 | AliRezaTaleghani | 121,n,Hangup() |
| 08:01.18 | AliRezaTaleghani | all is this |
| 08:01.26 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:01.43 | rishikesh | where to add this |
| 08:01.54 | AliRezaTaleghani | then, define this context for ur externenstion, for example in the time u define sip ext |
| 08:03.00 | rishikesh | but in my extension context its from-internal |
| 08:03.15 | rishikesh | where do i add those lines? |
| 08:03.26 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 08:04.07 | AliRezaTaleghani | u have 2 way |
| 08:04.12 | rishikesh | tell me |
| 08:04.19 | AliRezaTaleghani | 1- change from-internal to the new context |
| 08:04.38 | AliRezaTaleghani | 2- add the following line, in from-internal context |
| 08:04.48 | rishikesh | will it work for my extension if i change that |
| 08:04.49 | AliRezaTaleghani | include => onholdcon |
| 08:05.24 | AliRezaTaleghani | i preffer the second way, to save current dialing plans to |
| 08:05.31 | rishikesh | where can i find from-internal context |
| 08:05.42 | AliRezaTaleghani | /etc/asterisk/extentions.conf |
| 08:05.46 | AliRezaTaleghani | vs |
| 08:05.57 | AliRezaTaleghani | /etc/asterisk/extentions_additional.conf |
| 08:06.01 | AliRezaTaleghani | ;) |
| 08:06.21 | rishikesh | what about exten>= lines |
| 08:06.29 | rishikesh | where should i add those lines |
| 08:06.47 | AliRezaTaleghani | @ the end of the file, that u found the from-internal contex |
| 08:08.30 | *** part/#asterisk ox42 (0x42@devio.us) |
| 08:08.34 | rishikesh | from-internal-additional context is there but not from-internal |
| 08:08.50 | rishikesh | where should i add 121,1,Answer etc |
| 08:08.59 | E-bola | Do anybody know why some of my phones send asterisk: Got SIP response 403 "Use Proxy" |
| 08:12.53 | boodu | bye |
| 08:13.34 | AliRezaTaleghani | search the the context on that extentions*.conf for that special context |
| 08:17.14 | *** join/#asterisk ruyo (~psantos@a81-84-220-57.cpe.netcabo.pt) |
| 08:17.29 | rishikesh | i have found |
| 08:17.33 | rishikesh | from-internal |
| 08:17.38 | rishikesh | its on extension.conf |
| 08:20.10 | *** join/#asterisk lyetz (~lyetz@pool-71-177-220-108.lsanca.fios.verizon.net) |
| 08:22.23 | rishikesh | its not working for me |
| 08:22.30 | rishikesh | extension 121 is not available |
| 08:25.42 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 08:25.49 | rishikesh | extension 121 is not available |
| 08:27.41 | AliRezaTaleghani | what extention do u login with? |
| 08:27.44 | AliRezaTaleghani | is it sip too? |
| 08:27.55 | AliRezaTaleghani | if it's give me the out of: |
| 08:28.12 | AliRezaTaleghani | sip show users |
| 08:28.38 | AliRezaTaleghani | && did u reload new config?! |
| 08:28.52 | AliRezaTaleghani | rishikesh: && did u reload new config?! |
| 08:30.38 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 08:32.24 | stix | Hi guys. Can I change the rtp port range which asterisk uses? |
| 08:33.29 | E-bola | yes |
| 08:33.33 | Tim_Toady | yes stix in rtp.conf |
| 08:35.40 | stix | thanks :) |
| 08:36.11 | stix | is nat=yes in sip.conf useless if I don't have a sip_nat.conf? |
| 08:38.27 | Tim_Toady | sip_nat.conf? thats not a stabdart *conf file |
| 08:38.36 | Tim_Toady | standart |
| 08:40.17 | stix | okay, I just read about it here: http://blog.iwayvietnam.com/tuanta/2010/03/10/howto-setup-asteriskfreepbx-behind-nat/ |
| 08:40.31 | stix | it is a way to tell asterisk what your external IP is |
| 08:41.16 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 08:42.36 | stix | it is also mentioned here: http://www.voip-info.org/wiki/index.php?page_id=410&tk=4ad003c47dcc8f438097&comments_page=1 |
| 08:42.49 | Tim_Toady | not really, u can just add this in ur sip.conf: externhost or externip |
| 08:42.52 | *** join/#asterisk BANSAL (~bansal@117.199.119.131) |
| 08:43.58 | E-bola | its a freepbx file |
| 08:44.02 | E-bola | as it says in the guide... |
| 08:44.17 | Tim_Toady | stix read here for details how to configure sip: http://svnview.digium.com/svn/asterisk/branches/1.6.2/configs/sip.conf.sample?view=markup |
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| 08:51.55 | rishikesh | i restart my pc even |
| 08:54.27 | ectospasm | rishikesh: does the Asterisk CLI command,"dialplan show from-internal" show the target extension? |
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| 09:22.18 | *** join/#asterisk rishikesh (~Rishikesh@117.242.156.57) |
| 09:22.25 | rishikesh | hi |
| 09:23.23 | rishikesh | sip*CLI> dialplan show from-internal |
| 09:23.23 | rishikesh | [ Context 'from-internal' created by 'pbx_config' ] |
| 09:23.23 | rishikesh | <PROTECTED> |
| 09:23.24 | rishikesh | <PROTECTED> |
| 09:23.24 | rishikesh | <PROTECTED> |
| 09:23.24 | rishikesh | sip*CLI> |
| 09:23.26 | rishikesh | -= 0 extensions (0 priorities) in 1 context. =- |
| 09:23.30 | rishikesh | sip*CLI> |
| 09:26.10 | ectospasm | !pastebin | rishikesh |
| 09:26.18 | ectospasm | oops, not in this channel |
| 09:27.15 | ectospasm | rishikesh: 121 isn't listed there, what does your [from-internal] context in extensions.conf/extensions.ael look like? Use http://pastebin.com if you need to paste configuration. |
| 09:27.52 | *** join/#asterisk jetlag (~jetlag@pool-173-61-210-239.cmdnnj.east.verizon.net) |
| 09:29.30 | rishikesh | what to do? |
| 09:29.48 | rishikesh | i have extension no from 301-306 |
| 09:29.50 | *** join/#asterisk ruyo (~psantos@a81-84-220-57.cpe.netcabo.pt) |
| 09:29.58 | rishikesh | they all are using context=from-internal |
| 09:30.52 | rishikesh | in the from-internal context, i have added include >= radio |
| 09:30.53 | ectospasm | you need to separate the word "extension" from meaning a phone in your mind. |
| 09:31.09 | rishikesh | then what to do? |
| 09:31.34 | ectospasm | you see the extensions 301 through 306 in extensions.conf? |
| 09:31.43 | ectospasm | ...or are they in the radio context? |
| 09:31.45 | rishikesh | yes |
| 09:31.58 | rishikesh | i see in extension.conf |
| 09:31.58 | tzafrir | infobot, tell ectospasm about pb |
| 09:32.15 | *** join/#asterisk _omer (~omer@119.158.28.150) |
| 09:32.19 | tzafrir | ectospasm, that's how you send it in a private message |
| 09:32.24 | ectospasm | Ah, OK |
| 09:32.34 | ectospasm | I haven't frequented this channel in a while |
| 09:32.56 | kaldemar | ~pb |
| 09:32.57 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 09:32.57 | ectospasm | ...and I must have forgotten. |
| 09:34.22 | ectospasm | rishikesh: you should put the 121 priorities (steps) in the context containing 301-306. Show us your config in pastebin if you need to. |
| 09:35.08 | rishikesh | how? |
| 09:35.41 | ectospasm | paste your extensions.conf into the pastebin form, and send us the link. |
| 09:35.59 | ectospasm | some may allow you to pick a file to use instead of pasting |
| 09:38.16 | rishikesh | i use virtual machine |
| 09:38.23 | rishikesh | how do i copy paste? |
| 09:38.40 | ectospasm | I dunno, that'd depend on your VM implementation |
| 09:39.05 | rishikesh | i use to connect from putty through ssh |
| 09:40.48 | ectospasm | http://pastebin.com/puhvBZd4 |
| 09:40.55 | ectospasm | try that rishikesh |
| 09:42.41 | ectospasm | wait, I corrected it |
| 09:42.57 | ectospasm | rishikesh: try http://pastebin.com/e69esTTS |
| 09:44.02 | rishikesh | what is that ; other extensions |
| 09:44.17 | ectospasm | it's a comment, a place holder for your other extensions. Ignore it. |
| 09:45.11 | rishikesh | its the same thing as include >= radio |
| 09:45.34 | rishikesh | i have place that 121,n,answer in radio context in extension.conf |
| 09:46.39 | ectospasm | ah, I see. Unless you have 1 priority somewhere, 121 doesn't have a place to start. You need only one '1' priority (instead of 'n'), but not having one means Asterisk doesn't know where to begin when you dial 121 |
| 09:46.55 | rishikesh | its example |
| 09:46.56 | ectospasm | rishikesh: see this: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-5.html |
| 09:46.58 | rishikesh | i add 1 |
| 09:47.07 | rishikesh | and followed by answer |
| 09:48.32 | ectospasm | you'll have to use pastebin to copy and paste (from PuTTY) everything you're talking about, since I can't tell what's wrong this way. |
| 09:49.45 | E-bola | dont understand what you have to setup in a firewall to have sip clients work behind nat when your asterisk is somewhere on the internet |
| 09:50.21 | ectospasm | ~nat |
| 09:50.22 | infobot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 09:50.33 | E-bola | have setup all that |
| 09:51.35 | ectospasm | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions it may be more configuration than just setting up Asterisk and the firewall... |
| 09:53.50 | ectospasm | typically all you need is nat=yes and qualify=yes (or some value like 2000 for 2sec of keep-alive) |
| 09:55.38 | rishikesh | how do i select all in vi |
| 09:55.59 | E-bola | yep have that as well |
| 09:56.06 | E-bola | but it still depends on the firewall as far as i know |
| 09:56.20 | rishikesh | what is that place holder for other extenions? |
| 09:56.29 | E-bola | my problem is, i have 2 phones on the same lan behind the same nat firewall connected to the same asterisk server somewhere on the internet |
| 09:56.40 | E-bola | if 1 calls the other, it doesnt register it when you pick up the phone |
| 09:57.58 | ectospasm | rishikesh: it's just a comment, IGNORE IT |
| 09:58.14 | rishikesh | but i have done all the settings required |
| 09:58.19 | rishikesh | what should i do now? |
| 09:58.25 | rishikesh | i am not able to call 121 |
| 09:58.49 | rishikesh | i have been trying this for past one week |
| 09:58.50 | ectospasm | you need to paste the entire configuration to pastebin, giving us one line at a time is not helpful |
| 09:58.56 | rishikesh | i am not able to make this work |
| 09:59.15 | rishikesh | how do i copy paste from putty |
| 09:59.32 | ectospasm | rishikesh: if you need to, transfer extensions.conf to your workstation, and use notepad (or some other text editor) to select all of it so you may paste it. |
| 09:59.58 | ectospasm | rishikesh: if you highlight in putty with the mouse, it's automatically copied to the clipboard, if memory serves me correctly |
| 10:01.40 | ectospasm | E-bola: if both extensions in sip.conf have both nat=yes and qualify=yes, you shouldn't need to do anything else. qualify=yes keeps the connection open so the firewall knows where to send the traffice when one phone calls another. |
| 10:02.16 | ectospasm | you would need to set that in both contexts in sip.conf, I don't think having both in [general] will cut it. |
| 10:02.46 | ectospasm | ...but even then, some clients won't work. |
| 10:04.04 | E-bola | Set what in all contexts? |
| 10:04.15 | E-bola | i have qualify= and nat= on each peer |
| 10:05.24 | rishikesh | see this http://pastebin.com/emHWRWUm |
| 10:05.31 | rishikesh | extension.conf |
| 10:06.04 | E-bola | ectospasm: hmm i just configured a stun server on each phone (snopm 320) and that apparently made it work.... |
| 10:06.18 | *** join/#asterisk BANSAL (~bansal@117.199.125.229) |
| 10:06.44 | rishikesh | this extension_additiona.conf http://pastebin.com/f6xBtfp1 |
