| 00:06.08 | *** join/#asterisk Alfio (n=Amunoz@190.94.59.204) |
| 00:10.46 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 00:14.59 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 00:22.40 | *** join/#asterisk propellerhead (n=yogurt2u@host234.200-82-120.telecom.net.ar) |
| 00:24.38 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 00:49.28 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 00:57.59 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 01:02.45 | *** join/#asterisk bobJR (n=bob@adsl-153-104-205.tys.bellsouth.net) |
| 01:05.59 | *** join/#asterisk rgrasell (n=ryan@adsl-67-125-138-209.dsl.frsn02.pacbell.net) |
| 01:06.43 | rgrasell | My asterisk server isn't listening on ubp 5060. Can anyone help? |
| 01:07.18 | rgrasell | whenever a client tries to connect, it times out |
| 01:08.30 | Psychobilly | netstat -nltpu | grep 5060 |
| 01:08.32 | manxpower | What is the output of "netstat -an | grep 5060" on the server |
| 01:08.57 | rgrasell | nothing. no ports |
| 01:09.23 | manxpower | what is the output of "module unload chan_sip.so" and "module load chan_sip.so" use pastebin if the output is more than a line or two. |
| 01:09.25 | manxpower | ~pb |
| 01:09.26 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 01:09.38 | jaytee | yeah!!!! beat him!!!! hit him again!!!!! |
| 01:09.46 | manxpower | you might need "core" in front of "module" depending on the Asterisk version |
| 01:09.49 | jaytee | oops, wrong window ...sorry! |
| 01:10.31 | rgrasell | whoa: [Aug 2 18:10:15] NOTICE[6051]: chan_sip.c:16929 reload_config: Unable to load config sip.conf |
| 01:10.44 | rgrasell | its at /etc/asterisk/sip.conf |
| 01:11.55 | Psychobilly | check its permissions |
| 01:12.14 | rgrasell | who should own it? |
| 01:12.39 | Psychobilly | the user that runs asterisk |
| 01:12.52 | manxpower | pastebin the sip.conf |
| 01:12.56 | rgrasell | ok |
| 01:14.00 | rgrasell | http://pastebin.com/m3c1c4af6 |
| 01:14.07 | manxpower | obviously there could be a permissions issue as well. Check that |
| 01:14.10 | rgrasell | its very empty... i sit missing anything? |
| 01:14.17 | rgrasell | I want al my users in users.conf |
| 01:14.51 | manxpower | sorry, I can't help you at all with users.conf. Almost nobody except GUIs use it. |
| 01:15.08 | *** join/#asterisk xpot-mobile (n=james@75-162-174-215.slkc.qwest.net) |
| 01:15.42 | rgrasell | but that shouldnt ,atter for this whole deal should it? |
| 01:16.31 | Psychobilly | check perms |
| 01:17.46 | rgrasell | Psychobilly: do you know what the default user is for asterisk in the ubuntu repo version? or how can i check? |
| 01:18.04 | Psychobilly | ps aux | grep asterisk |
| 01:18.22 | Psychobilly | i think its using the user asterisk or nobody |
| 01:18.47 | rgrasell | Psychobilly: ok thanks let me check |
| 01:19.18 | rgrasell | yea its asterisk |
| 01:19.44 | Psychobilly | so chown asterisk /etc/asterisk/* and try again |
| 01:20.03 | rgrasell | yea i did that, i think its working. let me check... no errors so far |
| 01:20.42 | rgrasell | ok its working! thanks!! :) |
| 01:21.08 | *** part/#asterisk rgrasell (n=ryan@adsl-67-125-138-209.dsl.frsn02.pacbell.net) |
| 01:24.07 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 01:25.44 | *** join/#asterisk Kumbang (n=ketoprak@rusnas.paume.itb.ac.id) |
| 01:26.41 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 01:42.41 | *** join/#asterisk xpot-mobile (n=james@75-162-175-207.slkc.qwest.net) |
| 01:59.16 | hardwire | ahoy! |
| 02:00.14 | *** join/#asterisk wwalker (n=wwalker@72.249.1.66) |
| 02:05.51 | jaytee | ahoy? um....dude, this is the #asterisk channel, not the Royal Navy |
| 02:06.11 | box2 | YARRRR |
| 02:12.01 | *** join/#asterisk tfrew (n=tfrew@pool-72-82-51-228.rcmdva.east.verizon.net) |
| 02:25.26 | EugenMayer | hello. Iam using asterisk on a debian lenny with an AVM firtzcard 2.1 to sip out with ISDN. Currently i really have a bad sound quality on outbound. so what i hear is ok, but the other on the line hears me pretty bad |
| 02:25.30 | EugenMayer | the server is an newer ATOM |
| 02:25.33 | EugenMayer | with 2GB ram and nearly idling all the time |
| 02:25.36 | EugenMayer | Load avg is 0.1 |
| 02:39.00 | *** join/#asterisk riyadc (n=riyadc@riyadc.secure.citechco.net) |
| 02:48.37 | *** join/#asterisk Thazza (n=me@ntechm.lnk.telstra.net) |
| 02:55.58 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 02:59.09 | manxpower | what SIP device? |
| 02:59.24 | EugenMayer | Ekiga |
| 02:59.29 | EugenMayer | manxpower: Ekiga |
| 02:59.42 | manxpower | Are you using BRIStuff or mISDN? |
| 02:59.59 | EugenMayer | capi |
| 03:00.49 | manxpower | I'd search the mailing lists. BRI support has historically not been great, but that may have changed in the past year or so. DAHDI even has some BRI support now. |
| 03:00.51 | manxpower | ~mailingist |
| 03:01.03 | manxpower | ~mailinglist |
| 03:01.04 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 03:01.35 | EugenMayer | manxpower: what iam a bit confused |
| 03:01.38 | EugenMayer | BRI = capi ? |
| 03:02.16 | *** join/#asterisk Alfio (n=Amunoz@190.94.59.204) |
| 03:09.53 | *** join/#asterisk C4colo (n=DJpyro@66.185.111.33) |
| 03:12.28 | C4colo | if I have multiple Dial() commands to try in a row, how to I play constant music on hold without gaps |
| 03:13.13 | C4colo | if I do "StartMusicOnHold(class)" before the Dial commands then it just goes to ringing when the ringing progress packet comes in ... or I can use the m flag but then it gaps between each dial command |
| 03:16.35 | [TK]D-Fender | C4colo: dial them inside of a containing local channel |
| 03:17.09 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 03:17.36 | C4colo | ah |
| 03:17.41 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:20.55 | C4colo | ok, that will work, I'll have to move some stuff around, but it will do what I need |
| 03:20.58 | C4colo | thanks [TK]D-Fender |
| 03:30.36 | manxpower | C4colo: what tech (sip, analog zap, pri?) |
| 03:31.08 | C4colo | all sip |
| 03:32.33 | C4colo | and the local channel isn't working like I expected it to ... the channel variables I set before running the Dial command are not accessible |
| 03:33.17 | C4colo | I think I might have found something though |
| 03:33.20 | manxpower | try the /n to to the local channel. |
| 03:34.16 | C4colo | ah |
| 03:34.37 | C4colo | that might be it, when I was testing I saw the agi script was doing an answer at the beginning of the script even though it doesn't need to |
| 03:34.40 | C4colo | the call was already answered |
| 03:37.23 | C4colo | hmm,I took the answer out of the agi script and tried both with and without /n ... and either way the channel variables set before the dial(local/...) command are not being passed through |
| 03:37.36 | C4colo | I think the answer command in the agi script may have been causing the music on hold to stop |
| 03:37.51 | C4colo | I'm going to try without the Dial(local/...) and just use a Goto() |
| 03:42.40 | *** join/#asterisk BeeBuu (n=beebuu@125.95.249.88) |
| 03:45.20 | manxpower | what are you trying to accomplish? |
| 03:45.35 | *** join/#asterisk korcan (n=korcan@76.234.139.200) |
| 03:49.42 | [TK]D-Fender | C4colo: They are accessible, go read the CHANNELVARIABLES docs that tell you how inheretance works |
| 03:51.24 | C4colo | ok, trying that |
| 03:51.24 | *** topic/#asterisk by Corydon76-dig -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.2 (2009/08/02), Asterisk 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #aster |
| 03:52.46 | [TK]D-Fender | \o/ |
| 03:52.54 | [TK]D-Fender | Asterisk 8.6.7.5.3.0.9.! |
| 03:58.23 | C4colo | thanks, that worked |
| 04:01.39 | manxpower | MoH usually stops when the channel is answered. |
| 04:02.20 | [TK]D-Fender | And subsequent dials... well.. don't even have to go there I hope... |
| 04:05.47 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 04:05.59 | Qwell | #aster? |
| 04:06.09 | Qwell | Corydon76-dig: it ate the end of the topic |
| 04:07.25 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:07.33 | Corydon76-dig | Qwell: fantastic |
| 04:07.47 | *** topic/#asterisk by Corydon76-dig -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.2 (2009/08/02), Asterisk 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 04:07.52 | Qwell | I think you can remove the "Asterisk" for 1.6.0.10 |
| 04:08.11 | Corydon76-dig | Why? 1.6.0.11 isn't released yet |
| 04:08.31 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.2 (2009/08/02), 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 04:08.33 | Qwell | like that |
| 04:09.01 | Qwell | we can take -addons off soon :D |
| 04:09.19 | Corydon76-dig | Addons won't go anywhere until 1.6.3, remember? |
| 04:09.31 | Qwell | "soon" in the grand scheme of things.. |
| 04:09.40 | Corydon76-dig | Meh |
| 04:10.36 | Corydon76-dig | I'll be dead soon, in the grand scheme of things |
| 04:11.12 | manxpower | what's going to happen to -addons? |
| 04:11.33 | Corydon76-dig | It's getting merged into the core, so it won't be a separate package anymore |
| 04:12.13 | manxpower | but what about the whole "we don't want 2 codebases" thing? |
| 04:12.41 | Corydon76-dig | Eh? |
| 04:12.41 | box2 | in the grands scheme of things, maybe everyone forgot |
| 04:14.38 | [TK]D-Fender | Thought there were still licensing reasons for not including in the core including format_mp3 <- |
| 04:15.02 | manxpower | Corydon76-dig: As I understood it, Digium wanted to same codebase of the Open Source release of Asterisk and Asterisk Business Edition, but things like MP3 and spandsp and stuff like that could not be licensed into both versions. |
| 04:15.21 | Corydon76-dig | There are, but menuselect has given us the ability to set them as "off" by default |
| 04:15.38 | manxpower | Corydon76-dig: Ah! A very good point. |
| 04:16.26 | Corydon76-dig | Besides, ABE is mostly going bye-bye |
| 04:17.48 | manxpower | BTW, I accepted a job with http://nyigc.com/ |
| 04:21.29 | [TK]D-Fender | Corydon76-dig: Thought it was a legal distribution requirement, not just the fact you could choose not to include in compile |
| 04:29.23 | Corydon76-dig | [TK]D-Fender: nope, GPL difference |
| 04:29.33 | *** join/#asterisk martyndev (i=be18869a@gateway/web/freenode/x-vlxmdatdjorsvtsh) |
| 04:29.37 | martyndev | hi every1 :D |
| 04:29.56 | martyndev | i have a problem with my PRI, and i dont know if is asterisk problemor is a provider problem.. |
| 04:29.57 | [TK]D-Fender | Corydon76-dig: What makes them compatible for co-inclusion now? |
| 04:30.40 | martyndev | the pri is OK withzttol and I can send outbound calls ok.. but the inbound calls i can't recive :(. and in asterisk pri debug span not recive any message :( |
| 04:30.48 | martyndev | what do you think that is happened' |
| 04:30.48 | martyndev | ? |
| 04:31.09 | Corydon76-dig | [TK]D-Fender: that their build is disabled by default |
| 04:31.35 | [TK]D-Fender | Corydon76-dig: That satisfies being "semi-bundled"? |
| 04:33.37 | Corydon76-dig | [TK]D-Fender: They were always eligible for inclusion for the public GPL-Asterisk |
| 04:34.05 | Corydon76-dig | but they will never be distributed with ABE |
| 04:34.34 | martyndev | any comments about problem ? |
| 04:34.52 | [TK]D-Fender | Corydon76-dig: I'd rather have seen addon as a separate package only required by ABE and included in main then.... |
