| 00:00.53 | The_TiK | lmadsen, yeah that contains the extension i am calling from |
| 00:02.54 | The_TiK | i see DIALEDPEERNUMBER but it says it is broken |
| 00:03.08 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:06.14 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:08.31 | [TK]D-Fender | The_TiK: ${EXTEN} is the number you dialed (at least when the call is first accepted) |
| 00:09.03 | eXistenZ | pokes [TK]D-Fender |
| 00:10.40 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:12.17 | The_TiK | ok, ive got a call file and dials a phone number and then goes to an extension that plays a wav file |
| 00:12.42 | The_TiK | i just want to set monitor the call and set the filename to the number that was dialed |
| 00:13.06 | The_TiK | its all working except just setting the filename |
| 00:13.30 | [TK]D-Fender | The_TiK: then you might want to show us what you're doing. PASTEBIN is your friend. |
| 00:13.32 | [TK]D-Fender | ~pb |
| 00:13.33 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:13.34 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 00:15.44 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 00:16.00 | The_TiK | http://pastebin.com/d28b6a85 |
| 00:16.18 | *** join/#asterisk bluekelp (n=gabe@75-105-40-25.cust.wildblue.net) |
| 00:16.21 | ThatKidKel | Can anyone see the problem here? GotoIf($["${LAST_VERIFY}" > "2629743"]?s,12 : s,15) |
| 00:16.59 | [TK]D-Fender | The_TiK: And your call-file? |
| 00:17.46 | [TK]D-Fender | The_TiK: and that dialplan doesn't even DO anything |
| 00:18.25 | [TK]D-Fender | The_TiK: There is nothing to monitor. You aren't calling Dial after it. |
| 00:18.51 | The_TiK | yeah, the call file dials a number and then connects to it and plays the audio file that is created |
| 00:19.05 | *** part/#asterisk bluekelp (n=gabe@75-105-40-25.cust.wildblue.net) |
| 00:19.10 | The_TiK | http://pastebin.com/d723aa819 |
| 00:19.12 | The_TiK | thats the call file |
| 00:20.00 | The_TiK | so it uses iax2 to call a number and then connects it to that extension |
| 00:20.04 | [TK]D-Fender | The_TiK: You can't do a monitor after the callfile processes the channel you specified. |
| 00:20.29 | [TK]D-Fender | The_TiK: You need * to start monitoring BEFORE that call is placed out your itsp |
| 00:20.55 | *** part/#asterisk jarod14 (n=jarod14@ns1.viatelecom.com) |
| 00:20.55 | The_TiK | no, the call file is generated correctly, i just want to set the filename to the outgoing call |
| 00:21.05 | [TK]D-Fender | The_TiK: Which is something you can't do by placing a call to an IAX2 channel like that. |
| 00:21.23 | The_TiK | the wav monitor file is also created correctly, just the filename |
| 00:21.32 | [TK]D-Fender | The_TiK: I see what you're trying to do with the monitor but you have it on the wrong leg of the call. |
| 00:21.43 | The_TiK | it works this way though |
| 00:22.22 | [TK]D-Fender | The_TiK: exten => 661,5,Monitor(wav,${File1}) <- this will only record a call placed by the active channel. |
| 00:22.37 | [TK]D-Fender | the_but according to what you pasted, it never tries to dial out after. |
| 00:22.58 | The_TiK | it transfers the active channel to exten 611 |
| 00:23.06 | The_TiK | and then starts recording |
| 00:23.22 | The_TiK | thats how I understand it |
| 00:23.32 | [TK]D-Fender | TheYou understand incorrectly. |
| 00:24.00 | The_TiK | but its recording correctly, i just don't know what the outgoing number variable would be called to set the filename |
| 00:24.18 | [TK]D-Fender | The_TiK: you want to record what that outbound call through vitelity would hear/say. |
| 00:24.26 | The_TiK | yeah |
| 00:24.28 | [TK]D-Fender | the_it will not record anything. |
| 00:24.32 | The_TiK | it records fine |
| 00:28.08 | The_TiK | the recording is not what I am having a problem with, just setting the filename to number that is being called in the call file |
| 00:29.54 | [TK]D-Fender | The_TiK: Set another variable in your call-file with the number in it, and use that. |
| 00:30.14 | [TK]D-Fender | The_TiK: And I would like to see CLI output of one of these calls being processed from beginning to end. |
| 00:32.04 | The_TiK | ok, justa sec, i see what you are taking about now, about passing variable from the call file |
| 00:32.45 | [TK]D-Fender | The_TiK: That's one of several ways to do it. |
| 00:32.47 | *** part/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 00:33.49 | *** join/#asterisk routerweasel (n=4stfed@core.spokanecomputing.com) |
| 00:34.06 | *** part/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 00:40.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:43.55 | The_TiK | [TK]D-Fender, thanks for your help, this is what the call log looks like now |
| 00:43.55 | The_TiK | http://pastebin.com/df6d6607 |
| 00:44.41 | [TK]D-Fender | The_TiK: Go show me the recording now... |
| 00:45.13 | [TK]D-Fender | The_TiK: Because we can not see the Monitor actually START |
| 00:45.40 | The_TiK | it starts on line 9 |
| 00:45.57 | [TK]D-Fender | The_TiK: that doesn't actualy start recording... |
| 00:46.05 | [TK]D-Fender | The_TiK: Go show us the FILE. |
| 00:46.49 | *** join/#asterisk JohnnyBeGood (n=JohnnyBe@c-71-231-252-67.hsd1.wa.comcast.net) |
| 00:49.29 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6ad0579193f5cda9) |
| 00:50.10 | JohnnyBeGood | I need to open ports for asterisk, does 5060 needs to be TCP or UDP or both? |
| 00:50.40 | [TK]D-Fender | JohnnyBeGood: UDP, as well as 10000-20000 UDP |
| 00:51.48 | JohnnyBeGood | ok, tanks, some sites suggest TCP so I wasn't sure |
| 00:56.37 | JohnnyBeGood | what about sip_nat.conf if I have dyndns account that points to my home ip, do I need externhost=myaccountid.dyndns.com or externip=myaccountid.dyndns.com ? |
| 00:57.25 | [TK]D-Fender | jonGee, I dunno... that that an IP or a HOST you are providing? |
| 00:57.25 | JT | ~freepbx |
| 00:57.26 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 00:58.28 | JohnnyBeGood | its a host |
| 00:59.37 | [TK]D-Fender | SMRT |
| 01:03.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:03.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:06.49 | *** join/#asterisk BeeBuu (n=beebuu@61.142.247.242) |
| 01:08.31 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 01:20.53 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 01:20.54 | *** join/#asterisk Mad||Cow (n=madcowl@static-72-94-249-58.phlapa.fios.verizon.net) |
| 01:21.53 | *** join/#asterisk CVirus (n=Burzum@82.201.174.13) |
| 01:21.53 | Mad||Cow | Anyone ever seen an error like this: channel.c: Unable to find a codec translation path from ulaw to unknown |
| 01:22.49 | CVirus | I have a motherboard with 1 PCI slot .. so, I inserted a PCI card that can hold two PCI slots and connected two X100P cards to it ... and now only one of them is detected while the other is not ... any ideas ? |
| 01:23.21 | lanning | what does "lspci" show? |
| 01:23.39 | CVirus | the two cards |
| 01:23.45 | CVirus | but zttool can't |
| 01:23.48 | CVirus | it sees only 1 card |
| 01:24.59 | lanning | don't know. |
| 01:25.25 | CVirus | thanks |
| 01:28.16 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 01:28.16 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 01:30.01 | *** join/#asterisk fujin (n=aj@junglist.gen.nz) |
| 01:30.01 | *** join/#asterisk implicit_ (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 01:30.08 | fujin | so uh, I have a weird issue with an AGI scrip |
| 01:30.08 | fujin | t |
| 01:30.22 | fujin | occasionally they start to build up, forking asterisk, just now I had 4000 asterisk forks |
| 01:30.39 | fujin | my cluster failed over (out of memory), and I've disabled the AGI script |
| 01:30.42 | fujin | anyone seen an issue like that before? |
| 01:44.36 | *** join/#asterisk Dovid (n=Dovid@bzq-79-183-0-22.red.bezeqint.net) |
| 01:57.21 | *** join/#asterisk LoOoD (n=gman@64.201.247.2) |
| 02:05.24 | russellb | sharks sighted nearby, beware! |
| 02:06.30 | Dovid | russelb: R u that bored ? ;) |
| 02:08.25 | russellb | Dovid: :-p |
| 02:08.35 | russellb | very tired, so a little crazy |
| 02:11.48 | *** part/#asterisk fujin (n=aj@junglist.gen.nz) |
| 02:13.17 | Dovid | haha. join the club. |
| 02:13.39 | Dovid | got back form a wedding at 11:30 OM. its 5:10 AM and I am still working. the alarm still rings at 9:30 AM |
| 02:15.37 | *** join/#asterisk Mmurdock (n=vnjyjta@211.sub-97-50-230.myvzw.com) |
| 02:25.57 | *** join/#asterisk murdock_ut (n=murdock_@211.sub-97-50-230.myvzw.com) |
| 02:33.54 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 02:34.37 | bijit | is there any cheap like digium? |
| 02:34.53 | JT | what's the question? |
| 02:35.18 | bijit | i want to run asterisk @ home but can't afford a digium |
| 02:35.59 | bijit | so i was wondering if there is any digium like for me to use at home. |
| 02:36.13 | JT | what do you need to connect to at home? |
| 02:37.01 | bijit | i want to connect regular phone line to the computer so asterisk can handle my calls. |
| 02:37.28 | [TK]D-Fender | bijit: Cheaper alternative is the Linksys SPA-3102 |
| 02:39.35 | bijit | ty [TK]D-Fender |
| 02:48.04 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 02:49.48 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 02:53.20 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:53.48 | spokra | hey, does the snmp stuff work in 1.4.21.1? |
| 02:54.06 | russellb | it should |
| 02:54.13 | russellb | that's the idea, anyway |
| 02:54.26 | spokra | I get sub agent connected in the asterisk cli |
| 02:54.52 | spokra | but just end of mib when doing an snmpwalk |
| 02:55.05 | pcrane | does anyone know how to wait a random time in 1.2? |
| 02:55.23 | pcrane | in 1.4 you'd do: Wait(${RAND(5,30)}) |
| 02:55.30 | russellb | hm, tough one |
| 02:55.34 | spokra | voip-info doesn;t have anthing on configuring. found some other s via google but no luck |
| 02:55.39 | russellb | System would probably do it. |
| 02:55.44 | pcrane | about the only thing I can think of is something like: |
| 02:56.00 | pcrane | s,1,wait (1) |
| 02:56.00 | russellb | System(sleep `command to get random number but i am too tired to think`) |
| 02:57.04 | pcrane | s,2,Random(50:s,4) |
| 02:57.08 | pcrane | s,3,goto(s,1) |
| 02:57.15 | russellb | yeah, that would work ... |
| 02:57.16 | pcrane | s,4,Noop(done) |
| 02:57.20 | russellb | there isn't anything better built in |
| 02:57.28 | pcrane | ick |
| 02:57.40 | pcrane | I like the single command though |
| 02:57.48 | russellb | guess you should use 1.4 :) |
| 02:57.54 | pcrane | I've got no choice |
| 02:58.03 | pcrane | customer refuses to upgrade ;) |
| 02:58.19 | pcrane | the logic I've got was built for 1.4 |
| 02:58.25 | pcrane | having to adapt it to 1.2 |
| 02:58.35 | *** join/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 02:59.37 | *** part/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 03:00.29 | pcrane | thanks russellb |
| 03:00.50 | russellb | np |
| 03:00.53 | *** join/#asterisk JohnnyBeGood (n=micko113@c-71-231-252-67.hsd1.wa.comcast.net) |
| 03:02.21 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 03:15.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:20.40 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:22.38 | *** join/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 03:24.26 | *** part/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 03:33.29 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 03:34.38 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 03:38.46 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 03:44.17 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 03:46.17 | *** join/#asterisk Elijah` (n=elijah@70-101-193-201.dsl1.kgm.az.frontiernet.net) |
| 03:46.34 | Elijah` | Evening all... |
| 03:46.57 | pcrane | afternoon Elijah` ;) |
| 03:47.00 | TrentCreek | morning |
| 03:47.07 | Elijah` | having a little trouble getting phones to register with asterisk when outside my network :) |
| 03:47.09 | Elijah` | howdys :) |
| 03:47.38 | Elijah` | I think it's a NAT issue but not sure where to go from here... they register and work internally fine |
| 03:48.18 | TrentCreek | port forwarding |
| 03:48.24 | Elijah` | yep, got that... |
| 03:48.32 | Elijah` | forwarding 5060 UDP to the asterisk box.. |
| 03:48.33 | pcrane | sip or iax? |
| 03:48.36 | Elijah` | sip |
| 03:48.44 | pcrane | RTP ports? |
| 03:48.45 | *** join/#asterisk hi365_m (n=hi365@213.151.56.96) |
| 03:48.52 | Elijah` | and lsof -i shows that it is indeed listening... |
| 03:48.57 | Elijah` | bound to 0.0.0.0 |
| 03:48.59 | pcrane | forwarded? |
| 03:49.06 | TrentCreek | can you ping that port? |
| 03:49.17 | Elijah` | yes, 5060 is forwarded to the machine |
| 03:49.26 | Elijah` | umm... not sure let me try it |
| 03:51.40 | [TK]D-Fender | Elijah`: Need a lot more than that. Read up : |
| 03:51.41 | [TK]D-Fender | ~sipnat |
| 03:51.42 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:51.44 | [TK]D-Fender | ^^^^^^^^ |
| 03:52.56 | Elijah` | the host does repond do a ping, yes |
| 03:53.09 | Elijah` | so just forwarding 5060 is not enough to get a phone to register |
| 03:53.30 | Elijah` | let me check that link TK... |
| 03:54.09 | Elijah` | Aah, ok! |
| 03:54.32 | Elijah` | so lacking the bit of nat configuration there will keep a phone from registering? let me add that in quick.. |
| 03:56.19 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 03:57.39 | Elijah` | ok got it like that shows, let's try another register... |
| 03:59.35 | *** join/#asterisk stillbourne (n=stillbou@c-75-70-48-129.hsd1.co.comcast.net) |
| 04:01.00 | Elijah` | yeah no luck still, same problem.. |
| 04:01.20 | Elijah` | and running tcp dump on the port, you can see there's not a thing coming through |
| 04:02.43 | [TK]D-Fender | Elijah`: What have you got forwarded to you *? |
| 04:03.19 | [TK]D-Fender | Elijah`: And pastebin your sip.conf masking only passwords |
| 04:03.24 | [TK]D-Fender | ~pb |
| 04:03.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:03.26 | [TK]D-Fender | ^^^^^^^^^^^ |
| 04:03.38 | *** join/#asterisk hohum (n=dcorbe@apollo.wavelen.net) |
| 04:05.10 | Elijah` | Sorry was messing with it, 5060 and 10000-10005 are forwarded to the * |
| 04:05.37 | [TK]D-Fender | Elijah`: Enable SIP debug and pastebin the complete registration failure atttempt. |
| 04:06.11 | Elijah` | I have core set debug at 10, was not aware there was an sip debug feaure... |
| 04:06.15 | Elijah` | <--- newbie :P |
| 04:06.34 | TrentCreek | nooooooooooooobie! |
| 04:06.40 | [TK]D-Fender | Elijah`: "sip debug" <- |
| 04:07.22 | Elijah` | lol yeah that's me... but success! it finally registered, was the NAT :) |
| 04:07.30 | Elijah` | just took that time for the phone to retry |
| 04:08.47 | Elijah` | just made a call, awesome! |
| 04:09.11 | Elijah` | thanks guys... I really appreciate it: ) |
| 04:22.43 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 04:23.13 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 04:37.31 | *** join/#asterisk mmurdock (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 04:42.01 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 04:52.13 | *** join/#asterisk DragonBall-Z (n=Adnan@202.133.78.60) |
| 04:56.06 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-151-207.neo.res.rr.com) |
| 05:00.46 | *** join/#asterisk tompaw (n=tompaw@pav.vip.krakow.tompaw.net) |
| 05:01.15 | *** join/#asterisk Mawkee (n=root@200.152.186.45) |
| 05:01.38 | TrentCreek | OT: Anyone here gotten 802.1x supplicant working? |
| 05:02.19 | Mawkee | Anybody here with experience with Xorcom Astribank that can spare a couple of minutes on a doubt? |
| 05:03.18 | Strom_M | Mawkee: just ask your question... |
| 05:04.53 | *** join/#asterisk dbmoodb (n=ooos@unaffiliated/dbmoodb) |
| 05:05.20 | TrentCreek | he did ;-) |
| 05:07.02 | Mawkee | I tried with 3 xorcom astribanks, same problem. Brand new debian etch installation. All packages installed from updates.xorcom.com/rapid (via apt), step-by-step as the manual specifies. After everything is recognized, I create a simple dial plan to reach 3 phones connected to the bank. I dial from the second to the third. When I pick up the phone on the first phone and dial, the DTMF is played on both the second and the third line. |
| 05:07.22 | Mawkee | It's something that happens ALL the time. I tried with different installations, different astribanks |
| 05:07.30 | Mawkee | tried resseting the firmware |
| 05:08.01 | *** part/#asterisk dbmoodb (n=ooos@unaffiliated/dbmoodb) |
| 05:08.59 | Mawkee | Even more strange: If I dial from the second to the first, and pick up the third and dial, nothing is played on the first two channels |
| 05:10.22 | Mawkee | Anybody has at least an astribank that is plugged and working? Just for the reference, so I can say this is an isolated case? |
| 05:12.47 | Mawkee | I'll try something. Be back later if it doesn't work |
| 05:23.38 | Zochwar | interesting, answer() is only half-picking up the phone. The other line (analog) still rings, and the sound is fuzzy until i pick up on the other line. |
| 05:24.25 | *** join/#asterisk Mawkee (n=root@200.152.186.45) |
| 05:24.51 | Mawkee | Same old. Tried removing an extra Digium card that was on the mobo, but it didn't work. |
| 05:25.12 | Mawkee | I'm completely out of ideas. If somebody could give me a small idea to persuit, I would be really greateful |
| 05:28.06 | pcrane | Mawkee: We've had 3 xorcoms plugged in to debian lenny |
| 05:28.19 | Mawkee | pcrane: Never a problem similar to mine? |
| 05:28.20 | pcrane | nothing like that |
| 05:28.34 | pcrane | no |
| 05:29.02 | pcrane | did have a problem when the asterisk server stopped... it didn't boot up again after with the xorcoms still plugged in |
| 05:29.23 | pcrane | so, be aware of that |
| 05:29.36 | Mawkee | Hummmmmm |
| 05:29.39 | Mawkee | ok, thanks :) |
| 05:30.17 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 05:30.47 | *** join/#asterisk adorah (n=Michael@87.69.57.141.cable.012.net.il) |
| 05:31.02 | pcrane | when we had them, they plugged in and just worked |
| 05:31.07 | pcrane | do they work with the LiveCD? |
| 05:32.18 | Mawkee | Nope |
| 05:32.20 | Mawkee | Same error |
| 05:33.49 | pcrane | hm... that's odd |
| 05:33.57 | Mawkee | Really odd |
| 05:34.03 | Mawkee | I ran completely out of ideas |
| 05:34.21 | Mawkee | I unplugged everything, tried different usb jacks |
| 05:35.40 | TrentCreek | sip devices right? |
| 05:35.45 | Mawkee | nope |
| 05:35.49 | Mawkee | its a zap dev |
| 05:35.59 | TrentCreek | yuck |
| 05:36.13 | TrentCreek | Seems people ALWAYS having trouble |
| 05:36.42 | TrentCreek | that is why I use SIP and have fun |
| 05:37.21 | Mawkee | what sip device do you use? |
| 05:37.30 | Mawkee | for analog phones? |
| 05:37.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:38.02 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 05:40.59 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 05:43.14 | *** join/#asterisk Prof_Pardal (n=rsardinh@189.5.232.26) |
| 05:44.06 | Prof_Pardal | hi guys |
| 05:44.22 | Prof_Pardal | anybody have a astribank in production? |
| 05:45.26 | TrentCreek | I use Linksys with 2 lines |
| 05:45.35 | TrentCreek | It's based on the Sipura |
| 05:45.50 | TrentCreek | works like a charm, and never had a problem |
| 05:46.22 | TrentCreek | Well well..ANOTHER person with Zap having troubles |
| 05:46.26 | frogonwheels | TrentCreek: yeah my pap2t works pretty well |
| 05:46.35 | Prof_Pardal | pap2t works a lot |
| 05:46.45 | frogonwheels | TrentCreek: had a few configuration issues - but once set up it's rather nice. |
| 05:47.01 | TrentCreek | It's just the way to go |
| 05:47.18 | frogonwheels | you have to stand it up though, or it gets a bit warm |
| 05:47.40 | TrentCreek | I have not seen to many people that come here with sip problems, and it they do..it'susually solvable quickly |
| 05:47.48 | Prof_Pardal | my astribank is driving me crazy |
| 05:47.49 | TrentCreek | How warm? |
| 05:48.28 | Prof_Pardal | when 3 channels are open, and one of then press a dtmf |
| 05:48.35 | Prof_Pardal | everybody hears |
| 05:48.44 | *** join/#asterisk Mawkee (n=dan@189.5.232.26) |
| 05:49.06 | Prof_Pardal | have u seen it before? |
| 05:49.24 | Prof_Pardal | Mawkee, heya :) |
| 05:50.21 | Mawkee | Heya :) |
| 05:51.23 | *** join/#asterisk bmg505 (n=leon@196-209-78-27-tbnb-esr-2.dynamic.isadsl.co.za) |
| 05:53.59 | TrentCreek | seems you and Mawkee have a lot to discuss...almost same problems |
| 05:54.12 | Mawkee | He works with me |
| 05:54.23 | pcrane | that'd be the reason why ;) |
| 05:55.00 | Mawkee | hehehe |
| 05:55.01 | Mawkee | :-) |
| 05:55.05 | Mawkee | I'm calling Xorcom |
| 05:55.11 | Mawkee | maybe it's daytime there :-P |
| 05:55.42 | hi365_m | Mawkee: 5 minutes to 9 |
| 05:56.29 | Mawkee | hi365_m: Thanks! |
| 05:57.40 | Mawkee | Brb |
| 06:01.40 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 06:04.37 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 06:09.20 | Zochwar | Ok, so my problem was solved with a Wait(1) before Answer(), i only discovered that because the demo did it "Just for fun" |
| 06:10.56 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-151-207.neo.res.rr.com) |
| 06:15.51 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-75-1.singnet.com.sg) |
| 06:16.44 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:17.33 | Zochwar | Hm, anyone know if NT1 boxes (on ISDN lines) are interchangable, or do they need to be configured? |
| 06:17.50 | JT | they're generally just dumb modems |
| 06:18.04 | JT | might be a couple of dip switches or jumpers for termination, but that's about it |
| 06:19.42 | Zochwar | Ok, maybe i'll try swapping with another one, just to see if it helps |
| 06:24.04 | Kyoshi | when trying to use asterisk realtime, i want to use MySQL however, for some reason it's giving me an error saying the engine is not avialable, but if thats the case, mysqlclient is not working or installed properly. this is what confuses me cause i dont know the yum command to install it. anyone? |
| 06:31.40 | [netman] | Kyoshi: maybe the engine is innodb and ur installation only uses MyISAM? |
| 06:31.52 | Kyoshi | aroo? |
| 06:32.04 | Kyoshi | no dude im not install mysql server on this machine. its on a remote machine |
| 06:32.15 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 06:32.20 | Kyoshi | i only want to connect to it from my asterisk machine to use asterisk realtime |
| 06:33.05 | [netman] | check out whether the innodb is installed |
| 06:33.25 | [netman] | or maybe u could use another engine |
| 06:33.26 | Kyoshi | the CLIENT cares not which db storage method is used on the server. |
| 06:33.50 | Kyoshi | i dont see how the client would know the difference |
| 06:33.58 | [netman] | in fact, I don't know if really asterisk realtime uses innodb |