| 10:07.34 | ectospasm | E-bola: yeah, some endpoints require that. |
| 10:07.36 | rishikesh | this is sip.conf http://pastebin.com/v58QPvBD |
| 10:07.58 | E-bola | ectospasm: it must have been either stun or ICE that fixed it |
| 10:08.06 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 10:08.08 | E-bola | thats fine, but i dont understand why its necesary |
| 10:09.17 | ectospasm | rishikesh: the extensions separater is either '=' or "=>", not ">=" |
| 10:09.19 | E-bola | weird now it works even without a stun server, i must have changed something important in asterisk... |
| 10:09.31 | DND | guys any idea why my x-lite and PortGo shows 503 service unavailable whenever no one is answering the phone? |
| 10:10.28 | dlynes | rishikesh, if you're wanting much help on your config, you're probably going to be better off asking on #freepbx as well..it uses customized configs that not a lot of people in #asterisk understand, and with the new version, I believe it's migrated to freeswitch as well |
| 10:10.32 | DND | i mean after it rings for some time, it send busy tone then a message saying "The Number is not answering" then gives out 503 |
| 10:12.11 | DND | so the problem is after that i cannot dial anything unless restarting softphones to re-register |
| 10:12.11 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 10:14.58 | *** join/#asterisk DelphiWorld (~Delphi@41.200.0.40) |
| 10:15.00 | DelphiWorld | hi |
| 10:15.06 | DelphiWorld | how do i show my iax2 trunks? |
| 10:15.13 | DelphiWorld | tzafrir: :P |
| 10:15.30 | ectospasm | DelphiWorld: iax2 show peers should work, or maybe it's iax2 show users |
| 10:16.07 | DelphiWorld | ectospasm: this show my iax users, no? |
| 10:16.11 | DelphiWorld | i want to show iax trunks |
| 10:16.43 | ectospasm | iax2 show peers should work |
| 10:16.54 | ectospasm | works for me here. |
| 10:18.14 | DelphiWorld | ectospasm: do you have ast1.4? |
| 10:18.15 | *** join/#asterisk jetlag (~jetlag@pool-173-61-207-237.cmdnnj.east.verizon.net) |
| 10:19.24 | ectospasm | DelphiWorld: no, but I could in a second, gimme a moment |
| 10:19.42 | hrhrhr | it's the same in 1.4 |
| 10:19.46 | DelphiWorld | ectospasm: please, only if you could get 1.4 iax.conf file (default) |
| 10:20.19 | DelphiWorld | how do i list all my asterisk channels modules? |
| 10:20.58 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:22.05 | ectospasm | DelphiWorld: ls /usr/lib/asterisk/modules/chan* |
| 10:22.16 | DelphiWorld | lol ectospasm |
| 10:22.25 | DelphiWorld | ectospasm: do asterisk 1.4 have "include" in conf? |
| 10:22.39 | DelphiWorld | i don't want to kill my iax.conf file but i want to include other peers |
| 10:22.42 | ectospasm | or, you can run "module show like *chan* |
| 10:22.58 | *** join/#asterisk Dovid (~Dovid@194.98.133.158) |
| 10:23.05 | ectospasm | DelphiWorld: yeah, you can #include another file, or include => another context |
| 10:23.21 | DelphiWorld | ectospasm, example for include file please? |
| 10:23.25 | ectospasm | Asterisk 1.6.2 (and maybe others) let you use templates. |
| 10:23.39 | Dovid | anyone know what this error means ? SS7 got event: HDLC Bad FCS(8) on span 1/0 |
| 10:24.07 | ectospasm | DelphiWorld: #include filename.conf (if it's in /etc/asterisk/) or "#include /path/to/filename.conf" |
| 10:24.21 | DelphiWorld | ectospasm: thank you |
| 10:24.46 | ectospasm | Dovid: Bad Frame Check Sequence, it's when an HDLC frame fails the checksum and is discarded |
| 10:25.12 | ectospasm | usually one instance of that is nothing to be alarmed about |
| 10:25.25 | Dovid | it keeps coming up on the screen |
| 10:25.29 | DelphiWorld | ectospasm: doe i need to use ""? |
| 10:25.31 | ectospasm | if it happens frequently enough, you'll have audio or D-channel problems |
| 10:25.34 | ectospasm | DelphiWorld: no |
| 10:25.53 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:26.46 | Dovid | ectospasm: If it comes up every few seconds that means there is an issue on the line ? |
| 10:27.21 | hrhrhr | DelphiWorld: what kinda cfg are you gonna include? |
| 10:28.20 | DelphiWorld | hrhrhr: other iax2 peers |
| 10:28.59 | hrhrhr | i can understand wanting to keep the file separate but how many peers you gonna add? |
| 10:29.20 | DelphiWorld | hrhrhr: lol only 3 or 4;) |
| 10:30.03 | hrhrhr | is it worth it? :) |
| 10:30.29 | DelphiWorld | hrhrhr: yes, for me! |
| 10:30.29 | hrhrhr | considering they are jailed within the [context] anyway |
| 10:30.34 | DelphiWorld | hrhrhr: to kype it easy to read |
| 10:30.59 | DelphiWorld | and is not evean working for me if i show my iax peers or registry |
| 10:31.53 | hrhrhr | 'iax2 show peers' gives no output? |
| 10:32.18 | *** join/#asterisk wikii (~wiki.mir@119.160.105.172) |
| 10:32.21 | DelphiWorld | hrhrhr: yes |
| 10:32.47 | hrhrhr | tried restarting asterisk? |
| 10:33.00 | hrhrhr | it does that with me for some sip stuff occasionally |
| 10:33.10 | hrhrhr | seems to start in 'spaz' mode |
| 10:33.22 | SiNGLer | it enough to "iax2 reload" ot "iax reload" don't remember which one :) |
| 10:33.31 | wikii | [TD]-fender please check http://pastebin.ca/1922156 |
| 10:33.44 | hrhrhr | i've had some odd results with iax2 reload |
| 10:33.47 | hrhrhr | 9/10 it works |
| 10:33.53 | wikii | Singler please check http://pastebin.ca/1922156 |
| 10:34.05 | hrhrhr | if you're able to tho, a full asterisk restart might be worth a shot |
| 10:35.17 | DelphiWorld | hrhrhr: http://www.dpaste.org/7KtC/ |
| 10:35.48 | *** join/#asterisk slavon (~slavon@178.177.11.47) |
| 10:36.07 | SiNGLer | wikii: it's the same problem with fax? |
| 10:36.12 | hrhrhr | does that work? you may need trunk=yes |
| 10:36.31 | DelphiWorld | hrhrhr: lol trunk=yes require timing |
| 10:36.34 | wikii | Singler yeah |
| 10:36.41 | hrhrhr | you may also need a context per peer... |
| 10:36.42 | slavon | hello. where is asterisk module for jabber and gmail in RH asterisk repository? |
| 10:37.05 | SiNGLer | wikii: try setting same callerid as in phone, check if it works |
| 10:39.27 | wikii | soory cant get u |
| 10:40.36 | wikii | singler where i will change callerid |
| 10:40.37 | wikii | ? |
| 10:40.58 | SiNGLer | i'd do this: Set(CALLERID(num)=100) |
| 10:41.38 | wikii | ok i will set this in my dialplan |
| 10:41.38 | DelphiWorld | hrhrhr: could you configure it for me please;) |
| 10:42.11 | wikii | one more question singler.... i want to configure in asterisk |
| 10:42.23 | wikii | one more question singler.... i want to configure SMS in asterisk |
| 10:42.41 | SiNGLer | I never did SMS in asterisk |
| 10:42.57 | SiNGLer | so will not be able to help you |
| 10:43.00 | wikii | ok thankyou |
| 10:43.02 | slavon | for sms you may use misdn |
| 10:43.27 | wikii | wat is misdn |
| 10:43.28 | DelphiWorld | slavon: misdn for sms? |
| 10:43.55 | hrhrhr | DelphiWorld: i can barely configure my own pbx :P |
| 10:44.10 | DelphiWorld | hrhrhr: ? |
| 10:44.23 | slavon | sorry) chan_mobile) |
| 10:44.58 | slavon | or use dahdi if prov support it |
| 10:45.09 | slavon | in our projects we use external xml web services. ) |
| 10:45.24 | wikii | Slavon please send any howto link |
| 10:45.41 | slavon | http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 10:46.14 | wikii | Thankyou |
| 10:46.24 | slavon | np |
| 10:46.59 | E-bola | Is there anyway to make the asterisk console show longer lines? |
| 10:47.02 | E-bola | like for sip show peers |
| 10:47.40 | DelphiWorld | hrhrhr, ok, now how i should add iax users? |
| 10:47.51 | *** part/#asterisk wikii (~wiki.mir@119.160.105.172) |
| 10:49.25 | SiNGLer | DelphiWorld: check this link: http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf |
| 10:50.08 | DelphiWorld | SiNGLer: thank you a lot |
| 10:50.15 | SiNGLer | np |
| 10:53.00 | DelphiWorld | SiNGLer: iax users/trunks are confusing |
| 10:57.13 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 10:58.10 | DelphiWorld | yo elred_! |
| 10:58.40 | elred_ | salut Deathvalley122 |
| 10:58.43 | elred_ | err |
| 10:58.50 | elred_ | salut DelphiWorld |
| 10:59.05 | Deathvalley122 | -_- |
| 10:59.07 | DelphiWorld | elred_: lol |
| 11:05.37 | ruyo | Anyone knows where is the dahdi debug going? |
| 11:06.05 | ruyo | Or if I need to restart dahdi after doing "echo 1 > /sys/module/dahdi/parameters/debug" |
| 11:06.37 | *** join/#asterisk frek818_ (~frek818@rrcs-74-62-208-50.west.biz.rr.com) |
| 11:11.04 | *** join/#asterisk mbranca (daemon@mi-gw1.voismart.net) |
| 11:18.38 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 11:25.43 | DelphiWorld | ok |
| 11:25.48 | DelphiWorld | now i have a sip and a iax2 user |
| 11:25.56 | DelphiWorld | how do i configuere them to call betwan users? |
| 11:26.05 | DelphiWorld | coppice: :P |
| 11:28.10 | fauxalliance | dialplan DelphiWorld.. presumably they just _dial_ each other. |
| 11:28.49 | DelphiWorld | bhmmm |
| 11:28.53 | DelphiWorld | hmmm |
| 11:29.00 | DelphiWorld | how do i specify endpoint in app_dial? |
| 11:31.16 | fauxalliance | magic dialling wand? |
| 11:32.42 | fauxalliance | http://www.asteriskdocs.org/html/ch05s03s03.html @ DelphiWorld |
| 11:33.21 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 11:36.09 | DelphiWorld | Faustov: but see: |
| 11:36.28 | DelphiWorld | Faustov: i want to create a user with "101", that's both iax and sip so how ast identify it? |
| 11:36.47 | Faustov | WHAT |
| 11:36.54 | Faustov | ah, autocompletefail |
| 11:37.41 | SiNGLer | DelphiWorld: 101 can be both. In dialplan you must specify what you want to dial, ex exten => 101,1,Dial(IAX/101) |
| 11:38.07 | DelphiWorld | SiNGLer: so i should add also dial sip? |
| 11:38.20 | DelphiWorld | SiNGLer: sory my friend will comm online and explin to him please |
| 11:39.13 | SiNGLer | DelphiWorld: it depends what you want to do. if you want that dialing 101 would ring SIP, then you specify SIP/101, if you want IAX, then you specify IAX |
| 11:41.19 | *** join/#asterisk Rienzilla (rien@sinas.rename-it.nl) |
| 11:42.55 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:44.50 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 11:45.26 | Rienzilla | Hey everyone. I just got a new voip phone. I have a bunch of snom 300's and snom 360's, which are provisioned via dhcp. All these phones nicely get an address via dhcp, and then request their settings file from the webserver set in dhcpd.conf. But my new phone (snom M9) suddenly requests / (instead of /blah-{macaddress}.html). Do m9's configure themselves differently? |
| 11:50.42 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:53.10 | DelphiWorld | SUP DUDE sekil |
| 11:56.59 | *** join/#asterisk rishikesh (~Rishikesh@117.242.156.57) |
| 11:57.46 | rishikesh | somebody help me plz |
| 11:58.09 | rishikesh | i need to configure music on hold on ext no. 121 |
| 11:58.31 | rishikesh | like voicemail when i dial 121 i would like to listen to musiconhold |
| 12:00.10 | SiNGLer | rishikesh: pastebin your configuration |
| 12:00.41 | rishikesh | http://pastebin.com/v58QPvBD |
| 12:01.22 | SiNGLer | I don't see your configuration there |
| 12:01.57 | *** join/#asterisk Weazel (~deazel@62-90-210-148.barak.net.il) |
| 12:02.11 | rishikesh | see this http://pastebin.com/emHWRWUm |
| 12:02.19 | rishikesh | this extension_additiona.conf http://pastebin.com/f6xBtfp1 |