| 04:35.27 | [TK]D-Fender | martyndev: Comments? You've shown no debug for your problem and only the vagues of descriptions. |
| 04:35.32 | Corydon76-dig | [TK]D-Fender: I have no idea what you meant to say, but it's gibberish to me |
| 04:35.42 | [TK]D-Fender | martyndev: Don't ask for an autopsy without showing us the BODY |
| 04:36.25 | [TK]D-Fender | Corydon76-dig: have OS * INCLUDE the bits from Addons altogether and only separately package them for ABE |
| 04:36.52 | [TK]D-Fender | Corydon76-dig: As you are saying is "coming". IMO should have been done long ago based on the terms you've described |
| 04:37.11 | hardwire | waves to Corydon76-dig and [TK]D-Fender |
| 04:37.14 | hardwire | \\o |
| 04:37.16 | hardwire | o// |
| 04:37.19 | hardwire | \o/ |
| 04:37.28 | martyndev | [TK]D-Fender: what exactly do you need about it ?.. i'm using asterisk 1.2.14 with te410p digium card and e1 PRI |
| 04:37.41 | [TK]D-Fender | martyndev: DEBUG |
| 04:38.14 | [TK]D-Fender | martyndev: and 1.2.14 = ANCIENT |
| 04:38.15 | *** join/#asterisk mikea (n=mike@c-98-208-76-102.hsd1.ca.comcast.net) |
| 04:38.32 | hardwire | is surprised martyndevs TDM card hasn't lit itself on fire due to the ancientness of the asterisk/driver revision. |
| 04:38.49 | mikea | Any chance there's a GUI patch for FFA? |
| 04:38.50 | [TK]D-Fender | martyndev: Don't even dream of suppot for 1.2 unless you're on the latest, and if its a "bug" or "added feature needed", neither is happening. |
| 04:38.59 | Corydon76-dig | [TK]D-Fender: Yeah, basically I mentioned it casually in a meeting one day, and it was done less than 24 hours later |
| 04:39.13 | [TK]D-Fender | Corydon76-dig: You need to attend more of these meetings! |
| 04:39.24 | martyndev | right [TK]D-Fender . yes is old version, but i have other 6 pris works in the same asterisk .. |
| 04:39.42 | [TK]D-Fender | martyndev: Same answer. Get debug. |
| 04:39.52 | Corydon76-dig | [TK]D-Fender: I attend those meetings every week. It simply was a question that I asked, expecting a no and got a yes, instead |
| 04:40.06 | mikea | Corydon76-dig, What was the question? |
| 04:40.13 | *** join/#asterisk blkry (n=blkry@nc-65-41-180-145.dyn.embarqhsd.net) |
| 04:40.14 | Corydon76-dig | [TK]D-Fender: I expected legal reasons, and I got "Well, I don't know why we couldn't do that." |
| 04:40.33 | martyndev | right [TK]D-Fender , this morning i try enabling "pri debug span 6" in CLI Asterisk.. is that correct way to enable ? |
| 04:40.46 | Corydon76-dig | mikea: why we couldn't put addons back into the Asterisk package and release it as a single package |
| 04:41.01 | mikea | ah, nice |
| 04:41.07 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:41.21 | [TK]D-Fender | martyndev: its a start |
| 04:41.24 | Corydon76-dig | It specifically fixes versioning and build problems |
| 04:41.29 | mikea | Corydon76-dig, I was hoping it was "Why can't we add FFA to asterisk-gui". ;-) |
| 04:42.04 | Corydon76-dig | mikea: I'm sorry, I can't speak to vaporware |
| 04:42.11 | mikea | yeah, it was always kind of a pain to download and install both separately |
| 04:42.24 | mikea | when you usually need both |
| 04:42.44 | Corydon76-dig | The addons stuff is already done. What may or may not happen with asterisk-gui isn't. |
| 04:43.03 | martyndev | other suggest to see that debug [TK]D-Fender ? |
| 04:43.15 | mikea | Corydon76-dig, I gotcha, I was being factitious |
| 04:43.54 | [TK]D-Fender | martyndev: MOVE |
| 04:44.37 | martyndev | sorry [TK]D-Fender ? |
| 04:45.25 | [TK]D-Fender | martyndev: Go get the debug... |
| 04:54.25 | Qwell | Corydon76-dig: I was highly amused by that |
| 04:54.43 | Qwell | "Why can't -addons just be moved into Asterisk?" "uhh... uhh..." |
| 04:55.26 | *** join/#asterisk cpurn (n=cpurn@eth755.vic.adsl.internode.on.net) |
| 04:58.03 | manxpower | I don't think the Future Farmers of America would be a good addition to Asterisk. |
| 04:58.47 | mikea | manxpower, Free Fax for Asterisk :( |
| 04:58.56 | manxpower | never heard of it |
| 04:59.01 | Qwell | ~ffa |
| 04:59.02 | infobot | it has been said that ffa is ffa stands for free for all, or not CTF |
| 04:59.05 | Qwell | stupid bot |
| 04:59.08 | Qwell | ~faxforasterisk |
| 04:59.09 | infobot | methinks faxforasterisk is Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 04:59.17 | manxpower | nifty. |
| 04:59.22 | Qwell | manxpower: You hadn't heard of that? really? O.o |
| 04:59.23 | mikea | I just want to setup an inbound DID and have it email me tifs when a fax comes in :-( |
| 04:59.35 | manxpower | Qwell: I've been out of the Asterisk world for a while. |
| 04:59.47 | mikea | I can probably invest a ton of time into writing an app to do it, but I thought it'd just kind of work |
| 05:00.12 | mikea | It's cool, I'll muck with it tomorrow. |
| 05:00.15 | mikea | cya guys later |
| 05:02.07 | *** join/#asterisk shinao1 (n=shinao1@41.219.238.99) |
| 05:03.28 | martyndev | [TK]D-Fender: look "PRI Error: We think we're the CPE, but they think they're the CPE too." |
| 05:03.51 | [TK]D-Fender | martyndev: then stop being CPE |
| 05:04.21 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 05:05.58 | cpurn | is away: (Auto-Away after 10 mins) [BX-MsgLog On] |
| 05:06.27 | WindowsUser | bitchin |
| 05:07.46 | martyndev | what do you recomended [TK]D-Fender ? |
| 05:07.52 | [TK]D-Fender | martyndev: then stop being CPE <---------------- |
| 05:08.03 | martyndev | i try with pri_net and i recive this PRI Error: We think we're the network, but they think they're the network, too. |
| 05:08.39 | [TK]D-Fender | martyndev: I don't see any debug, nor a description of what you're connecting to. No configs. |
| 05:09.48 | manxpower | martyndev: one of three things. 1) Telco has a loopback on the line. Tell them to take it off. 2) You have a loopback on the line. Tell yourself to take it off. 3) you have pri_cpe set when it should be pri_net. |
| 05:10.03 | manxpower | martyndev: your problem is 1) or 2) |
| 05:10.09 | [TK]D-Fender | bets on 1/2 |
| 05:10.43 | martyndev | my provider is here and have a equimenet doind a loop with my asterisk |
| 05:10.49 | martyndev | checking the config is ok manxpower |
| 05:12.55 | [TK]D-Fender | martyndev: Stop F-ing looking back. All * is doing is TALKING TO ITSELF |
| 05:13.12 | manxpower | martyndev: I just told you the issue is 1) or 2). |
| 05:13.20 | [TK]D-Fender | martyndev: Congratulations, the card works. Now STOP |
| 05:13.43 | martyndev | [TK]D-Fender: right, let me talk with provider to stop the tests... |
| 05:14.13 | manxpower | Asterisk needs to figure out the line is looped back and tell the user about it. |
| 05:14.14 | martyndev | but, how can i know if is pri_cpe or pri_net [TK]D-Fender manxpower ? |
| 05:14.18 | drmessano | Any recommendations for a per-minute email to fax service? |
| 05:14.25 | manxpower | martyndev: you can't until the loop is taken off the line, |
| 05:14.39 | manxpower | in fact you can't do anything until the loop is taken off the line. |
| 05:18.25 | [TK]D-Fender | martyndev: If you're dealing with a telco provided circuit, then you are CPE |
| 05:21.03 | manxpower | [TK]D-Fender: maybe he doesn't understand what a loopback is. |
| 05:37.21 | jplank | its amazing how much faster you get things done, when you have a clear precise list of what your doing, and not working off the top of your head |
| 05:41.19 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 05:42.16 | [TK]D-Fender | bed time, later all |
| 05:43.12 | *** join/#asterisk b14ck (n=quassel@cpe-24-24-136-239.socal.res.rr.com) |
| 05:43.46 | b14ck | i've just installed asterisk (doing some more testing) on a centos 5.3 box (64-bit), and now, for some reason, after installing dahdi, and dahdi-tools, i cant use the dahdi command in asterisk |
| 05:43.49 | b14ck | wtf! |
| 05:44.12 | b14ck | dahdi_test runs fine, cat /proc/dahdi/1 shows Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: Linux26) 1" (MASTER) |
| 05:44.16 | b14ck | any ideas? |
| 05:45.56 | jplank | is dahdi missing from the cli? |
| 05:46.10 | b14ck | ya |
| 05:46.19 | b14ck | but nowhere else |
| 05:46.26 | b14ck | seems to be running fine on the system |
| 05:46.42 | jplank | I've only ever seen that with a screwed up system.conf or chan_dahdi.conf |
| 05:47.09 | jplank | was it a clean dahdi installation, or an upgrade from zaptel? |
| 05:47.26 | b14ck | a clean asteriski 1.6.1.1 install and a clean dahdi install |
| 05:47.29 | b14ck | on a brand new box |
| 05:48.25 | jplank | I can't say for sure since I don't use 1.6, but the only time I've ever seen that (assume dahdi is installed properly) is with a screwed up system.conf |
| 05:48.38 | b14ck | my system.conf only had loadzone=us |
| 05:48.40 | b14ck | :( |
| 05:49.08 | jplank | theres your problem |
| 05:50.03 | jplank | oh no cards installed? |
| 05:50.08 | b14ck | nope |
| 05:50.10 | b14ck | only dahdi_dummy |
| 05:50.53 | jplank | what about chan_dahdi.conf? |
| 05:51.19 | b14ck | totally normal |
| 05:51.24 | jplank | stupid question, but is it readable by asterisk? |
| 05:51.26 | b14ck | just the defaults |
| 05:51.28 | b14ck | yes |
| 05:51.36 | b14ck | atm its runing as root, ive been troubleshooting random problems with it |
| 05:52.30 | b14ck | bah |
| 05:52.34 | b14ck | i should just sleep on it |
| 05:52.35 | b14ck | ^^ |
| 05:52.36 | jplank | what if you do a reload chan_dahdi.so |
| 05:53.10 | b14ck | lemme check, recompilnig atm |
| 05:58.00 | drmessano | Did you install dahdi before compiling asterisk? |
| 05:58.13 | b14ck | no |
| 05:58.20 | b14ck | asterisk, then dahdi-linux, then dahdi-tools |
| 05:58.20 | drmessano | Thats your problem |
| 05:58.25 | drmessano | dahdi first |
| 05:58.27 | jplank | *better question |
| 05:58.28 | b14ck | really? |
| 05:58.29 | drmessano | then asterisk |
| 05:58.35 | drmessano | Yes |
| 05:58.35 | b14ck | cool--didnt know that before |
| 05:58.57 | drmessano | Asterisk cant compile for dahdi since it's missing as a dependency |
| 05:59.05 | b14ck | that would make sense |
| 05:59.22 | drmessano | Explains your problem 100% |
| 06:00.07 | b14ck | ok, trying again |
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| 08:16.51 | TSM | can asterisk decode inband dtmf? |
| 08:20.18 | *** join/#asterisk TommyBotten (n=tommy@217-14-12-26-dhcp-osl.bbse.no) |
| 08:22.04 | *** join/#asterisk Polysics (n=luca@li24-50.members.linode.com) |
| 08:22.07 | Polysics | hello |
| 08:22.34 | Polysics | i'm learning a lot with a configuration from scratch... make samples isn't that good after all :-) |
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| 08:26.22 | TommyBotten | Polysics: Make samples is just meant to show the diversity of asterisk |
| 08:26.27 | TommyBotten | not give you a good config :) |
| 08:28.35 | Polysics | so am i right in starting over after saving the samples somewhere? |
| 08:29.06 | Polysics | had a few bad minutes with trying to find out why my CLI wasn't working as expected any more |
| 08:29.32 | TommyBotten | Absolutely. That is how I made my startup config |
| 08:33.06 | Polysics | since i won't be using any hardware, i hope i can get at leasta playback :-) |
| 08:36.58 | TommyBotten | not any hardware at all? ;) |
| 08:37.36 | BeeBuu | how can i get the callerid from DTMF? |
| 08:37.50 | Polysics | no, i plan on using a VOIP provider for calls |
| 08:37.58 | Polysics | so far i'm focusing on internal works |