| 06:34.13 | Kyoshi | asterisk realtime doesnt care! |
| 06:34.14 | [netman] | Kyoshi: *there* is a difference |
| 06:34.27 | [netman] | e.g: transactions |
| 06:34.29 | Kyoshi | yes on the server side how it stores the data, yes you are right |
| 06:34.53 | Kyoshi | but accessing it from a client makes no difference just to do a reference |
| 06:35.05 | [netman] | no Kyoshi u r wrong |
| 06:35.14 | Kyoshi | i think the point you totally missed is the MySQL Client..... |
| 06:35.24 | Kyoshi | so you my friend are way off |
| 06:35.30 | [netman] | if the mysql client says "engine innodb" |
| 06:35.38 | Kyoshi | i asked for the YUM command to install the MySQL Client |
| 06:35.39 | [netman] | the engine must exists on the server |
| 06:35.40 | Kyoshi | thats all |
| 06:35.51 | Kyoshi | lets not go off trying to find fault in something that means nothing |
| 06:35.51 | JT | it seems [netman] will give his answer |
| 06:35.55 | JT | to a question no one asked |
| 06:35.59 | JT | whether we like it or not |
| 06:36.01 | Kyoshi | apparently |
| 06:36.14 | [netman] | Kyoshi: type mysql |
| 06:36.20 | [netman] | on ur machine |
| 06:36.35 | [netman] | if u see a mysql> prompt, that's all |
| 06:36.37 | Kyoshi | you think i have mysql installed? |
| 06:36.38 | JT | this isn't aol, use proper words |
| 06:36.51 | Kyoshi | ask me if i have it installed, i will tell you |
| 06:36.53 | Kyoshi | NO |
| 06:37.16 | [netman] | mysql-5.0.45-7.el5 |
| 06:37.20 | [netman] | yum install mysql |
| 06:38.45 | Kyoshi | why? |
| 06:39.58 | Kyoshi | why would i want to install mysql on this machine? i already told you it's installed on a dedicated machine. there will be 2 asterisk boxes that remotely connect to the mysql database server, which is a different physical machine. does this concept confuse you? what part of this dont you understand? is someone paying you to be this way and really annoy people cause you're worth every penny! |
| 06:40.34 | [netman] | 08:35 < Kyoshi> i asked for the YUM command to install the MySQL Client |
| 06:40.46 | [netman] | sorry for repeating, mouse issue |
| 06:41.43 | Kyoshi | correct and thats all |
| 06:42.02 | Kyoshi | if you';re "trying" to say "yum install -y mysql" will install the CLIENT only, then please clarify |
| 06:42.19 | [netman] | yes, that's just I mean |
| 06:42.28 | Kyoshi | english not your first language? |
| 06:42.39 | [netman] | if u want the client side, u need to install that package |
| 06:42.48 | [netman] | sorry dude, I'm Spanish |
| 06:43.02 | Kyoshi | dice espanola |
| 06:43.44 | Kyoshi | dice con migo |
| 06:45.03 | Kyoshi | bah |
| 06:45.11 | Kyoshi | mysql.i386 already installed |
| 06:45.14 | Kyoshi | up to date |
| 06:45.24 | Kyoshi | not the answer i needed |
| 06:45.53 | [netman] | so |
| 06:46.05 | [netman] | 08:36 < [netman]> if u see a mysql> prompt, that's all |
| 06:46.07 | [netman] | bad mouse :( |
| 06:46.53 | [netman] | I think the useful answer were answered already |
| 06:47.58 | Kyoshi | WARNING[14088]: config.c:1331 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 06:48.23 | Kyoshi | asterisk addons doesnt see the mysql client when using the menuselect |
| 06:48.27 | Kyoshi | thats a problem |
| 06:48.51 | [netman] | Kyoshi: have u properly set up the config parameters? |
| 06:49.04 | [netman] | host, user, password, db, etc....? |
| 06:49.22 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 06:49.25 | DarKnesS_WolF | Kyoshi: did u install the devel package correctly ? |
| 06:49.27 | DarKnesS_WolF | Koshatul: distro ? |
| 06:49.29 | [netman] | maybe asterisk couldn't connect to the mysql server |
| 06:49.50 | [netman] | devel? what for? |
| 06:50.07 | DarKnesS_WolF | Koshatul: ur compiling asterisk ? or using a package ? |
| 06:50.34 | DarKnesS_WolF | [netman]: if he is compiling as i can see from make menuselect then he should have devel pacakges for mysqlclient |
| 06:50.37 | Kyoshi | wolf: which devel package? |
| 06:50.43 | Kyoshi | ahh |
| 06:50.45 | Kyoshi | checking |
| 06:50.57 | [netman] | DarKnesS_WolF: I don't think he is compiling anything |
| 06:51.14 | DarKnesS_WolF | [netman]: how can he access the menuselect then ? |
| 06:51.44 | Kyoshi | mysql-devel installed |
| 06:51.47 | Kyoshi | not the problem |
| 06:52.30 | [netman] | DarKnesS_WolF: yes, if he really needs access the menuselect, he would have to install devel packages |
| 06:53.19 | Kyoshi | netman, you jut agreed with him but made it seem like he is wrong, nice twist |
| 06:53.28 | DarKnesS_WolF | Kyoshi: do u have anything like mysqlclient-devel ? |
| 06:53.41 | Kyoshi | but if your english is going to be this bad please netman, try not to help |
| 06:53.43 | [netman] | Kyoshi: I think u r not trying to compile anything |
| 06:54.05 | [netman] | but if u really try to compile, u need the devel libraries |
| 06:54.11 | Kyoshi | wolf: mysqlclient does not exist on the yum list |
| 06:54.25 | [netman] | mysqlclient is from Debian I guess |
| 06:54.34 | [netman] | Kyoshi: r u English? |
| 06:54.40 | DarKnesS_WolF | mmmm Kyoshi really i don't use redhat |
| 06:54.41 | Kyoshi | japanese |
| 06:55.16 | [netman] | Kyoshi: Spanish <- English -> Japanese ... I think it have something to do |
| 06:55.42 | Kyoshi | i speak english perfectly, it's not the english that is a problem with me, your english is horrible |
| 06:55.50 | [netman] | maybe Kyoshi |
| 06:55.59 | Kyoshi | english is my first language |
| 06:56.04 | [netman] | and u r enough smart to be the first one who told me |
| 06:56.09 | Kyoshi | then japanese, then chinese, then italian, spanish and german |
| 06:56.47 | Kyoshi | the problem i feel is that i cannot understand what you are trying to say |
| 06:56.55 | DarKnesS_WolF | Kyoshi: think about ur problem :-0 |
| 06:57.14 | [netman] | Kyoshi: maybe I didn't understand u from the beginning |
| 06:57.44 | DarKnesS_WolF | Kyoshi: is this a fresh installation ? or an upgrade ? |
| 06:57.52 | Kyoshi | fresh |
| 06:58.20 | [netman] | Kyoshi: can u connect to ur remote mysql server from the mysql client? |
| 06:58.32 | Kyoshi | yes |
| 06:58.41 | Kyoshi | thats why i think its an asterisk addon problem |
| 06:58.45 | [netman] | connect to the database also |
| 06:58.57 | Kyoshi | hmm |
| 06:58.58 | [netman] | and make queries |
| 06:59.05 | Kyoshi | that worked already |
| 06:59.27 | [netman] | r u sure asterisk is well configurated? |
| 07:01.05 | [netman] | is entering in real away mode |
| 07:02.57 | DarKnesS_WolF | Kyoshi: when u did compile asterisk-addon did it generated cdr_mysql ? and res_mysql ? |
| 07:14.01 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 07:18.25 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 07:19.51 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 07:26.15 | *** join/#asterisk mvanbaak (n=michiel@a82-95-228-138.adsl.xs4all.nl) |
| 07:26.48 | mvanbaak | bkruse: hey man |
| 07:27.17 | mvanbaak | my firewall at home died, so I left in a hurry last time we spoke (read, got disconnected because of that) |
| 07:27.35 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:27.58 | mvanbaak | I noticed you made some progress |
| 07:28.00 | mvanbaak | very nice |
| 07:31.19 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 07:33.55 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:35.21 | Prof_Pardal | hi |
| 07:35.29 | Prof_Pardal | finally question solved |
| 07:35.46 | *** join/#asterisk mvanbaak (n=michiel@a82-95-228-138.adsl.xs4all.nl) |
| 07:36.05 | Prof_Pardal | just only upgrade zaptel to svn version and astribank runs fine |
| 07:38.23 | *** join/#asterisk mvanbaak (n=michiel@a82-95-228-138.adsl.xs4all.nl) |
| 07:38.41 | *** join/#asterisk railsmunky (n=nick@collaboration.capuk.org) |
| 07:41.48 | railsmunky | Argh can anyone help. I'm having major problems with our PRI. I can receive incomming calls on any channel but all my outgoing calls fail saying the channel is in use |
| 07:44.20 | railsmunky | anyone there? |
| 07:45.05 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:46.23 | Strom_M | railsmunky: yeah |
| 07:46.33 | Prof_Pardal | railsmunky, yes |
| 07:46.47 | Prof_Pardal | can u give remote access? |
| 07:47.09 | Strom_M | pastebin your zapara.conf, your zaptel.conf, the relevant section of extensions.conf, and the CLI output of an inbound and outbound call at verbose 10 |
| 07:47.14 | Strom_M | remote access...lolol |
| 07:48.37 | Prof_Pardal | sorry, i'm installing a asterisk box right now and have no time to understand your problem via IRC |
| 07:48.51 | Prof_Pardal | i prefer see direct the CLI |
| 07:50.14 | railsmunky | sorry here... |
| 07:50.20 | railsmunky | I'll get some configs up for you |
| 07:52.16 | Prof_Pardal | ok, let see if we have time to solve it |
| 07:52.41 | Prof_Pardal | make a call and paste your CLI results |
| 07:53.02 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 07:53.08 | railsmunky | http://pastebin.com/m7dc1eef |
| 07:53.14 | railsmunky | that's the config |
| 07:54.20 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:56.16 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:56.34 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:56.39 | Prof_Pardal | hmm, sangoma... i ve never use a sangoma board |
| 07:56.57 | Prof_Pardal | i using digium a lot |
| 07:58.24 | Strom_M | Prof_Pardal: if you're so busy with your own asterisk problems, then perhaps it's best not to offer to help |
| 07:58.36 | Strom_M | railsmunky: where are the rest of the pastebins I requested? :) |
| 07:58.50 | railsmunky | Sorry looking for longs on the CHANUNAVIL :) |
| 07:59.05 | railsmunky | not all the channels are locked it seems |
| 07:59.36 | Strom_M | also, why are you using round robin for channel selection on your PRI? |
| 07:59.56 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:00.01 | railsmunky | i thought it might help. I've just done that now. Channel 1 was locked up so nothing would go out |
| 08:00.12 | Strom_M | heh |
| 08:00.27 | Strom_M | outbound calls are always supposed to hunt from the high-numbered channel down |
| 08:00.36 | railsmunky | ok. so G |
| 08:01.03 | railsmunky | they all appear to be cleared now :/ |
| 08:01.16 | Strom_M | by using G? |
| 08:01.21 | railsmunky | it's a good thing i guess - but i don;t like not knowing why |
| 08:01.24 | railsmunky | it was on g |
| 08:01.30 | railsmunky | i've set it to G now |
| 08:01.33 | Strom_M | ok |
| 08:01.50 | railsmunky | ok panic over for now. Do the configs look ok? |
| 08:01.52 | Strom_M | see if the problem returns -- my theory is that the telco didnt like your channel selection |
| 08:01.59 | Strom_M | lemme have another look |
| 08:02.00 | Strom_M | hang on |
| 08:02.02 | railsmunky | with starting on g? |
| 08:02.59 | Strom_M | apart from a lot of pointless settings in zapata.conf, it looks OK to me |
| 08:03.57 | railsmunky | aha :) |
| 08:04.24 | railsmunky | Thanks for your help. I'll keep an eye on it and see if anything else crops up |
| 08:08.07 | Prof_Pardal | railsmunky: did u try call with all 30 channels? |
| 08:09.03 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:15.01 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 08:27.46 | liri | if a meetme is running an AGI script in the background which endlessly loops on wait_for_digit() would it cause the participants of the conference to be cut out from each other? |
| 08:29.44 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 08:38.23 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 08:38.24 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 08:38.30 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:39.07 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 08:40.23 | phpboy | How can I check which call groups an extension belongs to? |
| 08:40.44 | phpboy | or rather, which queues and extension belongs to on the CLI |
| 08:41.23 | hi365 | (core?) show queues |
| 08:42.57 | *** join/#asterisk yasser202 (n=yasser@196.205.236.18) |
| 08:43.02 | yasser202 | \ |
| 08:47.14 | *** join/#asterisk Segnale007 (n=Segnale0@host15-242-dynamic.9-79-r.retail.telecomitalia.it) |
| 08:49.21 | phpboy | hi365: na, manually logged into |
| 08:55.23 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 09:01.07 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 09:01.31 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:01.58 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 09:06.01 | *** join/#asterisk kippi (n=kippi@untrust-gct.equinoxit.net) |
| 09:06.28 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 09:06.35 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 09:12.44 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 09:14.01 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 09:18.13 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-35ce1f704659afa3) |
| 09:22.08 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 09:45.23 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-0e190660ba8004ff) |
| 09:47.43 | *** join/#asterisk real_epicac (n=IceChat7@136.240.13.217.in-addr.dgcsystems.net) |
| 09:49.05 | *** join/#asterisk blq (i=Bl@dslb-088-066-227-019.pools.arcor-ip.net) |
| 10:02.04 | real_epicac | I want to send incoming calls to a specific sip-address to a specific context. Our asterisk registers that sip-address to sip-server. Can i configure that in sip.conf with a peer/user? |
| 10:04.16 | *** join/#asterisk kannan (n=kann@121.246.224.66) |
| 10:06.52 | kannan | hi to all.I am newly using the FXO card TDM2400p.I am using xlite softphone to make outgoing calls.When i dial the phone will ring in other end,if they pick up the call will hangup in asterisk. |
| 10:07.59 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 10:10.26 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 10:10.51 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:12.07 | pputman | kannan, I would use a callfile to determine if the problem is the card or a sip issue. |
| 10:12.27 | tzafrir_laptop | or 'originate' in the command-line |
| 10:13.04 | tzafrir_laptop | or even simpler: pastebin a trace of the call: core set verbose 3 |
| 10:13.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:13.17 | tzafrir_laptop | and show us what you see in the CLI |
| 10:13.47 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:13.52 | *** join/#asterisk dauhuber (n=dauhuber@213.189.154.108) |
| 10:13.54 | kannan | pputman:thank you.if i post any call log? |
| 10:14.08 | tzafrir_laptop | never realised asterisk 1.6.2 was released :-) (a post on asterisk-users mailing list) |
| 10:14.38 | tzafrir_laptop | jbot, tell kannan about pastebin |
| 10:14.52 | pputman | ~pastebin |
| 10:14.52 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:17.30 | kannan | i am using asterisk 1.2.27,zaptel 1.2.26 and libpri 1.2.5 |
| 10:17.47 | pputman | that might be your problem |
| 10:18.05 | kannan | i cant? |
| 10:18.31 | pputman | I'm not saying it won't work, its just those zaptel drivers are pretty outdated |
| 10:18.49 | pputman | like he said I would pastebin |
| 10:20.07 | kannan | then we try for the latest version only? |
| 10:21.21 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 10:21.35 | pputman | kannan, you might just have a misconfiguration. If you pastebin your logs we can tell more. |
| 10:21.53 | kannan | ok |
| 10:24.18 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 10:24.25 | tzafrir_laptop | pputman, it's latest zaptel 1.2 |
| 10:24.39 | tzafrir_laptop | kannan, ==^ |
| 10:25.18 | tzafrir_laptop | Not latest libpri, but you don't even need libpri anyway |
| 10:26.00 | pputman | yeah its just everytime ive ever had a problem with 1.2 zaptel drivers, an upgrade to the latest 1.4 fixed it. but then a lot of those times it was trixbox precompiled which have problems anyways. |
| 10:30.43 | kannan | pputman : just now post my logs in pastebin |
| 10:41.29 | pputman | kannan, it should have given you a link. |
| 10:42.31 | tzafrir_laptop | pputman, latest zaptel 1.2 have many of the bug fixes that latest zaptel 1.4 have |
| 10:43.47 | kannan | i cant.In which asterisk 1.4 version we are using that same conf which we use in 1.2ver |
| 10:46.29 | pputman | kannan, before you upgrade, send us the link to the pastebin |
| 10:47.06 | XnOSX | what is the command line for disable DTMF Debug in Asterisk CLI? |
| 10:51.14 | pputman | XnOSX, you can edit your /etc/asterisk/logger.conf and edit out dtmf from the line that says console, then do a logger reload |
| 10:51.37 | XnOSX | pputman ok thanx so much |
| 10:53.26 | XnOSX | pputman, hold on, but if i do this the DTMF Debug will be disable in Asterisk CLI? |
| 10:54.36 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:55.13 | pputman | XnOSX, yes, if you disable it from the console line. |
| 10:55.44 | XnOSX | ok |
| 11:21.30 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 11:23.04 | Mad|Cow | Anyone ever had any luck with spandsp? I'm trying to recieve a fax but keep getting the following: "channel.c: Unable to find a codec translation path from alaw to unknown" and "app_rxfax.c: Unable to restore read format on 'SIP/fax_user-0822e928'" Any ideas? |
| 11:23.39 | *** join/#asterisk exvito (n=exvito@89.181.47.143) |
| 11:26.48 | exvito | hi, i've just experienced the 2nd (in about 2 months) FXS line stuck at offhook state in a digium 4 port analog card. the only way i managed to bring it back to onhook was by stopping asterisk, reloading zaptel and restarting asterisk (no chan_zap.so reload would do it and no soft hangup possible as the channel did not seem to be in use) any ideas as on how to imrpove such reset ? |
| 11:29.02 | thomas | hey! |
| 11:29.21 | thomas | is it ok if i have many iax channels? why i have 32?! |
| 11:29.22 | thomas | http://paste.keks.be/67 |
| 11:29.26 | thomas | and why activ? |
| 11:31.46 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:34.09 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 11:35.15 | *** part/#asterisk exvito (n=exvito@89.181.47.143) |
| 11:37.11 | thomas | backup*CLI> iax2 show channels |
| 11:37.11 | thomas | Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format |
| 11:37.14 | thomas | (None) 193.108.19.245 (None) 07587/16954 00001/00001 00000ms -0001ms 0000ms unknow |
| 11:37.17 | thomas | (None) 193.108.19.245 (None) 07689/16937 00001/00001 00000ms -0001ms 0000ms unknow |
| 11:37.20 | thomas | what is this? |
| 11:37.22 | thomas | "none" ? |
| 11:50.16 | *** join/#asterisk Segnale007 (n=Segnale0@host15-242-dynamic.9-79-r.retail.telecomitalia.it) |
| 11:51.01 | thomas | <PROTECTED> |
| 11:51.28 | thomas | wrong window |
| 11:52.47 | *** join/#asterisk spike008t (n=spikie@ven69-2-82-228-116-153.fbx.proxad.net) |
| 11:52.56 | spike008t | Hi all |
| 11:56.48 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:56.58 | phpboy | hi |
| 11:57.03 | thomas | hi |
| 11:57.13 | phpboy | hi |
| 11:57.17 | thomas | :) |
| 11:57.17 | spike008t | hi lol |
| 11:57.23 | thomas | backup*CLI> iax2 show channels |
| 11:57.23 | thomas | Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format |
| 11:57.24 | spike008t | :) |
| 11:57.27 | thomas | (None) 193.108.19.245 (None) 08136/16989 00001/00001 00000ms -0001ms 0000ms unknow |
| 11:57.27 | phpboy | *giggle* |
| 11:57.30 | thomas | what is it? |
| 11:57.35 | thomas | channel "(None)" ? |
| 11:57.43 | liri | if a meetme is running an AGI script in the background which endlessly loops on wait_for_digit() would it cause the participants of the conference to be cut out from each other? |
| 11:58.10 | *** join/#asterisk droem (i=droem@p548EA537.dip0.t-ipconnect.de) |
| 11:58.11 | thomas | phpboy: you created php script for agi? |
| 11:58.22 | awk | liri out of intrest why not use a timeout |
| 11:58.23 | phpboy | *nod* |
| 11:58.48 | spike008t | hey i wanna know if the last version of iaxclient is stable under windows? |
| 11:58.56 | liri | awk: a timeout would mean that the "window" time for users to press the dtmf is limited to a specific time |
| 11:59.11 | awk | correct |
| 11:59.16 | awk | why would that be unlimited |
| 11:59.42 | liri | awk: I'd like to always listen to the participants dtmf's and when it happens perform some action (run a script) |
| 12:00.00 | *** join/#asterisk droem (i=droem@p548EA537.dip0.t-ipconnect.de) |
| 12:00.05 | liri | awk: is there a proper way of doing that? because using the agi background script seems to cut out the users on the conf |
| 12:00.29 | awk | what user? |
| 12:00.32 | awk | everyone? |
| 12:00.36 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 12:00.48 | awk | or the user waiting for the dtmf |
| 12:01.04 | liri | awk: everyone, not a specific user |
| 12:01.08 | liri | awk: user=participant in the conf |
| 12:01.15 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 12:06.03 | liri | awk: whats a proper way of doing that? |
| 12:09.58 | thomas | what is it? (None) 193.108.19.245 (None) 03149/17007 00001/00001 00000ms 0000ms 0040ms unknow |
| 12:10.05 | thomas | if i send the command iax2 show channels |
| 12:10.10 | thomas | the channel iss death?! |
| 12:10.16 | thomas | 1 active IAX channel |
| 12:13.17 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583423.dsl.bell.ca) |
| 12:14.32 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.40 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:14.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:15.08 | thomas | russellb: hello. :-) |
| 12:15.15 | russellb | waves |
| 12:15.18 | thomas | :/ |
| 12:15.28 | russellb | why the :/ ? |
| 12:15.38 | thomas | russellb: little question. have the new version .1 updated, |
| 12:15.59 | thomas | and i dont know why I have many (dead?) channels: |
| 12:16.01 | thomas | russellb: http://paste.keks.be/67 |
| 12:16.42 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:17.23 | [TK]D-Fender | thomas: Whats the actual problem? |
| 12:19.09 | thomas | [TK]D-Fender: hi. i dont know, is it a problem or no? backup*CLI> iax2 show channels > 32 active IAX channels (http://paste.keks.be/67) |
| 12:19.19 | russellb | are there active calls? |
| 12:19.20 | thomas | the channels is active? i have no calls |
| 12:19.21 | russellb | active registrations? |
| 12:19.25 | thomas | russellb: mom |
| 12:19.34 | [TK]D-Fender | thomas: Does actually DO anything bad? |
| 12:19.34 | russellb | i'm not your mother |
| 12:19.44 | thomas | russellb: aeh mom = moment - sorry :) |
| 12:20.53 | thomas | russellb: http://paste.keks.be/68 |
| 12:21.11 | thomas | [TK]D-Fender: i dont know!! |
| 12:21.26 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:21.37 | [TK]D-Fender | thomas: Just stop then. You're worrying about nothing. |
| 12:22.23 | thomas | [TK]D-Fender: hm. ok. and 32 active IAX channels isnt worrying ? |
| 12:22.24 | thomas | ok... |
| 12:22.27 | thomas | :) |
| 12:22.33 | thomas | [TK]D-Fender: thank you very much |