| 12:03.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:04.37 | SiNGLer | I guess you need to set up some test pbx, so that freepbx conf wouln't be in the way. Basically what you need is here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold I don't know where in freepbx conf it should be inserted |
| 12:06.02 | SiNGLer | oh, and your radio context is incorrect |
| 12:06.23 | SiNGLer | it says "exten >= 121,1,Answer" etc, but should be "=>" |
| 12:06.58 | Weazel | hey guys, i; |
| 12:07.04 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 12:07.35 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
| 12:07.54 | Weazel | hey guys, i'm having trouble with outbound fax on my PBX, looks like the system is sending the word "fax" instead of any numbers to the PRI - the fax is analog machine connected to an analog PBX which is connected via PRI to the asterisk... can anyone shed some light on this issue ? |
| 12:08.35 | Weazel | it seems it is displaying " Called g1/fax " I've searched around google, but only found people with the problem not with a solution |
| 12:08.54 | SiNGLer | Weazel: check if correct exten is comming from your analog pbx |
| 12:09.02 | SiNGLer | maybe it's only an error in dialplan |
| 12:09.05 | [TK]D-Fender | SiNGLer: Because that is what YOU put in your DIAL. |
| 12:09.13 | [TK]D-Fender | SiNGLer: Maybe you should look at what you're doing. |
| 12:09.17 | Weazel | SiNGLer: what do you mean? how can i check that? |
| 12:09.42 | SiNGLer | [TK]D-Fender: what are you talking? I was commenting rishikesh pastebin |
| 12:10.12 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:10.48 | [TK]D-Fender | rishikesh: Ok, then that was for you |
| 12:10.59 | Weazel | i think its an error in the dialplan aswell, but i can't seem to find the problem, inbound faxes are recieved fine, and in the CLI i see the call being dialed as it should but right when it says called g1/0524242424 it'll say |
| 12:11.10 | SiNGLer | [TK]D-Fender: chat snip: http://pastebin.com/2rnFbevn |
| 12:11.14 | Weazel | app_dial.c: -- DAHDI/2-1 is proceeding passing it to DAHDI/61-1 |
| 12:11.18 | Weazel | chan_dahdi.c: -- Redirecting DAHDI/61-1 to fax extension |
| 12:11.21 | Weazel | chan_dahdi.c: -- Hungup 'DAHDI/2-1' |
| 12:12.14 | *** join/#asterisk coppice (~chatzilla@m121-203-226-116.smartone-vodafone.com) |
| 12:12.35 | SiNGLer | Weazel: pastebin dialplan and verbose call log |
| 12:12.59 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 12:13.15 | [TK]D-Fender | HAHAHHAHAA |
| 12:13.18 | *** join/#asterisk AlHafoudh (~AlHafoudh@127.248.broadband15.iol.cz) |
| 12:13.23 | Weazel | there is the verbose |
| 12:13.25 | Weazel | http://pastebin.com/RnUErDXX |
| 12:13.26 | [TK]D-Fender | Someone actually posted a FreePBX dialplan here?! |
| 12:13.34 | drmessano | yes, sadly |
| 12:13.35 | [TK]D-Fender | Yup, it's a Monday all right.... |
| 12:13.42 | drmessano | and 3 lines of FAIL |
| 12:14.10 | Weazel | :( i'm pretty new to this whole thing, cut a noob some break |
| 12:14.34 | [TK]D-Fender | Weazel: Your inbound call appears to be from a FAX. |
| 12:15.08 | Weazel | [TK]D-Fender: but its an outgoing fax |
| 12:15.09 | [TK]D-Fender | Weazel: What Are you doing here exactly? |
| 12:15.17 | Weazel | i don't understand this behaviour |
| 12:15.49 | SiNGLer | I think it is automatic fax detection, which redirects call. |
| 12:16.10 | Weazel | i got 2 pbx connected via PRI cards 1 asterisk 1 analog old one, the fax is connected to the analog one, and when i try to dial a fax number out from the old pbx it goes through the pri of the asterisk and this is what happens |
| 12:16.30 | Weazel | it doesn't happend when a fax is recieved nor if i dial out from any other analog ext |
| 12:16.38 | Weazel | only when it is from a fax machine |
| 12:16.49 | [TK]D-Fender | Weazel: Disable fax detect on your channels leading to the other PBX |
| 12:17.26 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 12:17.34 | Weazel | u mean the faxdetect=incoming disable that ? |
| 12:17.47 | [TK]D-Fender | [08:16]<[TK]D-Fender>Weazel: Disable fax detect on your channels leading to the other PBX <---------------- |
| 12:18.29 | DelphiWorld | [TK]D-Fender: ast now in openwrt d-link dir300 |
| 12:18.42 | [TK]D-Fender | DelphiWorld: \o/ |
| 12:18.49 | DelphiWorld | [TK]D-Fender: :P |
| 12:20.21 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:20.22 | Weazel | [TK]D-Fender: if i disable my fax detect, wouldn't it affect my incoming faxes aswell ? |
| 12:20.43 | [TK]D-Fender | Weazel: Incoming from where? |
| 12:20.44 | Weazel | or can i put the faxdetect=incoming under a group=1 ? |
| 12:21.06 | [TK]D-Fender | Weazel: And you "group" isn't something you put other parms under in there |
| 12:22.01 | Weazel | from PRI of my main line |
| 12:22.08 | Weazel | that is connected to asterisk |
| 12:22.24 | Weazel | PRI > asterisk <PRI> Analog pbx |
| 12:22.55 | [TK]D-Fender | [08:17]<[TK]D-Fender>[08:16] <[TK]D-Fender> Weazel: Disable fax detect on your channels leading to the other PBX <---------------- |
| 12:23.03 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:23.45 | Weazel | (canceled the faxdetect and trying to send a fax) |
| 12:24.29 | Weazel | ok thanks i'm trying |
| 12:24.33 | Weazel | bless you D-fender |
| 12:29.15 | Dovid | hi |
| 12:29.31 | Dovid | on Asterisk when using ss7 is there any way to set the national or international indicator from the dial plan ? |
| 12:30.05 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 12:32.13 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 12:33.41 | Weazel | [TK]D-Fender: how can i set faxdetect=incoming only to specific channels |
| 12:33.45 | Weazel | ? |
| 12:33.58 | [TK]D-Fender | Weazel: Set parms. do "channel => ...". Change parms do som more "channel =>" |
| 12:34.36 | *** join/#asterisk markitoxs (~miranda83@lumison-gw.dub.ftuk.net) |
| 12:35.14 | tzafrir | Dovid, which ss7? chan_ss7? chan_dahdi? |
| 12:35.45 | markitoxs | Im trying to log all BYE messages, as i am inerested in the RTP-RxStat field, but a netcat does not capture that header, any suggestions? |
| 12:36.20 | Weazel | [TK]D-Fender: not sure how to do that :/ |
| 12:36.44 | Dovid | tzafrir: Chan_dahdi |
| 12:37.28 | *** join/#asterisk frek818_ (~frek818@rrcs-74-62-208-50.west.biz.rr.com) |
| 12:38.27 | tzafrir | Weazel, do you want to set it to all dahdi channels? All PRI channels? |
| 12:38.29 | hrhrhr | my understanding is with voice, * will choose the least cpu taxing codec. so g771 should be preferable over gsm, ya? |
| 12:38.34 | hrhrhr | g711 |
| 12:39.05 | tzafrir | hrhrhr, not exactly. Asterisk will attempt not to transcode at all |
| 12:39.31 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 12:39.47 | tzafrir | so if you e.g. call from gsm, gsm will get a preference |
| 12:40.11 | Weazel | tzafrir: only to channels 1 through 31, rest can be disabled |
| 12:40.32 | tzafrir | so basically: |
| 12:40.41 | tzafrir | faxdetect = incoming |
| 12:40.55 | tzafrir | chanel => 1-15,17-31 |
| 12:41.00 | hrhrhr | if i've replaced my voice files with g711, they're not gonna get used over gsm? ;/ |
| 12:41.05 | tzafrir | faxdetect = none |
| 12:43.01 | Weazel | thanks |
| 12:43.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:52.38 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 12:52.55 | *** join/#asterisk renshen (~Rino@93-63-217-144.ip29.fastwebnet.it) |
| 12:53.51 | hrhrhr | http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 12:53.56 | hrhrhr | anyone using these? |
| 12:54.02 | hrhrhr | lots of files are missing... |
| 12:54.42 | drmessano | Probably not many using the missing ones |
| 12:54.52 | drmessano | Due to their "missing" status |
| 12:54.55 | [TK]D-Fender | hrhrhr: WIKI = random otdated crap |
| 12:55.00 | [TK]D-Fender | outdated* |
| 12:55.19 | hrhrhr | awesome |
| 12:55.31 | hrhrhr | i guess i have to buy the voices then? |
| 12:56.23 | drmessano | Depends on what voices you need |
| 12:56.44 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:56.51 | drmessano | If you need something beyond the Asterisk standard voices, yes.. That's why people do voiceover work |
| 12:57.03 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:58.27 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 12:59.07 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 12:59.25 | Dovid | tzafrir: Chan_dahdi |
| 12:59.32 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 12:59.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:59.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:01.23 | [TK]D-Fender | hrhrhr: http://www.theivrvoice.com/ |
| 13:03.38 | russellb | http://www.digium.com/en/products/ivr/ |
| 13:04.16 | beek | hrhrhr: You're welcome to the voices in my head. |
| 13:09.11 | Dovid | anyone know Matthew Fredrickson name here ? |
| 13:11.38 | malcolmd | Dovid: creslin |
| 13:11.41 | stix | There is an encrypted version of SIP right? Where can I read about it? |
| 13:16.17 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 13:19.39 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 13:21.11 | *** join/#asterisk rrb3942 (~rbullock@208.34.96.186) |
| 13:22.03 | Dovid | thanks |
| 13:22.50 | ruyo | Does anyone have any idea why incomming calls from an analog line makes DTMF sounds when answering in an analog phone? |
| 13:23.36 | ruyo | Using a analog card (Openvox A400P) |
| 13:23.51 | [TK]D-Fender | ruyo: Perhaps your telco uses that to signal DID's delivered over that line. |
| 13:24.04 | [TK]D-Fender | ruyo: Your card doesn't make the other side send you DTMF. |
| 13:24.21 | tzafrir | ruyo, version of asterisk? |
| 13:24.30 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 13:25.06 | ruyo | I thought so, so I added bystdetect and cidsignalling=dtmf. Asterisk is 1.4.34 and dahdi is 2.3.0.1 |
| 13:25.37 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 13:25.47 | ruyo | Just a sec, I'll paste chan_dahdi.conf |
| 13:25.50 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 13:28.46 | ruyo | chan_dahdi.conf and system.conf: http://pastebin.com/4t64CGcJ |
| 13:32.22 | ruyo | I thought of echotraining using DTMF to adjust echo canceling, but even without it I get them. |
| 13:32.30 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 13:33.04 | [TK]D-Fender | ruyo: Do yuo have "usecallerid=yes"? |
| 13:33.33 | ruyo | I do. |
| 13:34.09 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:34.24 | ruyo | I've tried with "usecallerid=yes" and without it, but never "usecallerid=no". I don't know what the default is. |
| 13:35.36 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 13:36.55 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 13:37.46 | ruyo | By the way, analog phones connected to an analog gateway (analog->sip) doesn't have that problem. |
| 13:39.39 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 13:42.18 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
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| 13:46.14 | *** join/#asterisk peep (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 13:46.21 | peep | good morning all |
| 13:47.48 | peep | could anyone give a noob some help with a simple dial plan? |