| 08:41.26 | TommyBotten | BeeBuu: Not sure I understand you question. |
| 08:41.32 | TommyBotten | Ah, |
| 08:41.33 | TommyBotten | cool |
| 08:42.06 | BeeBuu | TommyBotten: callerid by FSK i can get it ,but by DTMF i can't |
| 08:47.52 | Polysics | erm... my SIP clients cannot connect any more :-( |
| 08:48.12 | Polysics | sip.conf is by the book |
| 08:54.23 | TommyBotten | Polysics: Check the CLI for debug information |
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| 09:03.42 | Polysics | what's strange is that nothing pops up in teh console |
| 09:03.45 | Polysics | nothing at all |
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| 09:09.28 | TommyBotten | use |
| 09:09.31 | TommyBotten | core set verbose 8 |
| 09:09.32 | TommyBotten | core set debug 8 |
| 09:09.37 | tzafrir_laptop | BeeBuu, what version of Asterisk? |
| 09:09.40 | TommyBotten | That will give you somewhat debug informatiion |
| 09:09.41 | TommyBotten | -i |
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| 09:11.12 | tzafrir_laptop | BeeBuu, also: when is the DTMF sent? before a ring? after the first ring? |
| 09:11.29 | tzafrir_laptop | if before a ring: is it after a polarity reversal? |
| 09:13.48 | defswork | anyone aware of any scams involving calls to Cuba ? |
| 09:13.54 | Hatrix | Hello, i am new in a company supporting various clients which use asterisk, we have troubles with one client specifically, it is a call-center, it runs right now 1.4.26, but we have all kinds of strange problems, one I hope will be fixed in revision 209955 is segfaults in app_queue. But what me worries is this: i was monitoring the system from the morning, nearly no load, it was nicely running with 0.6 cpu, now it's 3 hours later, we h |
| 09:14.20 | defswork | I've been asked to look into a proprietary pbx for company that have had thousands of calls to cuba appear on their bill |
| 09:15.49 | BeeBuu | tzafrir_laptop: 1.4.10 |
| 09:16.23 | BeeBuu | tzafrir_laptop: after the first ring. |
| 09:16.42 | tzafrir_laptop | This is only supported as of asterisk 1.6.0 |
| 09:17.12 | BeeBuu | tzafrir_laptop: can this help ?http://www.openvox.com.cn/bbs/viewtopic.php?t=324 |
| 09:17.24 | tzafrir_laptop | https://issues.asterisk.org/view.php?id=6683 |
| 09:18.22 | BeeBuu | tzafrir_laptop: someone give out a new wtdcm.c,does it work? |
| 09:18.41 | BeeBuu | wctdm.c |
| 09:18.51 | tzafrir_laptop | no, different issue |
| 09:19.13 | BeeBuu | tzafrir_laptop: are you sure? |
| 09:20.29 | tzafrir_laptop | I suppose |
| 09:20.56 | tzafrir_laptop | Driver issues may be related to the case with a polarity-related dtmf |
| 09:21.15 | Hatrix | And i forgot to mention that the CPU in top is showing right now 239 % ... |
| 09:21.21 | BeeBuu | tzafrir_laptop: em~~~ thanks,i will try it out |
| 09:21.40 | tzafrir_laptop | anybody aware of a backport of that feature to 1.4 ? |
| 09:21.49 | tzafrir_laptop | It would actually help me right now |
| 09:24.51 | TommyBotten | Hatrix: Strace the application to see what it does... if you manage :) |
| 09:26.11 | Hatrix | TommyBotten: I would, but this is production, I can not interrupt their working right now, but I am really worried, the cpu usage (from ps) climbs and is now round-about 33% ... |
| 09:26.34 | Hatrix | thing is that i have a hard-time reproduce this in a lab scenario, it drives me a lil bit of crazy |
| 09:27.10 | TommyBotten | Is it a 24/7 system? |
| 09:27.16 | Hatrix | (the cpu usage from top is between 120 and 260 %) ... whyever there is a difference ... |
| 09:27.33 | TommyBotten | And strace can be attached to a running process. Just be careful as it easily spins out of control |
| 09:28.50 | Hatrix | TommyBotten: no, i can do it starting from 9pm, then there are no calls anymore. I know that I can attach strace to running process, but it slows the damn thing down like a groundhog, and I do not want to give some thread-locking-problems a better chance to occure .. |
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| 09:30.11 | TommyBotten | true, true |
| 09:31.56 | TommyBotten | in that case, I would just monitor and prepare for restarts/extingiush fires etc. |
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| 09:37.16 | Hatrix | yeah, this is what I'm doing ... I have as well stopped verbose logging, now it seems to be more stable, at least the load went down a bit ... |
| 09:41.10 | Hatrix | 18 active calls / 41 active channels, should these numbers slow down a system considerably? seems not really much for me? |
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| 10:01.12 | tzafrir_laptop | BeeBuu, still here? |
| 10:01.31 | tzafrir_laptop | I backported it, but I don't have a place to test it |
| 10:04.27 | BeeBuu | tzafrir_laptop: yes,what's up? |
| 10:04.35 | BeeBuu | tzafrir_laptop: any suggestion? |
| 10:04.56 | tzafrir_laptop | ok. I have a backport to 1.4.21 . now trying to backport to 1.4.10 |
| 10:05.24 | BeeBuu | and what's the result? |
| 10:08.27 | tzafrir_laptop | bah, simply building 1.4.10 with a more recent gcc takes some workarounds... |
| 10:09.17 | BeeBuu | tzafrir_laptop: thanks for your help |
| 10:11.40 | BeeBuu | tzafrir_laptop: i had tried the wctdm.c in http://www.openvox.com.cn/bbs/viewtopic.php?t=324 ,it doesn't work~~~ |
| 10:12.33 | tzafrir_laptop | what card do you have? |
| 10:14.25 | BeeBuu | S400P |
| 10:14.58 | BeeBuu | zttool show: Wildcard S400P Prototype Board 1 |
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| 10:19.01 | BeeBuu | tzafrir_laptop: thanks a lot. i got to go~~~~ see you . |
| 10:19.22 | BeeBuu | will be back after 20 hours. |
| 10:19.50 | agx | is there a way to know if the "h" extension is called by a normal call or by a zombie channell? i'm having problem using GROUP_COUNT() inside of it. |
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| 10:22.32 | tzafrir_laptop | gives up building 1.4.10 . Too broken |
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| 10:29.05 | kraptv | This is slightly unrelated to asterisk: Has anyone had a great success with a cheap mobile device and a softphone? The fixed softphones are so expensive and the quality doesn't seem to be great. |
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| 10:29.10 | kraptv | examples would be the SIP softphones for iPhone. |
| 10:29.35 | tzafrir_laptop | kraptv, chan_bluetooth? |
| 10:30.03 | tzafrir_laptop | an analog ATA with a DECT phone? |
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| 10:31.05 | Cynagen | question |
| 10:31.20 | Cynagen | I'm looking to set up an Asterisk box |
| 10:31.52 | Cynagen | all i have are connexant modems, can they function as line receivers? |
| 10:32.14 | Cynagen | I'll be plugging a seperate phone into my unlocked PAP2 router |
| 10:32.46 | kraptv | yeah. an analog ATA/DECT is super cheap.. I was hoping something I could do transfers, etc. with. |
| 10:32.48 | Cynagen | also, can a dial have a script attached to it? |
| 10:33.18 | Cynagen | like say i dial a 10 digit, can i have it run a script to do a wget? |
| 10:34.26 | Cynagen | the documentation says there's the option for automation |
| 10:34.58 | defswork | that's not automation |
| 10:35.08 | Cynagen | can i lay out what my plan is? |
| 10:35.10 | defswork | and you can automate calls no problem |
| 10:35.17 | Cynagen | so you understand what i'm trying to accomplish? |
| 10:35.47 | Cynagen | the entire VoIP side of this is relatively easy to get set up and going |
| 10:36.06 | Cynagen | i have a google voice number, and as we all know, they just shut the open SIP down |
| 10:36.17 | Cynagen | and made it work with gizmo |
| 10:36.25 | Cynagen | now i don't wanna sign up with gizmo |
| 10:36.43 | Cynagen | i've got an IPKall number forwarded to my house, right now, direct to my PAP2 device |
| 10:37.05 | Cynagen | now my idea is the swap out the pap2 for an Asterisk box |
| 10:37.36 | Cynagen | handle all calls coming in from ipkall, to the pap2 through asterisk on a local network |
| 10:37.51 | Cynagen | so i have my FXS ports there |
| 10:38.17 | Cynagen | now, i have a path from DID to phone in the house |
| 10:38.34 | Cynagen | google voice forwards to the DID, and i have a local number |
| 10:38.39 | Cynagen | however, calling out will be a problem |
| 10:38.47 | Cynagen | i'd have to hit a webpage every time |
| 10:39.31 | Cynagen | i've done my homework (and a lot of packet sniffing) to get Google Voice's POST information to emulate the page effectively enough to connect a call without actually ever opening a browser |
| 10:39.34 | Cynagen | all done through wget |
| 10:39.57 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/session) |
| 10:40.46 | Cynagen | now, the idea is that i pick up the phone on the other end of the PAP2, dial as normal, the pap2 fowards the call to Asterisk, asterisk does the wget calls to sign me in, and then dial, and have it come in on my DID number, and from there, when asterisk receives the incoming from ipkall, finally connect it with my phone |
| 10:41.08 | yziquel | hi. i can manage to make SIP phone calls to external phones. However, after 15 or 30 secs, the connection drops off. Is this trouble usual? |
| 10:41.12 | Cynagen | really effin complicated, but it's all automated |
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| 10:46.48 | Cynagen | so can what i'm planning on doing be done? |
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| 11:08.24 | Hatrix | what is the best way to test call transfers? I have a bunch of scripts emulating client calls with sipp and as well create agents and take the calls. the scripts for the agents make logoffs, logons, and accept a random amount of calls. |
| 11:08.43 | Hatrix | But How do i Fit in call transfers in this scenario preferable with sipp? |
| 11:11.22 | yziquel | Anyone had calls automatically dropping off after 15 secs or so? |
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| 11:34.25 | Cynagen | so anyone got any idea if my google voice idea will work? |
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| 12:10.47 | ascii` | hello do you know any egyptian provider that provide free fixed/landline numbers for incoming calls? |
| 12:14.35 | Cynagen | ascii`: http://www.google.com/search?q=egyptian%20did%20voip |
| 12:14.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:17.45 | ascii` | Cynagen, found no good candidate in both voipproviderslist and voipinfo |
| 12:18.07 | ascii` | i tough that perhaps somebody has direct experience |
| 12:18.24 | Cynagen | sorry |
| 12:18.28 | Cynagen | would help more if i could |
| 12:18.45 | ascii` | nope, thanks :) |
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| 12:22.50 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:23.04 | ariel_ | Morning |
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| 12:44.51 | phr3ak | anyone using openvox card for incoming fax? |
| 12:45.54 | dovid | openvox sux IMHO |
| 12:45.58 | dovid | gave me to many issues |
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| 12:51.43 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 12:51.45 | Polysics | hello |
| 12:51.57 | Polysics | got a problem, an extension is there but it doesnt get found |
| 12:52.06 | Polysics | !pastebin |
| 12:52.09 | Polysics | ?pastebin |
| 12:52.10 | Polysics | oops |
| 12:52.12 | Polysics | :-) |
| 12:52.18 | Polysics | how do i get pastebin info? :-) |
| 12:53.00 | RypPn | ~pb |