| 12:23.15 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 12:27.07 | *** part/#asterisk dauhuber (n=dauhuber@213.189.154.108) |
| 12:27.44 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 12:28.20 | Mad|Cow | Anyone ever had any luck with spandsp? I'm trying to recieve a fax but keep getting the following: "channel.c: Unable to find a codec translation path from alaw to unknown" and "app_rxfax.c: Unable to restore read format on 'SIP/fax_user-0822e928'" Any ideas? |
| 12:33.32 | [TK]D-Fender | Mad|Cow: do you issue an explicit "Answer" before calling rxfax? |
| 12:34.06 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 12:35.25 | Mad|Cow | [TK]D-Fender: Yes i do. Is that not correct? |
| 12:36.00 | [TK]D-Fender | Mad|Cow: thats fine.. |
| 12:36.51 | *** join/#asterisk tompaw (n=tompaw@inet20909ng-0.eranet.pl) |
| 12:36.58 | Mad|Cow | [TK]D-Fender: The fax actually starts to go through... but when it seems to send, I see that message about unable to find a codec translation path |
| 12:37.08 | *** join/#asterisk CVirus (n=Burzum@196.218.41.31) |
| 12:42.05 | jeremy_g | hi |
| 12:42.11 | jeremy_g | does asterisk support IM and presence |
| 12:42.31 | *** join/#asterisk railsmunky (n=nick@82-70-165-142.dsl.in-addr.zen.co.uk) |
| 12:42.48 | [TK]D-Fender | jeremy_g: No, and yes (for devices registered to * and placing calls to it) |
| 12:43.26 | CVirus | I just ran zttest and it yields this .. Best: 0.000 -- Worst: 100.000 -- Average: 100.000000, Difference: 100.000000 |
| 12:43.28 | CVirus | what does it mean ? |
| 12:43.34 | CVirus | I have two FXO cards installed |
| 12:43.48 | CVirus | --- Results after 0 passes --- |
| 12:44.24 | railsmunky | Hey people. Right this morning panic out of the way... I'm having troubles with 0141 nubmers in the uk. Stripping the 0 from the number gives 141 and i'm getting a BT unrecognised number - so it's getting out. Every other number works 01274 down to 1274xxxxxx. I think it might be to do with the 141 on BT number to hide the ID. I'm wondering if sending a full international would work (but it's not) 441274? |
| 12:44.34 | railsmunky | Anyone any ideas? |
| 12:49.01 | railsmunky | i'm sending 441274 and that doesn't work |
| 12:49.14 | [TK]D-Fender | railsmunky: pastebin your dialplan and a complete call attempt at verbose 10 |
| 12:49.15 | [TK]D-Fender | ~pb |
| 12:49.19 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:49.21 | railsmunky | okedoke |
| 12:49.24 | *** join/#asterisk CleanerX (n=nix@p5B1331C1.dip0.t-ipconnect.de) |
| 12:49.45 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 12:50.31 | CVirus | I have a motherboard with 1 PCI slot ... so I'm using another PCI card that gives me two PCI slots and I inserted two FXO cards on them ... is there a problem with that ? |
| 12:50.38 | railsmunky | works |
| 12:50.38 | railsmunky | http://pastebin.com/m639afe1a |
| 12:51.24 | railsmunky | doesn't http://pastebin.com/mdeb9079 |
| 12:51.29 | *** join/#asterisk ManxPower (n=manxpowe@241.sub-70-222-228.myvzw.com) |
| 12:52.56 | railsmunky | [TK]D-Fender: its more what the telco (BT) expects rather than a dodgy dialplan |
| 12:53.11 | railsmunky | [TK]D-Fender: since they both make it out to the PRI and beyond |
| 12:53.41 | [TK]D-Fender | railsmunky: Go ask them then. |
| 12:53.58 | railsmunky | [TK]D-Fender: ok sorry :) |
| 12:54.19 | railsmunky | just wondering if anyone had done it before |
| 12:55.13 | ManxPower | done what? |
| 12:55.22 | railsmunky | setup asterisk on a BT pri |
| 12:55.54 | railsmunky | and had every number working (which it does) including 0141 (which it doesn't) :) |
| 12:56.01 | [TK]D-Fender | CVirus: You know, X100P's are bad enough all by themselves... and you want to try to be an even cheaper bastard and run them off a riser card? |
| 12:56.03 | ManxPower | railsmunky: There can't be much difference between a BT PRI and a PRI in any other alaw country |
| 12:56.25 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 12:56.30 | CVirus | [TK]D-Fender: LOL .. it's my boss who gave me the machine and the task |
| 12:56.34 | railsmunky | ManxPower: yeah that's what i figured. Do other countries have the 141 to hide the caller id? |
| 12:56.57 | ManxPower | railsmunky: Um, you don't use 141 to hide callerid on a PRI. |
| 12:57.09 | [TK]D-Fender | CVirus: then "good luck". |
| 12:57.10 | ManxPower | you use the Calling Pres and Caller Pres stuff |
| 12:57.29 | CVirus | [TK]D-Fender: Thanks :-) |
| 12:57.58 | CVirus | [TK]D-Fender: when I rmmod wcfxo and modprobe it again ... zttool can see them both ... but when I normal boot .. it sees only one of them .. any idea ? |
| 12:58.09 | ManxPower | railsmunky: you could easily emulate 141 in your dialplan. |
| 12:58.55 | [TK]D-Fender | CVirus: maybe you should make your own startup script to load them |
| 12:58.55 | ManxPower | CVirus: Around here "good luck" means "You're crazy and I'm not going to help you, but I'll try to be polite about it." |
| 12:59.21 | gr0mit | railsmunky, we have done a lot of PRI here |
| 12:59.41 | railsmunky | ManxPower: no that's the point i don't care about hiding it. The problem is that i can't call any number in glasgow :) |
| 12:59.51 | railsmunky | ManxPower: which has the area code 0141 |
| 12:59.56 | gr0mit | railsmunky, 141 works fine |
| 13:00.01 | railsmunky | ManxPower: that's the only thing i can think of |
| 13:00.51 | gr0mit | railsmunky, can you explain the issue again? |
| 13:00.56 | railsmunky | asterisk is doing what i does for every other area code in the uk. However when i dial a 0141 area code i get the telco saying they don't recognise the number |
| 13:00.58 | [TK]D-Fender | railsmunky: Again, call your telco and ask them why |
| 13:01.05 | railsmunky | yeah |
| 13:01.19 | *** join/#asterisk Mnabil_work (n=mnabil@196.218.41.31) |
| 13:01.37 | gr0mit | railsmunky, make sure you set the pridialplan=unknown |
| 13:01.46 | gr0mit | in your zapata.conf |
| 13:01.48 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:02.11 | gr0mit | if you want a working zaptata.conf file for a BT ISDN30 let me know. |
| 13:02.20 | ManxPower | railsmunky: like everything with PRI, you need to see what HANGUPCAUSE is. |
| 13:02.55 | gr0mit | ManxPower, ISDN here in UK operates very differently to the US |
| 13:03.12 | ManxPower | gr0mit: if it's ISDN, it has a HANGUPCAUSE |
| 13:03.17 | gr0mit | it does |
| 13:03.22 | railsmunky | where would i see that? |
| 13:03.35 | gr0mit | but I think the problem is the dialplan and the settings in zapata.conf |
| 13:03.38 | ManxPower | and that HANGUPCAUSE will tell you why the telco is rejecting the call. |
| 13:03.43 | gr0mit | and 141 works fine |
| 13:04.03 | ManxPower | railsmunky: after the Dial, do a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) then you see it in the dialplan. |
| 13:04.12 | railsmunky | great. i'll try that |
| 13:04.17 | gr0mit | railsmunky, please pastebin your zapata.conf file and we can take a look for you |
| 13:04.22 | ManxPower | railsmunky: then put the output of the failed call on pastebin.ca |
| 13:05.25 | gr0mit | railsmunky, you need to send the normal digits to line. |
| 13:05.37 | gr0mit | not in internat format or without a leading zero |
| 13:05.53 | ManxPower | gr0mit: thank you for volunteering to help railsmunky. I can go back to regular work. |
| 13:05.54 | gr0mit | i.e. if you are calling a local number dial the 6 or 7 digits |
| 13:06.10 | railsmunky | when i send full codes eg... 01274XXXXXX i get a fail from the teclo |
| 13:06.15 | gr0mit | yup |
| 13:06.17 | railsmunky | i have to send 1274XXXXXX |
| 13:06.30 | gr0mit | so you need to check what is in your zapata.conf |
| 13:06.43 | gr0mit | and make sure you have pridialplan=unknown |
| 13:06.44 | railsmunky | http://pastebin.com/m55a19e65 |
| 13:07.29 | gr0mit | yup. so make sure you have pridialplan=unknown, then you can dial all calls normally |
| 13:07.47 | gr0mit | it is a quirk of BT's implementation of ISDN |
| 13:08.07 | railsmunky | so include the leading 0> |
| 13:08.08 | railsmunky | ? |
| 13:08.10 | gr0mit | yup |
| 13:08.31 | liri | is it possible to enable feature codes (catching DTMF) for participants on conference? |
| 13:10.11 | railsmunky | gr0mit: http://pastebin.com/m432ac1c7 |
| 13:10.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:10.30 | railsmunky | gr0mit: is that correct. I've set that in my zapata.conf |
| 13:10.55 | gr0mit | i think you need a lowercase g |
| 13:10.59 | [TK]D-Fender | liri: Instructions seeme to say "no". |
| 13:11.16 | railsmunky | gr0mit: no that's just which channels to use |
| 13:11.21 | gr0mit | but yes, that is the sort of thing |
| 13:11.37 | gr0mit | well, let me paste my zapata.conf |
| 13:11.40 | gr0mit | 1 sec |
| 13:11.42 | railsmunky | ok ta |
| 13:13.01 | gr0mit | http://www.pastebin.ca/index.php |
| 13:13.29 | railsmunky | think i need more than that :) |
| 13:13.42 | gr0mit | doh! |
| 13:13.50 | gr0mit | http://www.pastebin.ca/1065507 |
| 13:14.20 | railsmunky | gr0mit: great thanks! |
| 13:14.25 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 13:14.26 | liri | [TK]D-Fender: there just has to be a way to do it :) |
| 13:14.44 | gr0mit | then in the dialplan, you need exten _9X.,1,Dial(ZAP/g1/${EXTEN:1},60) or similar |
| 13:14.46 | [TK]D-Fender | liri: You have the source, just like the rest of us, get busy. |
| 13:14.50 | hsv-al | hello fellow internet addicts - are we all looking forward to another long & glorious week of internet addiction, coffee sipping, burning eyes @ 8am, and staring at our monitors? |
| 13:14.53 | ManxPower | liri: sure there is, go modify the code. |
| 13:15.06 | gr0mit | exten => _9X.,1,Dial(ZAP/g1/${EXTEN:1},60) |
| 13:15.09 | gr0mit | i mean |
| 13:15.27 | liri | I thought it requires a modification of configuration such as extension |
| 13:16.40 | [TK]D-Fender | liri: ... read the app's instructions. that functionality does not exist. You'll need to code all of it yourself at source level. |
| 13:17.04 | gr0mit | railsmunky when you set your outbound callerid you need to send BT the same number of digits as they are sending you for incoming calls |
| 13:19.05 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 13:19.20 | gr0mit | working yet, railsmunky ?? |
| 13:20.07 | railsmunky | gr0mit: no and my hour is up :/ I'll give it a go out of hours. Could it be to do with the nationalprefix etc.. too? |
| 13:20.52 | gr0mit | have you restarted asterisk? |
| 13:21.02 | railsmunky | gr0mit: yep |
| 13:21.11 | gr0mit | not just reloaded? |
| 13:21.20 | railsmunky | gr0mit: yeah |
| 13:21.22 | *** join/#asterisk glaz (n=strke@206.223.238.2) |
| 13:21.36 | gr0mit | this is a BT ISDN30, right? |
| 13:21.44 | railsmunky | gr0mit: yeah with 18 channels |
| 13:22.18 | gr0mit | ok, well the zapata file i send you is from our own working BT line |
| 13:22.23 | gr0mit | so it defo works! |
| 13:22.55 | jeremy_g | my * box is registering with a remote sip proxy but it is required to have a certain display name? how do i set that. is there any parameter in sip.conf that can be changed. calleridname=desired display name <-- is this valid in sip.conf before register => line |
| 13:23.02 | railsmunky | gr0mit: it looks better than mine so i'm going to migrate those settings across tonight and alter my dialplan accordingly. We'll see if it makes a difference. |
| 13:23.13 | gr0mit | ok, well i am about this everning |
| 13:23.24 | hsv-al | Motorola Engineering 3GSM Test Mobile |
| 13:23.25 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:23.32 | gr0mit | just ping so if you get stuck let me know |
| 13:23.35 | railsmunky | gr0mit: great thanks for your help now, and i'll get on this evening and give it a go. |
| 13:24.06 | gr0mit | hsv-al, and....? |
| 13:26.51 | jeremy_g | why doesnt anyone dare to solve my problem |
| 13:27.06 | [TK]D-Fender | jeremy_g: I answered this yesterday. You can't <- |
| 13:27.07 | jeremy_g | a custom display field for * registering with another sip proxy |
| 13:27.23 | ManxPower | jeremy_g: maybe because someone came here asking the same question several days ago, and the answer seemed to be "you can't do that in Asterisk" |
| 13:27.55 | jeremy_g | ManxPower, [TK]D-Fender: ok, that means i ll have to change the code |
| 13:28.01 | ManxPower | your only real option would to install SER, which lets you do most anything you want to do. |
| 13:29.00 | liri | [TK]D-Fender: actually on voip-info.org Asterisk+cmd+MeetMe page there's a command about the conf-background.agi script "this does not work with non-Zap channels in the same conference" |
| 13:29.19 | liri | [TK]D-Fender: so maybe this functionality exist but it requires the channel to be Zap and no SIP |
| 13:30.11 | *** join/#asterisk canapa (n=canapa@80.120.254.214) |
| 13:31.05 | jeremy_g | liri:you have to compile some kernel module, uncomment some clock line in the Makefile somewhere |
| 13:31.08 | ManxPower | liri: so as you can see you can't do what you want to do. |
| 13:31.31 | jeremy_g | liri:damn, some object module .o not kernel module |
| 13:31.42 | jeremy_g | but thats too old |
| 13:33.08 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:33.14 | hsv-al | getting an error on this line: |
| 13:33.53 | liri | possibly I can connect to the AMI and wait on/filter events for the conference? |
| 13:34.06 | hsv-al | exten => _X.,n,Set(WHO= ${HOTDESK_PHONE_STATUS(${LOCATION})}) |
| 13:34.14 | hsv-al | where's the error in that?? |
| 13:34.28 | [TK]D-Fender | hsv-al: first thing, leading space.... |
| 13:35.30 | hsv-al | _X.,n,Set(WHO=${ |
| 13:35.32 | hsv-al | ahhh, there it is |
| 13:35.35 | ManxPower | hsv-al: I don't see an error message! |
| 13:35.48 | hsv-al | works oddly enough after space closed between = $ |
| 13:36.05 | creativx | asterisk loves the whitespace |
| 13:36.07 | ManxPower | hsv-al: now you know why we never add extra white space. |
| 13:36.22 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 13:37.02 | [TK]D-Fender | liri: Go try. |
| 13:39.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:41.11 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:45.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:45.56 | Mad|Cow | I'm getting the following in my debug: "chan_sip.c: ** Our prefcodec: 0x0 (nothing)" Anyone know how to set this to ulaw? |
| 13:46.19 | ManxPower | Mad|Cow: looks like you didn't allow= any codecs |
| 13:46.51 | Mad|Cow | ManxPower: Is it refering to my sip.conf? |
| 13:47.17 | ManxPower | Mad|Cow: yes, all SIP stuff is in sip.conf |
| 13:47.52 | hsv-al | manx power: |
| 13:47.53 | hsv-al | http://i29.tinypic.com/34djw44.jpg |
| 13:48.07 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 13:49.07 | Mad|Cow | manxpower: I have allow=ulaw and allow=alwa in [general], will that not do it? |
| 13:50.55 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 13:51.37 | ManxPower | alwa is not a valid codec. |
| 13:52.05 | ManxPower | You do not generally want to enable both ulaw and alaw. |
| 13:52.07 | Mad|Cow | ManxPower: sorry... it says alaw |
| 13:52.42 | ManxPower | Mad|Cow: put a copy of your sip.conf, masking only yhe passwords, on pastebin.ca |
| 13:52.46 | Mad|Cow | ManxPower: I actually only want to use ulaw... but i would be happy with either at the moment |
| 13:52.56 | Mad|Cow | ManxPower: roger |
| 13:54.06 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:54.33 | ManxPower | still waiting on that pastebin |
| 13:57.17 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-79-65-218.sw.biz.rr.com) |
| 13:58.44 | Mad|Cow | ~pastebin |
| 13:58.47 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:58.53 | *** join/#asterisk CVirus (n=Burzum@196.218.41.31) |
| 13:58.57 | CVirus | pci:0000:02:09.0 wcfxo+ 1057:5608 Wildcard X100P |
| 13:59.01 | CVirus | pci:0000:02:0e.0 wcfxo- 1057:5608 Wildcard X100P |
| 13:59.10 | CVirus | Why is my second FXO card not handled y wcfxo ? |
| 13:59.53 | CVirus | by* |
| 13:59.57 | Mad|Cow | ManxPower: sorry... http://pastebin.com/d767b365a |
| 14:00.02 | *** part/#asterisk MarkWD (n=Mark@rrcs-67-79-65-218.sw.biz.rr.com) |
| 14:01.07 | Mad|Cow | ManxPower: I'm storing most of my sip users in mysql...... the sip user fax_user is who we are interested in... |
| 14:02.13 | Mad|Cow | ManxPower: in my disallow, i have all. and in allow, i have ulaw;alaw |
| 14:02.42 | liri | what is the Present Menu option in MEETME_OPTS aka 's' |
| 14:03.02 | CVirus | Do I need a sound card for asterisk to output any sounds ? |
| 14:03.33 | jaytee | CVirus, only if you're making calls from the server which most people don't |
| 14:03.56 | ManxPower | Mad|Cow: I cannot help you then, as I don't do Realtime |
| 14:04.12 | ManxPower | Mad|Cow: STOP IT!!! ALLOW ONLY ONE CODEC! |
| 14:04.36 | ManxPower | In many versions of asterisk allow=ulaw;alaw is invalid |
| 14:05.14 | CVirus | jaytee: thanks |
| 14:05.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:05.40 | Mad|Cow | manxpower: I'll give it a shot. thanks :-) |
| 14:06.54 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 14:10.28 | Mad|Cow | ManxPower: ever seen anything like this? "channel.c:3059 set_format: Unable to find a codec translation path from ulaw to unknown" |
| 14:12.13 | waverly360 | Hey guys. I'm running asterisk 1.2.14, with zaptel-1.4.9.2, libpri-1.2.7 and for some reason asterisk can't see the pri. When I attempt to do a "pri show span 1" I get a "no such command". I'm using a sangoma a101d card with wanrouter version 3.2.5. wanrouter status shows the pri as up and connected. Sangoma's website claims that my error is due to a misconfigured/misinstalled libpri, but the instructions for libpri are very straightforward, so |
| 14:13.12 | ManxPower | waverly360: use 1.2 zaptel for 1.2 asterisk and libpri |
| 14:13.35 | ManxPower | don't expect Asterisk to even find zaptel 1.4 |
| 14:14.04 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:14.09 | *** join/#asterisk gfather (n=gg@86.108.97.14) |
| 14:14.15 | gfather | hello guys |
| 14:14.22 | waverly360 | ...y'know now that you mention it...it does seem silly to use 1.4... |
| 14:14.33 | waverly360 | scratches his head. |
| 14:14.57 | waverly360 | I thought I had compiled with the 1.2 version...let me check my notes... thanks Manx |
| 14:16.05 | gfather | like i was reading some stuff , and like i can connect to the asterisk from any wifi point |
| 14:16.24 | gfather | but can i dail for free by usinf an ip or something |
| 14:17.22 | gfather | like lets say i have a wifi phone , an i want to call the server , but i dont have priveleges to get into the netwrok like a local |
| 14:17.49 | gfather | so can i dail the ip or something , and will it ring , like im getting a phone call , not as a local |
| 14:17.51 | gfather | ? |
| 14:18.01 | gfather | or by a software on the pc ? |
| 14:18.10 | gfather | is that possible ? |
| 14:18.38 | ManxPower | gfather: none of us have any idea what you are talking about. |
| 14:18.39 | creativx | dail |
| 14:18.44 | creativx | he wants to dail!! |
| 14:18.49 | creativx | duhhhh ManxPower.. |
| 14:19.08 | gfather | ok i have some writing mistakes |
| 14:19.19 | gfather | <ManxPower> ok ill explain better |
| 14:19.50 | gfather | i can connect a wifi phone from any wifi point to the asterisk server , right |
| 14:20.11 | ManxPower | gfather: in theory yes, in practice it does not work very well. |
| 14:20.25 | ManxPower | ~wifiphones |
| 14:20.28 | ManxPower | ~wifi |
| 14:20.29 | jbot | hmm... wifi is see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
| 14:20.52 | gfather | but this phone has to be already configured with the server |
| 14:20.57 | ManxPower | well that was useless |
| 14:21.10 | ManxPower | gfather: all phones have to be configured for the server. |
| 14:21.16 | ManxPower | If you want to use that server. |
| 14:21.21 | gfather | yes |
| 14:21.54 | gfather | but if i want to use the phone , and its not configured with the server |
| 14:22.13 | ManxPower | gfather: exactly what do you want to do with the phone if you don't have it configured? |
| 14:22.26 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 14:22.43 | gfather | i want to call , using the ip , or a domain name , without configuring it |
| 14:23.02 | ManxPower | gfather: your question has NOTHING to do with Asterisk. |
| 14:23.02 | gfather | like lets say , the company i want to call have asterisk |
| 14:23.12 | ManxPower | You must contact tech support for the phone you purchased. |
| 14:23.25 | gfather | nah man , ur not getting my point |
| 14:23.29 | gfather | like lets say , the company i want to call have asterisk |
| 14:23.46 | gfather | they have configured it , so if u have a wifi phone |
| 14:24.00 | gfather | u can call them for free on www.company.com |
| 14:24.25 | ManxPower | gfather: OK. Then contact the tech support for your phone, as this has nothing to do with Asterisk. I cannot tell you how to dial by IP address or SIP URI as how you do that is different for every phone. |
| 14:24.50 | ManxPower | gfather: so do whatever you have to do to make the phone call www.company.com |
| 14:25.07 | gfather | im asking if it can be done on asterisk |
| 14:25.17 | gfather | im asking if i can do this option |
| 14:25.21 | ManxPower | gfather: no, it cannot be done on Asterisk. |
| 14:25.40 | gfather | but its easy , its not that big thing to do |
| 14:25.42 | ManxPower | Fortunatly it has NOTHING WHATSOEVER TO DO WITH ASTERISK. |
| 14:26.06 | gfather | im sure u still dont get what im asking |
| 14:26.38 | gfather | lets say its a new feature for asterisk , can it be done ? |
| 14:26.55 | ManxPower | gfather: what you are asking is "how do I use WiFi SIP Phone <unknown model> <unknown brand> may a direct call dialed using the IP address to a specific SIP server" |
| 14:27.09 | gfather | <ManxPower> no im not asking that |
| 14:27.35 | file | chan_sip can accept calls from unauthenticated devices... if configured to do so |
| 14:27.37 | gfather | may a direct call dialed using the IP address to a specific SIP server" |
| 14:27.46 | gfather | thats right |
| 14:27.47 | ManxPower | gfather: Then exactly what are you asking? What SIP server the company you are calling does not matter. ANY SIP server, from Microsoft, cisco, or open source will work. |
| 14:28.18 | gfather | what im asking is ( may a direct call dialed using the IP address to a specific SIP server") |
| 14:28.20 | ManxPower | gfather: I cannot help you further. Best of luck. |
| 14:28.55 | gfather | like can i do that for my company , or others do it by asterisk |
| 14:28.55 | ManxPower | You obviously do not know how Asterisk works, how SIP works, and how WiFi SIP Phones work and I do not have a days worth of time to teach you. |
| 14:28.59 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 14:29.27 | gfather | man i know ur here to help |
| 14:29.28 | *** part/#asterisk ManxPower (n=manxpowe@241.sub-70-222-228.myvzw.com) |