| 13:48.12 | *** join/#asterisk hardcore (~hardcore@2001:470:1f0b:761::1) |
| 13:48.20 | pabelanger | ~ask |
| 13:48.20 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:48.26 | pabelanger | peep: ^^ |
| 13:48.39 | *** join/#asterisk telnettech (~chatzilla@216.49.139.56) |
| 13:48.49 | hardcore | :) |
| 13:48.56 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 13:49.32 | peep | heh, see total noob! |
| 13:50.16 | ruyo | I bet infobot is here against his will. |
| 13:50.39 | [TK]D-Fender | ~infobot |
| 13:50.40 | infobot | [tk]d-fender, i love abuse, feed me!, or whack, yo |
| 13:51.17 | ruyo | Or not. |
| 13:51.37 | [TK]D-Fender | peep: PASTEBIN is your friend. Show us what you're doing, what happens and where you don't like the oucome. |
| 13:51.39 | [TK]D-Fender | ~pb |
| 13:51.39 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:51.41 | [TK]D-Fender | ^^^^ |
| 13:52.39 | pabelanger | infobot: have a cookie |
| 13:52.40 | infobot | Ta. *Munch* |
| 13:52.49 | peep | http://pastebin.ca/1923321 - in this simple custom context im attempting to write a line into a DB when the call is created. Should be simple but if the far end hangs up before the cleanup and disconnect lines are hit a ton of zombie connections get left open on the mysql server. the voip-info entry for mysql() recommends cleaning up the connections in the h extension but there were no examples |
| 13:52.59 | *** join/#asterisk andrechr (~andrechr@nrcc-l3-gateway.nrcc.noklab.com) |
| 13:53.01 | peep | basically - I know im doing it wrong, I just dont know WHAT im doing wrong! |
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| 13:58.27 | Weazel | i get "Set(REALCALLERIDNUM=9000127" how do i add a prefix of 3 to this callerid ? |
| 13:58.33 | marcompile | so does anyone know how can I use multiple soundcards with chan_alsa |
| 13:58.47 | FlashDeluxe | Hi @ all! Ive got a problem, if i want to execute "dahdi_genconf" i get the message "Empty configuration -- no spans" but is see the card mit lspci |
| 13:59.10 | [TK]D-Fender | Weazel: Show us the call |
| 13:59.43 | pabelanger | Weazel: Set(REALCALLERIDNUM=39000127")? Or are you looking to do it dynamically? |
| 14:00.18 | [TK]D-Fender | peep: What is the point of doing those 2 commands AFTER the dial? |
| 14:00.39 | [TK]D-Fender | peep: Oh, and your "h" won't ever get called because you have no "1" priority |
| 14:01.31 | peep | [TK]D-Fender - I originally was calling them after line 5 but if the call was terminated before it hit line 6 the connection would stay open, also DOH! |
| 14:02.33 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:02.49 | upb | one way to solve this would be to write a mysql proxy in Visual Basic .NET and put it between your MySQL server and Asterisk |
| 14:02.54 | [TK]D-Fender | peep: SHOW us the call. |
| 14:03.49 | [TK]D-Fender | peep: You just want to add one record, right? System() out and call mysql directly. 1 line. |
| 14:10.11 | tzafrir | FlashDeluxe, what's the output of: dahdi_hardware |
| 14:11.54 | FlashDeluxe | tzafrir: pci:0000:08:00.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card |
| 14:12.00 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:12.03 | hrhrhr | russellb: those are both yank voices; digium sell uk ones? |
| 14:12.15 | tzafrir | FlashDeluxe, the driver is not loaded |
| 14:12.40 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:13.04 | FlashDeluxe | ok and why? |
| 14:14.13 | FlashDeluxe | tzafrir: do you mean zaphfc? |
| 14:14.20 | peep | [TK]D-Fender - I only want to write one line to the DB but the DB is on another server so I dont think system would work(?) Ill generate some calls now and post them to pastebin |
| 14:14.25 | tzafrir | FlashDeluxe, yes |
| 14:14.42 | tzafrir | dahdi_genconf modules; /etc/init.d/dahdi restart |
| 14:15.00 | upb | peep: ofcourse it would work but its an incredibly stupid solution |
| 14:15.02 | russellb | hrhrhr: no |
| 14:15.07 | [TK]D-Fender | peep: You can call whatever you want from CLI. Incluuding calls to apps taht will issue MySQL commands on other servers. Scripting is scripting |
| 14:15.12 | upb | the overhead of executing mysql client is pretty big |
| 14:15.46 | [TK]D-Fender | peep: What happens when you close off your DB BEFORE you dial? |
| 14:16.35 | FlashDeluxe | tzafrir: it seems that zaphfc doesn exist and that i have no timing (ztdummy?) device http://paste.debian.net/85162/ |
| 14:18.22 | *** join/#asterisk peep_ (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 14:18.27 | peep_ | [TK]D-Fender: in your opinion then is it a better practice to just call an external script rather than trying to use mysql() ? |
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| 14:18.43 | tzafrir | upb, so is the overhead of opening a connection to a different server |
| 14:19.46 | [TK]D-Fender | peep_: What is the result of my previous corrections? |
| 14:20.18 | upb | tzafrir: the mysql client would need to open it anyway versus being able to use a connect pool |
| 14:22.07 | *** join/#asterisk Polysics (~Luca@host118-66-dynamic.30-79-r.retail.telecomitalia.it) |
| 14:22.12 | Polysics | hello |
| 14:22.22 | Polysics | where can i find docs for the Bridge AMI command ,please? |
| 14:22.45 | tzafrir | FlashDeluxe, what version of dahdi-linux is it? |
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| 14:23.04 | FlashDeluxe | tzafrir: dahdi-linux-complete-2.3.0.1+2.3.0 |
| 14:25.11 | Polysics | bridge is in tehre but is not documented :-) |
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| 14:26.27 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 14:28.52 | tzafrir | Polysics, asterisk -rx 'manager show command Bridge' |
| 14:29.51 | peep_ | [TK]D-Fender: http://pastebin.ca/1923348 - with the changes you made it seems to work! The only problem is that sometimes clear is getting called without any resultid after it and I get an error like whats after the "####' in this pastebin |
| 14:30.04 | Polysics | tzafrir, many thanks |
| 14:30.18 | peep_ | [TK]D-Fender: Do I even need to do a clear for an insert statement since its not returning a result set? |
| 14:30.26 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:31.07 | Polysics | btw, does bridge do what i think it does? :-) |
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| 14:31.21 | [TK]D-Fender | peep_: that was fixing the "h" exten. That was a clear first step, but that cannel var is dead byt he time you hit it. What about the suggestion to clean it up BEFORE the dial? |
| 14:31.27 | Polysics | put two channel in communication with each other |
| 14:32.45 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:36.27 | Polysics | argh, ehm... it might be that Bridge actually does NOT do what i think it does :-) |
| 14:36.40 | Polysics | i am using that for a FollowMe implementation |
| 14:37.17 | Polysics | i was under the impression that user SIP/1001 could call an extension, that would Originate a call between 1002 and a n IVR context |
| 14:37.38 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 14:37.41 | Polysics | if 1002 presses 1 or whatever to accept, 1002 gets bridged to 1001 |
| 14:38.03 | Polysics | but i am getting channel not found for 1001 (the original caller) |
| 14:38.22 | Polysics | all of this is implemented in Adhearsion, but i suppose the principles are there anyway |
| 14:38.51 | Polysics | am i babbling or there is any logic in my idea? :-) |
| 14:38.58 | peep_ | [TK]D-Fender: I changed it to attempt to clear right before the dial but the resultid still seems to be null a good portion of the time - http://pastebin.ca/1923356 |
| 14:41.11 | [TK]D-Fender | peep_: exten => _X.,n,MYSQL(Clear ${resultid}) <-- shouldn't this be ${connid} ? |
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| 14:43.56 | ruyo | Anyone knows the DAHDI equivalent of "Bellcore/Telcordia Caller ID Scheme" of a Grandstream analog gateway? |
| 14:44.16 | *** join/#asterisk peep (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 14:44.22 | FlashDeluxe | hi, asterisk doesn`t appear in /etc/init.d/ can somebody help me? Its Version 1.6.2.10 |
| 14:44.31 | Polysics | FlashDeluxe, on debian? |
| 14:44.42 | Polysics | you need to copy the init script from contrib |
| 14:44.44 | peep | [TK]D-Fender: and that is what we call a little too much reliance on copy/paste XD |
| 14:44.54 | FlashDeluxe | Polysics: yes |
| 14:44.56 | Polysics | open it, change a few variables at the top, and chmod it |
| 14:45.01 | ruyo | FlashDeluxe, did you "make config"? |
| 14:45.46 | Polysics | [TK]D-Fender, sorry to address you directly, but i am stumped, isn't Brdige supposed to do what i think it does? :-) |
| 14:46.00 | FlashDeluxe | ruyo: no -.-* after make config it works, thanks =) |
| 14:46.16 | Polysics | ruyo, lol, i suck, i was doing it by hand |
| 14:47.00 | ruyo | It kind of says inside a big box after doing "make". :P |
| 14:47.21 | Polysics | then i kind of suck :-) |
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| 14:49.14 | Polysics | and still can't figure out what i am doing wrong ,if anything |
| 14:50.15 | [TK]D-Fender | Polysics: Dial + M(). You made this complicated for nothing |
| 14:50.21 | peep | [TK]D-Fender: http://pastebin.ca/1923368 - ok now that variable is being called correctly but i am still seeing these warnings |
| 14:50.25 | [TK]D-Fender | Polysics: Never needed originate or bridge in the first place |
| 14:50.46 | Polysics | [TK]D-Fender, i was trying to implement the whole thing in Ruby. still, i see your point |
| 14:50.50 | *** join/#asterisk Zebaril (diogoserra@bl7-79-9.dsl.telepac.pt) |
| 14:51.00 | Polysics | just for educational purposes, what does Bridge do then? |
| 14:51.29 | [TK]D-Fender | Polysics: What it says |
| 14:51.37 | Polysics | no it doesn't :-) |
| 14:51.48 | [TK]D-Fender | Polysics: Takes 2 channels. Bridges them together |
| 14:51.53 | Polysics | ok, i just needed to know that, i am at least using the correct function |
| 14:52.05 | Polysics | i will give myself 1 more hour then move to Dial+M() |
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| 14:53.03 | [TK]D-Fender | Polysics: No, so far you are using a shit-ton of unnecessary crap |
| 14:53.22 | Polysics | but one of them is the correct command :-) |
| 14:53.35 | [TK]D-Fender | Polysics: Congratulations one finding today's masochism outlet. |
| 14:53.40 | Rienzilla | 1 |
| 14:53.43 | peep | lol |
| 14:53.51 | [TK]D-Fender | on* |
| 14:55.54 | [TK]D-Fender | peep: For a test, add a 5 second wait between your INSERT and the clear. Just as a sanity check to see if there is a race condition, etc |
| 14:57.44 | *** join/#asterisk dulloa (Dave@201.215.245.213) |
| 14:57.59 | peep | [TK]D-Fender: Same error, is it possible that there just isnt anything to clean up in memory after an insert? |
| 14:58.02 | dulloa | hi ! |
| 14:58.38 | [TK]D-Fender | peep: Wouldn't make sense not to be able to clean up... Keep that CLI option open... |
| 14:59.43 | peep | [TK]D-Fender: Yeah I agree, just seems to be a lack of documentation on this particular command. In the meantime, I will remove the clear line and watch memory utilization to see if we start running off the rails |