| 12:53.01 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 12:54.01 | Polysics | http://pastebin.com/m6f539b70 |
| 12:54.11 | Polysics | here is my sip.conf and extensions.conf |
| 12:54.30 | Polysics | what i ma trying to achieve is dial 9 on my softphone and hear the "hello world" |
| 12:54.36 | Polysics | i am on 1.6.1.1 stable |
| 12:54.48 | Polysics | shouldn't this simple setup work? |
| 12:54.59 | phr3ak | can it detect the fax all time or only in the first several seconds ? |
| 12:55.37 | [TK]D-Fender | Polysics: No. |
| 12:55.40 | Polysics | my softphone registers all right, at least from "sip show peers" it is there |
| 12:55.55 | Polysics | [TK]D-Fender, that's a good thing to know in itself :-) |
| 12:56.08 | Polysics | why not, or wher eto look, please? |
| 12:57.04 | [TK]D-Fender | Polysics: Because you aren't looking where your calls are going |
| 12:57.36 | Alfio | Polysics you need to allow the context phones to dial into your incoming context |
| 12:57.59 | Polysics | so my SIP context should be "incoming"? |
| 12:58.56 | Polysics | or is it something else? |
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| 12:59.40 | Polysics | no, just using "incoming" as sip context doesn't allow the extension to go through anyway |
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| 13:00.28 | Polysics | i'm baffled |
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| 13:02.18 | Polysics | sip show settings does show "incoming" as context |
| 13:02.19 | Alfio | Polysics try to include the context phones under your incoming context in your extensions.conf. include=> phones |
| 13:02.23 | [TK]D-Fender | Polysics: You're phone is looking in [phones, and your exten is in [incoming] |
| 13:02.31 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:03.10 | [TK]D-Fender | Polysics: This is the equivalent of * telling you "There are no socks in the top drawer!". It does not know or care that they are in the 2nd |
| 13:03.48 | [TK]D-Fender | Polysics: s/you're/Your |
| 13:04.02 | Polysics | adding include => phones at the beginning of [incoming] raises a complaint on reload |
| 13:05.38 | *** join/#asterisk tAnk (i=tank@the.matrix.has-you.net) |
| 13:05.41 | [TK]D-Fender | Polysics: And I see no reason you should be using INCLUDEs nor your revised configs |
| 13:06.38 | Polysics | [TK]D-Fender, then what is the "recommended" solution, please? |
| 13:07.04 | [TK]D-Fender | Polysics: Make the context you tell your phones to use and put your extens there. |
| 13:07.32 | [TK]D-Fender | Polysics: Start making new contexts as you begin creating different layers of access, IVR's, etc |
| 13:08.14 | Polysics | like this? http://pastebin.com/m4c86bb73 |
| 13:09.18 | Polysics | ok, that solves it |
| 13:09.46 | Polysics | what is the use for the "context=" directive in sip.conf, then? after all, every phone has its context |
| 13:10.03 | Polysics | is it used if there is no specified context in the phone definition? |
| 13:10.10 | [TK]D-Fender | Polysics: For calls that are NOT phone a phone you've defined. |
| 13:10.17 | [TK]D-Fender | from* |
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| 13:12.28 | pablus | hi!, i have a question about analogic and digital phones.., is possible to reuse all those phones if i change the pbx by asterisk? |
| 13:13.00 | [TK]D-Fender | pablus: Depends what phones. |
| 13:13.22 | [TK]D-Fender | pablus: And whether its with the money for the hardware it'll take to drive them |
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| 13:20.48 | stix | Hi guys. I have this php-script containing: $channel = shell_exec("asterisk -rx 'core show channels' |grep '^SIP/$extension'"); - When I run the script from the command-line it works, but when I run it from a browser I get: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 13:20.55 | stix | what is going on there? |
| 13:21.37 | [TK]D-Fender | stix: Permissions |
| 13:21.46 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:22.00 | stix | [TK]D-Fender, which permissions? Not manager.conf ? |
| 13:22.09 | [TK]D-Fender | stix: Probably running as "apache" <- |
| 13:22.15 | [TK]D-Fender | FILE permissions |
| 13:22.27 | stix | oh okay, I'll check |
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| 13:23.55 | mrkiko | Hi all!!! I have to do a foolish thing. I need to let asterisk say to me the typed number, but with personalizzed digit files. How can I do this? |
| 13:24.16 | stix | asterisk is owning the file and httpd is running as user asterisk |
| 13:25.04 | [TK]D-Fender | mrkiko: Go make your own recordings for the digits and a dialplan loop to go digit by digits and playback your personalized recordings |
| 13:25.16 | yziquel | when making an outbound call, the voice drops to zero, while the SIP session is still active. What should I do? |
| 13:25.37 | mrkiko | [TK]D-Fender. I've done the first part. I just can't immagine the cycle... |
| 13:26.34 | [TK]D-Fender | mrkiko: Set endcounter = length of input. set position to 1. split off char{X} from input. |
| 13:26.41 | [TK]D-Fender | mrkiko: This is programming 101. |
| 13:27.11 | stix | mrkiko, you can also create a custom language (where the digit-files are stored) and then change the language to that in your dialplan. |
| 13:28.13 | mrkiko | oh... I'll search on the book may be how to create such a language. Yes, I understand the logic of the cycle, I may write it in C in a second. but I don't immagine it in the asterisk language... |
| 13:29.17 | pablus | TK D-Fender : searching in internet, i found a hardware for this task --> ATA is the name .. is possible to reuse all the digital and analogic phones with this? or depens?? |
| 13:30.22 | [TK]D-Fender | mrkiko: "core show application gotoif" <- |
| 13:30.29 | *** join/#asterisk maskas (n=maskas@179.111.181.58.dynamic.max.com.pk) |
| 13:30.31 | [TK]D-Fender | mrkiko: This ain't Raw-Cat Sigh Hence |
| 13:30.46 | maskas | hello, could anyone help me with a pri related problem? |
| 13:30.51 | [TK]D-Fender | pablus: NO. Digital phones speak their own special language. |
| 13:30.57 | ariel_ | pablus: analog phones yes, even using what is called channel bank. But digital no |
| 13:31.24 | [TK]D-Fender | pablus: So I told you that for those you MIGHT be able to reuse though depending on the MODEL |
| 13:31.33 | [TK]D-Fender | pablus: And the equipment to let you do so is not cheap |
| 13:31.42 | mrkiko | thank you |
| 13:32.15 | Loki | Hey. I have a question. What is the best way of setting outbound callerids? I have mutiple DID |
| 13:32.24 | maskas | I send an outgoing call via a pri over a certain channel, the remote end signals to asterisk to move the call to another channel. The call connects but audio is on way. I can hear called party but they can not hear me. This is using asterisk1.4.26 |
| 13:32.30 | maskas | also when the calls hangs up, the following error is shown: WARNING[11188]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found?, and chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/3 on span 1 |
| 13:32.35 | [TK]D-Fender | Loki: "core show application set", "core show function CALLERID" |
| 13:32.35 | Loki | 's, and want to be able to use them without making mutiple extenions for a user |
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| 13:39.40 | pablus | TK D-fender, so if not possible, can I to replace digital phones by IP phones..? |
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| 13:41.07 | [TK]D-Fender | pablus: Of course you can buy something else. Why would owning a bicycle stop you from replacing it with a car? |
| 13:41.07 | [TK]D-Fender | pablus: and I asked you WhAT MODEL of digital phone you have twice now. |
| 13:41.38 | pablus | TK D-fender .. our digital phones are Meridian... |
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| 13:42.08 | [TK]D-Fender | pablus: that is a BANNER brand, not a model. Might help to know how MANY you have to support as well |
| 13:42.16 | vk4akp | Hi |
| 13:42.23 | vk4akp | Anyone help with this error please? |
| 13:42.26 | vk4akp | [Aug 3 23:09:44] WARNING[18410]: chan_sip.c:12282 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '6016d9b45ee5e72e22966a971c9d5427@122.104.157.145'. Giving up. |
| 13:42.53 | pablus | TK D-fender .. ops, let me see |
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| 13:49.07 | stix | [TK]D-Fender, have you any other suggestions regarding the permissions-issue? |
| 13:49.32 | [TK]D-Fender | stix: Verify the files are there, prove the user things run as. |
| 13:49.50 | [TK]D-Fender | stix: make bash script to do your call for this |
| 13:52.09 | E-bola | I have features.conf automon => *3 and added wW to my dial command, as well as having DYNAMIC_FEATURES=automon in [Global] yet if i press *3 during a call absolutely nothing happens |
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| 13:52.19 | E-bola | Anybody got an idea how i can debug that? |
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| 13:53.16 | stix | Files are there and other things are working in the script. What do you mean by "prove the user things run as"? |
| 13:53.16 | [TK]D-Fender | E-bola: Did you restart * since changing these? |
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| 13:53.39 | [TK]D-Fender | stix: What part of that wasn't clear as day? Go call a bash script that DUMPS what user things are running as, file & folder locations, permissions, etc |
| 13:54.26 | E-bola | [TK]D-Fender: yep |
| 13:54.42 | E-bola | I even tried binding *3 to a button on my snom phone, since i read somewhere u had to press it very quickly |
| 13:54.48 | E-bola | This is on asterisk 1.4 |
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| 13:55.27 | [TK]D-Fender | E-bola: You do. And "binding to a button" does not tell me the phone works the way you'd like it to. Prove your DTMF mode is correct (#1), then do it BY HAND |
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| 13:57.22 | stix | [TK]D-Fender, consider this: we are on different levels here and we are not all speaking native english. So no, it is not cleas as day... |
| 13:58.06 | [TK]D-Fender | stix: I said check permissions. Prove the user the script is running as. Do that in your script.. same with the file dump. |
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| 13:59.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:05.41 | yziquel_ | seems my asterisk uses port 3030. how come? how can i set this up? |
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| 14:09.36 | [TK]D-Fender | yziquel_: Grep your configs |
| 14:10.36 | stix | [TK]D-Fender, thank you. Asterisk was running as root somehow - better check my init script. |
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| 14:14.15 | E-bola | [TK]D-Fender: Works now |
| 14:14.15 | E-bola | Hmm is there no settings at all for it? |
| 14:14.16 | [TK]D-Fender | E-bola: Settings for what? |
| 14:14.23 | E-bola | i see its running a sox bin with lots of parameters, is that not defineable anywhere? |
| 14:14.42 | E-bola | automon recording |
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| 14:15.15 | [TK]D-Fender | E-bola: By default I'd say nothing likely beyond "vi res_features.c" |
| 14:15.38 | E-bola | hmm ok, i found a macro that supposedly lets u control it ur self, couldnt get it to work so far though |
| 14:15.58 | E-bola | i added apps => *9,caller,Macro,apprecord under [applicationmap] in features.conf |