| 14:29.45 | gfather | ok lets say i have an asterisk server |
| 14:30.13 | gfather | i want to call through the ip ,or through the domain name |
| 14:30.22 | *** join/#asterisk naitram (n=naitram@216.77.58.40) |
| 14:30.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:30.32 | gfather | but not by configuring it from the start |
| 14:30.55 | gfather | but not by configuring the phone from the start |
| 14:31.12 | naitram | is there a way to set a custom call back event to an AMI client? I need to fire an event when busy is detected in my dial scripts |
| 14:32.20 | gfather | <@file> chan_sip can accept calls from unauthenticated devices... if configured to do so , so it can work ? |
| 14:32.44 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 14:33.22 | gfather | so what im asking can be done |
| 14:33.42 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-22a69f431f09f96d) |
| 14:34.07 | gfather | u dont have to be authenticated to call from the net |
| 14:34.47 | *** join/#asterisk spokra (n=spokra@host093-179-142.sea0.speakeasy.net) |
| 14:38.17 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 14:39.42 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 14:43.12 | *** part/#asterisk naitram (n=naitram@216.77.58.40) |
| 14:43.15 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:43.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:43.49 | gfather | :S |
| 14:46.11 | *** part/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 14:47.09 | *** join/#asterisk ManxPower (n=manxpowe@241.sub-70-222-228.myvzw.com) |
| 14:52.21 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 15:00.52 | hsv-al | http://i2.cdn.turner.com/cnn/2008/CRIME/07/08/missing.mother.ap/art.jpg.hans.jpg |
| 15:01.07 | hsv-al | Reiser, known in programming circles for his ReiserFS file system |
| 15:01.14 | ManxPower | .part #not-really-funny-stuff |
| 15:01.20 | ManxPower | oops |
| 15:01.22 | hsv-al | all your reiser fs are belong to jail! |
| 15:01.22 | hsv-al | # NEW: Attorney says Reiser 'went right to' spot where body was found |
| 15:01.23 | hsv-al | # KTVU: Hans Reiser admits strangling wife; he previously denied killing he |
| 15:01.25 | bijit | for me to have a working dns server do I have to register a domain name? |
| 15:01.53 | ManxPower | bijit: no. |
| 15:02.40 | bijit | but if i just add any name will its be seen outside my LAN? |
| 15:07.50 | ManxPower | bijit: no. |
| 15:09.00 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 15:09.20 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 15:09.28 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 15:11.25 | ManxPower | You can run a name server with no domains configured, this is called a "caching nameserver" |
| 15:14.01 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:17.40 | *** join/#asterisk uTx (n=unix@modemcable074.229-82-70.mc.videotron.ca) |
| 15:19.25 | *** join/#asterisk TorrK (n=Torr@nat.jged.com) |
| 15:21.35 | *** join/#asterisk kj4acm (i=Ritalin@c-68-47-239-88.hsd1.tn.comcast.net) |
| 15:21.48 | macros73 | What amount of time lapse should there be, end to end, on a typical SIP call before the parties get annoyed? On some of my test calls it seems like a full second before they hear me. |
| 15:22.55 | kj4acm | could someone recommend some IP phones for asterisk use in the $200-300 range? (in the united states). It'll be fore a 4 extension small business. could possible grow to 6-8 extensions in a couple years. voip phone->asterisk->POTS |
| 15:23.15 | *** join/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-138-96.ph.ph.cox.net) |
| 15:23.17 | M1s3ry | kj4acm, polycom |
| 15:23.40 | *** part/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-138-96.ph.ph.cox.net) |
| 15:23.41 | kj4acm | m1s3ry: cool. that's what i was looking at. was curious about aastra, linksys and cisco too |
| 15:24.02 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:24.16 | macros73 | kj4acm: I have good experiences with the Aastra 480i in a home setting, fwiw. |
| 15:24.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:24.57 | macros73 | kj4acm: People I call from home via Vitelity/Asterisk/Aastra claim they perceive no difference in call quality from a traditional POTS call. |
| 15:27.24 | kj4acm | macros73: that's good to hear. you're doing voip->pots->call ? I was using a digium tdm400 that went fxs->card->asterisk->voip provider and had some echo problems at times |
| 15:27.58 | macros73 | kj4acm: No, just a SIP trunk on my end |
| 15:28.00 | *** join/#asterisk ACiDV (n=joel@246-192.hy.cgocable.ca) |
| 15:28.04 | kj4acm | i'm going to let this guy use this card and he'll do voip phones->asterisk->pots and i worry about echo. i guess using a channel bank is the way to go to solve that? |
| 15:28.23 | kj4acm | he doesn't want to use a voip provider |
| 15:29.03 | macros73 | kj4acm: Beyond my experience, currently. I haven't used any of the fxo/fxs stuff yet. |
| 15:30.13 | kj4acm | how do you like the 480i |
| 15:32.14 | macros73 | kj4acm: Love it. I have the CT version, comes with a wireless mobile handset that pairs with the base station. Crystal clear call quality unless I have torrents running. (No QoS on my current router.) |
| 15:32.36 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:32.38 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:33.40 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 15:33.46 | macros73 | kj4acm: (Oh, also, using g711 at home, too, which probably helps the call quality. That may not be feasible at your client site, depending on bandwidth available. But the 480i is supposed to support g729a directly. I'll test that tonight. |
| 15:34.06 | suma | macros73: GSM is also good |
| 15:34.13 | suma | no need of paying license |
| 15:34.20 | macros73 | http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB73-068D0DC3/03/hs.xsl/18230.htm#dl_instructions |
| 15:34.50 | macros73 | suma: When I tried GSM here, users gave it a thumbs down. 3/5 on call quality, though I admit we have needy users here. |
| 15:34.58 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 15:36.04 | *** join/#asterisk oilinki (n=oil@ppp-124-120-6-247.revip2.asianet.co.th) |
| 15:36.06 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 15:36.30 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 15:36.34 | macros73 | and they are used to the good life...ISDN PRI voice on our current phone system. :D |
| 15:37.53 | *** join/#asterisk jeffgus (n=jeffgus@216.86.199.4) |
| 15:40.17 | nr4q | polycom 650 > 560... right ? |
| 15:41.34 | *** join/#asterisk U-238 (n=u-238@pdpc/supporter/base/u-238) |
| 15:41.59 | nr4q | funny at voipsupply the 560 is more than 650 |
| 15:42.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:42.47 | U-238 | hello everyone, I am writing an asterisk module and I would like to have a function be executed constantly once every second (the function checks for changes in a mysql database) |
| 15:43.00 | U-238 | is there some way I can register this function to make asterisk run it constantly |
| 15:43.01 | U-238 | ? |
| 15:43.50 | CVirus | looks like there's noway to get two X100P cards on the riser card to work |
| 15:43.54 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 15:43.58 | [TK]D-Fender | nr4q: 560 = GIGABIT. Thats why its pricey |
| 15:44.03 | CVirus | been digging for the past 2 days |
| 15:44.19 | *** join/#asterisk adr3nalin3 (n=plasma@72-164-59-242.dia.static.qwest.net) |
| 15:44.34 | nr4q | [tk]d-fender: ah. looked like the 650 had more features. gigabit's not needed |
| 15:44.49 | [TK]D-Fender | U-238: And what would it do if there is a change? |
| 15:44.57 | U-238 | make a call |
| 15:45.04 | [TK]D-Fender | nr4q: Then you clearly should not be looking at the 560 at all |
| 15:45.11 | U-238 | if there is an outgoing channel available |
| 15:45.13 | nr4q | clearly :) |
| 15:45.15 | Qwell | CVirus: why not just get a multi-port card? |
| 15:45.18 | U-238 | otherwise it does nothing |
| 15:45.23 | U-238 | cos it will just try again next time |
| 15:45.26 | Qwell | Digium TDM410 - save yourself the trouble |
| 15:45.34 | [TK]D-Fender | U-238: you should write a stored procedure for your database and have that call a script. this is not *'s job |
| 15:45.46 | [TK]D-Fender | Qwell: His time clearly ahs no value |
| 15:46.11 | U-238 | other than by creating a text file in asterisk's spool folder |
| 15:46.17 | U-238 | what is the best way to get a script to make a call? |
| 15:46.43 | U-238 | I also want * to report status back to mysql like whether the call was successful, etc |
| 15:47.34 | macros73 | Sounds like someone is making an autodialer. :D |
| 15:47.41 | U-238 | yes, it is |
| 15:47.46 | U-238 | :) |
| 15:47.52 | macros73 | "Hello, are you happy with the size of your Wii?" |
| 15:47.55 | U-238 | dials and plays a message and hangs up |
| 15:48.29 | macros73 | Actually that could be nifty. Tie it into Nagios for example and get a call if a critical system goes down. |
| 15:48.38 | macros73 | Unless the critical system is the * system. |
| 15:48.43 | U-238 | haha |
| 15:49.03 | [TK]D-Fender | U-238: call file or AMI Originate |
| 15:49.23 | U-238 | what is AMI originate? |
| 15:49.34 | [TK]D-Fender | ~ami |
| 15:49.35 | jbot | hmm... ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 15:49.47 | U-238 | ok cool |
| 15:49.56 | U-238 | and I want to limit the number of concurrent calls that it makes |
| 15:50.02 | U-238 | because currently with call files |
| 15:50.15 | U-238 | if there are too many call files some of the calls just dont happen |
| 15:50.28 | [TK]D-Fender | U-238: this is up to whatever script you create to ensure |
| 15:50.37 | U-238 | ok |
| 15:51.30 | U-238 | and finally, |
| 15:51.49 | U-238 | does anyone know of an autodialler that can make calls which are scheduled in a mysql database? |
| 15:51.56 | U-238 | so I don't need to make one myself :) |
| 15:52.03 | [TK]D-Fender | U-238: No. Get busy |
| 15:52.11 | U-238 | ok |
| 15:52.13 | U-238 | I will |
| 15:53.07 | U-238 | if anyone's curious, the program is to call people and remind them to take medication at certain times |
| 15:53.17 | U-238 | thanks for the help |
| 15:53.28 | *** join/#asterisk adr3nalin3 (n=plasma@72-164-59-242.dia.static.qwest.net) |
| 15:54.13 | [TK]D-Fender | U-238: Why would it place a call based on a DB entry changing instead of "scheduled" as you now describe? |
| 15:54.16 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:54.25 | U-238 | oh |
| 15:54.26 | dominic1 | I need some help with the devstate function |
| 15:54.29 | U-238 | well the db entry doesnt change |
| 15:54.32 | U-238 | what I mean is |
| 15:54.40 | U-238 | the program constantly checks the db |
| 15:54.48 | U-238 | for calls scheduled in the past |
| 15:55.10 | dominic1 | I want to do the following: if somebody is called the devstate should be set to ringing |
| 15:55.17 | [TK]D-Fender | U-238: I think you need to completely rethink your approach.... |
| 15:55.24 | U-238 | ok |
| 15:55.26 | dominic1 | if the user picks up the call the devstate should be inuse |
| 15:55.33 | U-238 | well I'm open to suggestions |
| 15:55.36 | U-238 | basically |
| 15:55.44 | U-238 | I want to schedule for calls to be made at certain times |
| 15:55.49 | U-238 | and play a certain message |
| 15:55.56 | U-238 | I don't mind too much how its done |
| 15:55.58 | [TK]D-Fender | dominic1: Why are you using devstate to duplicate what basic presence should already be doing and fully capable of? |
| 15:56.25 | [TK]D-Fender | U-238: Just go sit and think on it and let us know when you reach an * problem about it. |
| 15:56.47 | *** join/#asterisk furibondox (n=linux_us@host219-81-static.38-79-b.business.telecomitalia.it) |
| 15:57.01 | dominic1 | special configuration: I have some users which have a few virtual numbers |
| 15:57.30 | dominic1 | and every number should have it's own devstate and not be associated with the device |
| 15:57.45 | dominic1 | I need a kind of virtual devstate |
| 15:58.17 | uTx | not sure if this is the place to ask, but I need someone to code a new fuction is asterisk |
| 15:59.00 | furibondox | hi, i've a problem looking report in freepbx: if i call an internal extension from another and the first is unavailable, no log is been tracked in freepbx... how can i log this type of call? |
| 15:59.20 | M1s3ry | furibondox, #freepbx |
| 15:59.42 | [TK]D-Fender | ~freepbx |
| 15:59.43 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:59.45 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 15:59.49 | M1s3ry | FYI you'll probably get better help there |
| 15:59.52 | [TK]D-Fender | dominic1: Ok... thats different.... |
| 16:00.40 | [TK]D-Fender | dominic1: to change the state to inuse, I'm use "M()" in your dial to the actual device to trigger the update |
| 16:00.46 | uTx | should I ask in dev? |
| 16:00.51 | [TK]D-Fender | M1s3ry: s/better/ant/ |
| 16:00.55 | [TK]D-Fender | any* |
| 16:01.10 | M1s3ry | lol |
| 16:01.20 | [TK]D-Fender | uTx: You could TRY there... but this may be "bounty" territory |
| 16:01.35 | dominic1 | but what's with the ringing and escpecially the ringing and Inuse state? |
| 16:01.45 | CVirus | is it possible to install asterisk on two servers and 1 FXO card on each and then my dial plan would be like if an incoming call goes to line no. 1 then an IVR menu is initialized and if the user press the # key he gets the dial tone of line no.2 which is on the other asterisk server ? |
| 16:01.52 | [TK]D-Fender | dominic1: What do you mean, "whats with"? |
| 16:02.05 | uTx | is there a channel for that |
| 16:02.06 | [TK]D-Fender | CVirus: Sure |
| 16:02.06 | uTx | ok |
| 16:02.07 | uTx | thanks |
| 16:02.29 | CVirus | [TK]D-Fender: what shall I read about to do so ? |
| 16:02.32 | M1s3ry | CVirus, you'll require more than just 1 fxo on each side though... unless you peer them with sip/iax |
| 16:02.33 | [TK]D-Fender | uTx: For bounties, check the WIKI |
| 16:02.41 | dominic1 | how can I set the state to ringing, If I use a macro in the dial command nobody can see if a extension has been picked up |
| 16:02.45 | *** join/#asterisk oron (n=oron@local.xorcom.com) |
| 16:02.53 | [TK]D-Fender | CVirus: Its all dialplan + whatever kind of channels you'll se tup between your boxes. |
| 16:03.33 | CVirus | thanks |
| 16:03.44 | [TK]D-Fender | dominic1: you seem to have missed it. that macro gets called if they ANSWER so you can change your state |
| 16:04.25 | dominic1 | cool thank you very much for that information |
| 16:05.00 | dominic1 | that will help me a lot with my problem! |
| 16:05.08 | [TK]D-Fender | dominic1: You're welcome. |
| 16:06.04 | *** part/#asterisk oron (n=oron@local.xorcom.com) |
| 16:06.35 | uTx | dev kicked me |
| 16:06.49 | [TK]D-Fender | uTx: lol |
| 16:06.52 | Qwell | no, they muted you |
| 16:06.57 | uTx | ahh |
| 16:06.57 | [TK]D-Fender | uTx: Must have earned it. |
| 16:07.08 | uTx | feels like a kick |
| 16:07.08 | [TK]D-Fender | uTx: So what do you need exactly? |
| 16:07.23 | uTx | I need the caller to know the state of his call |
| 16:07.24 | [TK]D-Fender | uTx: If you can still see, then you aren't gone yet :) |
| 16:07.33 | uTx | Ie ringing connect, so on |
| 16:07.55 | [TK]D-Fender | uTx: Please describe a complete scenario |
| 16:08.06 | uTx | ok I call some one |
| 16:08.07 | [TK]D-Fender | uTx: And HOW you are to be indicating things. |
| 16:08.10 | Qwell | and please use complete sentences |
| 16:08.16 | Qwell | grammar applies here as well |
| 16:08.17 | uTx | I need a IVR to say ringing |
| 16:08.17 | ManxPower | uTx: The only way that you MIGHT be able to do that is using AMI, but really, since nobody ever needs this feature, perhaps you are confused. |
| 16:08.18 | [TK]D-Fender | uTx: DETAILS <--- |
| 16:08.37 | ManxPower | uTx: you cannot get those sorts of states in the dialplan |
| 16:08.48 | [TK]D-Fender | ManxPower: this scenarios has more holes than a brick of swiss chees... I'd wait on it.. |
| 16:08.54 | ManxPower | The dialplan will BLOCK until the call ends. |
| 16:09.07 | ManxPower | [TK]D-Fender: oh, I'm not waiting around to help him in this foolish task. |
| 16:09.13 | uTx | when the user answers it has to say answered |
| 16:09.21 | [TK]D-Fender | ManxPower: I meant for him to clarify... |
| 16:09.29 | ManxPower | uTx: Asterisk may not be the solution for you. |
| 16:09.35 | [TK]D-Fender | uTx: start over, your process description is really weak. |
| 16:09.45 | uTx | I know |
| 16:09.50 | uTx | let me try again |
| 16:10.09 | Qwell | Enter is NOT punctuation. Please don't hit enter after every word. |
| 16:10.12 | ManxPower | uTx: maybe you can come back when you know what you want to do, rather than just having a vague idea of what you want to do. |
| 16:10.21 | uTx | you know in the console you see call progress |
| 16:10.31 | uTx | and then call bridged when connected |
| 16:10.32 | *** part/#asterisk ManxPower (n=manxpowe@241.sub-70-222-228.myvzw.com) |
| 16:10.48 | uTx | I need the caller to know all this by way of IVR messages played only to him |
| 16:11.02 | [TK]D-Fender | uTx: Sorry, still meaningless. |
| 16:11.21 | uTx | so he would be calling someone, and he would he a ring plus hear a ivr message saying ringing |
| 16:11.22 | [TK]D-Fender | uTx: And stop calling "audio" you're playing back as "IVR messages" |
| 16:11.31 | [TK]D-Fender | uTx: IVR means he's INPUTTING something. |
| 16:11.58 | [TK]D-Fender | uTx: "core show application dial" <- Go read the instructions. |
| 16:11.59 | uTx | no just a sound file played back, when rining it says ringing |
| 16:12.14 | [TK]D-Fender | uTx: Like the tone isn't obvious enough? |
| 16:12.25 | uTx | and when the call gets connect a audio file says call connected |
| 16:12.40 | [TK]D-Fender | uTx: go read the instructions to DIAL. |
| 16:12.46 | uTx | I did |
| 16:12.54 | uTx | this is not possible |
| 16:13.00 | uTx | you can see it in the console |
| 16:13.09 | [TK]D-Fender | uTx: Sure it is. use a MoH class that plays that recording. |
| 16:13.11 | uTx | like making progress |
| 16:13.33 | [TK]D-Fender | uTx: and use "M()" or something to play the "answered" bit. |
| 16:13.42 | Qwell | M is called |
| 16:13.53 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:14.00 | uTx | ok and for ringing bit |
| 16:14.16 | uTx | I need to catch FAS |
| 16:14.18 | uTx | and no ring |
| 16:15.03 | uTx | I do it now by having the console open |
| 16:15.03 | uTx | but a audio message is better |
| 16:15.21 | uTx | no need to open the console to see if a call should be ringing |
| 16:15.26 | uTx | and if I got the connect |
| 16:16.00 | [TK]D-Fender | uTx: And why do you need it to say this to you? Football games don't ahve this kind of play-by-play... |
| 16:16.31 | [TK]D-Fender | uTx: You know its ringing because you hear it ringing. you know its answered because it STOPS ringing... |
| 16:16.49 | uTx | how about if your calling kenya |
| 16:16.50 | nr4q | should I get a discount for having 2 BRI's over the cost of a single BRI ? |
| 16:16.58 | uTx | and the provide is not sending a ring |
| 16:17.04 | uTx | when they should be |
| 16:17.05 | nr4q | or would tha tbe a fractional |
| 16:17.05 | Qwell | nr4q: probably not |
| 16:17.18 | uTx | or there is FAS |
| 16:17.23 | uTx | it is for testing routes |
| 16:17.24 | [TK]D-Fender | uTx: If * knows, then YOU can force ringing. |
| 16:17.31 | Qwell | I doubt you could get a fractional T1 with just 4 channels |
| 16:17.38 | Qwell | 8-12 is more likely |
| 16:18.01 | uTx | I need to know what the farend is doing |
| 16:18.07 | uTx | play by play |
| 16:18.17 | uTx | like a debug in audio form |
| 16:18.27 | Qwell | but.. |
| 16:18.39 | Qwell | if...Asterisk doesn't get any indication of what the far end is doing |
| 16:18.40 | [TK]D-Fender | uTx: * either thinks the line is ringing, or has been answered, thats it. |
| 16:18.51 | Qwell | how...is Asterisk going to know what to play? |
| 16:19.09 | uTx | in the console I see making progress |
| 16:19.19 | uTx | which = ringing |
| 16:19.20 | Qwell | then you're getting indications from the far end |
| 16:19.22 | uTx | normally |
| 16:19.29 | uTx | and call briged |
| 16:19.34 | uTx | = call is answered |
| 16:19.41 | [TK]D-Fender | uTx: Again, you can do this with M() |
| 16:19.47 | uTx | ok I know |
| 16:19.53 | uTx | for the answered part |
| 16:19.58 | uTx | how about the ringing |
| 16:20.07 | uTx | making progress part |
| 16:20.14 | [TK]D-Fender | uTx: and force ringing announcement via MoH class while "ringing" |
| 16:20.29 | uTx | I don't want to force |
| 16:20.38 | uTx | I want to see what the carrier is giving me |
| 16:20.42 | *** join/#asterisk ManxPower (n=manxpowe@114.sub-70-222-203.myvzw.com) |
| 16:20.56 | [TK]D-Fender | uTx: You don't seem to understand... if you get "making progress", then you WILL be ringing. Its going to come next no matter what. |
| 16:21.14 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:21.14 | lmadsen | [TK]D-Fender: well, unless the world ends of course |
| 16:21.15 | uTx | yes |
| 16:21.26 | uTx | but I want a audio saying so |
| 16:21.49 | [TK]D-Fender | uTx: Its like announcing "apple pushed off the table", instead of "aspple falling". "Apple falling" already knows its been pushed and is pointless. It will also "fall" when pushed". Anything more is pedantic and worthless |
| 16:22.33 | uTx | ok I see |
| 16:22.48 | uTx | but some times you get no ring |
| 16:22.50 | [TK]D-Fender | uTx: And this isn't just a "module". You will need a complete recoding of the dial mechanism. |
| 16:22.59 | uTx | and it's hard to know the PDD of the call |
| 16:23.11 | [TK]D-Fender | uTx: PDD? |
| 16:23.22 | uTx | post dial delay |
| 16:23.40 | [TK]D-Fender | uTx: What are you dialing over, btw? |
| 16:23.57 | ManxPower | uTx: not hard at all. You know the PDD if you know the tech used to dial the call |
| 16:24.10 | x86 | gah |
| 16:24.12 | [TK]D-Fender | uTx: Something tells me that due to zone indications, audio progress is impossible, so * won't give you anything anyways if the zone is foreign |
| 16:24.21 | ManxPower | PPD on analog will be .3 - .5 (depending on how you configure it) per DTMF digit. |
| 16:24.33 | uTx | how about if your calling kenya |
| 16:24.35 | Qwell | PPD? |
| 16:24.37 | ManxPower | PPD on non-analog is instant. |
| 16:24.39 | uTx | and the ping is 845 ms |
| 16:24.41 | x86 | I keep getting random dropped calls on SIP phones --> asterisk --> channel bank --> analog phones |
| 16:24.51 | Qwell | oh, nevermind |
| 16:25.00 | ManxPower | uTx: that is .845 seconds, is it really that important. |
| 16:25.09 | ManxPower | x86: turn off busydetect and callprogress. |
| 16:25.38 | [TK]D-Fender | ManxPower: thats FXS, not FXO |
| 16:25.45 | uTx | It is an interesting project |
| 16:25.51 | uTx | any takers |