| 15:00.50 | [TK]D-Fender | peep: Open eyes is a good thing... I'd google up some other code samples using this to see if there is any useful commentary |
| 15:01.20 | dulloa | Hi, I purchased a codec and let me know when I send the license |
| 15:01.26 | FlashDeluxe | mhhh i ve got another problem, dahdi show channels doenst show any channels :( (th pseudo channel is there) |
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| 15:02.49 | p3nguin | dulloa: How will we know when you have sent the license? |
| 15:04.07 | ruyo | FlashDeluxe, have you done "dahdi_cfg -vvv"? |
| 15:04.41 | FlashDeluxe | ruyo: yes it says http://paste.debian.net/85173/ |
| 15:04.47 | FlashDeluxe | looks ok for me |
| 15:05.34 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:05.37 | ruyo | What card is that? |
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| 15:06.04 | *** join/#asterisk sa000 (fressh@unaffiliated/sa000) |
| 15:06.09 | FlashDeluxe | a no-name-very-cheap hfc card ;) |
| 15:06.11 | *** part/#asterisk sa000 (fressh@unaffiliated/sa000) |
| 15:06.33 | *** join/#asterisk sa000 (fressh@unaffiliated/sa000) |
| 15:06.38 | FlashDeluxe | with a cologne chip on it |
| 15:07.00 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:07.23 | p3nguin | So it smells nice, too? |
| 15:07.23 | sa000 | hi! in terms of jitter, packet loss or delay do asterisk box has some functionality to minimize these problem effects ? |
| 15:07.59 | ruyo | I never used dahdi with BRI cards, but if that's a 1 port and dahdi represents both audio channels plus the D channel, looks ok. |
| 15:08.47 | peep | [TK]D-Fender: Thanks for your patience dude, I know exactly how not fun it can be dealing with noobs |
| 15:09.22 | FlashDeluxe | yes the dahdi_cfg looks good but asterisk doesnt seem to know the card or the driver i think.. |
| 15:09.22 | bbryant | ~book |
| 15:09.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:12.34 | FlashDeluxe | dahdi show status tells me that it knows the card :S |
| 15:13.28 | ChannelZ | dahdi_genconf makes only a fragment of a config file which you need to include in your chan_dahdi.conf |
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| 15:13.54 | ChannelZ | (I'm assuming that's what you meant above by dahdi_cfg?) |
| 15:13.59 | ruyo | FlashDeluxe, if you do "module reload chan_dahdi.so" what does it say? |
| 15:14.04 | [TK]D-Fender | peep: There is a difference between "noob" and "twit". Inexperience can be corrected :) keep it up... |
| 15:14.35 | *** part/#asterisk dulloa (Dave@201.215.245.213) |
| 15:14.44 | ruyo | FlashDeluxe, by the way, do you have chan_dahdi.conf made? |
| 15:16.38 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.95.159) |
| 15:17.03 | tzafrir | FlashDeluxe, what's the output of lsdahdi ? |
| 15:17.04 | FlashDeluxe | ahhhh, ChannelZ: You`re right, it wasn`t configured in a right way... ruyo: the config file was there, but as i said, i configured it in a wrong way |
| 15:17.49 | FlashDeluxe | seems to work now, thanks for help @ everybody :) |
| 15:18.03 | FILLVAIO3 | Hello guys. Is there possible to change CALLERID(all)=name<number> in queue for each member different? |
| 15:18.44 | Kobaz | anyone have any documentation on how to use/set connected number information |
| 15:18.56 | sa000 | anyone comments on my question ? |
| 15:22.48 | fauxalliance | sa000, asterisk has some... the rest is inherently TCP/IP. |
| 15:23.49 | ChannelZ | except RTP is UDP |
| 15:23.52 | p3nguin | Why would it be TCP when our signalling and media are on UDP? |
| 15:23.54 | [TK]D-Fender | FILLVAIO3: Where? When? |
| 15:24.40 | sa000 | fauxalliance: thanks. Can i get the exact doc to read about this? |
| 15:25.03 | fauxalliance | your s/tcp/udp/protocol stack |
| 15:25.10 | fauxalliance | ;-) |
| 15:25.26 | Polysics | one last thing before i go to the macro method |
| 15:25.33 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:25.39 | Polysics | is there a way to make a channel wait indefinetly while listening to some audio? |
| 15:25.42 | sa000 | fauxalliance: i mean related to asterisk |
| 15:25.42 | ChannelZ | sa000: Asterisk does have a jitter buffer, see sip.conf for adjusting it. Most end points do it themselves as well |
| 15:25.56 | fauxalliance | p3nguin, UDP falls under the /IP part ;-) |
| 15:26.13 | fauxalliance | TCP/IP, learn how it fits together, there is no escape. |
| 15:26.14 | ChannelZ | fauxalliance: there is no error correction or packet ordering in UDP |
| 15:26.27 | fauxalliance | thanks for clarifying that for sa000 |
| 15:26.28 | Naikrovek | ChannelZ: no, but there is in RTP I believe |
| 15:26.31 | ChannelZ | fauxalliance: packets arrive when they arrive, in whatever order, if they arrive at all |
| 15:26.46 | fauxalliance | indeed. that was my 'best effort' |
| 15:26.59 | Naikrovek | that's the cool feature about UDP |
| 15:27.07 | Naikrovek | as a programmer UDP is so much easier |
| 15:27.12 | ChannelZ | Naikrovek: yes but fauxalliance is saying that TCP/IP handles it which is not correct |
| 15:27.47 | Naikrovek | ChannelZ: ah. yes. UDP has no native error correction or retransmission facilities. if packet is dropped, it's gone. hope you didn't love it and cherish it |
| 15:27.54 | sa000 | ok |
| 15:28.19 | ChannelZ | UDP is more like standing on a street corner yelling, rather than having a one-on-one conversation |
| 15:28.27 | Naikrovek | udp multicast is like that |
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| 15:28.40 | fauxalliance | don't put words in my mouth ChannelZ... what is not managed by asterisk, is INHERENTLY handled at by the protocol stack... ie... the TCP/IP suite. what ever is in there is in the RFC's... start there sa000 |
| 15:28.51 | ChannelZ | in terms of the person you're talking to maybe hears you sometimes, maybe doesn't |
| 15:28.57 | Naikrovek | yeah |
| 15:29.10 | fauxalliance | 'best effort' != guaranteed delivery |
| 15:29.16 | Naikrovek | it's like talking to your wife while she's trying to watch TV and surf the web at the same time |
| 15:29.21 | Naikrovek | maybe she'll hear you, maybe she won't |
| 15:29.38 | fauxalliance | send the divorce paperwork registered mail Naikrovek ;-) |
| 15:29.41 | sa000 | http://www.youtube.com/watch?v=hmaQXwWWO9o .. In this video this guy is calculating jitter and also other traffic but whats the point than with comparision in terms of jitter,delay or packet loss. Let say asterisk with free switch. I mean its all up to network ?? |
| 15:29.44 | Naikrovek | lol |
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| 15:30.18 | FILLVAIO3 | <[TK]D-Fender> For example: i have a queue with two members, first is SIP/101 with prority 1, second is IAX2/79261234567. For each of them i need to set specific CALLERID(all) before queue will call to them. Is there possible? |
| 15:30.29 | sa000 | i mean there is no role of pbx than with respect to jitter,delay , packetloss |
| 15:30.46 | fauxalliance | sa000, barely,,, at the codec level. |
| 15:30.55 | fauxalliance | you can play with some 'windows' |
| 15:31.56 | bradleyd | I am trying to get the SIPCALLID from a BRIDGEPEER in the dialplan. I cant find any builtin functions to do this |
| 15:32.09 | sa000 | <fauxalliance> you can play with some 'windows': which windows ? |
| 15:32.17 | fauxalliance | tx and rx ;-) |
| 15:32.39 | fauxalliance | is all out of context, therefore, coffee. |
| 15:33.01 | ChannelZ | Asterisk de-jitters receiving media, if it's in the media stream |
| 15:33.42 | ChannelZ | As set in sip.conf or iax.conf if you're using one or the other |
| 15:33.45 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 15:34.33 | p3nguin | bradleyd: How about SIP_HEADER? If the info you are looking for is in the header, this function should be able to extract it. |
| 15:34.37 | fauxalliance | chapter 15, section four of 'the book' sa000 |
| 15:35.40 | *** join/#asterisk BANSAL (~bansal@117.207.83.70) |
| 15:35.55 | Polysics | is upgrading from 1.6.11 to 1.6.2 configure, make and make install for the most part? |
| 15:36.23 | FILLVAIO3 | [TK]D-Fender: For example, i have a queue with two members, first is SIP/101 with prority 1, second is IAX2/79261234567. For each of them i need to set specific CALLERID(all) before queue will call to them. Is there possible? |
| 15:37.23 | fauxalliance | FILLVAIO3, saw that a moment ago... please clear your clipboard. |
| 15:37.25 | [TK]D-Fender | FILLVAIO3: Not with those CHANNEL TYPES. Only 1 channel type lets you "do stuff" selectively. Guess which. |
| 15:37.29 | *** join/#asterisk thansen (~thansen@125.sub-97-36-164.myvzw.com) |
| 15:37.39 | [TK]D-Fender | Polysics: There is no 1.6.11 |
| 15:37.54 | Polysics | [TK]D-Fender, i mistyped, it is 1.6.1.11 |
| 15:37.55 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 15:38.34 | [TK]D-Fender | Polysics: Make sure that config differences don't pose issues and that 3rd party/addon modules are supported/replaced |
| 15:39.32 | sa000 | fauxalliance: The future of asterisk ? |
| 15:39.37 | Polysics | [TK]D-Fender, aside from third party stuff (which is only the official Asterisk addons), does that mean i can take a look at CHANGES, then have a go at it and eventually fix the config on the go? |
| 15:39.57 | fauxalliance | sa000, v2 |
| 15:39.58 | [TK]D-Fender | Polysics: Unless the changes crash you out immediately. |
| 15:39.58 | bradleyd | p3nguis: I tried that but it tells me that it is only for sip calls, but the call is a sip call. It is an outbound call |
| 15:40.29 | sa000 | fauxalliance: sorry but i didnt understandt this short cut v2? |
| 15:40.31 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 15:40.52 | fauxalliance | <PROTECTED> |
| 15:41.12 | fauxalliance | ~book |
| 15:41.13 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:41.28 | sa000 | yes got it |
| 15:42.19 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 15:45.51 | sa000 | fauxalliance: thanks. reading |
| 16:10.32 | sa000 | thanks all |
| 16:10.33 | sa000 | bye |
| 16:11.13 | Polysics | ok, time to make install over 1.6.1.11 and see what happens |
| 16:11.19 | Polysics | aka. "pull the pin and throw" |
| 16:11.35 | Chainsaw | Polysics: The world as we know it will end! |
| 16:12.32 | Polysics | Chainsaw, maybe not, but my server might :-) |
| 16:14.56 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.95.159) |
| 16:15.00 | FILLVAIO3 | Hi again |
| 16:16.36 | FILLVAIO3 | Dows anybody know how to change CDR cols values before its will be writed to mysql db? |
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| 16:23.10 | b14ck | FILLVAIO3, do you want to modify a particular value? Or just change which fields are written to the database all together? |
| 16:24.36 | Polysics | apparently the world survived, and so did my server |
| 16:25.05 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 16:25.36 | FILLVAIO3 | b14ck: i need to change [src] value before its writen into table |
| 16:25.55 | Qwell | changing src seems to be a bit silly.. why not set the userfield? |
| 16:26.29 | b14ck | FILLVAIO3, yah, not sure exactly. Not sure why you'd want to do that either =) |
| 16:27.23 | FILLVAIO3 | Ok. thanx |
| 16:27.25 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:28.53 | FILLVAIO3 | But what about more then one collumn to write my own vars? is that possible? |