| 14:16.08 | E-bola | but like *3 before, *9 does nothing |
| 14:16.26 | *** join/#asterisk voipamn (n=voipman@194.42.129.177) |
| 14:16.28 | voipamn | sup all |
| 14:17.03 | voipamn | I'm having random issues with IAX2, i dont see any problems with iax2 debug... just retrying to send-tx. |
| 14:17.39 | voipamn | any idea where to start with ? |
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| 14:18.05 | wwalker | anyone successfully used WaitForSilence from an AGI? It works as expected in the dialplan, but seems to not work in my AGI (for an outbound call originated via AMI) |
| 14:18.37 | ariel_ | retrying to send-tx? are you hearing any issues? do you have dropped calls? It could be your network connection. |
| 14:18.43 | *** join/#asterisk mattboll (n=mattboll@lns-bzn-51f-81-56-151-195.adsl.proxad.net) |
| 14:18.53 | mattboll | hi |
| 14:19.12 | [TK]D-Fender | E-bola: Feel free to actually show something |
| 14:19.49 | voipamn | ariel_ any chance is a fw on one side ? |
| 14:20.11 | voipamn | i have checkpoint 500W at one end... and i dont' think it can handle IAX very well.. but that's just a guess |
| 14:20.49 | [TK]D-Fender | wwalker: its an app like any other. Your AGI must not be processing the call under the same circumstance |
| 14:21.04 | E-bola | [TK]D-Fender: http://pastebin.com/m49706e27 |
| 14:21.06 | ariel_ | voipamn: yes it could be a fw issue, but are you dropping calls? missing sound, jitters? |
| 14:21.19 | voipamn | hmm.. a bit.. |
| 14:21.27 | E-bola | Sory for not showing pastebin stuff right away, but i was just looking for a way to try and debug it myself, instaid of having you spent time looking at it :) |
| 14:21.54 | voipamn | ariel_, when a call fails, i see it "arriving" to the other server but cannot be esteblished.. |
| 14:22.04 | mattboll | I've configured a grandstream gxp 280 with asterisk, I can talk but I can't heard any idea how to find what is wrong ? (I've tried with 2 gxp280 same problem so it may be a configuration problem) |
| 14:22.23 | [TK]D-Fender | E-bola: Well you were commenting on changes you made and there was nothing for us to comment on. And in here you didn't include everything. try again |
| 14:22.50 | [TK]D-Fender | mattboll: maybe some detail about where * and your phone are.... |
| 14:23.07 | yziquel_ | [TK]D-Fender: do not know what to grep for. I still do not understand why my calls go out on port 3030. |
| 14:23.16 | ariel_ | voipamn: when these calls can't be esteblished you need to setup iax2 debug and see what it says, if you have any debugs post them to pastebin and let us see them |
| 14:23.16 | E-bola | [TK]D-Fender: I've included all the lines i consider relevant, what other parts would need to see? |
| 14:23.27 | [TK]D-Fender | yziquel_: that would be RTp <- |
| 14:23.34 | ariel_ | mattboll: natting issue? are they in the same network or over the internet? |
| 14:23.36 | [TK]D-Fender | yziquel_: look at rtp.conf |
| 14:23.49 | [TK]D-Fender | E-bola: I do not see what the CALL processes. |
| 14:23.50 | voipamn | alright will do |
| 14:24.31 | mattboll | asterisk server is far away, but I can talk (and hear me from an other phone) it is just that I don't hear |
| 14:25.18 | yziquel_ | [TK]D-Fender: my rtp.conf is really basic: http://paste.lisp.org/display/84666 |
| 14:25.28 | [TK]D-Fender | mattboll: .... far away? Could you please describe something useful about the NETWORKING between the two... |
| 14:25.43 | voipamn | voipamn, should i trunk=yes in both machines ? and jitterbuffer=yes as well? |
| 14:25.54 | [TK]D-Fender | yziquel_: What protocol is your call in? |
| 14:26.04 | E-bola | [TK]D-Fender: you mean you want to see the dial cmd? |
| 14:26.14 | yziquel_ | [TK]D-Fender: SIP over RTP/UPD |
| 14:26.20 | [TK]D-Fender | E-bola: everything that sets up the call |
| 14:26.41 | E-bola | exten => 131,1,Dial(SIP/Bruger1,,WwTt) |
| 14:26.46 | voipamn | oops ariel_* |
| 14:26.47 | E-bola | thats all, im just testing so far |
| 14:26.50 | [TK]D-Fender | yziquel_: Well the outbound source port is not under your control the only ones that are, are the iNBOUND ports |
| 14:26.53 | voipamn | ariel_, should i trunk=yes in both machines ? and jitterbuffer=yes as well? |
| 14:26.54 | mattboll | phone -> switch -> internet -> asterisk |
| 14:27.06 | [TK]D-Fender | E-bola: ENTIRE F-ING OUTPUT! |
| 14:27.09 | mattboll | I've got some other phones running |
| 14:27.33 | [TK]D-Fender | mattboll: read : |
| 14:27.35 | [TK]D-Fender | ~sipnat |
| 14:27.36 | infobot | [~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 |
| 14:27.36 | E-bola | [TK]D-Fender: hehe dude i must be clueless :) You want the output from the console while when the call connects? |
| 14:27.38 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:27.48 | [TK]D-Fender | E-bola: Entire CLI output |
| 14:28.27 | *** join/#asterisk Defraz (n=T0tal@c72co-edge-router.fuzecore.com) |
| 14:28.32 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 14:28.43 | ariel_ | voipamn: trunk=yes only if you have a timing device install like a digium board or ztdummy |
| 14:29.13 | E-bola | [TK]D-Fender: i doubt it will help any, ile try to dig around some more on my own first, no need to steal your time unless absolutely necesary :) |
| 14:29.37 | [TK]D-Fender | E-bola: You doubt? I'm asking because I EXPECT to see a screw-up in there. |
| 14:30.07 | [TK]D-Fender | E-bola: And if you knew what to look for you'd already have solved your problem. |
| 14:30.16 | E-bola | http://pastebin.com/m4fe1ffc0 |
| 14:30.20 | [TK]D-Fender | E-bola: Your asking us for help means you should be showing us. |
| 14:30.54 | [TK]D-Fender | E-bola: that is the ENTIRE call? |
| 14:31.18 | E-bola | yes |
| 14:31.20 | [TK]D-Fender | E-bola: pastebin your entire dialplan. |
| 14:32.14 | yziquel_ | [TK]D-Fender: are you saying that when I make an outbound call, i have no control over which UDP port is chosen for RTP? |
| 14:32.45 | E-bola | http://pastebin.com/m6a6fa203 |
| 14:32.47 | [TK]D-Fender | yziquel_: You have no control over the source port period. only the destination |
| 14:35.35 | [TK]D-Fender | E-bola: restarted since making that addition? |
| 14:35.50 | E-bola | Yep, i restart after all changes |
| 14:35.54 | mattboll | [TK]D-Fender: it works for other phones (and i've just re-rechecked my config) |
| 14:36.11 | [TK]D-Fender | mattboll: time for you to show SIP debug for your failed call... |
| 14:36.11 | E-bola | im abit uncertain about how to use the DYNAMIC_FEATURES setting |
| 14:36.13 | [TK]D-Fender | ~pb |
| 14:36.13 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 14:36.16 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 14:36.29 | mattboll | [TK]D-Fender: there is no failed call⦠|
| 14:36.43 | [TK]D-Fender | mattboll: I'd consider a lack of voice "failed" |
| 14:36.45 | mattboll | It is just that I don't hear anything |
| 14:36.52 | mattboll | ok |
| 14:37.35 | *** join/#asterisk bongfrog (n=quassel@c-98-245-211-157.hsd1.co.comcast.net) |
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| 14:38.49 | [TK]D-Fender | E-bola: So your caller can use automon off that exten, but not apps? |
| 14:39.06 | mattboll | http://pastebin.com/d69fc5ca9 |
| 14:39.19 | mattboll | it is a call to 33 which is my echo-test |
| 14:39.49 | [TK]D-Fender | mattboll: SIP DEBUG <---- |
| 14:39.52 | E-bola | [TK]D-Fender: yep, thats how it is atm |
| 14:40.06 | E-bola | automon works fine, i cant get anything defined under [applicationmap] to work |
| 14:44.13 | [TK]D-Fender | E-bola: http://pastebin.com/m6a6fa203 <- in this PB I see you didn't name it right |
| 14:45.10 | [TK]D-Fender | E-bola: it is your most recent PB. the earlier version was named right |
| 14:45.20 | *** join/#asterisk clintc (n=clintc@n128-227-127-135.xlate.ufl.edu) |
| 14:45.22 | [TK]D-Fender | E-bola: Confirm, reload, retest |
| 14:45.23 | E-bola | didnt name what right? the applicationmap feature? |
| 14:45.23 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:45.49 | mattboll | http://pastebin.com/d66d7675b |
| 14:45.53 | [TK]D-Fender | E-bola: [tester] <- not the macro |
| 14:46.00 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 14:46.27 | E-bola | ohhh ya i know, i tried changing the app to a goto in the features.conf |
| 14:46.44 | E-bola | but shouldnt it more or less log something in CLI regardless of what its set to? i'd even take an error |
| 14:46.51 | E-bola | its just not responding to *9 at all |
| 14:46.59 | E-bola | i tried mapping it to something else as well, no go either |
| 14:47.02 | E-bola | only automon works |
| 14:47.49 | [TK]D-Fender | E-bola: Ok, I'm out of ideas for now... |
| 14:47.55 | [TK]D-Fender | mattboll: Do you hear the playback? |
| 14:48.13 | mattboll | [TK]D-Fender: no :( |
| 14:48.36 | [TK]D-Fender | mattboll: check your firewalls, and describe the networking on your server side |
| 14:50.09 | mattboll | asterisk is on a vserver, i don't have access to every firewalls but it works for the other phones |
| 14:51.06 | [TK]D-Fender | mattboll: any of those other phones on the same network as your GS'? |
| 14:51.17 | [TK]D-Fender | mattboll: Pastebin yuor server's sip.conf masking only passwords |
| 14:51.49 | mattboll | the same switch (even tried the same cable) |
| 14:51.55 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
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| 14:54.50 | mattboll | there is 550 line, so I copied the most relevant part : http://pastebin.com/m734734bb |
| 14:55.00 | mattboll | 112 works and 115 does'nt |
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| 14:59.55 | E-bola | [TK]D-Fender: solved it :) |
| 15:00.03 | E-bola | My co-worker had created the dialplan from scratch |
| 15:00.04 | [TK]D-Fender | mattboll: Ok, I see no real config difference. PB the [general] section |
| 15:00.13 | [TK]D-Fender | E-bola: and...? |
| 15:00.19 | E-bola | he had [global] instaid of [globals] you and I both missed that i guess :) |
| 15:00.32 | [TK]D-Fender | E-bola: So you weren't showing me the actual configs? |
| 15:00.34 | E-bola | i found out, since it worked when i did a manual set on the extension it worked |
| 15:00.50 | E-bola | [TK]D-Fender: Ofcourse i was u just missed it, http://pastebin.com/m6a6fa203 |
| 15:01.07 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 15:01.12 | [TK]D-Fender | E-bola: Hrm... oh well |
| 15:01.53 | E-bola | :) |
| 15:01.57 | afink | you're probably going to have to cry yourself to sleep tonight aren't you TK? |
| 15:01.58 | E-bola | who cares, it works now hehe |
| 15:02.04 | [TK]D-Fender | afink: nope |
| 15:02.14 | afink | =P |
| 15:02.15 | [TK]D-Fender | afink: I sleep the sleep of the righteous :p |
| 15:02.54 | coppice | don't the righteous want their sleep back? |
| 15:03.43 | [TK]D-Fender | coppice: No, the void has been left with guilt and they don't want to compound it by trying to take it back :) |
| 15:04.19 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:04.19 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:04.20 | h00k | So, my company was testing a Linksys phone system for a client and found a bug in their firmware, now I have been given the task to research Asterisk to potentially deploy for our client |
| 15:04.37 | [TK]D-Fender | mattboll: Peer audio RTP is at port 172.16.2.116:5004 <- doesn't look too good here... |
| 15:06.04 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:06.06 | vk4akp | Nite Phreaks. |