| 16:25.57 | x86 | ManxPower: yeah but if I kill busydetect, I'll get voicemails with dialtone or reorder tone whenever someone calls me and hangs up when they get my voicemail greeting |
| 16:26.00 | ManxPower | no, FXO port |
| 16:26.09 | ManxPower | x86: it sucks to be you. |
| 16:26.13 | [TK]D-Fender | uTx: You'll have to pay serious bucks to a consultant for a job like that. |
| 16:26.26 | uTx | can you do it |
| 16:26.31 | x86 | ManxPower: ... thanks for the help ... |
| 16:26.35 | uTx | we can discuss outside the channel |
| 16:26.43 | [TK]D-Fender | ManxPower: he is not using FXO, there is no "call progress" failure potential for that. |
| 16:26.44 | jeev | got IBM corporate to offer $200 discount on another system since they ruined my order after a week of fighting |
| 16:27.16 | x86 | [TK]D-Fender: right |
| 16:27.18 | ManxPower | [TK]D-Fender: I never said he's using FXO or FXS. I simply said you know the PDD on an FXO. |
| 16:27.47 | ManxPower | there is no PPD on FXS normally |
| 16:27.53 | ManxPower | sotty PDD |
| 16:28.19 | [TK]D-Fender | ManxPower: I was answering about x86's problem... you are fragmenting again. Go caffeinate! |
| 16:28.38 | ManxPower | [TK]D-Fender: there is no solution to x86's problem |
| 16:28.47 | x86 | :( |
| 16:28.59 | [TK]D-Fender | x86: Go max out your debug and show us something. |
| 16:29.02 | ManxPower | if you enable callprogress or busydetect you WILL have random disconnections. |
| 16:29.29 | ManxPower | You can increase busy count to reduce the random disconnects, but it won't eliminate them and it comes at a cost of a longer time takend to detect hangup. |
| 16:29.44 | x86 | ManxPower: I don't have callprogress set on any spans, although busydetect was set one a couple... i just disabled it |
| 16:30.06 | x86 | hmm |
| 16:30.19 | x86 | I wonder if these channelbanks support disconnect supervision |
| 16:30.24 | ManxPower | goes off to buy a Sawsall |
| 16:30.43 | defswork | x86: without it analog it's a mare |
| 16:31.08 | [TK]D-Fender | x86: Doesn't work that way on FXS... |
| 16:31.48 | [TK]D-Fender | x86: TELEPHONES don't CUT the circuit or reverse polarity because telephones to not GENERATE any. |
| 16:32.21 | [TK]D-Fender | x86: your CB can TELL your phone that IT has given up, but then that'd be at *'s request. |
| 16:32.33 | x86 | hmm |
| 16:32.40 | [TK]D-Fender | x86: And the only reason to do that is * had a normal reason to stop a call... like the SIP phone hanging up |
| 16:32.47 | ManxPower | busydetect for FXS is just plain stupid |
| 16:33.02 | x86 | hmm |
| 16:33.31 | ManxPower | x86: what is the signalling set to for the channels going to the CB? |
| 16:33.35 | [TK]D-Fender | ManxPower: More like "apples & oranges". How many MPG do you get on a parachute? |
| 16:34.06 | d-tech | depends on how big the bananas are? |
| 16:34.08 | x86 | ManxPower: kewlstart |
| 16:34.19 | ManxPower | fxs_ks or fxo_ks? |
| 16:34.56 | x86 | fxo, naturally, since they are fxs ports |
| 16:35.38 | ManxPower | x86: yes, but some people don't know that. |
| 16:36.10 | x86 | would it even work if the channel bank did FXO signalling and I was doing FXS? heh |
| 16:36.14 | ManxPower | and since you have been here for YEARS and still tried to use busydetect -- I assume you are still not very familiar with Asterisk even after years of being here. |
| 16:36.20 | x86 | didn't even think that would work at all |
| 16:36.44 | adr3nalin3 | could someone please help me setup a dial plan with FXO zap channels? I have been asterisk-gui spoon fed in the past but since gui won't work with my hardware I need to put down the mouse. thanks. |
| 16:36.45 | x86 | ManxPower: actually dude, you told me to use busydetect to solve my problem with dialtone voicemails |
| 16:36.56 | x86 | about a year ago |
| 16:37.04 | Qwell | I doubt it was ManxPower |
| 16:37.15 | [TK]D-Fender | adr3nalin3: ... |
| 16:37.17 | [TK]D-Fender | ~book |
| 16:37.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:37.20 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 16:37.22 | x86 | I'm 97.5% sure it was manx |
| 16:37.27 | elguero | x86: What CB are you using? |
| 16:37.34 | x86 | elguero: Rhino CB24 |
| 16:37.56 | elguero | x86: I have a Rhino and found in the docs that it should be fxo_ls |
| 16:37.59 | x86 | that particular one hasn't been swapped out for a decent channel bank yet... (read Adit 600) |
| 16:38.14 | x86 | elguero: ls and ks are practically the same |
| 16:38.20 | x86 | elguero: it's not a signalling issue |
| 16:38.43 | elguero | x86: I had been using kewlstart as well and then one day I found that out.... okay... I was just offering a suggestion... it eliminated a lot of messages in the logs for me |
| 16:38.52 | adr3nalin3 | [TK]D-Fender: I have it in my hand, maybe I'm just thick but I really didn't understand it |
| 16:38.54 | [TK]D-Fender | x86:signalling=fxols <---- |
| 16:39.05 | [TK]D-Fender | adr3nalin3: What part isn't working? |
| 16:39.21 | adr3nalin3 | incoming, outgoing calls |
| 16:39.33 | adr3nalin3 | [TK]D-Fender: I'm gonna read through it again |
| 16:39.39 | [TK]D-Fender | adr3nalin3: pastebin what your dialplan, and the CLI output of the failed attempt |
| 16:40.21 | x86 | elguero: I hear what you're saying, and I appreciate your effort, but I've tried ls before with the same results, and I'm not getting any messages in the logs other than normal clearing messages |
| 16:40.31 | NovceGuru | holy hell the polycom 2x firmware takes 10 minutes to boot |
| 16:40.36 | elguero | x86: okay... no problem |
| 16:41.52 | [TK]D-Fender | NovceGuru: You're doing something horribly wrong then. |
| 16:42.02 | [TK]D-Fender | NovceGuru: 2.2.0 takes < 2 minutes for my IP 501 |
| 16:42.20 | [TK]D-Fender | NovceGuru: Did you just upgrade keepiong old configs? |
| 16:43.22 | NovceGuru | [TK]D-Fender: I've reset the 2 options in the reset menu except for format filesystem (from what I read that'll screw you) |
| 16:43.40 | [TK]D-Fender | NovceGuru: Nope. |
| 16:43.48 | nr4q | anyone know if asterisk supports polycom hd voice? |
| 16:43.59 | [TK]D-Fender | NovceGuru: You should only upgrade your SIP btw, not BR for 99% of cases |
| 16:44.07 | [TK]D-Fender | nr4q: 1.6 does. |
| 16:44.08 | NovceGuru | nr4q: it's just a codec, but i've been wondering my self |
| 16:44.11 | *** join/#asterisk Segnale007 (n=Segnale0@host59-121-dynamic.182-80-r.retail.telecomitalia.it) |
| 16:44.14 | [TK]D-Fender | nr4q: 1.4 in passthrough only |
| 16:44.15 | nr4q | my googlefu is not very strong |
| 16:44.19 | nr4q | fender thanks |
| 16:44.26 | M1s3ry | we have it working... it sounds nice |
| 16:44.49 | Qwell | NovceGuru: it's *amazing* |
| 16:44.50 | NovceGuru | [TK]D-Fender: I just let a trixbox do the upgrade for me when I was toying with it, I've since moved away from trixbox |
| 16:45.07 | Qwell | even in ulaw, the 650s sound *great* |
| 16:45.28 | NovceGuru | Qwell: yeah I heard it at a demo, I almost didn't believe it |
| 16:45.31 | Qwell | a good deal of it is just from the vastly superior audio hardware (speake, mic) that was used |
| 16:45.34 | NovceGuru | I still don't since I haven't got mine setup yet |
| 16:45.49 | NovceGuru | yeah I figured a lot of it was just a nice speaker |
| 16:45.59 | [TK]D-Fender | NovceGuru: Guess that means you have no clue how they're set up... time to go downlaod the admin guides & firmware and actually LEARN something. |
| 16:46.11 | phpboy | how would I go about getting an agent id in my dialplan? |
| 16:46.11 | Qwell | of course, you can absolutely tell the difference when you're using g722. |
| 16:46.13 | NovceGuru | nice hardware + a wideband codec probably = amazningness |
| 16:46.27 | phpboy | provided the agent is in fact logged on |
| 16:46.29 | [TK]D-Fender | phpboy: what does "getting an agentID" mean? |
| 16:46.40 | NovceGuru | [TK]D-Fender: yeah I just got it yesterday, trying to get away from ciscos |
| 16:46.54 | NovceGuru | although I could probably upgrade those in my sleep now *quivvers* |
| 16:47.23 | *** join/#asterisk nortex (n=chatzill@96.226.10.187) |
| 16:47.55 | M1s3ry | plans on disassembling a few 650's and using the speakers for his car... |
| 16:48.37 | Strom_M | M1s3ry: although polycom and pioneer both start with a 'p'.... |
| 16:48.50 | NovceGuru | M1s3ry: seems expensiv e:P |
| 16:49.44 | M1s3ry | NovceGuru, but who else has a car that is G.722 capable? |
| 16:50.10 | [TK]D-Fender | M1s3ry: ..... it isn't the SPEAKER that is G.722 capable... |
| 16:50.26 | [TK]D-Fender | SMRT |
| 16:50.36 | M1s3ry | understood... however I nvr said I'd just use the speakers |
| 16:50.51 | M1s3ry | wait |
| 16:50.52 | M1s3ry | yes i did |
| 16:50.58 | [TK]D-Fender | M1s3ry: Then you've said nothing at all :) |
| 16:51.03 | M1s3ry | sighs |
| 16:51.14 | nortex | Hello all! I have a simple question, I have a sip soft phone that after it hung up the sip channel failed to drop, any idea first how to force it to drop? This Asterisk 1.4.18 if that helps. |
| 16:51.14 | NovceGuru | haha |
| 16:51.50 | nr4q | yikes... BRI digium card seems pricey |
| 16:53.58 | Qwell | nr4q: looks fairly low on telephonydepot.com |
| 16:54.10 | Qwell | do remember, it is, of course, a quad port card |
| 16:54.36 | M1s3ry | nr4q, digium's resellers can get you a deal as well |
| 16:55.27 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 16:55.30 | [TK]D-Fender | nortex: "soft hangup [channel]" |
| 16:56.20 | nortex | [TK]D-Fender: Tried that, but the channel is actually marked to destroy, but won't until we restart asterisk |
| 16:57.00 | [TK]D-Fender | nortex: What's the consequence? |
| 16:57.02 | _Sam-- | hey what do i need to make my followme work? http://www.pastebin.ca/1065680 |
| 16:57.04 | x86 | I've seen that happen before, but that was way back in like 1.2.9 |
| 16:57.10 | nortex | [TK]D-Fender: Here is the sip show channels information. 192.168.2.115 2107 6e2a51c7179 00102/00000 0x0 (nothing) No (d) Tx: ACK |
| 16:57.55 | [TK]D-Fender | nortex: And in "show channels"? |
| 16:58.02 | nr4q | qwell: guess it's cheaper than buying a channel bank and t1 card too |
| 16:58.14 | nortex | [TK]D-Fender: Consequence? Of restarting asterisk, droping active calls in a call center :) |
| 16:58.36 | [TK]D-Fender | nortex: No, of that channels itting for a while. |
| 16:58.49 | nortex | [TK]D-Fender: The call actually does not appear in show channels. |
| 16:59.06 | [TK]D-Fender | nortex: Sounds like you can afely ignore it... |
| 16:59.08 | nortex | [TK]D-Fender: Oh, well it keeps a call center agent from getting more calls. |
| 16:59.30 | [TK]D-Fender | nortex: the restart the phone and let it re--register. |
| 16:59.42 | [TK]D-Fender | nortex: firewall off the IP, etc.. |
| 17:00.30 | nortex | [TK]D-Fender: The queue sees the sip device in use. Tried that, the only work around at this point is to force the ip address to change. When the phone re-registers it comes right back in to the channel that is not destroyed. |
| 17:00.55 | x86 | nortex: what version of asterisk, out of curiosity |
| 17:00.56 | *** join/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 17:01.00 | nortex | [TK]D-Fender: THe firewall ] |
| 17:01.01 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 17:01.20 | nortex | [TK]D-Fender: The firewall may work. |
| 17:01.30 | nortex | x86: 1.4.18 |
| 17:01.35 | [TK]D-Fender | nortex: Go try it. |
| 17:02.26 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 17:02.37 | x86 | nortex: interesting... I had the same problem with sip channels and 1.2.9 (and perhaps 1.2.13?) way back in the day, went away as soon as I upgraded to 1.4 |
| 17:03.03 | x86 | nortex: might try upgrading asterisk / zaptel / libpri (if you use it), and see if that helps... 1.4.21.1 is out |
| 17:03.30 | [TK]D-Fender | nortex: "restart when convenient" and sleep on it :) |
| 17:03.38 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 17:04.05 | nortex | x86: I have wondered about that, this thing has a call center solution on it so they have certain "custom" changes in the code preventing us from running a newer version. |
| 17:04.24 | nortex | [TK]D-Fender: I wish it would just go away :) |
| 17:04.30 | x86 | nortex: ah, so call your vendor and bitch, don't come here ;) |
| 17:04.31 | [TK]D-Fender | nortex: It can! |
| 17:04.40 | ipstatic | Could someone help me and tell me why my AEL dialplan is not compiling? http://pastebin.com/d1176d482 |
| 17:05.17 | *** join/#asterisk railsmunky (n=nick@5acd4099.bb.sky.com) |
| 17:05.20 | nortex | x86 I'm in the middle, the client calls me and the vendor. |
| 17:05.37 | nortex | XI jsut have my doubts they can fix it. |
| 17:06.08 | x86 | then rip it out an put a new solution in ;) |
| 17:06.20 | x86 | nortex: are you an inbound call center or outbound? |
| 17:06.28 | x86 | I'm an outbound call center |
| 17:06.44 | nortex | x86: so close to doing that, it is an inbound center with plans to start doing dialer campaigns |
| 17:06.53 | x86 | running asterisk at 7 different offices with about 200 users... 90% analog and 10% SIP |
| 17:06.58 | [TK]D-Fender | ipstatic: Set(CURRENT_QUEUE=CUT(${queues},\,,${z})); <-- |
| 17:06.58 | j0 | [TK]D-Fender: may i buy you a beer for all the help you've given me and this channel? |
| 17:07.10 | [TK]D-Fender | ipstatic: Set(CURRENT_QUEUE=${CUT(${queues},\,,${z})}); |
| 17:07.19 | [TK]D-Fender | ipstatic: Don't forget to REFERENCE your function calls |
| 17:07.43 | [TK]D-Fender | ipstatic: Nothing says it quite like PayPal :) |
| 17:07.56 | j0 | [TK]D-Fender: pm me your e-mail |
| 17:08.06 | j0 | and no, i don't expect any more support from you :) |
| 17:08.44 | *** join/#asterisk hi365_m (n=hi365@213.151.61.31) |
| 17:09.39 | [TK]D-Fender | ipstatic: Look at the difference between the 2 lines... |
| 17:09.52 | ipstatic | yeah I see now |
| 17:10.09 | [TK]D-Fender | ipstatic: the GET the value of a function you have to put ${} around it,. |
| 17:10.18 | nortex | x86: Do you use or recommend a dialer. |
| 17:10.33 | [TK]D-Fender | ipstatic: the only time you don't is when you're setting it, at which point its on the left side of the "=" |
| 17:10.47 | [TK]D-Fender | nortex: Polycom phones are great at dialing! |
| 17:11.19 | nortex | [TK]D-Fender: Thanks smart ass :) |
| 17:11.27 | [TK]D-Fender | nortex: All part of the service :p |
| 17:12.25 | j0 | [TK]D-Fender: have lunch on me too. :) |
| 17:12.30 | x86 | nortex: don't use one, but there are a few free ones out there |
| 17:12.33 | ipstatic | [TK]D-Fender: I am still getting the same errors when running aelparse |
| 17:12.43 | [TK]D-Fender | ipstatic: new pastebin please... |
| 17:13.29 | [TK]D-Fender | j0: Thanks :) |
| 17:14.46 | j0 | np. cheers! |
| 17:15.37 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 17:16.06 | Mike8861 | hello everyone |
| 17:16.20 | ipstatic | [TK]D-Fender: http://pastebin.com/d31256f22 |
| 17:16.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:17.11 | [TK]D-Fender | ipstatic: While(${z} < FIELDQTY(${queues},\,)) <- here again you are referencing a function without ${} |
| 17:18.06 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 17:18.12 | Mike8861 | does LinuxMCE's asterisk works like native asterisk ? |
| 17:18.25 | Mike8861 | or it is simlar to trixbox ? |
| 17:19.12 | [TK]D-Fender | Mike8861: Why don't you go look at their decription. |
| 17:19.29 | [TK]D-Fender | Mike8861: and * is jsut *. Whether its with more crap is besides the point. |
| 17:19.42 | Mike8861 | i have read: it uses Ubuntu's package of asterisk |
| 17:20.08 | Mike8861 | not understand what does that mean.....does that mean it is problematic ? |
| 17:20.18 | NovceGuru | Mike8861: im sure its config will be wiped if you do anything by hand, if thats what you want to hear |
| 17:20.36 | [TK]D-Fender | Mike8861: it means its just a precompiled version. |
| 17:20.38 | Mike8861 | NovceGuru: thank you very much |
| 17:20.50 | [TK]D-Fender | NovceGuru: No reason to claim that... |
| 17:20.56 | *** join/#asterisk erreur404 (n=erreur40@tri59-1-87-88-208-131.dsl.club-internet.fr) |
| 17:20.57 | Mike8861 | [TK]D-Fender: thanks, got it. |
| 17:21.14 | NovceGuru | that pretty much goes for anything that uses freepbx/web front end? (I'm not pro, just my experience) |
| 17:21.29 | [TK]D-Fender | NovceGuru: I don't see any implication of it coming with FreePBX yet.. |
| 17:21.38 | Qwell | LinuxMCE? |
| 17:21.45 | [TK]D-Fender | NovceGuru: Do you see something I don't? Care to link me? |
| 17:21.57 | [TK]D-Fender | Qwell: KuBuntu offshoot |
| 17:22.08 | [TK]D-Fender | Qwell: Apparently. |
| 17:22.10 | NovceGuru | [TK]D-Fender: Seemed years ago I played with FreePBX and I made some hand changes and it wiped them after I updated something in the GUI |
| 17:22.13 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 17:22.29 | NovceGuru | could very well be wrong, just my general experience with any gui frontend |
| 17:22.33 | [TK]D-Fender | NovceGuru: Yes, and you keep throwing "FreePBX" into this conversation. Who said LinuxMCE CAME WITH IT? |
| 17:23.41 | Mike8861 | [TK]D-Fender: LinuxMCE looks like some kind of central management more then a Media Center |
| 17:23.42 | nortex | NovceGuru: Some GUI's don't. |
| 17:24.26 | Mike8861 | [TK]D-Fender: it even take control of TV, lights, VCR, security cams |
| 17:24.26 | Mike8861 | and VOIP as well |
| 17:24.29 | [TK]D-Fender | Mike8861: ok,fine,sure... and the reason you are looking at it at all is...? |
| 17:24.35 | NovceGuru | [TK]D-Fender: oh, I never said it did |
| 17:24.56 | NovceGuru | just my general experience with GUIs, but /clear and forget I said it! :) |
| 17:24.59 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 17:25.03 | CVirus | Isn't there anyway that I could change the IRQ numbers of certain devices from the OS not from the BIOS as my BIOS doesn't support that |
| 17:25.37 | [TK]D-Fender | CVirus: FUBAR <--- |
| 17:25.49 | Mike8861 | i cannot sleep without worry with that LinuxMCE take control of my home = = |
| 17:26.04 | NovceGuru | apprently linuxmce does use it though wiki.linuxmce.org/index.php/Asterisk-LinuxMCE |
| 17:26.10 | [TK]D-Fender | CVirus: Go do something productive with your time. Does your boxx only pay you for successes? If not he's wasted the money better spent on a card that will WORK <- |
| 17:27.06 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:27.10 | [TK]D-Fender | NovceGuru: Asterisk 1.2.7.1 <-- cutting edge! |
| 17:27.24 | NovceGuru | BLEEDING |
| 17:27.50 | [TK]D-Fender | NovceGuru: No, thats the edge.... HE'LL bleed ;) |
| 17:28.06 | hsv-al | d-fender |
| 17:28.10 | [TK]D-Fender | ~emo |
| 17:28.10 | jbot | /wrists |
| 17:28.12 | hsv-al | updated date: September 2009 |
| 17:28.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:28.13 | [TK]D-Fender | :D |
| 17:28.31 | [TK]D-Fender | hsv-al: Ah, they got chan_fluxcapacitor.so! |
| 17:28.35 | ipstatic | [TK]D-Fender: I am still getting errors: http://pastebin.com/d4a261ccd |
| 17:28.35 | Mike8861 | maybe can complie 1.6 on linuxmce |
| 17:28.42 | hsv-al | Diablo3's not coming out |
| 17:28.45 | hsv-al | for 1.5 years argh! |
| 17:28.47 | [TK]D-Fender | Mike8861: Sure you can. |
| 17:29.23 | Mike8861 | hsv-al: no anymore |
| 17:29.25 | NovceGuru | I can't imagine the upkeep on the linuxmce project |
| 17:29.38 | NovceGuru | unless they have more developers then when it was first released, which it probably does |
| 17:29.46 | hsv-al | mike? |
| 17:29.48 | hsv-al | date got moved up? |
| 17:30.05 | hsv-al | 2010? |
| 17:30.34 | Mike8861 | hsv-al: http://www.blizzard.com/diablo3/ |
| 17:30.57 | Mike8861 | hsv-al: Diablo 2 has just got updated |
| 17:31.31 | [TK]D-Fender | hsv-al: Time to rock out on DNF! |
| 17:31.45 | hsv-al | dude, dont get happy d-fender |
| 17:31.51 | hsv-al | DNF delayed what? 8 years? |
| 17:31.55 | hsv-al | that game is a fantasy |
| 17:32.01 | hsv-al | to put 3d realms into public conciousness, marketing tactics |
| 17:32.07 | Mike8861 | hsv-al: maybe blizzard will remain MOOMOO in Diablo3 |
| 17:32.11 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:32.16 | Mike8861 | cows are cute! |
| 17:32.24 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 17:32.28 | hsv-al | wirts leg+asterisk 1.4 book + horadric cube |
| 17:33.02 | Mike8861 | 0.0!!! |
| 17:35.18 | Mike8861 | its raining outside |
| 17:36.06 | Mike8861 | anyone have a TAPI SIP client that works on windows xp ? |
| 17:36.17 | Mike8861 | cannot find one that works. |
| 17:36.35 | *** join/#asterisk metfan2007 (n=jc@201.103.114.140) |
| 17:37.27 | NovceGuru | astapi? |
| 17:37.54 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com) |
| 17:38.03 | Mike8861 | NovceGuru: thanks, i will install it now! |
| 17:38.09 | metfan2007 | Hi all!! I just upgraded an Asterisk box from 1.2 to 1.4. I checked all the UPGRADE.txt file, but I'm receiving "[Jul 8 12:34:19] WARNING[7536]: app_queue.c:3002 try_calling: The device state of this queue member, Agent/8279, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings" messages every call... any idea? |
| 17:39.11 | putnopvut | metfan2007: are you using SIP queue members? |
| 17:41.29 | metfan2007 | putnopvut: No, there are a mix of remote IAX trunks and Zap channels |
| 17:42.29 | metfan2007 | putnopvut: in my queues.conf I have only Agetn/XXX entries |
| 17:42.47 | putnopvut | metfan2007: ahh. Okay, that's actually a problem that has been reported on the bugtracker. |
| 17:42.53 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:43.05 | putnopvut | It's not a major issue, but there's a small race condition between the device state thread and the queue application. |
| 17:43.20 | putnopvut | Let me find the bug number. |
| 17:43.34 | metfan2007 | putnopvut: Oh, ok ok |
| 17:44.00 | putnopvut | metfan2007: http://bugs.digium.com/view.php?id=12771 |
| 17:44.45 | hsv-al | Mark: |
| 17:44.48 | hsv-al | http://www.apple.com/retail/bridgestreet/ |
| 17:44.59 | hsv-al | dont know if your a apple fanboi, iphone 2 8am friday :) |
| 17:45.14 | putnopvut | Ah, I'm not much of an apple fanboi, but thanks :) |
| 17:45.19 | Qwell | hsv-al: openmoko.org |
| 17:45.26 | Qwell | that > iPhone |
| 17:45.37 | hsv-al | fully working sip client? :) |
| 17:45.40 | hsv-al | y/n? |
| 17:45.44 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 17:46.15 | hsv-al | there's a full blown sip client coming out for 3G iphone 2 |
| 17:46.28 | hsv-al | im just getting data service on it, no phone service |
| 17:47.03 | jpcansa | is there a way to set up a sip to an HP IpaqVoiceMessenger 510 ?? |
| 17:47.12 | hsv-al | qwell , never heard of it, but interesting read |
| 17:47.37 | Mike8861 | http://www.fring.com/iphone |
| 17:48.09 | Mike8861 | hsv-al: u can try Fring |
| 17:48.57 | hsv-al | well, they really opened up development api's to the iphone 2 |
| 17:49.07 | hsv-al | lots of community developed stuff is going to come out |
| 17:49.14 | Qwell | can it do more than one thread? |