| 16:29.16 | b14ck | FILLVAIO3, yah, you can actually make your own CDR rules (what columns you want, etc.) |
| 16:29.25 | b14ck | You can configure it in the cdr_custom.conf file |
| 16:29.56 | b14ck | You can find some examples online: http://www.asteriskguru.com/tutorials/cdr_custom_conf.html |
| 16:30.22 | b14ck | So you can add custom fields there, then edit your MySQL DB to add those fields, and write whatever you want to them. |
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| 16:32.09 | FILLVAIO3 | Ok. thanx, i will try this now!!! |
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| 16:42.27 | FILLVAIO3 | b14ck: i have configure cdr_mysql.conf, and there [columns] with [alias =>]. Can i use this config for my purposes? |
| 16:45.20 | FILLVAIO3 | working! |
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| 16:46.41 | b14ck | =) |
| 16:47.03 | FILLVAIO3 | thanx! |
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| 16:52.50 | FILLVAIO3 | but how to make any action after queue ends? |
| 16:54.58 | [TK]D-Fender | like? |
| 16:55.51 | FILLVAIO3 | any, for example run macro |
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| 17:03.29 | [TK]D-Fender | FILLVAIO3: "h" <- Asterisk Standard Extension |
| 17:07.08 | FILLVAIO3 | [TK]D-Fender: thanx again! Is possible to set h extension for all contexts globaly? |
| 17:08.55 | [TK]D-Fender | FILLVAIO3: Extensions are in contexts they do what you put in them. |
| 17:09.03 | [TK]D-Fender | FILLVAIO3: There is no "global". |
| 17:09.18 | [TK]D-Fender | FILLVAIO3: Then again, maybe you should think about INCLUDE's |
| 17:10.13 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 17:10.50 | b14ck | my allergies are killing me today |
| 17:10.51 | b14ck | lame |
| 17:11.56 | FILLVAIO3 | [TK]D-Fender: i know about includes, and try it early. But i don't understand why i can't use in different extensions includes more than one time |
| 17:12.22 | v1s | if I have Ringing -> Wait(1) - Congestion in my context what should happen if some one callls? and if I replace it with Hangup what should happen or be different? |
| 17:13.32 | v1s | FILLVAIO3: why do u want to use includes more then one time in one extension? |
| 17:13.43 | FILLVAIO3 | [TK]D-Fender: Sorry, need to go. thanx for help! Good day for ya. |
| 17:14.20 | [TK]D-Fender | v1s: Depends and what your call is coming in on. |
| 17:14.59 | *** join/#asterisk voip_troll (~les@96.51.239.24) |
| 17:15.07 | v1s | [TK]D-Fender: like if I am using ipkall number |
| 17:15.40 | voip_troll | Is there a way to load an audio file that is going to be played back multiple times (outbound calls) to eliminate the disk i/o overhead? |
| 17:16.03 | v1s | [TK]D-Fender: I am trying to do call back but I want the line to just give a busy signal with out having to answer. But both seem to just be ringing |
| 17:16.36 | [TK]D-Fender | v1s: Perhaps you should be SHOWING us. |
| 17:16.39 | [TK]D-Fender | ~pb |
| 17:16.39 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:16.45 | v1s | voip_troll: I read an article somewhere where u can setup a ram disk then use it to play the files to or from. |
| 17:17.06 | [TK]D-Fender | voip_troll: RAMFS <- |
| 17:18.10 | v1s | [TK]D-Fender: http://pastebin.com/V2tqas6y |
| 17:18.52 | [TK]D-Fender | v1s: where is the CALL to look at? |
| 17:19.10 | voip_troll | [TK]D-Fender: Thanks :) |
| 17:19.19 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
| 17:19.52 | v1s | voip_troll: maybe check out tmpfs also |
| 17:20.07 | v1s | http://en.wikipedia.org/wiki/Tmpfs |
| 17:20.27 | pabelanger | +1 for tmpfs |
| 17:20.41 | voip_troll | Yea :) |
| 17:24.24 | v1s | [TK]D-Fender: what u want the full debug or v+10 ? |
| 17:25.20 | voip_troll | The problem with tmpfs is it uses swap... which puts me back into problems with disk io |
| 17:25.36 | *** join/#asterisk hugorebelo (~hugo@187.35.137.180) |
| 17:25.52 | v1s | voip_troll: I think it only goes to swap if it runs out of mem |
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| 17:26.15 | v1s | Everything stored in tmpfs is temporary in the sense that no files will be created on the http://en.wikipedia.org/wiki/Hard_drive; however, swap space is used as backing store in case of low memory situations. On http://en.wikipedia.org/wiki/Reboot_%28computer%29, everything in tmpfs will be lost. |
| 17:27.18 | voip_troll | ah, didn't see that in the dox I'm looking at. Thanks :) |
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| 17:32.34 | ruyo | When setting an option in chan_dahdi.conf, it remains for all channels until told otherwise, right? |
| 17:33.09 | ruyo | Or does everyting reset when "channels => X" is used? |
| 17:33.22 | ruyo | *channel |
| 17:34.12 | [TK]D-Fender | ruyo: No reset |
| 17:35.47 | ruyo | It'd give me hope if you told otherwise. :P |
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| 17:40.46 | Naikrovek | ah, TF2. you are so fun |
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| 17:48.17 | *** join/#asterisk mnuzaihan (~tohyttym@bb116-14-139-133.singnet.com.sg) |
| 17:48.22 | mnuzaihan | hi. I have a question. |
| 17:48.29 | seanbright | ~ask |
| 17:48.29 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:48.37 | mnuzaihan | ok. ;) |
| 17:49.06 | mnuzaihan | is there a way to reroute to another trunk if the quality of the current trunk is sub-optimal? |
| 17:49.25 | mnuzaihan | something like LCR but more for quality instead |
| 17:50.43 | mnuzaihan | for example, if the latency is too high. |
| 17:51.22 | mnuzaihan | i'm using SIP for trunking to upstream providers. |
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| 17:53.44 | leifmadsen | mnuzaihan: sounds like RTCP stuff, which isn't well supported in Asterisk at the moment. There is a branch oej_ was/is working on that might be good for you to test. |
| 17:54.22 | mnuzaihan | oej_ is the name of the repository branch? |
| 17:54.32 | mnuzaihan | ok.. got it. |
| 17:54.35 | mnuzaihan | thanks. :D |
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| 17:55.02 | oej_ | It's "pinefrog", mnuzaihan |
| 17:55.02 | leifmadsen | mnuzaihan: maybe this one... http://svn.asterisk.org/svn/asterisk/team/oej/pinefrog-1.4/ -- I can't tell because oej_ has crazy naming that doesn't help you in determining what the branch actually does |
| 17:55.16 | oej_ | That's exactly the branch |
| 17:55.35 | oej_ | There are README files in most of the branches to explain, leifmadsen |
| 17:56.04 | leifmadsen | oej_: you have a lot of branches -- it still makes it hard to search through all the README files to find what I'm looking for or for recommending to people to test |
| 17:56.20 | leifmadsen | if there were 3 or 4 branches it would be less of a problem, but I see like 20-30 in there |
| 17:56.48 | leifmadsen | you need an index.html file in there or something |
| 17:56.56 | oej_ | Right. I've started to add some tags to the crazy names |
| 17:57.07 | leifmadsen | less crazy names would be helpful :) |
| 17:57.29 | oej_ | The crazy names will still be there, but pinefrog would be pinefrog-rtcp-1.4 |
| 17:57.57 | WIMPy | Has there ever been any activity towards supporting G4 fax with Asterisk? |
| 17:59.27 | leifmadsen | not to my knowledge |
| 18:00.23 | WIMPy | That's the impression google gave me. That's bad. :-( |
| 18:01.56 | leifmadsen | bad is a relative term |
| 18:03.11 | WIMPy | Well, that makes it hard to sell as a business communication system. |
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| 18:05.31 | leifmadsen | in some markets, perhaps |
| 18:06.04 | leifmadsen | I haven't had a problem selling it to my business customers |
| 18:06.12 | WIMPy | It didn;te bother me so far, as I only used * for IVR type of stuff and at home. |
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| 18:06.31 | leifmadsen | I've not once had someone ask me about G4 faxing in the 6+ years I've been doing asterisk |
| 18:06.38 | bradleyd | during the hangup, I can get the correct channel from BRIDGEPEER--${BRIDGEPEER} if I core show channel BRIDGEPEER I can see SIPCALLID but can figure out how to get it via dialplan |
| 18:07.06 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 18:07.13 | leifmadsen | bradleyd: use DumpChan() to find channel variables, otherwise you might want one of the SIP...() functions or CHANNEL() function |
| 18:07.16 | WIMPy | But I've been asked for a 20 branch company for a "normal" phone system. And they are still using fax. |
| 18:07.47 | leifmadsen | why can't you just keep the fax separate on their own lines? Why complicate things? |
| 18:08.09 | leifmadsen | or route the calls from Asterisk through to the fax machines directly |
| 18:08.20 | bradleyd | I will try DumpChan(), since I tried SIPCHANINFO and SIP_HEADER(Call-ID) to no avail |
| 18:08.54 | leifmadsen | bradleyd: it is highly likely you'll find what you need in a dialplan function rather than a channel variable |
| 18:08.55 | WIMPy | Because a fax to e-mail type thing would be a very good argument, saving the cost of the fax machines. |
| 18:09.02 | leifmadsen | ~hylafax |
| 18:09.03 | infobot | A telecommunication system for UNIX systems. URL: http://www.hylafax.org |
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| 18:09.55 | WIMPy | Yes, but with iaxmodem that will only do G3 fax, if I get it correctely. |
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| 18:10.28 | leifmadsen | shrugs |
| 18:10.46 | bradleyd | leifmadsen: thats what I thought, but the disconnect seems to be that a customer calls in--we park them..then we spool a call that hits a outbound context to an agent. I need the sip call id from that leg |
| 18:10.54 | bradleyd | thanks for the reply |
| 18:12.12 | WIMPy | I found capi4hylafax, but that requires the AVM capi to do G3 fax and I'm not sure it it could be connected to cahn_capi. |
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| 18:21.05 | bradleyd | No application 'DumpChan' for extension (call_agent, h, 4) |
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| 18:23.12 | KavanS | ok - running into a number that I dial that is busy routinely - suggestions for a macro that detects for busy, and redials automatically for me? - maybe a pause of 2-3 secs |
| 18:23.16 | KavanS | someone else doing this I assume? |
| 18:24.12 | t_dot_zilla | i'm repeatedly getting this error: queue frame: Exceptionally long voice queue length queuing to .... and it's causing CPU to skyrocket |
| 18:24.21 | t_dot_zilla | does anyone know what is causing it? |
| 18:24.45 | KavanS | n/m - cmd retrydial |
| 18:24.47 | KavanS | reading now... |
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| 18:29.19 | *** part/#asterisk voip_troll (~les@96.51.239.24) |
| 18:30.02 | p3nguin | bradleyd: If the 'h' extension has been executed, there is no more channel for DumpChan() to dump. |
| 18:30.38 | Micc | KavanS, I think there is also a new callback feature in 1.8 when a line is busy it retries and calls you back when it goes through. |
| 18:30.54 | KavanS | Micc, doh :( running 1.4 here |
| 18:31.08 | KavanS | reading some more online - looks like there's something that can be done with AGI scripting and a "call" file |