| 15:07.56 | [TK]D-Fender | swats vk4akp with a copy of "2600" |
| 15:08.00 | [TK]D-Fender | vk4akp: Damn right@ |
| 15:08.09 | [TK]D-Fender | :p |
| 15:08.18 | jake[work] | any idea what would cause this? http://www.insperia.com/bad-dtmf.wav i'm on 1.4.25.1 and using a TE110P |
| 15:09.11 | [TK]D-Fender | jake[work]: Interrupt loss, bad EC, etc |
| 15:09.50 | jake[work] | would it show on outbound too? outbound sounds clear |
| 15:09.51 | *** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 15:11.24 | [TK]D-Fender | jake[work]: Verify the rest. |
| 15:16.06 | Gnarfy | Heya, still struggling a bit with CDR. My question is: if i set cdr.conf -> unanswered=yes, i get all the info i want to see for each dial, but these outgoing dials do not take over the accountcode or the custom field i've set in the dialplan, anyway to get it to do this? |
| 15:18.37 | [TK]D-Fender | Gnarfy: Show us |
| 15:19.35 | mattboll | [TK]D-Fender: Yeah it's good now, thank you so much ! |
| 15:19.41 | jake[work] | <PROTECTED> |
| 15:20.02 | [TK]D-Fender | mattboll: What did you find? |
| 15:20.12 | [TK]D-Fender | jake[work]: nothing I'm aware of |
| 15:20.59 | mattboll | audio rtp port, with 41000 (like an other phone) it works |
| 15:21.55 | jake[work] | dahdi_test comes back 99.9% + |
| 15:22.10 | Gnarfy | The relevant extensions.conf bit and cdr result: http://pastebin.com/m763fa234 |
| 15:22.12 | jake[work] | <PROTECTED> |
| 15:22.20 | dovid | [TK]D-Fender: Ever use mtr ? |
| 15:22.32 | jake[work] | what else are you suggesting? EC? |
| 15:22.34 | [TK]D-Fender | mattboll: Good to hear |
| 15:22.49 | [TK]D-Fender | dovid: Nope |
| 15:23.06 | dovid | great tool to test over the internet if you have QoS issues |
| 15:31.06 | Gnarfy | [TK]D-Fender: it might just be my lack of knowledge on how CDR works, i know each dial gets its own uniqueid - but it would be great if i can make those outgoing dials get to take over the accountcode from the dialplan... Or is this just how that option works and will i have to mess about with ForkCDR/ResetCDR or a custom AGI call after each dial? |
| 15:31.44 | [TK]D-Fender | Gnarfy: I'm grey on this... |
| 15:31.53 | Gnarfy | ah oke :) |
| 15:33.12 | Gnarfy | I'll go an read the sourcecode, maybe that will give me a clue :) |
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| 15:40.07 | *** part/#asterisk MACscr (n=Mark@98.214.100.212) |
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| 15:50.15 | beek | is back... tanned, rested and ready. |
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| 15:56.50 | yziquel_ | is 'port' differen from 'bindport' in sip.conf? Available information on the web is sort of confusing. |
| 15:57.12 | jake[work] | [TK]D-Fender: http://insperia.com/bad-dtmf.wav voice seems to be fine. |
| 15:57.20 | jake[work] | (this is a new new file) |
| 15:58.36 | [TK]D-Fender | yziquel_: "port" is for peers and specifies the TARGET's destport. BINDPORT defines the port * LISTENS on. |
| 15:58.44 | [TK]D-Fender | is out to lunch... |
| 15:59.05 | manxpower | neither option is required for most people |
| 15:59.28 | jake[work] | anyone else have any ideas? http://insperia.com/bad-dtmf.wav |
| 15:59.50 | yziquel_ | manxpower: but setting them to 5060 does no harm, i guess. |
| 16:00.09 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 16:00.10 | manxpower | yziquel_: unless the remote side uses a high port as the source port. |
| 16:00.20 | javb | i've done Asterisk implementations up to 120 sip ua, and up to 50 concurrent sip channels. With 4 T1, and FULL load. (96 TSM/SS7/T1 PRI calls at the same time) All this JUST with Asterisk pbx 1.4 on CentOS 5.0. My question, what are the best practices and considerations on bigger implementations? Any resources around there? |
| 16:00.34 | manxpower | do a "netstat -an" (windows of *nix) and notice the source port of connections are not what you would expect. This is normal |
| 16:01.01 | *** part/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 16:01.06 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 16:01.15 | surfdue | anyone firmiliar with the old lucent phone systems |
| 16:01.18 | manxpower | since port= only applies to peers, usually the far side would not have a high port, but if it goes thru NAT then it could. |
| 16:01.27 | eppigy | TRABAJO |
| 16:06.54 | coppice | jake[work]: something is muting the DTMF. Each little blip in that file is a few milliseconds of real DTMF |
| 16:07.12 | jake[work] | do you think it's before asterisk? |
| 16:07.32 | jake[work] | hear how it sounds broken? |
| 16:07.53 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:08.25 | coppice | yes. something is detecting the start of the DTMF and muting the channel |
| 16:08.47 | jake[work] | coppice: ok - tnx |
| 16:09.32 | tzafrir_laptop | jake[work], where is that coming from? |
| 16:09.42 | tzafrir_laptop | voip or pstn? |
| 16:09.45 | jake[work] | pstn |
| 16:09.52 | *** join/#asterisk galeras (n=galeras@186.80.190.38) |
| 16:10.05 | jake[work] | it's going though PRI -> Avaya -> IVR -> Asterisk |
| 16:10.09 | tzafrir_laptop | try ztmonitor / dahdi_monitor to listen to it before it gets into asterisk |
| 16:10.16 | manxpower | coppice: Can you tell if it's an issue of inband DTMF over a compressed codec or would that sound different? |
| 16:10.28 | tzafrir_laptop | assuming it does not get muted by the PRI adapter itself |
| 16:10.32 | jake[work] | the output is from the Monitor app |
| 16:10.52 | tzafrir_laptop | Monitor is recorded inside Asterisk |
| 16:10.56 | jake[work] | yep |
| 16:10.57 | coppice | it could be DTMF through a VAD, but DTMF through a codec just sounds like DTMF |
| 16:11.01 | manxpower | jake[work]: you are missing part of the diagram |
| 16:11.07 | tzafrir_laptop | ztmonitor / dahdi_monitor is before it enters Asteriskj |
| 16:11.12 | jake[work] | yep |
| 16:11.14 | voipamn | i'm trying to connect my two machine with sip instead of iax, but it false can anyone have a look http://pastebin.com/d4822e63d |
| 16:11.20 | voipamn | tzafrir_laptop :) |
| 16:11.22 | jake[work] | manxpower: what part? |
| 16:11.28 | sumasuma | in the Dial command, what is the unit of timeout ? |
| 16:11.39 | manxpower | voipamn: you should ask on the channel for your GUI |
| 16:11.39 | tzafrir_laptop | seconds |
| 16:11.44 | sumasuma | milliseconds or seconds ? |
| 16:12.03 | manxpower | jake[work]: Avaya -> IVR -> Asterisk Your calls magically get to an IVR? |
| 16:12.20 | sumasuma | tzafrir_laptop: thanks |
| 16:12.22 | voipamn | manxpower, i still asterisk :) |
| 16:12.26 | voipamn | it's even |
| 16:12.28 | manxpower | ~freepbx |
| 16:12.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:12.39 | jake[work] | manxpower: it's digital tie lines |
| 16:12.58 | *** join/#asterisk xa0z (n=Interex@75-129-230-28.dhcp.mtvr.il.charter.com) |
| 16:13.00 | jplank | manxpower, did anyone in my office get in contact with you yet? |
| 16:13.11 | manxpower | jake[work]: Dude, if you want a fix you need to tell us details. A T-1? An E-1? a CAS T-1? A PRI? E&M/Wink? |
| 16:13.42 | xa0z | If I have a DTA/ATA such as the pap2, and I have a single extension but two channels, will the phone "beep" for callwaiting still or do I need to do something else to have that happen? |
| 16:13.48 | manxpower | jplank: yup! Their messaged ended up in my spam folder (all important messages end up in my spam folder Ugh). |
| 16:13.59 | jake[work] | <PROTECTED> |
| 16:14.54 | manxpower | jake[work]: Is the IVR is on Asterisk and not the Avaya? |
| 16:14.55 | jplank | Let me know what you have all that stuff done, or if they bust your balls about anything. Once thats complete, I have some stuff I need to ship down to you (hard phones) |
| 16:15.14 | jake[work] | the IVR is a custom IVR (not asterisk or avaya) using dialogic boards |
| 16:15.21 | manxpower | jplank: Awesome |
| 16:15.39 | manxpower | looks over his glasses at jake[work] |
| 16:15.49 | jake[work] | hey - it's not mine |
| 16:15.56 | jake[work] | or my recommendations |
| 16:15.57 | jplank | oh, email your IP when you get a chance, I'm sure its dynamic, but I could change the ACL as needed |
| 16:17.05 | jake[work] | i was just wondering if there was anything in asterisk or dahdi that could cause that problem |
| 16:17.39 | manxpower | jplank: done |
| 16:18.00 | manxpower | jake[work]: unlikely |
| 16:18.12 | manxpower | Not when using a "tie line" |
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| 16:23.28 | jplank | got it |
| 16:23.44 | jplank | (FYI you do come up as spam for me) |
| 16:26.59 | *** join/#asterisk ethicx (n=ethicx@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 16:27.59 | xa0z | :/ |
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| 16:41.20 | kn0x | any reason running asterisk with realtime priority would crash a system? |
| 16:41.43 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 16:42.20 | tzafrir_laptop | kn0x, crash it (panic and alike) or hang it? |
| 16:42.30 | tzafrir_laptop | Also: what kernel version? |
| 16:42.48 | tzafrir_laptop | does it still answer pings? |
| 16:42.50 | kn0x | 2.6.26-2-amd64 / debian patchs |
| 16:42.59 | kn0x | and full system reboot with no logs |
| 16:43.01 | tzafrir_laptop | is the manager interface still usable? |
| 16:43.03 | brah | Hi. I have Digium Wildcard TDM400P, and autobooting zaptel.ko, ztdummy.ko and wctdm.ko crashes my system. Loading them manually however doesn't |
| 16:43.09 | jake[work] | tzafrir_laptop: what format is stream.raw in? i'm trying to read it in windows |
| 16:43.31 | kn0x | tzafrir_laptop: no kernel panic or the like |
| 16:43.55 | tzafrir_laptop | kn0x, hmm.... 2.6.26 should not allow a real-time process to take 100% CPU time after a while |
| 16:44.03 | tzafrir_laptop | maybe you have two such processes? |
| 16:44.40 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 16:44.50 | tzafrir_laptop | jake[work], raw stream? What was the original codec used? |
| 16:44.51 | *** join/#asterisk many (n=many@dslb-188-098-011-162.pools.arcor-ip.net) |
| 16:45.00 | kn0x | tzafrir_laptop: there are other 'RT' processes if thats what you mean |
| 16:45.04 | jake[work] | just used dahdi_monitor |
| 16:46.11 | kn0x | tzafrir_laptop: the problem went away either after I turned off highpriority=yes or stopped using chanspy |
| 16:46.11 | kn0x | I don |
| 16:46.11 | kn0x | I don't know which was the culprit |
| 16:46.33 | tzafrir_laptop | jake[work], sox -s -w -r 8000 -c 1 stream.raw stream.wav |
| 16:46.36 | kn0x | and other than setting up a lab sometime in the feature.. i can't experiment on the box becuz its production |
| 16:46.43 | jake[work] | tnx |
| 16:47.01 | tzafrir_laptop | jake[work], or use latest dahdi_monitor from svn, that is able to write wav files |
| 16:49.07 | jake[work] | tnx |
| 16:49.47 | jake[work] | ok - prb is definately in asterisk |
| 16:50.04 | jake[work] | dahdi_monitor audio is clean |
| 16:52.45 | *** join/#asterisk asasas (n=asasasas@c-68-61-183-129.hsd1.mi.comcast.net) |
| 16:57.55 | manxpower | jake[work]: how did you determine that there was a DTMF problem? |
| 16:58.39 | jake[work] | a huge amount of cutomer complaints. plus we tested it and continually get bounced to the invalid number route |
| 16:59.35 | manxpower | You heard the blips telepathically or via some other means? |
| 16:59.57 | manxpower | I meant how did you determine it was blips and not some other issue? |