| 17:49.24 | hsv-al | didnt really look deeply at it, but from what i read |
| 17:49.26 | hsv-al | i wouldnt be surprised |
| 17:52.22 | *** join/#asterisk ta^3 (n=tacvbo@189.136.42.104) |
| 17:53.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:53.37 | *** join/#asterisk redax (i=redax@82.141.129.7) |
| 17:53.40 | redax | hi, |
| 17:55.11 | redax | is it possible to have a SIP phone possible to display the called party's name with asterisk? |
| 17:55.47 | Datax | yes |
| 17:55.51 | redax | not for incoming call, but the outgoing... |
| 17:56.07 | Datax | ah |
| 17:56.17 | adr3nalin3 | [TK]D-Fender: I got incoming calls working ok I am just having problems with outgoing calls would you still be willing to take a look at my extensions.conf? |
| 17:56.29 | Datax | you want person A to see person B's name when they dial person B's number ? |
| 17:56.40 | redax | yep. |
| 17:57.07 | Datax | mmhh, never tried, I know it works with Cisco callmanager but you probably don't care much for CCM :op |
| 17:57.25 | ACiDV | redax check bugs 8824 ... Remote (called) Party Identification |
| 17:58.24 | *** join/#asterisk JCJC (n=JCJC@netblock-72-25-115-165.dslextreme.com) |
| 18:00.10 | redax | ACiDV: wow.. |
| 18:06.13 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 18:06.14 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:07.32 | adr3nalin3 | Could someone please help me with outbound calling via zap channels? Everytime I try to make a call I get Call from '1211' to extension '92238100' rejected because extension not found. |
| 18:07.39 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:07.44 | [TK]D-Fender | adr3nalin3: pastebin the CLI output of your failed attempt and the relevant setiocn of your dialplan |
| 18:08.00 | [TK]D-Fender | adr3nalin3: You clearly don't ahve an exten in the proper context to match that # dialed |
| 18:08.21 | [TK]D-Fender | adr3nalin3: pastebin the failed attempt at verbose 10, SIP debug enabled, along with your dialplan. |
| 18:08.48 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 18:11.35 | [TK]D-Fender | ~cpid |
| 18:11.36 | jbot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 18:11.52 | *** join/#asterisk serialthrilla (n=noemail@adsl-71-131-145-38.dsl.sntc01.pacbell.net) |
| 18:13.03 | [TK]D-Fender | Qwell: Not sure how to read Mantis on this one... is this to be merged into the 1.6 RELEASE? |
| 18:13.11 | Qwell | what? |
| 18:13.33 | Qwell | oh |
| 18:13.36 | Qwell | sure, eventually |
| 18:13.46 | Qwell | all new features eventually end up in 1.6.x |
| 18:15.29 | etfonhomey | [TK]D-Fender, do you know of a way in the Polycoms to keep the handsfree volume persistant through a reboot of the phone? There's a sip.cfg option to do it between calls, but the volume is always restored to default after a reboot. |
| 18:16.10 | adr3nalin3 | [TK]D-Fender: here is the sip debug and I am getting the dialplan: http://pastebin.com/m144fbfa0 |
| 18:16.15 | [TK]D-Fender | etfonhomey: it should stick on the phone local settings |
| 18:16.49 | [TK]D-Fender | etfonhomey: Either way... WTH are you rebooting Polycoms? Do them right the first time! |
| 18:16.54 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:17.14 | hsv-al | lol, this email squeaked through spamasassin |
| 18:17.19 | etfonhomey | [TK]D-Fender, actually, I just found this: http://tinyurl.com/5c6jg8 |
| 18:17.24 | hsv-al | this is the icing on the cake |
| 18:17.36 | hsv-al | and the url doesnt even work, (SANDEEP PEHMDAYPEEZ) - C14l15 Tabs . . . . now $39.99 only exclusively through http://www.cialis-pharma-india-ceuticials.com $39.99. Don't think twice before you go. (SANDEEP PEHMDAYPEEZ) |
| 18:17.39 | adr3nalin3 | [TK]D-Fender: Here is the dialplan --> http://pastebin.com/m72ef2558 |
| 18:18.19 | [TK]D-Fender | adr3nalin3: Looking for 92238100 in default (domain 192.168.150.249) SIP/2.0 404 Not Found <- So... what line in your dialplan (pastebined) do you think is supposed to match that? |
| 18:19.04 | adr3nalin3 | [TK]D-Fender: I am trying to dial out a trunk zap line |
| 18:19.12 | *** join/#asterisk Xamusk (n=Xamusk@189.1.136.223) |
| 18:19.21 | [TK]D-Fender | adr3nalin3: so far your entire [default] context has NOTHING in it. |
| 18:19.36 | Xamusk | so, to use asterisk as an ATA would I need an FXS or an FXO board? |
| 18:19.37 | [TK]D-Fender | adr3nalin3: Look athe the context your SIP peer is using and realize that its empty. |
| 18:19.46 | [TK]D-Fender | Xamusk: * is not an ATA |
| 18:19.58 | Xamusk | but can't it be used like one? |
| 18:20.49 | *** join/#asterisk yarekt (n=fx3@unaffiliated/yarekt) |
| 18:20.55 | adr3nalin3 | [TK]D-Fender: I see I meant to include number-plan-custom-1 |
| 18:21.08 | Xamusk | I mean, to connect a normal telephone to a SIP account? |
| 18:21.49 | [TK]D-Fender | Xamusk: You can have * connect to an ITSP, and use an ATA to connect to * if you want |
| 18:22.39 | Xamusk | huh, that wouldn't solve my problem |
| 18:24.19 | [TK]D-Fender | Xamusk: Would help if you told us what kind of problem you actually have. |
| 18:25.27 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:27.00 | jaytee | facepalms |
| 18:27.19 | jaytee | *headdesks* |
| 18:27.27 | jaytee | ok, time for a smoke |
| 18:28.32 | yarekt | erm, hello, im just wondering, this asterisk thing, some sort of viop platform? |
| 18:28.55 | nr4q | how would a channel bank and T1 card stack up against a tmd400 for call quality on POTS ? |
| 18:29.35 | *** join/#asterisk railsmunky (n=nick@5acd4099.bb.sky.com) |
| 18:30.28 | yarekt | as im looking for something which is PC to PC, not necessarily to normal phones, but with echo canceling and that |
| 18:31.34 | nr4q | yarekt: it's a voip based pbx software package |
| 18:31.46 | nr4q | yarekt: what do you want to accomplish by linking two PC's together? |
| 18:32.33 | yarekt | nr4q, basically i would like something like skype, minus all the features |
| 18:32.49 | Qwell | so then just use skype |
| 18:33.33 | yarekt | Qwell, are you trying to put me off from developing my own software? |
| 18:33.36 | nr4q | yarekt: if you are only calling a couple different people like Qwell said use skype or just use a softphone |
| 18:34.06 | Qwell | yarekt: no, you never mentioned anything about developing anything |
| 18:34.23 | [TK]D-Fender | yarekt: And even if you did, this has nothing to do with * so far. |
| 18:34.43 | [TK]D-Fender | yarekt: If you want kess than Skype offers, * is definiitlely the wrong direction |
| 18:34.45 | [TK]D-Fender | less* |
| 18:35.13 | yarekt | well, thats all i came here to ask, and oh, id rather trade in better voice support for half bakes skype features |
| 18:35.51 | M1s3ry | yarekt, you'll be using asterisk if you want some form of routing calls with features built within. if you just want softphone to softphone, easily done without asterisk, if you want softphone to IVR to call queue (<--example) then asterisk is what your looking for |
| 18:36.08 | [TK]D-Fender | yarekt: Well I've never heard of * as being "better voice support" so I think this is still not what you're looking for. |
| 18:36.23 | yarekt | fair enough |
| 18:36.28 | yarekt | thanks for your time |
| 18:36.56 | [TK]D-Fender | yarekt: Hope you find whatever it is you're looking for. |
| 18:39.51 | [TK]D-Fender | NEXT!@@!@ (c) BKW |
| 18:40.54 | *** part/#asterisk furibondox (n=linux_us@host219-81-static.38-79-b.business.telecomitalia.it) |
| 18:41.41 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:43.38 | *** join/#asterisk SwK (n=SwK@user-69-73-16-126.knology.net) |
| 18:47.08 | *** join/#asterisk gcarrillog (n=gcarrill@201.151.84.209) |
| 18:47.24 | gcarrillog | hi |
| 18:47.47 | NovceGuru | wow, sip<mac>.cfg for polycoms is 122kbyte |
| 18:48.16 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 18:49.00 | gcarrillog | sorry my english is't good |
| 18:49.15 | gcarrillog | i have an spa400 |
| 18:49.41 | gcarrillog | today when i start asterisk have the next error |
| 18:50.01 | gcarrillog | Forbidden - wrong password on authentication for REGISTER for 'spa400' |
| 18:50.45 | *** join/#asterisk rdgr (n=rich@82-32-1-139.cable.ubr01.azte.blueyonder.co.uk) |
| 18:52.07 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:54.36 | jaytee | gcrarrillog, the password set in the spa400 does not match what is set as secret="password" in your sip.conf |
| 18:54.50 | adr3nalin3 | [TK]D-Fender: I included the context I needed to for outgoing calls in the default context, Now when I try to make a call out I get Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 18:55.19 | [TK]D-Fender | adr3nalin3: And error messages like that are worthless without all of the CLI output of the attempt |
| 18:55.30 | gcarrillog | i havent modified the configuration files |
| 18:55.55 | adr3nalin3 | [Jul 8 13:52:47] WARNING[4269]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 18:56.05 | [TK]D-Fender | adr3nalin3: THE ENTIRE DAMN CALL |
| 18:56.29 | adr3nalin3 | [TK]D-Fender: That was the whole thing. |
| 18:56.47 | gcarrillog | the spa400 does not have passwd |
| 18:56.59 | [TK]D-Fender | adr3nalin3: No, it isn't. That is generated because you called DIAL in the dialplan |
| 18:57.09 | [TK]D-Fender | adr3nalin3: make sure you're at verbose 10 |
| 18:57.18 | [TK]D-Fender | adr3nalin3: and pastebin the ENTIRE call. |
| 18:59.07 | Elijah` | Another Q guys... you shouldn't have to forward ports to the phones should you? |
| 18:59.17 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 18:59.20 | Elijah` | phones are ringing but got no sound... |
| 18:59.25 | [TK]D-Fender | Elijah`: No. |
| 18:59.30 | adr3nalin3 | [TK]D-Fender: entire call http://pastebin.com/m57d0035 |
| 19:00.12 | Elijah` | so what's the most likely cause for no sound then? I've got 5060 and the RTP ports forwarded to * |
| 19:00.41 | [TK]D-Fender | adr3nalin3: Wow, first a SIP probelm on dial, now a ZAP problem. Are you going to get consistent any time soon? |
| 19:00.56 | [TK]D-Fender | adr3nalin3: -- Executing [s@macro-trunkdial:2] Dial("SIP/1211-09dfe930", "Zap/7/14023064109/515-223-8100") <--- that is really not the format to dial out a zap channel |
| 19:01.11 | [TK]D-Fender | Elijah`: pastebin your sip.conf |
| 19:01.20 | Elijah` | ok :) |
| 19:02.07 | *** join/#asterisk iNetForce (n=f@74.246.21.235) |
| 19:02.51 | iNetForce | In the appliance when I do date -s 07081457 it takes the time fine but it shows UTC |
| 19:02.57 | iNetForce | i am in eastern -5 |
| 19:03.09 | Qwell | iNetForce: "the appliance"? |
| 19:03.10 | iNetForce | after a while the appliance changes the time itself. |
| 19:03.15 | iNetForce | si |
| 19:03.22 | [TK]D-Fender | Qwell: Don't forget to set it to "broil" ;) |
| 19:03.27 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583423.dsl.bell.ca) |
| 19:03.38 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:03.43 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:03.48 | iNetForce | root:~> date |
| 19:03.48 | iNetForce | Tue Jul 8 18:50:59 UTC 2008 |
| 19:03.54 | Qwell | which appliance? |
| 19:03.58 | iNetForce | it is 14:58 here |
| 19:04.00 | iNetForce | AA50 |
| 19:04.09 | *** join/#asterisk s0lid (n=s0lid@58.69.137.28) |
| 19:04.10 | iNetForce | i am GMT -5, Eastern |
| 19:04.11 | Qwell | The AA50 intentionally stores date in UTC. |
| 19:04.34 | Qwell | You need to upload your timezone files if you want it to have proper time in other applications. (this is all documented) |
| 19:04.38 | iNetForce | does it have an NTP server somewhere? It keeps changing the time to whateve it likes |
| 19:04.47 | iNetForce | I did uploaded New york |
| 19:04.53 | Qwell | yes, it uses the ntp server that you specified |
| 19:04.56 | iNetForce | applied changes and restart the unit |
| 19:05.04 | iNetForce | i took the NTP out |
| 19:05.10 | iNetForce | using the GUI |
| 19:05.11 | Qwell | Call support. |
| 19:05.27 | s0lid | hi |
| 19:05.35 | s0lid | what does asterisk mean when i get this error message |
| 19:05.36 | s0lid | Got SIP response 500 "Server Internal Error" |
| 19:06.29 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:06.31 | [TK]D-Fender | s0lid: Let me guess, got a bunch of Polycom phones by any chance? |
| 19:07.26 | *** join/#asterisk gfather1 (n=enforcer@79.173.244.73) |
| 19:07.31 | gfather1 | hello guys |
| 19:09.10 | gfather1 | anyone here ? |
| 19:11.28 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 19:11.37 | [TK]D-Fender | gfather : I hope you're not expecting a personal reflxive "Hello" from all 275 people in here :) |
| 19:11.41 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 19:11.45 | [TK]D-Fender | (oops, just lost a few!) |
| 19:11.51 | gfather1 | that would be very lovely :) |
| 19:11.58 | gfather1 | and i would be very happy |
| 19:12.10 | gfather1 | but its hard to be done :) |
| 19:12.22 | [TK]D-Fender | gfather1: You're "special".... just not that kind :) |
| 19:12.29 | Elijah` | ok here we go, http://elijah.pastebin.com/m2286001d |
| 19:12.30 | gfather1 | looooooooool |
| 19:12.46 | s0lid | [TK]D-Fender, nope i experience this with my VOIP provider |
| 19:12.56 | s0lid | i have ATA for my UA |
| 19:13.01 | gfather1 | <[TK]D-Fender> i had a question today , but i did not get the answear i was hoping for |
| 19:13.08 | [TK]D-Fender | s0lid: Are you experiencing any actual problems because of it? |
| 19:13.36 | s0lid | [TK]D-Fender, yes i get circuit-busy from my provider but what they say they never saw the call came to there switch |
| 19:14.21 | [TK]D-Fender | Elijah`: pastebin a failed call with SIP debug enabled |
| 19:14.41 | [TK]D-Fender | s0lid: Please provide CLI output with SIP debug in a pastebin. |
| 19:14.43 | [TK]D-Fender | ~pb |
| 19:14.44 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:14.44 | Elijah` | aah, ok sure... |
| 19:14.46 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 19:15.07 | gfather1 | can i get unauthenticated calls from the net ( like a pc software or wifi phone ) ? |
| 19:15.09 | [TK]D-Fender | gfather1: Well go ahead and ask... |
| 19:15.37 | [TK]D-Fender | gfather1: Yes. set "allowguest=yes" under [general] and se a context & codecs. |
| 19:16.04 | [TK]D-Fender | gfather1: and un-authed calls wil go to that context. |
| 19:16.28 | gfather1 | ok , then how will this be done , will it accpet it as a local phone , or will it be like normal call ? |
| 19:17.03 | [TK]D-Fender | gfather1: Every call to * is jsut like every other call |
| 19:17.15 | gfather1 | like if i ave a wifi phone , and i want to call , can i make it by ip , or domain name ? |
| 19:17.42 | gfather1 | like i want to call the company , ill call it through www.company.com |
| 19:17.51 | [TK]D-Fender | gfather1: Normally if you have a WiFi phone you'll put ACCOUNT CREDENTIALS into it. Why on earth would you send in calls from it un-authed? |
| 19:18.17 | gfather1 | ok , i want to make something like , people who dont work in the company , and wants to call , |
| 19:18.31 | [TK]D-Fender | gfather1: Fine, then do as I advised |
| 19:18.53 | gfather1 | thy can call for free by any wifi phone , by calling www. ....... |
| 19:18.59 | [TK]D-Fender | gfather1: and they will call you via "extension@youriporhost" |
| 19:19.13 | gfather1 | yes something like that :) |
| 19:19.55 | gfather1 | so it can be done easlly |
| 19:20.30 | nr4q | fender: thanks for the help |
| 19:20.35 | *** part/#asterisk nr4q (i=Ritalin@c-68-47-239-88.hsd1.tn.comcast.net) |
| 19:20.53 | gfather1 | <[TK]D-Fender> man thats very nice |
| 19:21.02 | gfather1 | that was the answear i hoped for :) |
| 19:23.21 | s0lid | [TK]D-Fender, here's the link from pastebin http://pastebin.com/mbfb0e98 |
| 19:23.56 | s0lid | [TK]D-Fender, i paste the debug log from the calling being make until it hangup |
| 19:24.03 | [TK]D-Fender | s0lid: that is not SIP debug .. |
| 19:24.16 | [TK]D-Fender | s0lid: SIP DEBUG"<- go start it from CLI. |
| 19:24.37 | s0lid | ok wait |
| 19:24.46 | [TK]D-Fender | s0lid: and I said the entire call. |
| 19:25.43 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:26.20 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 19:29.59 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 19:30.48 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 19:31.57 | s0lid | [TK]D-Fender, ok here's the link again http://pastebin.com/m2b533223 |
| 19:32.26 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 19:32.34 | ghenry | what does hint do again? for presense? |
| 19:32.48 | [TK]D-Fender | ghenry: Yes |
| 19:32.59 | ghenry | ok |
| 19:33.05 | ghenry | and _s-? |
| 19:33.10 | ghenry | catch all status? |
| 19:33.21 | ghenry | when looking for BUSY etc. |
| 19:33.25 | Xamusk | [TK]D-Fender, actually I wanted to have a print server, wireless router, pptp client and an ATA in a single machine |
| 19:33.57 | [TK]D-Fender | s0lid: Contact: <sip:192.168.90.24> <-- ya have not correctly set your system up to work from behind NAT |
| 19:34.07 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:34.23 | [TK]D-Fender | ghenry: No, taht is just pattern matching in the dialplan. |
| 19:34.28 | s0lid | [TK]D-Fender,oh sorry change the ip for security purposes |
| 19:34.34 | s0lid | *chaned |
| 19:34.38 | [TK]D-Fender | ghenry: No such thing as "looking for busy" |
| 19:34.39 | s0lid | changed* |
| 19:35.00 | s0lid | [TK]D-Fender, 192.168.90.24 are public |
| 19:35.17 | ghenry | status of exten I meant |
| 19:35.39 | [TK]D-Fender | ghenry: what you showed jsut there has nothing to do withs tatus of anything |
| 19:36.03 | ghenry | nm, I need to spend time to explain myself ;-) |
| 19:36.06 | Elijah` | ok here we go, sorry I had to get a remote phone on first |
| 19:36.07 | Elijah` | http://elijah.pastebin.com/d12d806a |
| 19:36.16 | Elijah` | that's the pastebin of the sip debug :) |
| 19:36.30 | ghenry | i meant [TK]D-Fender exten => s,2,Goto(s-${DIALSTATUS},1) |
| 19:36.40 | [TK]D-Fender | s0lid: Unmask it and paste me real debug. I'm not going to go around guessing what I can trust. |
| 19:36.41 | ghenry | NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER |
| 19:36.55 | ghenry | and exten => _s-.,1,Goto(s-NOANSWER,1) |
| 19:36.56 | [TK]D-Fender | ghenry: tahts a Goto. It ain't Raw-Cat Science. |
| 19:37.06 | ghenry | it was the _s- I was asking aboot |
| 19:37.28 | ManxPower | ghenry: pattern match, starting with "s-" plus 1 or more of any character |
| 19:37.34 | [TK]D-Fender | ghenry: that will catch any other exten starting with "s-" and more chars. |
| 19:37.52 | ghenry | doh, I wasn't thinking of strings. |
| 19:38.02 | ghenry | I thougth it was a special thing or something. |
| 19:38.04 | ghenry | thanks. |
| 19:38.56 | gfather1 | <[TK]D-Fender> i want to bulid a new server |
| 19:39.18 | gfather1 | any recomendation of stuff and products i should buy |
| 19:39.21 | [TK]D-Fender | gfather1: You have my official blessing. |
| 19:39.32 | gfather1 | <[TK]D-Fender> thanks :) |
| 19:39.42 | [TK]D-Fender | gfather1: would help if I had a list of needs and expectations. |
| 19:40.01 | gfather1 | well i have 7 lines from the phone company |
| 19:40.11 | gfather1 | wich i would like to connect to the server |
| 19:40.25 | gfather1 | and make the local phones al ip based and stuff |
| 19:41.09 | [TK]D-Fender | gfather1: where are you located, and what kind of "lines"? |
| 19:41.14 | gfather1 | im looking also for a good quality normal price phones |
| 19:41.21 | gfather1 | im located in jordan |
| 19:41.25 | gfather1 | city amman |
| 19:42.06 | Elijah` | :) |
| 19:42.46 | [TK]D-Fender | gfather1: Polycom's phones are the best choice out there, but Linksys SPA phones may be much more cost effective where you are. |
| 19:43.24 | gfather1 | any site to compare prices , or a shopping site |
| 19:43.34 | gfather1 | so i know what prices ranges im thinking off |
| 19:43.38 | [TK]D-Fender | gfather1: If you're talking straight analo lines, Either a Sangoma A200d loaded up to 8 ports, or a simial TDM800 with HWEC |
| 19:43.51 | gfather1 | oks |
| 19:43.55 | gfather1 | ill save the models |
| 19:43.55 | [TK]D-Fender | gfather1: I don't know any places with pricing that is relevant to you. |
| 19:44.06 | gfather1 | any good site |
| 19:44.17 | gfather1 | and ill find my way for ordering them |
| 19:44.30 | [TK]D-Fender | gfather1: only good sites I know are on the other end of the planet. |
| 19:44.38 | [TK]D-Fender | gfather1: www.telephonydepot.com |
| 19:46.00 | *** join/#asterisk s0lid (n=s0lid@58.69.137.28) |
| 19:46.01 | Xamusk | [TK]D-Fender, is an ATA actually doable with *? |
| 19:46.04 | gfather1 | well these days u can order anything from anywhere |
| 19:47.03 | [TK]D-Fender | Xamusk: * is not an ATA. You can USE an ATA with * though. And you can use PCI hardware interfaces to plug analog phones into * directly. |
| 19:47.16 | hsv-al | wtf, spikelee may start an isp called: "thaplanet.com" |
| 19:47.16 | gfather1 | Digium tdm800 is very cool , i saw it the other day |
| 19:47.24 | gfather1 | ill chek now the sangoma |
| 19:47.25 | [TK]D-Fender | gfather1: Of course you can... its a question of shipping, duties, etc. |
| 19:47.39 | gfather1 | yes |
| 19:47.43 | [TK]D-Fender | gfather1: Which is why its usually best to look local |
| 19:47.59 | gfather1 | well they are not available local i cheked |
| 19:48.09 | Xamusk | [TK]D-Fender, what I meant was to connect an analog telephone to * and make * connect to a SIP provider |
| 19:48.11 | gfather1 | i found digium in dubai |
| 19:48.40 | [TK]D-Fender | Xamusk: Yes you can definitely do that and I'd typically reocmmend using an ATA to let analog phones be usable by * |
| 19:48.50 | Xamusk | hum, ok then |
| 19:51.13 | s0lid | [TK]D-Fender, sorry i got disconnected a while ago |
| 19:51.19 | s0lid | [TK]D-Fender, have you seen the logs? |
| 19:51.32 | [TK]D-Fender | s0lid: Unmask it and paste me real debug. I'm not going to go around guessing what I can trust. <------------ |
| 19:53.07 | Elijah` | [TK]D-Fender, here's the two pastebins of my setup now... the sip.conf: http://elijah.pastebin.com/m2286001d and an sip debug of a failed call: http://elijah.pastebin.com/m4c5f9800 |
| 19:53.25 | Elijah` | I'm the one with the phones that will ring but there's no sound :) |
| 19:53.50 | anonymouz666 | [TK]D-Fender: do you know if PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 supports distinctive ring? |
| 19:54.15 | ManxPower | anonymouz666: all polycoms support changing the ring on the fly using SIP HEaders |
| 19:54.38 | anonymouz666 | ManxPower: good to know. thanks for the information. |
| 19:55.09 | ManxPower | I should say All Polycom SoundPoint IP phones do |
| 19:55.33 | ManxPower | Polycom has other phones, not SoundPoint IP that would not support this, but you would not normally be using these with Asterisk anyway. |
| 19:56.27 | s0lid | ok wait |
| 19:57.53 | Elijah` | :) |
| 19:58.14 | hsv-al | is there a way to search/retrieve naptr records |