| 18:31.18 | Micc | KavanS, I think there are some similar scripts or dialplan examples on voip-info.org that may do the same thing. |
| 18:32.13 | Micc | KavanS, also I wouldn't recommend using 1.8 in production for a while.It'll take a little time to work out the kinks I'm sure. |
| 18:32.14 | bradleyd | nod, just figured that out |
| 18:32.30 | KavanS | Micc, yeah was thinking that...1.4 has been uber reliable for us - don't want to hop into the mix yet |
| 18:33.31 | Micc | KavanS, 1.6 has been reliable for us in the last 8 or 9 months. Before that it was crashing all the time. |
| 18:34.11 | Micc | I can still cause a crash with fax for asterisk on occasions though. |
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| 18:38.36 | [TK]D-Fender | [14:31]<KavanS>reading some more online - looks like there's something that can be done with AGI scripting and a "call" file <- Raw dialplan can do this |
| 18:39.01 | KavanS | [TK]D-Fender, interesting - trying to find more info on voip-info.org - not seeing the example I'm looking for - any suggestions on keyword/search terms? |
| 18:39.19 | [TK]D-Fender | KavanS: Originate() |
| 18:42.20 | Micc | TKD-Fender, is Originate in 1.4? |
| 18:42.52 | p3nguin | The dialplan app is not, but that doesn't matter a whole lot. |
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| 18:45.31 | leifmadsen | Micc: nope |
| 18:45.49 | leifmadsen | Micc: like p3nguin said though , just the dialplan app -- you can still do it via Manager and such |
| 18:46.20 | Micc | leifmadsen, which means TKD-Fender just sent KavanS on a wild goose chase. |
| 18:46.22 | [TK]D-Fender | [14:42]<Micc>TKD-Fender, is Originate in 1.4? <- no, but you can also call asterisk directly for this |
| 18:46.24 | p3nguin | I was recently told to run it with the System() app. I must say it works perfectly. |
| 18:46.24 | leifmadsen | KavanS: why not make it simple and just check the ${DIALSTATUS} channel variable and if it is BUSY loop back and try something else |
| 18:46.47 | [TK]D-Fender | p3nguin: thats what I just alluded to |
| 18:46.53 | KavanS | leifmadsen, was thinking that - just enabled busydetect=yes on chan_dadhi, hoping to see if it will work well |
| 18:47.06 | leifmadsen | why do you need busydetect? |
| 18:47.24 | leifmadsen | sounds like you're using analog |
| 18:47.26 | KavanS | article I'm reading suggests to enable it - maybe I am barking up the wrong tree |
| 18:47.32 | KavanS | yep, the trunk dialing outwards is analog |
| 18:47.53 | leifmadsen | I don't know.... test and see what works |
| 18:49.46 | KavanS | ok, I will research a bit more - thanks for the direction on this guys, got a lot to read |
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| 19:11.06 | ariel_ | hello folks |
| 19:11.17 | ariel_ | hardwire: you around? |
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| 19:18.04 | t_dot_zilla | i'm repeatedly getting this error: queue frame: Exceptionally long voice queue length queuing to .... and it's causing CPU to skyrocket |
| 19:18.13 | t_dot_zilla | does anyone know what is causing it? |
| 19:19.50 | Diffen2 | Evning, I have started to use SNOM 320 phones and now it seems like the audio in the calls can be online for more 20 sec. then the audio disaperas. |
| 19:19.55 | Diffen2 | Maximum retries exceeded on transmission 2d3afd11-8d309abc@192.168.12.36 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. |
| 19:19.55 | Diffen2 | [Aug 23 21:17:20] WARNING[5099]: chan_sip.c:1998 retrans_pkt: Hanging up call 2d3afd11-8d309abc@192.168.12.36 - no reply to our critical packet (see doc/sip-retransmit.txt). |
| 19:21.44 | frek818_ | Diffen2, Are these phone on the same local network as the Asterisk box? |
| 19:21.57 | Diffen2 | frek818 no |
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| 19:30.28 | b14ck | pokes frek818_ |
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| 19:54.16 | mnuzaihan | hi, i have another question on RTCP. Between me and the upstream would be no issues, but if the upstream has another trunk upstream, will RTCP on my end detect the degradation of quality? |
| 19:55.16 | mnuzaihan | or maybe this question is best answered in the mailing list. :/ |
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| 20:07.06 | citywok | So my Asterisk install crashed after a reload, it appears as though the SIP module may have locked up. http://pastebin.com/LBddkL0i |
| 20:07.22 | citywok | 1.6.1.11 (i know, i plan to upgrade to 1.6.2 soon) |
| 20:07.41 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 20:11.47 | Chainsaw | citywok: That's sad, yes. Without upgrading to the newest available 1.6.2.... not much is going to change. |
| 20:15.10 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:15.26 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:17.55 | pabelanger | citywok: Or test with 1.6.2 and see if it is fixed, if not report a bug, get fix then backport to 1.6.1.11 |
| 20:18.37 | citywok | It doesn't happen every reload (they are scheduled once an hour), but it seems to happen once every week or so. Though typically it happens at the midnight reload and not in the middle of the day. |
| 20:23.03 | citywok | I'll do another test upgrade and see if everything i have is compatible with 1.6.2 and see if i can upgrade this week. |
| 20:23.50 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:25.41 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:25.59 | fofware | hello all |
| 20:27.28 | fofware | one question, in asterisk 1.4 [mydevice] regexten=2000 need something more to work? |
| 20:29.17 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 20:31.02 | pabelanger | fofware: What are you trying to do? |
| 20:31.42 | fofware | I trying to define a device like example in sip.conf for xlite |
| 20:32.08 | fofware | and regextension with some numberr |
| 20:32.43 | fofware | but I if define with this way extension number not found |
| 20:33.09 | [TK]D-Fender | fofware: Feel free to show us |
| 20:33.14 | fofware | and if i defeine [9999] , blblblblblb |
| 20:33.14 | pabelanger | fofware: Then why not use the existing example with sip.conf for xlite? |
| 20:33.43 | fofware | the extension work |
| 20:34.12 | fofware | I'm ussing the example but it don't work im my case |
| 20:34.52 | [TK]D-Fender | BBIAb |
| 20:42.06 | *** join/#asterisk Diffen2 (~diffen2@c-2875e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 20:44.53 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:44.58 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 20:45.13 | fofware | [TK]D-Fender: http://www.pastebin.ca/1923610 |
| 20:57.18 | *** join/#asterisk DeVilSoulBlacK (~aandaluz@190.151.25.203) |
| 20:57.34 | Diffen2 | Hello, i have a problem that i have a bad feeling about. When i dial out from a device the ACK that should be sent to the server are sent to a completley different address... |
| 20:59.05 | DeVilSoulBlacK | hi any one have url to shared how to enable skype under asterisk 1.6.0.10 |
| 20:59.06 | Diffen2 | Seems like the invite, session in progress and the ok to the device are correct. but the ack are sent way off |
| 20:59.11 | DeVilSoulBlacK | no from zero |
| 21:01.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:01.27 | fofware | <PROTECTED> |
| 21:01.46 | pabelanger | ~skypeforasterisk |
| 21:01.47 | infobot | it has been said that skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 21:01.49 | Qwell | DeVilSoulBlacK: http://downloads.digium.com/pub/telephony/skypeforasterisk/README |
| 21:01.56 | Kobaz | aaaxxxeterisk |
| 21:01.58 | [TK]D-Fender | fofware: SHOW us the error. |
| 21:02.14 | [TK]D-Fender | fofware: And where exactly are you expecting it to CREATE this extension and what are you expecting it to do? |
| 21:03.12 | fofware | [TK]D-Fender: when I call from other extension the error is extension not found |
| 21:03.34 | [TK]D-Fender | fofware: And what do you expect it to DO? |
| 21:04.49 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 21:05.18 | DeVilSoulBlacK | qwell i get this "chan_skype.c:869: confused by earlier errors, bailing out" |
| 21:05.48 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 21:05.53 | fofware | [TK]D-Fender: handle _request_invite: Call from device1 to extension 2002 rejected because extension not found |
| 21:05.53 | fofware | I expect the extension ring, like if I define the extension with number for example [2002] |
| 21:06.14 | [TK]D-Fender | fofware: [2002] is NOT an "extension" |
| 21:06.31 | [TK]D-Fender | fofware: You seem to have some fundamental misunderstandings about * and those parameters |
| 21:06.58 | [TK]D-Fender | fofware: "regexten" will in no way make an extension that will actually dial anything at all |
| 21:07.11 | DeVilSoulBlacK | i have asterisk 1.6.0.10 |
| 21:07.17 | fofware | right it is a device, sure many thiings i don't understand very well |
| 21:07.32 | Qwell | DeVilSoulBlacK: without seeing much more information, nobody will be able to help |
| 21:07.41 | [TK]D-Fender | fofware: Completely forget about "regexten". You do not need it. |
| 21:07.53 | fofware | [TK]D-Fender: ok, I did understan bad |
| 21:08.00 | DeVilSoulBlacK | skype for asterisk ist possible work under zaptel ? |
| 21:08.02 | Kobaz | DeVilSoulBlacK: i have an 87 oldsmobile... can you fix it? |
| 21:08.06 | DeVilSoulBlacK | or i need update to dahdi |
| 21:08.23 | Qwell | DeVilSoulBlacK: Asterisk 1.6 doesn't work with Zaptel... |
| 21:08.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:10.14 | citywok | there was a comic last week about what it meant if you capitalized random letters in your name, or used l33t speak. DeVilSoulBlacK i suggest you go find it. |
| 21:10.42 | fofware | [TK]D-Fender: thanks a lot, I don't understand very well yet about extensions, devices, etc |
| 21:10.53 | [TK]D-Fender | fofware: Time to go make your dialplan. Also to realize you did not set the CONTEXT for your devices. this is where their calls should land so you can process them |
| 21:11.10 | [TK]D-Fender | fofware: the dialplan (extensions.conf) is 95% of Asterisk. |
| 21:11.35 | [TK]D-Fender | fofware: the few lines it takes for a SIP phone to be "usable" is nothing at all really. the real work is PROCESSING your calls |
| 21:11.54 | *** join/#asterisk io_error (debian-tor@gateway/tor-sasl/ioerror/x-29167786) |
| 21:12.31 | fofware | [TK]D-Fender: ok, width dial plan I have less problems, but now I'm cheking all parameters for sip.conf |
| 21:12.44 | beardy | fofware: Setting the context where calls to your devices will land is _very_ important. |
| 21:13.13 | io_error | I have a question I cannot find the answer to in the docs. When someone calls in and is directed to voicemail, I want only the user's greeting played; I want to get rid of the automated message that says "Please leave your message after the tone" etc. How can I do that? |
| 21:13.25 | io_error | And, i only want to do this for some extensions |
| 21:13.32 | [TK]D-Fender | io_error: "core show application voicemail" <- |
| 21:14.30 | io_error | Ah, there it is. What a strange place for it to be. :) Thanks! |