| 16:59.58 | jake[work] | blips? |
| 17:00.02 | jake[work] | oh |
| 17:00.21 | jake[work] | it's a straight Read cmd |
| 17:00.25 | manxpower | (11:08:13 AM) coppice: yes. something is detecting the start of the DTMF and muting the channel |
| 17:00.32 | jake[work] | yeah |
| 17:00.52 | jake[work] | when i use dahdi_monitor, no blips |
| 17:01.00 | jake[work] | when I use Monitor - blips |
| 17:01.35 | manxpower | OK. |
| 17:01.43 | jake[work] | the Read cmd takes in 14 digits. but it seldom gets the correct digits |
| 17:02.15 | jake[work] | i checked mantis and didn't see anything obvious in this version |
| 17:02.52 | manxpower | Did you call in with a SIP phone to hear the recorded file or did you play the file in an external media player? |
| 17:03.00 | jake[work] | external media player |
| 17:06.15 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:12.44 | *** join/#asterisk kpettit (n=keith@63.99.176.242) |
| 17:14.54 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 17:23.36 | tzafrir_laptop | jake[work], so Asterisk is to blame, here |
| 17:24.03 | jake[work] | i think so. if the audio sounds good on dahdi_monitor |
| 17:24.31 | tzafrir_laptop | If you enable debug of a high enough value you'll probably even see the not to enable and disable muting |
| 17:25.20 | jake[work] | i'm less worried that the audio is weird. dtmf detection is not reliable |
| 17:25.32 | Qwell | jake[work]: describe your setup in full |
| 17:29.27 | jake[work] | HP DL380 TE digium card (using wct4xxp) |
| 17:29.46 | jake[work] | asterisk 1.4.25.1 |
| 17:30.24 | jake[work] | dahdi 2.1.0.2 |
| 17:30.38 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 17:30.57 | buttons840 | i will never encounter the same unique id twice will i? |
| 17:31.13 | Qwell | buttons840: you can across instances. ie; if you restart Asterisk |
| 17:31.25 | jake[work] | Qwell: i can give you more... what else are you looking for? |
| 17:31.33 | Qwell | jake[work]: call flow, etc |
| 17:31.37 | jake[work] | ok |
| 17:32.08 | *** join/#asterisk s14ck (n=s14ck@190.77.92.226) |
| 17:32.11 | jake[work] | can we start from the box itself (considering we know the audio is good from there)? |
| 17:32.47 | Qwell | start at chan_dahdi, explain the rest |
| 17:32.52 | jake[work] | ok |
| 17:33.05 | buttons840 | i initiate several calls using async originates, i will then use unique id's to map events to their originate in the script logic; if a unknow unique id arises, it will querry asterisk using getvar for a special unique channel variable which will allow me to map the new uniqueid to the appropriate channel in the future.... anyone see any flaws in this process? |
| 17:33.55 | jake[work] | ok - zapata.conf is configured fxs_ks |
| 17:34.33 | jake[work] | zapata.conf is 1,1,0,esf,b8zs |
| 17:34.42 | jake[work] | er zaptel.conf |
| 17:35.49 | jake[work] | call goes into asterisk and hits the Read cmd |
| 17:35.57 | jake[work] | listens for 14 digits |
| 17:35.59 | Qwell | s,1,Read()? |
| 17:36.17 | jake[work] | yes |
| 17:36.27 | jake[work] | ok |
| 17:36.29 | jake[work] | i get you |
| 17:36.36 | jake[work] | i can pull the dialplan |
| 17:36.37 | jake[work] | one sec |
| 17:37.08 | jake[work] | ok |
| 17:37.48 | jake[work] | Answer, Read(var from ext ivr), Read(CLID via DTMF) |
| 17:37.56 | jake[work] | Answer() again :) |
| 17:38.09 | Qwell | show me dialplan. pastebin |
| 17:38.15 | jake[work] | ok |
| 17:41.17 | jake[work] | http://pastebin.ca/1516951 |
| 17:41.36 | jake[work] | pbpstart gets to getacctnum via a mgr redirect |
| 17:42.37 | *** join/#asterisk Defraz (n=T0tal@c72co-edge-router.fuzecore.com) |
| 17:43.15 | Qwell | in get-account-number-a? |
| 17:43.33 | jake[work] | i just updated it - was missing the invalid route |
| 17:44.18 | jake[work] | it gets stuck in get acct num,i |
| 17:44.55 | jake[work] | a bunch of calls going there |
| 17:45.58 | jake[work] | here is what Monitor sounds like: http://www.insperia.com/bad-dtmf.wav |
| 17:46.54 | Qwell | yeah, something is muting them for sure |
| 17:47.05 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 17:47.08 | Qwell | no question |
| 17:47.20 | jake[work] | but it's fine at dahdi_monitor |
| 17:47.22 | jake[work] | no muting |
| 17:47.44 | Qwell | it's happening in asterisk |
| 17:47.47 | jake[work] | yep |
| 17:49.00 | *** join/#asterisk KazaLite (n=KazaLite@94-193-98-124.zone7.bethere.co.uk) |
| 17:49.31 | KazaLite | hi all... is there some .h file in asterisk from where I can find version of asterisk on which my code is running? |
| 17:50.34 | [TK]D-Fender | KazaLite: "show version" in CLI |
| 17:50.49 | jake[work] | Qwell: i'll pm you the dahdi_monitor output link |
| 17:51.32 | WindowsUser | KazaLite: did you look in the includes directory yet? |
| 17:51.54 | KazaLite | i mean there is change in function prototy of ast_config_load() |
| 17:52.13 | KazaLite | it took only 1 parameter in askterisk 1.4, but it takes 2 in 1.6 |
| 17:52.39 | KazaLite | so i need to know some #define that could tell me asterisk version so that my correct function gets called |
| 17:52.41 | WindowsUser | if you looked at the includes directory you'd see includes/asterisk/version.h |
| 17:52.55 | KazaLite | ahh great... thans |
| 17:55.22 | KazaLite | there is not version.h in 1.6 |
| 17:55.26 | KazaLite | it is in 1.4 |
| 17:56.34 | KazaLite | renamed to ast_version.h |
| 17:56.41 | KazaLite | in 1.6 |
| 17:57.20 | KazaLite | now thats stupid thing to change the name of file on which I am gonna rely to find version of asterisk!! |
| 17:57.28 | *** join/#asterisk s14ck (n=s14ck@190.77.92.226) |
| 17:58.29 | [TK]D-Fender | KazaLite: "show version" in CLI <---------- |
| 17:59.33 | KazaLite | [TK]D-Fender: show me sample code for doing what you are saying:) |
| 17:59.58 | KazaLite | you know there are compile time switches? things like #if define() sort of things? |
| 18:00.14 | KazaLite | those tiny things that help you in coditional compilation? |
| 18:00.30 | [TK]D-Fender | KazaLite: "asterisk -rx 'show version' > OMMFGZ" |
| 18:01.03 | jake[work] | Qwell: should i call in and open a ticket? |
| 18:01.57 | KazaLite | [TK]D-Fender: did you ever write some C code?{please dont mind} |
| 18:02.15 | [TK]D-Fender | KazaLite: Depends WHERE this code goes. |
| 18:02.21 | manxpower | KazaLite: Development questions are better asked on #asterisk-dev |
| 18:02.54 | KazaLite | manxpower: Thanks. Thats what i was thinking that I might on some wrong place |
| 18:02.59 | manxpower | screams obscenities at his printer. |
| 18:03.15 | jake[work] | the printer backs off |
| 18:04.43 | *** join/#asterisk Rickardo (n=chatzill@200.204.124.211) |
| 18:06.22 | Rickardo | Hello guys... anyone had this problem: "Everyone is busy/congested at this time"? I have a SIP trunk with Call Manager Express... |
| 18:07.03 | [TK]D-Fender | Rickardo: Meaningless error message |
| 18:07.11 | [TK]D-Fender | Rickardo: Go look at actual SIP DEBUG for the call |
| 18:08.14 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 18:09.14 | timeshell | Greetings. Was wondering where you specify to answer a DAHDI channel on the 2nd or 3rd ring? |
| 18:10.23 | WindowsUser | wait(5) and then answer it?:) |
| 18:10.44 | Rickardo | is there something special what i should looking for in debug message??? |
| 18:13.25 | buttons840 | there isn't any build in function in asteisk which would all you to queue up outbound calls to various extensions is there? i've been looking at the queues, but it appears they are use for incoming calls being routed to a number of agents? |
| 18:15.12 | buttons840 | as i've mentioned before i have a client that wants to make outbound phone calls to those who have appointments scheduled with him. so it will dial out to a variety of (indeterminable in advance) numbers, and then execute a dial plan once connected |
| 18:15.30 | buttons840 | i just wanted to double check that there isn't any built in functionality that might make my life easier? |
| 18:16.33 | timeshell | WindowsUser Duh. I should have figured that out. Thanks. |
| 18:21.35 | timeshell | Ok. Here's my issue. We have our main line forwarded after hours to a after hours service. However, our line still rings just once even though the call is forwarded. The problem is, * is still picking up the line as if it were a call. |
| 18:21.36 | WindowsUser | buttons840: im not aware of any queuing for outgoing calls. it would be handy tho |
| 18:21.53 | timeshell | I want * to here at least 2 rings before answering. |
| 18:21.56 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 18:21.58 | timeshell | s/here/hear |
| 18:22.06 | [TK]D-Fender | timeshell: WindowsUser's suggestion works |
| 18:22.13 | timeshell | I just tried it |
| 18:22.15 | timeshell | It didn't |
| 18:22.19 | [TK]D-Fender | timeshell: wait longer :) |
| 18:22.38 | [TK]D-Fender | timeshell: Or also just change your dialplan to maintain its own schedule |
| 18:22.44 | [TK]D-Fender | timeshell: And jsut ignore the call. |
| 18:22.53 | [TK]D-Fender | timeshell: And why aren't you handling your own after-hours? |
| 18:22.54 | timeshell | wait(5) and answer "IF" the line is still ringing? |
| 18:22.57 | timeshell | or just answer anyway. |
| 18:23.17 | WindowsUser | wait more than 5 seconds? |
| 18:23.19 | timeshell | [TK]D-Fender Is that your business? |
| 18:23.51 | [TK]D-Fender | timeshell: :p |
| 18:23.57 | jblack | timeshell: Yes, it is, actually. |
| 18:24.05 | timeshell | lol |
| 18:24.10 | jblack | He can give the best advice by understanding the context of your questions |
| 18:24.23 | [TK]D-Fender | timeshell: if the line is no longer ringing and you haven't answered after the wait the call should disconnect |
| 18:24.49 | timeshell | jblack, Actually, no it's not in that context. Whether we forward to another service using the providers' phone system or using our own really is none of his business. |
| 18:25.04 | [TK]D-Fender | timeshell: MNow remember you have to wait > than the inter-ring delay for * to ack that its no longer rining. this may mean up to 7 s |
| 18:25.31 | manxpower | I believe there is also a waitforring |
| 18:25.51 | timeshell | Ok, what's happening is * is picking up a call, but nothing is there. Just dial tone. It forwards that call anyway to an inbound queue. All the inbound agents phones now ring indefinitely because there's nobody here to pick it up. |
| 18:25.51 | [TK]D-Fender | manxpower: I believe thats audio only post-answer, no? |
| 18:26.01 | manxpower | Asterisk does not do "number of rings" it does time. In the USA a typical ring cycle is 6 seconds, IIRC |
| 18:26.02 | [TK]D-Fender | timeshell: then you needed a little something xtra |
| 18:26.15 | manxpower | [TK]D-Fender: I think you can do waitforring before answer. |
| 18:26.37 | [TK]D-Fender | manxpower: then you'll wait forever.. |
| 18:26.50 | [TK]D-Fender | manxpower: then again.. if the line is no longer ringiing it should simply die |
| 18:27.02 | [TK]D-Fender | manxpower: As soon as * grasps that its game over. |
| 18:27.07 | [TK]D-Fender | manxpower: Which is kinda the goal. |
| 18:27.23 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 18:27.25 | [TK]D-Fender | timeshell: Running your own hours should be viable... |
| 18:27.37 | wwalker | anyone seen this before? this is a connection between asterisk and a commercial sbc or media gateway. All the google hits talk about grandstream handsets. rtp.c: RTP Read too short (-213, expecting 96184) |