| 19:58.17 | hsv-al | with an * func? |
| 19:58.56 | [TK]D-Fender | Elijah`: Contact: <sip:300@127.0.1.1> <- Line #239 & 249. You are telling them to conact LOCALHOST... |
| 19:59.27 | Elijah` | aah... ok that should be taken care of with the NAT setup though I thought? |
| 20:00.51 | Elijah` | aha, I wonder if that is bcause that phone has nat=yes set? |
| 20:00.52 | [TK]D-Fender | Elijah`: undo yourtemplating, set codecs across the board and set canreinvite=no, nay=yes for ALL of them except remote *'s / ITSPs. those last ones should only be "nat=no" |
| 20:00.52 | M1s3ry | oh man... I love it when I apply a patch... yet fail to remember to recompile. :/ |
| 20:01.16 | [TK]D-Fender | anonymouz666: You should be shot for using that firmware revision on a 601 anyways :p |
| 20:01.40 | Elijah` | ok so all remote phones set nat=yes on |
| 20:01.55 | Elijah` | internal phones too? |
| 20:02.03 | [TK]D-Fender | Elijah`: just do it. |
| 20:02.29 | [TK]D-Fender | Elijah`: And next time also set verbose to 10. I know its executing dialplan in there.... |
| 20:02.51 | Elijah` | verbose... on the sip debug or core set debug? |
| 20:03.16 | *** join/#asterisk tompaw (n=tompaw@pav.vip.krakow.tompaw.net) |
| 20:03.25 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 20:03.31 | [TK]D-Fender | Elijah`: basic "set verbose 10" |
| 20:03.38 | [TK]D-Fender | Elijah`: yes, I still want SIP debug. |
| 20:04.30 | Elijah` | ok got all that except the "set codecs across the board" not sure what you mean by that? |
| 20:04.39 | Elijah` | I'm sorry I'm new to this :( |
| 20:05.08 | [TK]D-Fender | Elijah`: disallow=all" followed by allow= for only the codec that theys hould be using. |
| 20:06.26 | Elijah` | which should be what, g711u? |
| 20:06.50 | *** join/#asterisk rycar (n=rycar@66-17-9-220.biz.bkfd.arrival.net) |
| 20:08.36 | Elijah` | I'll do g711u, as far as I know that's what they're using.. |
| 20:08.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:09.27 | [TK]D-Fender | Elijah`: Just make up your mind. |
| 20:09.38 | rycar | my caller ID shows up as 000-000-0000. It used to work, so I think the phone company is sending the data properly. Any ideas where I should start to troubleshoot this problem? |
| 20:09.50 | rycar | (incomming caller ID that is) |
| 20:12.50 | *** join/#asterisk andreadb7474 (n=andrea@195.94.142.68) |
| 20:13.38 | *** join/#asterisk gfather1 (n=enforcer@79.173.209.130) |
| 20:13.59 | gfather1 | im back , man that took along time to get my nick back |
| 20:14.06 | andreadb7474 | hi guys, there someone can help me about a trouble |
| 20:14.35 | ManxPower | ~ask |
| 20:14.36 | jbot | i heard ask is 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. |
| 20:14.56 | gfather1 | <[TK]D-Fender> man the Sangoma A200 can take up to 24 |
| 20:15.34 | gfather1 | and its cheaper than the 4 port diguim |
| 20:15.47 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 20:15.55 | gfather1 | is it similar in quality ? |
| 20:16.03 | ManxPower | gfather1: It's less than US$350? |
| 20:16.32 | gfather1 | its $173.90 :) |
| 20:16.46 | ManxPower | gfather1: that would be with NO modules. |
| 20:17.00 | ManxPower | $350 is a fully pupulated 4-port analog Digium card. |
| 20:17.08 | *** join/#asterisk Edder_ (n=edder@201.192.8.198) |
| 20:17.14 | andreadb7474 | Ok I've upgraded asterisk from 1.2 to 1.4 and all go better, but i've problem with moh that chomp, obviously i'have loaded first ztdummy module and seems to work correctly |
| 20:17.17 | ManxPower | They put little puppydogs in the analog modules. |
| 20:17.23 | gfather1 | im talking about the Sangoma A200 |
| 20:17.34 | gfather1 | http://www.telephonydepot.com/ProductDetails.asp?ProductCode=105%2D052%2DA200BRM&Show=TechSpecs |
| 20:17.41 | gfather1 | its less price and more ports |
| 20:18.06 | ManxPower | gfather1: Digium and Sangoma have similar prices for similar products. Neither is significantly cheaper or more expensive than the other. |
| 20:18.06 | andreadb7474 | but when i hold the call i hear music choppy |
| 20:18.19 | ManxPower | If you are seeing a big difference, then you are not comparing similar products. |
| 20:18.21 | Qwell | gfather: that card you linked it a 4 port card |
| 20:18.24 | funxion | I've got a weird problem. I'm using linksys spa942's when using the conference button on the phone to conference another call in the 2 remote ends loose audio and the audio received audio on the originating handset get garbled. Anyone have an idea? |
| 20:18.40 | Qwell | (with no modules) |
| 20:19.04 | ManxPower | andreadb7474: First you say it does not work, then you say it works, then you say it's choppy. WHAT is the issue you are having? |
| 20:19.11 | gfather1 | From 2 to 24 ports supported ,then does it have something messing ? |
| 20:19.15 | Qwell | yes |
| 20:19.24 | ManxPower | gfather1: yes, you have to buy the ports |
| 20:19.25 | Qwell | entire cards are missing |
| 20:19.33 | Qwell | it's multiple cards that "pretend" to be one |
| 20:19.45 | andreadb7474 | music on hold is choppy |
| 20:19.58 | ManxPower | andreadb7474: and what happens if you remove ztdummy? |
| 20:20.04 | ManxPower | it's not required for MoH |
| 20:20.16 | andreadb7474 | same things (choppy) |
| 20:20.34 | gfather1 | ah i see , there is an option to include the card |
| 20:20.37 | ManxPower | andreadb7474: did you remove the mpg123 stuff for 1.4, as it's not recommended. |
| 20:20.39 | gfather1 | damn |
| 20:20.41 | gfather1 | :) |
| 20:20.51 | Elijah` | [TK]D-Fender: Here's the sip debug output: http://elijah.pastebin.com/m4c4bdb57 |
| 20:21.00 | gfather1 | so what would u recomend guys , diguim or sengoma ? |
| 20:21.01 | Elijah` | the call did not drop that time, but still no sound |
| 20:21.18 | ManxPower | gfather1: It does not really matter |
| 20:21.30 | ManxPower | You'll have the same analog issues no matter what card you use. |
| 20:21.50 | andreadb7474 | I've not loaded mpg123 stuff, i use oggvorbis format |
| 20:22.04 | gfather1 | <ManxPower> analog issues ? |
| 20:22.29 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 20:22.44 | Edder_ | hi, somebody can help me with a configuration with sipura devices |
| 20:23.04 | ManxPower | yes. Analog lines will have issues with dialing time, hangup issues, etc. Just the standard stuff |
| 20:23.15 | Edder_ | yes |
| 20:23.29 | Edder_ | the most is dialing time |
| 20:23.30 | ManxPower | andreadb7474: put a copy of musiconhold.conf on pastebin.ca |
| 20:23.46 | Edder_ | sure, give me a minute |
| 20:23.49 | gfather1 | arent those fixed ? |
| 20:24.08 | funxion | did anyone read my question |
| 20:24.15 | ManxPower | gfather1: "fixed", your TELCO has to fix the hangup issues, there is no real way to "fix" the dialing time as it takes time to send DTMF to the telco. |
| 20:24.31 | ManxPower | If you don't want analog issues then use a PRI |
| 20:25.04 | gfather1 | ah , i though its from the astrisk , not from the telco side |
| 20:25.16 | ManxPower | Asterisk is limited by the analog line. |
| 20:25.51 | gfather1 | wel i dont think i can where i live |
| 20:25.59 | ManxPower | many of the advanced features are not supported on analog lines like answer, hangup supervision, DTMF transmission time, sending Callerid info, etc |
| 20:26.06 | andreadb7474 | what's pastebin.ca ? |
| 20:26.09 | gfather1 | they will bust my ass and will think im against the goverment |
| 20:26.12 | ManxPower | ~pb |
| 20:26.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:26.18 | andreadb7474 | however are only 2 rows |
| 20:26.19 | ManxPower | andreadb7474: you can go there and see. |
| 20:26.23 | andreadb7474 | mode= files |
| 20:26.35 | andreadb7474 | directory=/var/lib/asterisk/sounds/voip_sounds/moh/23 |
| 20:26.47 | ManxPower | andreadb7474: any valid musiconhold will have at least THREE lines. |
| 20:26.53 | funxion | I've got a weird problem. I'm using linksys spa942's when using the conference button on the phone to conference another call in the 2 remote ends loose audio and the audio received audio on the originating handset get garbled. Anyone have an idea? |
| 20:27.17 | andreadb7474 | what i've lost? |
| 20:27.25 | ManxPower | the MoH class, at least. |
| 20:28.07 | andreadb7474 | sorry |
| 20:28.14 | andreadb7474 | little mistake |
| 20:28.18 | ManxPower | funxion: sounds like the classic nat+reinvite or the classic RTP packet size issues. |
| 20:28.30 | funxion | no nat |
| 20:28.41 | ManxPower | andreadb7474: A single mistake can cost you tens of thousands of dollars when dealing with telecom, so you should be more careful. |
| 20:28.43 | gfather1 | <ManxPower> do u think a pri is available in jordan ? |
| 20:28.47 | funxion | could you provide insight to rtp packet size issue? |
| 20:29.10 | ManxPower | funxion: some linksys devices default to 30ms (.30) packet sizes, which do not work with Asterisk. |
| 20:29.17 | ManxPower | Asterisk expects a 20 ms RTP packet size. |
| 20:29.23 | funxion | ahh |
| 20:29.35 | funxion | Ill try that now |
| 20:29.38 | [TK]D-Fender | Elijah`: Reliably Transmitting (NAT) to 74.84.3.183:29490: Contact: <sip:elijah@127.0.1.1> You are giving them the wrong address again. Fix your peer |
| 20:29.40 | andreadb7474 | i try to fix it immediately |
| 20:30.41 | Elijah` | I know but how would I change that? |
| 20:31.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:31.17 | ManxPower | Elijah`: just like every single other NAT Asterisk user in the world. localnet= and externip= in sip.conf [general] |
| 20:31.32 | ManxPower | what the values are depends on your network |
| 20:32.10 | ManxPower | unless your issue is with a DEVICE behind NAT and Asterisk is not behind NAT, in which case it's totally different. |
| 20:32.10 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:32.12 | Elijah` | yes, it's set to localnet=192.168.0.0 |
| 20:32.25 | ManxPower | Elijah`: that is not a valid localnet, as it has no mask |
| 20:32.36 | ManxPower | You DID go read sip.conf.sample, right? |
| 20:32.43 | Elijah` | oops it has 255.255.0.0 on the end |
| 20:32.50 | Elijah` | yeah, I just din't type the whole thing out sorry |
| 20:32.54 | lesouvage | I have a dtmf problem. It seems to work fine but when I press the buttom just a little bit longer then sometimes this doubles the input for the digit with failure as a result. Is there a variable that can be set to avoid this problem? |
| 20:33.04 | ManxPower | Elijah`: ok, now put a copy of your sip.conf on pastebin.ca as we can no longer trust what you say. |
| 20:33.05 | Elijah` | localnet=192.168.0.0/255.255.0.0 |
| 20:33.27 | Elijah` | LOL ok |
| 20:33.49 | funxion | ManxPower Tried it and it didnt make a difference |
| 20:33.55 | ManxPower | Elijah`: you need to be more careful. A single wrong config setting could allow random people to use your server to make free calls anywhere |
| 20:34.06 | ManxPower | funxion: then I cannot help you further. |
| 20:34.15 | funxion | thnx anyways |
| 20:34.22 | ManxPower | Elijah`: well, free to them, cost to you. |
| 20:34.33 | Elijah` | http://elijah.pastebin.com/d5c9f91e3 |
| 20:35.12 | andreadb7474 | I've fixed it but, the music is still as before, feels bad |
| 20:35.27 | Elijah` | see if I've seriously screwed up something.... hehe |
| 20:35.37 | Edder_ | ManxPower: can you help me, with my issue? |
| 20:35.46 | ManxPower | Elijah`: Sorry, I cannot help you with users.conf issues nor with AsteriskGUI or with AsteriskNOW. |
| 20:35.58 | ManxPower | Edder_: where is your question? |
| 20:36.14 | ManxPower | ~ask |
| 20:36.14 | jbot | it has been said that ask is 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. |
| 20:36.39 | Edder_ | my problem is with analog lines and sipuras |
| 20:36.48 | ManxPower | Edder_: I cannot help you. |
| 20:36.48 | Elijah` | that was sip.conf... |
| 20:37.04 | ManxPower | Elijah`: but it uses template settings from users.conf |
| 20:37.06 | Edder_ | ManxPower: ok, thanks |
| 20:37.20 | ManxPower | that's what all the (!) and (phones) crap is. |
| 20:37.30 | Elijah` | aah... no no that's a template |
| 20:37.39 | Elijah` | right? |
| 20:37.42 | Edder_ | any reference about dialing tones? |
| 20:38.03 | Elijah` | and is defined right there in sip.conf... |
| 20:38.51 | Elijah` | the (phones) just refers up to what's defined in [phones] as I understood it... |
| 20:39.16 | Elijah` | so you don't have to define that again for each phone :P |
| 20:39.31 | ManxPower | Elijah`: OK, I'll edit it out then. Here: http://elijah.pastebin.com/m5455f3f4 |
| 20:39.39 | ManxPower | Elijah`: THOSE ARE TEMPLATES |
| 20:39.57 | ManxPower | I just said I can't help you with them. I doubt anyone here can help you with them. |
| 20:40.26 | ManxPower | you should of course remove the port=5060 as well |
| 20:40.29 | andreadb7474 | perhaps can help the fact that i hear choppy also the ring when i transfer a call ? |
| 20:40.30 | Elijah` | ohhh, ok I'm sorry misunderstood |
| 20:40.34 | ManxPower | since the remote port is frequently NOT 5060 |
| 20:40.51 | Elijah` | aaah, good idea didn't think of that :P |
| 20:40.56 | ManxPower | but I mist now go back to paying work |
| 20:41.15 | Elijah` | lol yes ty, I will try that :) |
| 20:43.13 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:44.05 | *** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-3bfa09c95d84f116) |
| 20:46.47 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 20:52.05 | *** join/#asterisk snowy_owl (n=snowy_ow@201-048-005-105.static.ctbctelecom.com.br) |
| 20:55.12 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 20:55.23 | variable_office | what do i have to do in order to get the PGSQL() app running? |
| 20:55.44 | snowy_owl | yeeeaaah my people! Hi.. is there a way to REdial to another route when the first attempt wasnt sucessful? Obs: without returning an error to caller before trying the second time. |
| 20:56.18 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 20:56.21 | *** part/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 20:59.54 | ManxPower | snowy_owl: yes |
| 21:00.36 | *** join/#asterisk browser (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 21:00.50 | ManxPower | But a newbie would never figure it out, you should go read The Asterisk Book, as well as extensions.conf.sample, the docs for the Dial statement, especially the DIALSTATUS and HANGUPCAUSE variables. You cannot do this on FXO ports. |
| 21:01.00 | ManxPower | ~book |
| 21:01.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:01.17 | hsv-al | lol |
| 21:01.23 | hsv-al | page 151 |
| 21:01.27 | hsv-al | Anti-boyfriend code |
| 21:01.29 | hsv-al | lulz |
| 21:01.42 | ManxPower | snowy_owl: It took me 2 years to write a failover, alternate route system that I, my client, and my users were happy with. |
| 21:01.58 | ManxPower | hsv-al: it has nothing to do with the anti-boyfriend option |
| 21:02.30 | ManxPower | anti-girlfriend/anti-boyfriend are for INCOMING calls, snowy_owl is talking about OUTGOING calls. |
| 21:02.46 | hsv-al | i wasnt reading his statement, i was just sifting |
| 21:02.48 | hsv-al | through old shit |
| 21:03.30 | ManxPower | hsv-al: so Tech Tourettes? |
| 21:04.00 | hsv-al | conditional branching, via caller ID |
| 21:04.24 | hsv-al | make custom messages for all friends/family / business acquaintances |
| 21:04.34 | snowy_owl | ManxPower: my system was builded over openser, but actually Im trying to use asterisk to solve some problems. This way, I need to implement the failover route in this system too. |
| 21:05.07 | ManxPower | snowy_owl: good. |
| 21:05.42 | ManxPower | You found VoIP providers are not as reliable as they say they are. |
| 21:06.13 | hsv-al | manxpower, what is a good provider that can provide lets say 10 pstn #'s |
| 21:06.18 | hsv-al | , ie: callcentric, at a good price |
| 21:06.22 | snowy_owl | ManxPower: anyway, tks for your answer. I have been looking for this information all day and I hadnt find it yet. But now, I now that is possible. It is enough |
| 21:06.29 | *** part/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 21:06.33 | hsv-al | i heard of some company called aretta |
| 21:06.36 | hsv-al | no one else? |
| 21:07.20 | lesouvage | I'm using the A option (Dial("SIP/040135000-081e5ff0", "SIP/003159111111@0501111111|40|A(/var/lib/asterisk/sounds/aankondiging)m"), where "aankondiging" is the announcement to be plaied for the one who is called. The problem is that the anouncement starts before the one called picked up the phone and, when the one who is called takes his or her time, the anouncement is finished when the... |
| 21:07.22 | lesouvage | ...phone is picked up. Is there a solution for this problem (except from improving/changing the source code) |
| 21:07.45 | ManxPower | lesouvage: the solution is to stop using analog |
| 21:08.02 | ManxPower | Analog provides no way for the calling device to know when the far end was answered. |
| 21:08.13 | ManxPower | It's not just analog, it's analog FXO or channelized T-1 FXO |
| 21:08.28 | lesouvage | ManxPower: I'm a total sipper on this box |
| 21:08.50 | ManxPower | lesouvage: what is [003159111111] |
| 21:09.04 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 21:09.14 | ManxPower | or whatever the device that hands calls to the PSTN |
| 21:09.34 | lesouvage | ManxPower: That is actually my phone number (the number called) with some adjustments for the irc channel. |
| 21:09.36 | ManxPower | lesouvage: This applies to ANY DEVICE WITH ANALOG LINES, not only Asterisk |
| 21:09.46 | ManxPower | So, so what is [0501111111] then? |
| 21:09.55 | browser | #linux |
| 21:10.37 | ManxPower | somewhere, somehow you are connecting to the PSTN. How are you doing this? |
| 21:10.55 | lesouvage | That is the name of the entry of the sip trunk and the phonenumber of this sip trunk (again with some adjustments) |
| 21:11.12 | ManxPower | lesouvage: how are you connecting to the PSTN? |
| 21:11.46 | lesouvage | ManxPower: I'm using the services of a siptrunk provider. |
| 21:12.03 | ManxPower | lesouvage: there is no such thing as a "sip trunk". |
| 21:12.07 | ManxPower | It is a peer or a provider. |
| 21:12.25 | ManxPower | lesouvage: then contact your provider and tell them to stop answering the call before dialing the PSTN |
| 21:13.55 | lesouvage | ManxPower: Thanks for the last line, that must be the problem. |
| 21:14.58 | ManxPower | I would change SIP providers, any provider that unconditionally answers the call is run by morons that know nothing about telecom |
| 21:15.44 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 21:15.53 | lesouvage | ManxPower: what is wrong with using the word SIP trunk. I have as many outgoing and incoming lines as my bandwidth can handle. From a user perspective that really looks like a trunk. |
| 21:16.41 | lesouvage | ManxPower: I think I first try a decent conversation, but it seems realy hard to find a proper sip provider. |
| 21:17.26 | ManxPower | lesouvage: there is no protocol difference |
| 21:17.39 | ManxPower | with IAX2 trunking there is an actual difference in the protocol. |
| 21:18.05 | ManxPower | there is no difference in the protocol in SIP if you send one call, 100 calls or 1000 calls at the same time to the same place. there is no protocol difference. |
| 21:18.13 | ManxPower | "sip trunking" is a MARKETING term. |
| 21:19.06 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:19.50 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:20.09 | lesouvage | ManxPower: I totally agree, but marketing is what gives speed to the cashflow. I have seen the best voip startup going down the drain because of lack of marketing. |
| 21:20.32 | ManxPower | When you say "SIP trunking" we know you are a newbie, just reciting marketing material. Much like if you went to a place that only sold Cannon brand copiers and asked for a "Xerox machine". They will immediately know nothing about copy machines and would be a good person to sell that old outdated, over priced copier to. |
| 21:20.43 | ManxPower | lesouvage: this is a technical channel, this is not a marketing channel |
| 21:21.16 | [hC] | I am still curious what you would call what most people refer to as a "sip trunk" |
| 21:21.20 | [hC] | ie, a trunk via sip. |
| 21:21.24 | ManxPower | ...they will immediatly know you know nothing... |
| 21:21.31 | ManxPower | [hC]: they are just vomiting up marketing terms |
| 21:21.55 | [hC] | ManxPower: well, i refer to my sip "trunks" as sip trunks, and I know what im talking about. |
| 21:22.04 | [hC] | ManxPower: It just happens to be the easiest way to describe it |
| 21:22.13 | ManxPower | [hC]: it's a provider, SIP provider, or ITSP (generic terms) It's also a peer/friend (Asterisk terms) |
| 21:22.18 | [hC] | ManxPower: not meaning "Trunking" in the sense that IAX2 has trunking, of course. |
| 21:22.21 | ManxPower | [hC]: describe *what*? |
| 21:22.48 | ManxPower | It's the same SIP regardless of if you send the call to a provider or to a SIP phone. |
| 21:22.53 | [hC] | ManxPower: a connection to the PSTN via sip. |
| 21:22.55 | lesouvage | ManxPower: You start the marketing talk and I'm expert enough to know that you are totaly right from a technical point of view. |
| 21:23.07 | snowy_owl | A see a lot of people saying "sip trunking" instead of "sip for telephones". The RFC 3372 defines it. |
| 21:23.09 | ManxPower | [hC]: I just told you. |
| 21:23.11 | [hC] | ManxPower: I realize. In traditional telco terms, a "trunk" was simply a connection to the PSTN. |
| 21:23.34 | ManxPower | [hC]: and the SIP service provider has many trunks, but there are no trunks between Asterisk and the provider |
| 21:24.13 | snowy_owl | using other words: SIP T has ISDN information inside his body. It isnt another protocol. |
| 21:24.16 | [hC] | ManxPower: If by trunks you mean a D channel with several B channels, as in a PRI, i think you are even misusing the word, because traditional vendors refer to a single FXO line as a trunk as well. |
| 21:24.46 | ManxPower | Heck, even using "traditional" usage, a trunk is a single voice channel. a "sip trunk" as unlimited voice channels. |
| 21:25.02 | Strom_M | a PRI is technically a trunkgroup, not a trunk |
| 21:25.09 | ManxPower | [hC]: each one is a single channel carrying a single voice call. |
| 21:25.17 | lesouvage | ManPower: If I have to explain a customer what can be done with a SIP telephone subscription I tell him or her it is a bundle with an undefined number of in or outgoing lines. From a technical point of view kind of nonsense but non techies understand it. |
| 21:25.21 | ManxPower | I don't use the telco term trunk either |
| 21:25.40 | [hC] | ManxPower: all this is, is nitpicking details to determine the purest of meanings, in their technical sense. |
| 21:25.41 | ManxPower | lesouvage: And they are going to understand the term "sip trunk" any better? |