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| 21:33.38 | fofware | beardy: yes, I'm looking for more simple way to configure asterisk in embed devices and use less settings are possible to save space and processor time |
| 21:34.32 | *** join/#asterisk m0t3jl (~petr.mote@193.85.113.247) |
| 21:34.59 | pabelanger | fofware: look into templates for sip.conf, will help cut down on redundant settings |
| 21:35.58 | fofware | pabelanger: yes I looked it |
| 21:36.58 | [TK]D-Fender | Not worth it |
| 21:37.19 | [TK]D-Fender | couple of bytes of a sip config file should not screw you |
| 21:37.30 | [TK]D-Fender | If you're that tight, just shoot yourself now and be done with it |
| 21:43.45 | beardy | That's a bit drastic.. |
| 21:44.59 | fofware | [TK]D-Fender: I have working Asterisk over devices with only 4Mb of flash and processor of 180Mhz, and now I lost sounds and VM because os is biger than last year but work fine for home use, now is needed devices with 8Mb of flash to get a good PBX for home users |
| 21:45.18 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:49.49 | fofware | [TK]D-Fender: I know you don't like asterisk running on embed device, but it could be good, every body have a DSL router, so if asterisk run on many of them, will be good for every body, special for peoples that live in countryes where comunicatios are too spencive |
| 21:50.18 | fofware | mc |
| 21:51.30 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 21:52.54 | [TK]D-Fender | fofware: I didn't say embeded was bad |
| 21:53.18 | [TK]D-Fender | fofware: I said if you think you need to use SIP TEMPLATES then you are penny pinching to a pathetic degree |
| 21:54.14 | fofware | [TK]D-Fender: :o) |
| 21:56.18 | DeVilSoulBlacK | is away: auto-away |
| 21:57.00 | DeVilSoulBlacK | is back (gone 00:00:42) |
| 21:58.13 | Qwell | DeVilSoulBlacK: Turn that off. |
| 22:00.57 | *** part/#asterisk charlesgmoore (416b3b43@gateway/web/freenode/ip.65.107.59.67) |
| 22:03.46 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 22:11.11 | Trixboxer | Hi, Is there a away where I can avoid the call cut due to extension becoming unreachable at the time of call transfer ? like the call will bounce back to the person who had transferred it |
| 22:11.39 | Qwell | it's unreachable.. what would you like it to do? |
| 22:12.12 | Trixboxer | it will ring back to the person who had transferred it, like for eg |
| 22:12.54 | Trixboxer | call came on 1000 and the person at 1000 transfers it to 2000 but 2000 is not registered yet then the call will again ring at 1000 |
| 22:13.15 | Trixboxer | currently its hanging up |
| 22:13.23 | Trixboxer | I dont want to use voicemail |
| 22:13.27 | pabelanger | Trixboxer: its called an attended transfer |
| 22:14.07 | Trixboxer | pabelanger, hmm do I need to do something special for it ? |
| 22:16.23 | pabelanger | if you are using features.conf, look at atxfer |
| 22:17.17 | Trixboxer | Im using hard IP phones and the people are used to habit of pressing transfer button... so can asterisk take care of it automatically ? |
| 22:18.47 | pabelanger | Trixboxer: Not unless you some how remap the 'transfer' button to dial asterisk first. Otherwise, the transfer is handled outside of asterisk control |
| 22:19.19 | Trixboxer | ok, will check it out.. thanks mate |
| 22:19.22 | [TK]D-Fender | Trixboxer: Change their habits. I recommend elctro-shock |
| 22:19.45 | citywok | Yea, changing habits is hard. I have told like 30 people to dial the phone, THEN pick up the handset. that's a hard habit to break. |
| 22:20.05 | Qwell | what? why? |
| 22:20.32 | citywok | Some people take too long to dial while looking up the number, so the phone times out and dials whatever they have entered. (it's set to 5 seconds) |
| 22:20.40 | Trixboxer | [TK]D-Fender, hahaha eclctro-shock so their hairs go like * |
| 22:20.50 | Qwell | so fix your phone dialplan |
| 22:21.24 | citywok | Yea, I need to just set the timeout to 10 seconds I think. |
| 22:21.43 | citywok | People need to learn how to dial faster. not pressing a button for 5 seconds, what are you doing? lol. |
| 22:21.49 | [TK]D-Fender | Timeouts? Pardon? |
| 22:22.23 | citywok | The dial timeout on the Aastra phones. if you don't press a button for X seconds with the handset off-hook it assumes you are done dialing and sends the dialed digits out. |
| 22:22.29 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 22:22.34 | [TK]D-Fender | citywok: Excellent |
| 22:22.58 | citywok | Set to 5 I've had a 2 or 3 people complain that it's too fast, out of 150. lol. |
| 22:23.01 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 22:24.16 | citywok | What POE switches do people use/recommend? We're considering switching away from softphones and replacing the 100 agents with Aastra 31i's, but we'd have to replace our switches to support all the POE. I've seen people say the 48 ports only have enough power for 24 phones, and the 24 ports put out jsut as much power as the 48's, so i'm thinking a bunch of cisco 24 port POE's. |
| 22:25.00 | citywok | Apparently the Aastra's register w/ the POE switch @ 15 watts even though they dont use that much, and the 48 port cisco's i've seen only support 24 @ 15 or 48 @ 7.7. |
| 22:27.07 | Chainsaw | citywok: We use a Brocade FES4802-ILP. |
| 22:27.22 | Chainsaw | citywok: (You may see them as Foundry in second-hand channels) |
| 22:29.27 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 22:30.40 | citywok | Yup, I see that. big 2U sucker. I'm not sure i want to carry 2 of those to the Philippines in my suit case. lol. Does it power all 48 ports @ 15W? i don't see any mention of it. |
| 22:32.10 | *** join/#asterisk variable_office (~variable_@10.61.pool.border1.ch1.iswan.net) |
| 22:32.51 | variable_office | Asterisk is not sending a 180 message back to the upstream device when it is acting as a sip client. It just goes straight from 100 to 200 |
| 22:32.54 | Chainsaw | citywok: It has a 600W PSU, so based on my calculations it should come close. |
| 22:32.59 | variable_office | how can I get it to send the 180? |
| 22:34.54 | fofware | about DIAL... If I mix a SIP/devices with IAX2/externalPbx/, for example DIAL(SIP/6000&SIP/6001&IAX2/otherPBX/2020) sips devices are in local PBX ring one time and when DIAL to IAX2, my local pbx think the call was answered and still ringing on external, Exists any way to fix that, maybe ussing SIP/OtherPBX/2020 ? |
| 22:36.09 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 22:37.32 | fofware | DIAL to more than one PBX only work in this way? |
| 22:38.11 | [TK]D-Fender | fofware: What your other PBX does isn't *'s problem... make it not answer before actually answering on the farther end |
| 22:38.58 | Chainsaw | citywok: Update, 480W available for PoE; so 10 watts per port. It negotiates for 15.4, the next step down is 7.7; do you know the actual consumption per unit pelase? |
| 22:39.00 | Chainsaw | citywok: please? |
| 22:39.44 | citywok | Chainsaw: The actual consumption "depends on how many modules you have attached", so the phones register with the switch for a full 15 |
| 22:40.08 | citywok | i think they use something like 7 or 8 |
| 22:40.34 | Chainsaw | citywok: Okay. 10 watts per port should cover it. |
| 22:40.35 | fofware | [TK]D-Fender: so you think i have an error in dial plan of second PBX? with propper one must be work perfect? |
| 22:41.07 | citywok | Does the switch not mind if you register 48 ports @ 15, since they are only using 7 or 8? The cisco's won't allow the extra ones to negotiate the POE. |
| 22:42.03 | Chainsaw | citywok: I'd have to test that. Worth asking Brocade perhaps? |
| 22:42.43 | [TK]D-Fender | fofware: "local pbx think the call was answered" <- it WAS answered.. and the remote PBX is sitting in the way of its separate outbound call |
| 22:44.10 | citywok | Chainsaw: Yea, i've asked our vendor to ask aastra how many watts these things actually use on POE, I'll check about brocade/foundry for the wattage stuff. |
| 22:45.44 | Chainsaw | citywok: It's likely you'll end up with 2U kit if you want that power budget though. I just don't see a way to do all that in a 1U box without liquid cooling. |
| 22:46.29 | variable_office | anyone know about asterisk never sending out 180 sip ringing messages? |
| 22:47.18 | pabelanger | variable_office: network routing issue? |
| 22:47.33 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 22:47.33 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:47.46 | variable_office | pabelanger, No I tried putting a tcpdump right on the interface |
| 22:47.51 | variable_office | it skips it entirely |
| 22:47.56 | citywok | Chainsaw: yea, i'm tempted to use 24 port cisco's, that way if i lose a switch i don't get completely wiped out. |
| 22:47.57 | *** join/#asterisk EvilPsych (~EvilPsych@76.177.92.118) |
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| 22:48.14 | pabelanger | variable_office: pb at SIP debug from Asterisk |
| 22:48.21 | Chainsaw | citywok: I would look further then just Cisco though. Tends to be overpriced for what you get. |
| 22:49.06 | variable_office | pabelanger, pretty sure I checked sip debug, but let me double check |
| 22:49.43 | thehar | Anyone ever manageded a *&*$ing Panasonic D500 "hybrd' pbx? heh |
| 22:49.58 | *** join/#asterisk m0t3jl (~petr.mote@193.85.113.247) |
| 22:51.16 | variable_office | pabelanger, nope straight from 100 to 200 |
| 22:51.37 | citywok | Chainsaw: Yea, i'd go with older 10/100 24 port POE switches and use my existing RPS's to power them. We've got mostly 48 port 10/100 switches now running on the RPS's |
| 22:51.44 | pabelanger | variable_office: pastebin it, so we can all review it |
| 22:52.21 | Chainsaw | citywok: Okay :) |
| 22:53.20 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 22:53.40 | variable_office | pabelanger, http://pastebin.ca/1923755 |
| 22:54.42 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 22:57.07 | pabelanger | variable_office: what does your Dial syntax look like? |
| 22:58.10 | variable_office | pabelanger, standard ... Dial(SIP/) |
| 22:58.50 | variable_office | pabelanger, or standard ... Queue(xxx,t,,,600) |
| 23:00.01 | pabelanger | variable_office: indications.conf loaded? |
| 23:00.45 | variable_office | pabelanger, there is an indications.conf present. looking for a specific value? is there a way that i can check if it was implicitly loaded? |
| 23:01.02 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 23:01.51 | variable_office | pabelanger, I just upgraded from 1.6.2.8 to 1.6.2.11 and there was no change |
| 23:01.57 | pabelanger | variable_office: *CLI> indications show us |
| 23:02.04 | pabelanger | should tell if they are loaded. |
| 23:02.31 | pabelanger | variable_office: So, this used to work with .8 but stopped when you upgraded to .11? |
| 23:02.52 | variable_office | pabelanger, no I have no idea if it ever worked, but I just tried upgrading to .11 in the hopes that it would |
| 23:03.10 | pabelanger | ok |
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| 23:09.50 | variable_office | pabelanger, pabelanger btw indications was/is loaded |
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| 23:33.06 | variable_office | pabelanger, does your asterisk send a 180 ringing? |
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