| 21:25.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:26.19 | [hC] | "I would like to purchase a sip trunk" "I would like to purchase a connection between my server and your network that provides me access to the pstn" |
| 21:26.22 | ManxPower | [hC]: sort of like teachers that don't allow "r u mi bff?" TXTing crap in papers? |
| 21:26.27 | [hC] | Which are you going to say? |
| 21:26.42 | ManxPower | [hC]: I would like a SIP account, SIP service |
| 21:27.03 | [hC] | ManxPower: but which sip service? I sell many sip based services. |
| 21:27.17 | ManxPower | [hC]: It's a MARKETING term. I could care less what people call stuff in marketing, but this is a technical channel where accuracy is REQUIRED to be able to do anything correctly |
| 21:27.46 | [hC] | ManxPower: i could see why you would get upset if there was such a thing as sip trunking, and people were confusing the two making it difficult to provide technical assistance. |
| 21:28.11 | ManxPower | At least with over use of T/t Dial options, the user will get hit with a $10,000 phone bill and that is their punishment. |
| 21:28.15 | [hC] | ManxPower: however, that is just not the case. |
| 21:28.52 | lesouvage | ManxPower: no, but if I compare it with a ISDN30 (lines, options costs) they understand what I mean. From a user perspectie it is just making a phone call. Lets just rest the case and stick to the technical perspective. Btw: is there an Asterisk marketing irc channel? |
| 21:29.07 | ManxPower | lesouvage: I'm talking technical./ |
| 21:29.19 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-71fef7382d9074d3) |
| 21:29.39 | [hC] | lesouvage: there is not a marketing channel. there is a business mailing list, however. |
| 21:29.47 | ManxPower | [hC]: lazyness carries over to other parts too. |
| 21:30.03 | bkruse | [hC]: LOL@asterisk-biz |
| 21:30.17 | ManxPower | [hC]: Users think "IAX trunking" is the same as "SIP trunking" |
| 21:30.25 | ManxPower | THAT is why we should not use the term here. |
| 21:30.50 | ManxPower | I constantly have people tell me they have an IAX2 trunk when they don't. |
| 21:31.05 | [hC] | bkruse: :) |
| 21:31.26 | ManxPower | [hC]: So with SIP, if you have SIP service connecting you to the PSTN and call it trunking, what do you think people are going to think an IAX2 trunk is? |
| 21:31.43 | [hC] | I TOTALLY agree with frustration over the "marketing" sense of the word trunk with IAX. |
| 21:31.57 | ManxPower | [hC]: that's caused by the SIP Trunking |
| 21:32.16 | ManxPower | I mean they are both VoIP protocols, the terms should mean the same thing, right? |
| 21:32.43 | [hC] | ManxPower: It all boils down to how strict you want to be, personally.... I guess. I still refer to my sip connections as "trunks" and i dont really care. I know the difference. |
| 21:33.05 | ManxPower | [hC]: but the person here might noty know the difference. |
| 21:33.33 | ManxPower | This is a tech channel, can't we at least use accurate terms? |
| 21:34.01 | Nugget | Similarly, I refer to Linux as "poo" :) |
| 21:34.18 | ManxPower | Nugget: all you crazy *BSDers do that. |
| 21:34.28 | Nugget | it's part of the license. |
| 21:34.35 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 21:35.10 | Qwell | Nugget: yeah, that 4th clause... |
| 21:35.23 | Qwell | 4) User must refer to Linux as 'poo'. |
| 21:35.59 | Nugget | yup |
| 21:36.05 | lesouvage | ManxPower: The A option in the Dial command should work properly, meaning that the announcement normally starts when the phone is picked up. I'm writing an e-mail now to my provider now asking to get his sip messages right. |
| 21:36.35 | ManxPower | lesouvage: correct |
| 21:36.51 | [hC] | ManxPower: I have the answer |
| 21:36.54 | [hC] | SLURPEE TIME :) |
| 21:37.14 | [hC] | (is it hot as balls for any of you guys today too?) |
| 21:37.23 | ManxPower | [hC]: stop using the term "sip trunk"? |
| 21:37.31 | lesouvage | hC About the meaning of life, the universe and anything else? |
| 21:37.39 | [hC] | ManxPower: no.. slurpees, man. :) |
| 21:38.00 | [hC] | Am I crazy to want to purchase a g4 "lampshade" imac because i think its cool? |
| 21:38.15 | [hC] | I have one for sale here for $200 that im tempted to pick up |
| 21:38.22 | *** join/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
| 21:39.38 | ipso | Is there a way in the queue system that when an agent is automatically logged out (for not answering a call) that a dialplan entry or macro can be called? Or is parsing log entries the only way to know about an auto-logout? |
| 21:40.23 | [hC] | ipso: i dont know specifically but you can likely trap the event using the manager |
| 21:41.05 | ipso | [hC]: Thanks, I'll take a look |
| 21:42.00 | *** join/#asterisk railsmunky (n=nick@5acd4099.bb.sky.com) |
| 21:46.48 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:46.53 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 21:47.52 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:48.07 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:50.27 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:51.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:54.05 | *** part/#asterisk fogo (n=fogo@72.8.104.15) |
| 21:54.09 | lesouvage | ManxPower: I just tried a sip "friend" of an other sip provider and I still have the same problem. It is possible that they both have there sip messages not in order but it could also be an Asterisk dial cmd a parameter issue. I have no more sip "friends" available. Are you really sure the A parameter should work properly if the provider has his sip messages in place (sorry for asking again... |
| 21:54.11 | lesouvage | ...but I seem t have two sip providers who don't have their messages in order) |
| 21:54.18 | *** join/#asterisk rvhi (n=chatzill@udp197017uds.hawaiiantel.net) |
| 21:54.39 | rvhi | hi, i tried to compile app_curl, have libcurl installed, but it still won't compile |
| 21:57.00 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 21:57.31 | rycar | how do I debug zaptel hardware? I am trying to find out if my phone company is actually passing me callerID information |
| 21:58.22 | *** part/#asterisk gcarrillog (n=gcarrill@201.151.84.209) |
| 21:58.22 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 21:59.06 | Qwell | rycar: be a little more specific |
| 21:59.09 | Qwell | what hardware? |
| 21:59.13 | *** part/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 21:59.38 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:02.58 | *** join/#asterisk schemathings (n=schemath@68-244-39-103.area3.spcsdns.net) |
| 22:04.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:06.32 | rycar | Qwell: Digium Wildcard TE110P T1/E1 Card |
| 22:06.45 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 22:06.57 | teknoprep | is there a patch for 1.4.18 that allows for sip tcp ? |
| 22:07.08 | jaytee | rvhi, what linux distro are you running? |
| 22:07.19 | rvhi | jaytee: ubuntu |
| 22:07.37 | teknoprep | i personally love centos 5 for asterisk installs |
| 22:07.40 | teknoprep | makes life easy |
| 22:08.03 | rvhi | all our servers use ubuntu, it is almost impossible to change |
| 22:08.18 | teknoprep | why would it be impossible to change for just one server ? |
| 22:08.57 | rvhi | nobody knows how to manage centos |
| 22:09.03 | teknoprep | heh |
| 22:09.05 | rvhi | lots of commands are different |
| 22:09.18 | teknoprep | nah not the commands |
| 22:09.23 | teknoprep | maby the init structure |
| 22:09.29 | rycar | you just type yum instead of apt-get |
| 22:09.37 | rvhi | can't convince people to switch just because app_curl can't compile |
| 22:09.43 | teknoprep | lol |
| 22:09.47 | teknoprep | but its a phone server |
| 22:09.54 | jaytee | rvhi, what version of Ubuntu? Gutsy, Hardy? |
| 22:09.58 | teknoprep | and ubuntu honestly is borked in my opinion |
| 22:09.59 | rvhi | 7.0.4 |
| 22:10.01 | teknoprep | for servers |
| 22:10.13 | jaytee | Edgy? your shittin me? |
| 22:10.19 | teknoprep | its an absolute great desktop os for ppl wanting to switch from windows |
| 22:10.35 | rvhi | because all other servers run on the one |
| 22:10.35 | rycar | centos for servers, ubuntu for desktop |
| 22:10.42 | rvhi | we settle on this one |
| 22:10.45 | *** join/#asterisk MmixX (i=mmixx@ded-138-69.eglobalreach.net) |
| 22:10.54 | rvhi | tried to use 8.0.4 on a new server |
| 22:11.03 | rvhi | all voice prompts are broken |
| 22:11.06 | jaytee | you need to install libcurl?-gnutls-dev and replace the ? with whatever version of libcurl you've got installed. |
| 22:11.11 | teknoprep | rvhi why not just try ? |
| 22:11.22 | teknoprep | i mean hell RedHat puts ALOT of work into making a good product |
| 22:11.30 | teknoprep | then ppl take that and release it as CentOS |
| 22:11.33 | rvhi | i have apt-get install libcurl3-gnutls-dev |
| 22:12.07 | teknoprep | i run my centos 5.1 install inside of vmware with the kernel-vm |
| 22:12.25 | teknoprep | it runs perfectly with no skips |
| 22:12.45 | jaytee | I run production * on RHEL 5 and my test environment on CentOS 5. I use Ubuntu for one of my desktops at home and at work but I'd never use Ubuntu as a production server. |
| 22:13.00 | teknoprep | agreed |
| 22:13.02 | teknoprep | i know ppl that do |
| 22:13.13 | teknoprep | but they don't have long enough support cycle per version IMHO |
| 22:13.23 | teknoprep | you have to keep upgrading |
| 22:13.23 | jaytee | rvhi, you ran apt-get libcurl3-gnutls-dev and it still won't compile? |
| 22:13.30 | rvhi | right |
| 22:13.55 | jaytee | try on the Ubuntu forums, you're missing some dependencies most likely. |
| 22:14.02 | rvhi | if i manually add app_curl.so in the Makefile, it works |
| 22:14.03 | teknoprep | soooo i am am looking for sip tcp |
| 22:14.07 | teknoprep | for 1.4.18 |
| 22:14.15 | teknoprep | i was hoping for an RPM with it already compiled |
| 22:14.18 | teknoprep | but thats askign alot |
| 22:14.19 | jaytee | teknoprep, nope, ain't gonna happen. |
| 22:14.25 | teknoprep | dan |
| 22:14.26 | teknoprep | damn |
| 22:14.27 | jaytee | try 1.6 beta |
| 22:14.35 | teknoprep | yeah but i use freepbx on ALL of my systems |
| 22:14.49 | jaytee | sucks to be you! :-) |
| 22:14.55 | teknoprep | hey its not my systems |
| 22:15.05 | jaytee | ok, so it sucks to be "them" |
| 22:15.07 | teknoprep | i have to install this stuff as a consultant for the casual Windows IT admin |
| 22:15.24 | teknoprep | ssh is like omfg to them |
| 22:15.37 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 22:15.46 | jaytee | I'm a casual Windows/Linux/VOIP/TDM IT admin. |
| 22:15.50 | teknoprep | maby i will try to do the sipX proxy to asterisk |
| 22:15.58 | jaytee | I'm using sipX |
| 22:16.05 | teknoprep | so i can integrate MS Com Server 2007 |
| 22:16.21 | jaytee | as a proxy for UDP to TCP to get to Exchange Unified Messaging |
| 22:16.24 | teknoprep | yup |
| 22:16.31 | jaytee | it works fine |
| 22:16.32 | teknoprep | is it nice ? |
| 22:16.42 | jaytee | Unified Messaging is nice, yeah |
| 22:16.46 | teknoprep | cool |
| 22:17.04 | Qwell | teknoprep: who doesn't have long enough support? Ubuntu? |
| 22:17.18 | teknoprep | Qwell, yeah thats how i feel |
| 22:17.19 | Qwell | are you kidding? the current version is supported until 2013 |
| 22:17.31 | teknoprep | Qwell, it is ? well prove me wrong then lol |
| 22:17.43 | Qwell | http://www.ubuntu.com/getubuntu/download |
| 22:17.47 | jaytee | teknoprep, here's a link you might find handy for sipX to Exchange UM using * (or FreePBX in your case). http://blog.lithiumblue.com/2007/10/accessing-exchange-2007-unified.html |
| 22:17.58 | Qwell | Ubuntu 8.04 LTS Server Edition - Supported to 2013 |
| 22:18.07 | mvanbaak | ubuntu schmubuntu |
| 22:18.20 | teknoprep | jaytee, already have it but thanx man |
| 22:18.42 | teknoprep | sipX runs inside of VMware fine ? |
| 22:18.57 | jaytee | I really like Ubuntu, don't get me wrong but I just think the bleeding edge platform isn't where you should focus your efforts installing a production VOIP system. |
| 22:19.13 | mvanbaak | ubuntu is still linux ... |
| 22:19.26 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:19.56 | jaytee | yeah, but it's always based on Debian unstable. I'd rather run * on Debian stable than on the latest rev of Ubuntu. |
| 22:20.13 | mvanbaak | I'd rather not run linux at all |
| 22:21.01 | jaytee | mvanbaak, there's a Windows fork of * IIRC but I think you have to compile it using Visual Studio and not sure how well it runs. Probably get alot of BSOD's. |
| 22:21.19 | Qwell | jaytee: No such thing. |
| 22:21.21 | mvanbaak | jaytee: asterisk runs fine on *BSD |
| 22:21.41 | teknoprep | hey to use Exchange UM do i have to upgrade exchange to 2007 ? |
| 22:21.42 | jaytee | Qwell, just a sec |
| 22:22.01 | mvanbaak | teknoprep: call your local MS rep |
| 22:22.12 | teknoprep | becuase i was just thinking of trying out MS Communications Servers 2007 |
| 22:22.16 | teknoprep | with exchagne 2003 |
| 22:22.23 | teknoprep | well that question wasto jaytee |
| 22:22.53 | jaytee | teknoprep, yes you need to run Exchange 2007 |
| 22:23.00 | mvanbaak | what I've heard is that you need a seperate box for the UM stuff |
| 22:23.11 | teknoprep | jaytee, have you ever run MS Communication Server 2007 |
| 22:23.25 | jaytee | mvanbaak, if you have alot of users yes but you can run it on the main Exchange box |
| 22:23.40 | mvanbaak | define 'alot of users' |
| 22:23.43 | mvanbaak | 50 ? |
| 22:24.07 | jaytee | teknoprep, I've tested it with their Speech Server component but not done alot with it. Very, very steep learning curve. |
| 22:24.23 | teknoprep | jaytee, for the end user ? |
| 22:24.37 | jaytee | mvanbaak, yeah around that point I'd start thinking of moving UM to a separate box |
| 22:24.49 | mvanbaak | jaytee: omg |
| 22:24.56 | jaytee | teknoprep, no the learning curve is for the person integrating it. |
| 22:25.03 | teknoprep | jaytee, oh |
| 22:25.09 | mvanbaak | we run 1500 users on the some p4 as we run our mail platform and our dns server |
| 22:25.18 | mvanbaak | of course this is asterisk |
| 22:25.31 | teknoprep | 1500 user exchange box on a p4 ? |
| 22:25.33 | wishes | gmorning |
| 22:25.58 | jaytee | mvanbaak, I'm thinking UM could handle up to a 100 or more users easily but we don't plan on testing that theory. Hardware is relatively cheap and we get discounts on liceninsing because we're a non-profit. |
| 22:26.06 | mvanbaak | asterisk + postfix + courier-[(imap)|(pop)] |
| 22:26.06 | Qwell | I wonder if Exchange would run in wine. |
| 22:26.11 | Qwell | sorry, that was random |
| 22:26.37 | mvanbaak | jaytee: how about rackspace at a colocation ? |
| 22:27.19 | teknoprep | Qwell, doubt it |
| 22:27.23 | jaytee | Qwell, there's a 32 bit version of Exchange 2007 you could try but it's only for developers to test. MS recommends running the 64 bit version of Server 2K3 with Exchange 2007 64bit. |
| 22:27.39 | mvanbaak | gheh |
| 22:27.51 | mvanbaak | the 64 bit version of 2k3 is emulation |
| 22:27.58 | mvanbaak | the kernel is still 32 bit |
| 22:28.32 | jaytee | really? hmmm, well it's been awhile since I looked at the source code :-) |
| 22:29.30 | mvanbaak | the cpu driver has a bunch of hooks to allocate more mem |
| 22:29.40 | mvanbaak | but all the instructions are still 32 bit |
| 22:30.04 | mvanbaak | 64 bit in windows is only used to overcome the memory limit in 32 bit |
| 22:30.16 | Kobaz | when i do a sip reload, my audiocodes gateway goes to status: UNKNOWN.... every other sip device keeps it's reg status... why is that? |
| 22:30.27 | jaytee | mvanbaak, you have some documentation to back up that claim? |
| 22:30.27 | mvanbaak | ms calls it 64 bit, linux calls it 32bit pae |
| 22:30.58 | mvanbaak | jaytee: not here. I have it at work |
| 22:31.38 | jaytee | ok, well maybe next time your on at work and I'm here you can shoot me a link, I'd be interested in reading about that in more detail. |
| 22:31.50 | teknoprep | is asterisk 1.6 going to be a while before its finished? |
| 22:32.23 | mvanbaak | jaytee: if I remember I will |
| 22:32.29 | Kobaz | never mind, i had a host setting in the sip conf |
| 22:32.35 | mvanbaak | teknoprep: it's in beta |
| 22:32.44 | teknoprep | mvanbaak, yup |
| 22:32.54 | mvanbaak | it will be here somewhere this year I'm for sure |
| 22:32.57 | jaytee | and my 50-100 users statement is a highly conservative guesstimate based on slower hardware. On a Quad Xeon with a load of RAM I'd feel comfortable pushing it to 200-500 users as long as I've got teamed NICs running at 1GB. |
| 22:34.36 | jaytee | beta9 of 1.6 seems fairly stable in testing so far. I've not run into any really serious problems with it but i still need to test using Digium T1 hardware to cover all the bases. |
| 22:35.09 | *** part/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 22:36.29 | mvanbaak | gheh |
| 22:36.43 | mvanbaak | running 1.6 svn is not as nice |
| 22:36.50 | mvanbaak | you have to checkout dahdi |
| 22:36.52 | *** join/#asterisk arcanine (n=saxon_m2@202.138.162.67) |
| 22:37.01 | *** join/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 22:37.06 | mvanbaak | and compiling that is not as easy as zaptel yet |
| 22:37.34 | *** join/#asterisk mgroman (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 22:37.41 | *** join/#asterisk mookid (i=mookid@httpd.info) |
| 22:37.46 | mgroman | lol |
| 22:37.58 | mgroman | oops wrong channel, sorry |
| 22:38.35 | *** part/#asterisk mgroman (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 22:39.37 | mvanbaak | gheh |
| 22:39.39 | wishes | gmornin |
| 22:39.44 | mvanbaak | EPIC FAIL |
| 22:40.15 | wishes | that wasnt an epic fail |
| 22:40.23 | mvanbaak | and I'm off to bed |
| 22:40.38 | wishes | and epic fail would be like pasting your asterisk config with sip provider and passwords |
| 22:40.39 | mvanbaak | well, this is: http://dl.ziza.ru/other/072008/08/fail/026_fail.jpg |
| 22:40.54 | jaytee | epic fail? I'm thinking aparthied |
| 22:42.00 | mvanbaak | latero all |
| 22:42.06 | jaytee | nite |
| 22:43.16 | *** join/#asterisk macros73_ (n=cs@c-67-165-65-27.hsd1.pa.comcast.net) |
| 22:43.44 | macros73_ | I think my idea crashed FreeNode. |
| 22:45.43 | wishes | lies |
| 22:47.33 | macros73_ | anyone here setup asterisk failover in the past? |
| 22:48.08 | macros73_ | or should I look at openser if I'm serious about it? |
| 22:48.24 | *** part/#asterisk ROARJ (n=joseph_2@201.216.146.69) |
| 22:53.18 | macros73_ | anyone even alive here? |
| 22:53.20 | *** join/#asterisk NovceGuru (n=NovceGur@oh-65-40-70-180.sta.embarqhsd.net) |
| 22:54.22 | Strom_C | no |
| 22:54.24 | Strom_C | we're all dead |
| 22:54.29 | Strom_C | please check the number and dial again |
| 22:54.38 | Strom_C | if you need help, hang up and then dial your operator |
| 22:54.41 | Strom_C | this is a recording |
| 22:54.46 | Strom_C | two one three four |
| 22:55.04 | Qwell | What, no SIT? |
| 22:55.05 | lesouvage | macros73: there is a lot of info available for implementing failover mechanism. It depends on the scale of the implemenation you are thinking about. |
| 22:55.14 | lesouvage | I'm (still) not dead ;-) |
| 22:55.19 | Strom_C | Qwell: "this is a recording" negates the need for an SIT :) |
| 22:55.30 | Qwell | I see |
| 22:57.58 | *** join/#asterisk GwaiLo (n=xabbu@unaffiliated/gwailo) |
| 23:02.04 | *** join/#asterisk Zemmad (n=calibisi@cpe-66-69-164-90.sw.res.rr.com) |
| 23:04.35 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:06.15 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 23:18.28 | *** join/#asterisk jtknapp (n=skip@216.160.91.74) |
| 23:18.44 | *** part/#asterisk jtknapp (n=skip@216.160.91.74) |
| 23:20.14 | *** join/#asterisk railsmunky (n=nick@5acd4099.bb.sky.com) |
| 23:21.58 | *** join/#asterisk Che-Anarc (i=Che-Anar@bb-82-108-13-47.ukonline.co.uk) |
| 23:28.12 | *** join/#asterisk schemathings (n=schemath@68-244-39-103.area3.spcsdns.net) |
| 23:31.09 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 23:32.34 | *** join/#asterisk GwaiLo (n=xabbu@unaffiliated/gwailo) |
| 23:33.14 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 23:34.14 | MatBoy | when I have my trunk settings allright (sip trunk) and I can use this sipp account normally using a sip client, and when I use it as a trunk with asterisk and the password seems to be not OK, in what direction should I seach ? |
| 23:35.50 | mookid | What are the limits on the number of concurrent users/sessions ? |
| 23:36.40 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 23:37.23 | mookid | How many conference calls / how many people in each conference can it handle? |
| 23:37.51 | mookid | has anyone tested this? |
| 23:45.00 | rycar | =~centos52bug |
| 23:45.11 | rycar | ~centos52bug |
| 23:45.11 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 23:47.14 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 23:47.31 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 23:49.39 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:49.46 | CanWood | Hey folks. I'm writing a dialplan and it uses Authenticate(1234). Is there a way to disable the playing of the "Please enter your password now" recording? (other than deleting or replacing the recording itself) |
| 23:51.57 | deeperror | CanWood, do you want to just read in digits? |
| 23:52.40 | CanWood | yes, but deny access to the next line if it's wrong (not in a db or file) |
| 23:53.50 | deeperror | How about 2 lines Read() and GotoIf(${FOO}=db(this/that/key)?here:there) |
| 23:53.59 | outtolunc | edit app_authenticate.c search for prompt = "agent-pass"; |
| 23:54.10 | deeperror | haha or that |
| 23:55.01 | CanWood | both good approaches. I'll explore deeperror's suggestion first though, so I don't need to merge code changes each time I recompile. Thanks for the suggestion outtolunc |
| 23:55.37 | CanWood | I like how Authenicate() handles it and am trying to avoid reinventing the wheel with teh "try three times then deny" logic |
| 23:56.05 | *** join/#asterisk Great_Anta_Baka (i=c636cafa@gateway/web/ajax/mibbit.com/x-3b974c22ab3c7502) |
| 23:56.17 | outtolunc | it wouldn't be hard at all to mod the app to passin the file you want played |
| 23:56.43 | deeperror | outtolunc, but then if you upgrade you have to patch the code |
| 23:56.50 | CanWood | I agree it wouldn't be tough, but it would mean that any t.... what he said :) |
| 23:57.02 | outtolunc | that or get it commited to trunk |
| 23:57.17 | outtolunc | this is how asterisk grows |
| 23:57.34 | CanWood | I noo much of a n00b to presume that my suggestions should be comitted |
| 23:58.29 | outtolunc | suggestions never get commited, patches (code) does, if it passes <G> |
| 23:58.38 | outtolunc | er +t |
| 23:59.20 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 23:59.20 | *** mode/#asterisk [+o bkruse] by ChanServ |