| 18:27.46 | manxpower | [TK]D-Fender: after an answer you would wait forever since the line isn't ringing anymore. |
| 18:27.46 | timeshell | Asterisk should try to detect if there is a dial tone when it answers before doing anything else. If it answers and there's only dial tone, it should disconnect. |
| 18:28.05 | manxpower | timeshell: should but Asterisk does NOT detect dialtone like that |
| 18:28.16 | [TK]D-Fender | manxpower: thing is he WON'T be answering... |
| 18:28.25 | [TK]D-Fender | manx he wants to let it "ring out" if its going to. |
| 18:28.42 | manxpower | [TK]D-Fender: 1,Wait (5) n,waitforring done |
| 18:28.45 | [TK]D-Fender | manxpower: And ... we should have an app for that... jsut in case ;) |
| 18:34.20 | timeshell | As I google "waitforring", I get a little confused. Some uses suggest it is used before Answer, others after Answer. What does it actually do? |
| 18:35.00 | manxpower | how about "core show application waitforring"? |
| 18:35.13 | manxpower | timeshell: don't answer the line if you don't have to. |
| 18:35.31 | timeshell | http://www.voip-info.org/wiki/view/Asterisk+cmd+WaitForRing |
| 18:35.45 | manxpower | The wiki is usually wrong. Try the actual docs in the CLI |
| 18:36.00 | timeshell | Still doesn't tell you to use it before or after "Answer" |
| 18:36.23 | *** join/#asterisk [Rated-R] (n=sdfcfvdg@user-24-96-25-7.knology.net) |
| 18:36.52 | manxpower | if you are so worried about it try it both ways. My advice is to not use Answer with WaitForRing. Never. Ever. |
| 18:37.29 | timeshell | :| |
| 18:37.38 | manxpower | The whole point of the app is to avoid answering the line when you get "phantom rings", answering the line would make that pretty useless. |
| 18:37.38 | timeshell | WaitForRing answers? |
| 18:37.49 | manxpower | YOU DO NOT WANT AN ANSWER!!!!!!!!!Q!!!! |
| 18:37.52 | timeshell | Well, that's the way I get it. |
| 18:38.04 | timeshell | lol |
| 18:38.18 | manxpower | The whole point of this is that you don't want to answer when you only get one ring |
| 18:38.19 | timeshell | Don't get upset. I'm trying to figure out why other people are placing answer before it. |
| 18:38.28 | timeshell | That's what's confusing me. I agree with you. |
| 18:38.37 | manxpower | timeshell: Because the Wiki is usually wrong. |
| 18:39.06 | [TK]D-Fender | timeshell: If you answer, then the line stops ringing. |
| 18:39.21 | [TK]D-Fender | timeshell: that is INSANE to think that you should wait for MORE rings |
| 18:39.27 | timeshell | I agree!!! |
| 18:39.41 | timeshell | Like I said, I'm trying to figure out why other people are placing answer before it. |
| 18:39.47 | timeshell | That makes no sense |
| 18:44.30 | timeshell | K, WaitForRing seems to do it. Thanks manxpower. |
| 18:45.42 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 18:48.04 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 18:49.04 | [TK]D-Fender | timeshell: So what did you do with it? Wait,(5) then WaitforRing? |
| 18:49.05 | *** join/#asterisk Defraz (n=T0tal@c72co-edge-router.fuzecore.com) |
| 18:49.21 | timeshell | No, just WaitForRing |
| 18:49.31 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 18:50.34 | [TK]D-Fender | o<O |
| 18:55.48 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 18:56.21 | *** join/#asterisk Darlanth (n=darlanth@wsip-24-234-164-49.lv.lv.cox.net) |
| 18:56.25 | Darlanth | hello |
| 18:56.36 | Darlanth | has anyone used ip0020 phones here |
| 18:56.45 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 18:58.01 | awkfu | If I compile DAHDI and DAHDI-Tools from source, then do I still need to compile zaptel? |
| 18:59.27 | tzafrir_laptop | awkfu, no |
| 18:59.50 | tzafrir_laptop | zaptel and dahdi cannot run together on the same system |
| 18:59.53 | awkfu | tzafrir_laptop thx |
| 19:00.01 | [TK]D-Fender | Darlanth: About as many as the last time you asked. STILL looks like cheap junk... |
| 19:00.02 | tzafrir_laptop | (as in: the modules zaptel and dahdi) |
| 19:00.15 | *** join/#asterisk p3nguin (i=BuhPjX1J@i.got.my.network.from.a2infotech.com) |
| 19:01.00 | exsync | What does the "Domain" on a Linksys/Sipura under system information refer to? |
| 19:01.36 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 19:01.43 | exsync | My IP's are static, gateways static, is that just fluff? |
| 19:02.03 | *** join/#asterisk MarkBest (n=chatzill@66.236.6.194.ptr.us.xo.net) |
| 19:02.45 | MarkBest | [TK]D-Fender Thanks for you help a few weeks ago. I got everything up and running in regards to my Page() features. |
| 19:02.59 | [TK]D-Fender | MarkBest: You're welcome. |
| 19:05.46 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 19:05.58 | ayeso | how can I disconnect an active SIP channel from the console? |
| 19:06.14 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 19:06.29 | *** join/#asterisk tdg911 (n=tdg911@75-131-246-217.static.slid.la.charter.com) |
| 19:07.37 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 19:07.52 | *** part/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 19:10.08 | Kobaz | blarg |
| 19:10.21 | Kobaz | i have a purely lan based sip system |
| 19:10.26 | ayeso | it was 'soft hangup <channel>' if anyone was interested |
| 19:10.26 | Kobaz | and if the internet goes out, the phones go out with it |
| 19:10.36 | Kobaz | even though nothing requires the internet, and all the phones are local |
| 19:13.37 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:15.09 | ayeso | Kobaz: Well that's odd, sounds like its not purely lan based then.. when your internet connection is down can you ping the phones from the * server? |
| 19:15.27 | Kobaz | yeah |
| 19:15.47 | *** join/#asterisk javb (n=javb@190.94.122.69) |
| 19:16.08 | ayeso | whats your network toplolgy like? |
| 19:16.10 | Kobaz | i have a 24 port sip gateway that all the phones are connected to |
| 19:16.16 | Kobaz | when the net goes down |
| 19:16.21 | Kobaz | 4 phones stay reachable |
| 19:16.29 | Kobaz | but the ping is like 1000ms according to asterisk |
| 19:16.33 | Kobaz | it's like. instant |
| 19:16.49 | Kobaz | as soon as the outside network is unplugged.. 80% of the phones go byebye |
| 19:17.02 | Kobaz | a truck hit a phone pole outside, and the net went down |
| 19:17.05 | ayeso | all phones and * server on the same subnet? |
| 19:17.09 | Kobaz | same subnet |
| 19:17.50 | ayeso | what is the vendor for the gateway? |
| 19:17.53 | Kobaz | citel |
| 19:17.58 | Kobaz | i think it's a citel problem, honestly |
| 19:18.04 | ayeso | i agree |
| 19:18.14 | Kobaz | I've never had this problem with polycom once i enabled dnsmanager |
| 19:18.14 | elguero | Deeewayne: Hey... I know you are probably real busy... I was curious if you had a chance to look at that pri trace in regards to the b channel messaging service support on a NI2 switch |
| 19:18.15 | ayeso | if you have support with them id get on the horn |
| 19:18.46 | *** join/#asterisk maskas (n=maskas@134.146.70.202.dynamic.max.com.pk) |
| 19:18.49 | ayeso | your phones dont know your * server via DNS do they? |
| 19:18.50 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-142-149-133.stat.centurytel.net) |
| 19:18.51 | *** join/#asterisk nauticalthinker (n=mratliff@phoenix.warren-wilson.edu) |
| 19:18.53 | maskas | hello, could anyone help me with a pri related problem? |
| 19:18.54 | jeff_phillips | hello |
| 19:18.56 | Kobaz | nope, ot's all via ip |
| 19:18.57 | maskas | I send an outgoing call via a pri over a certain channel, the remote end signals to asterisk to move the call to another channel. The call connects but audio is on way. I can hear called party but they can not hear me. This is using asterisk1.4.26 |
| 19:19.00 | maskas | also when the calls hangs up, the following error is shown: WARNING[11188]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found?, and chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/3 on span 1 |
| 19:19.01 | nauticalthinker | nick nautical |
| 19:19.14 | jeff_phillips | Going back to a project that I had once looked into before... |
| 19:19.16 | ayeso | Kobaz: ok good, yeah Id call them and ask WTF is this thing doing |
| 19:19.22 | Kobaz | yeah |
| 19:19.24 | Deeewayne | elguero, see pm |
| 19:19.29 | nauticalthinker | anyone know why zapata.conf is missing from /etc/asterisk/? |
| 19:19.50 | maskas | nauticalthinker: its chan_dahdi.conf now |
| 19:20.10 | jeff_phillips | For cable modem users who have a cable telephone line bundled with their cable service, has anyone successfully gotten the DOCSIS PacketCable NCS phone service setup as a trunk in Asterisk without converting to analog & back? |
| 19:20.24 | nauticalthinker | thank you! |
| 19:20.51 | jeff_phillips | I see info on using asterisk to host the service at the cable operator's plant, but nothing about using it in place of the MTA at the customer side |
| 19:20.57 | *** part/#asterisk nauticalthinker (n=mratliff@phoenix.warren-wilson.edu) |
| 19:22.39 | Kobaz | okay, the citel has an external dns server in network settings, but dns is disabled |
| 19:22.44 | Kobaz | i cleared out the dns servers entirely |
| 19:23.43 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 19:26.22 | ayeso | Kobaz: i dont think its a DNS issue if you are peering via IP address, can you see the mac address table on that thing? |
| 19:26.44 | ayeso | Kobaz: or the arp table? |
| 19:29.10 | [TK]D-Fender | jeff_phillips: I can't imagine their modem acting as a proxy for you instead of as an ATA |
| 19:29.25 | [TK]D-Fender | jeff_phillips: this is contrary to telco's "Dumb clients only" policies |
| 19:29.53 | jeff_phillips | well their sales rep keeps trying to cram a line down my throat. I keep telling him I'll only take one for this specific purpose--he's well aware of it |
| 19:31.20 | *** join/#asterisk Nullslash (n=Null@24.164.133.222) |
| 19:31.25 | manxpower | jeff_phillips: What specific endpoint do you have? |
| 19:31.47 | Nullslash | Hi. |
| 19:33.06 | [TK]D-Fender | jeffthis is expressly the sort of stuff they don't want to hear about |
| 19:33.32 | jeff_phillips | None yet -- the cable company wants me to take a phone line from them and I keep telling them I don't want anything analog but they want to send me an MTA with an analog RJ-11 jack on it and I keep refusing to accept it |
| 19:34.28 | jeff_phillips | Something stupid about them having to extend the cable line an extra block to reach our building, it costing too much for the installation, them and their own red tape not being able to bill us for the install costs directly... having to amortize it out over the service contract, but not offering a service we want that is expensive enough to cover it |
| 19:34.47 | jeff_phillips | so they want me to add a phone line onto the bill too that I don't want and told them I will only take if i can use it as a trunk on my PBX |
| 19:35.02 | jeff_phillips | (we've been going back and forth with them for a long while over this.) |
| 19:35.53 | jeff_phillips | funny thing was the contract we signed for the installation actually expired before they got around to doing the install, so now they have the physical installation done and nothing on paper to obligate us to buy anything from them at all |
| 19:36.09 | jeff_phillips | so they are really upset and want me to sign something and I keep telling them not unless you give me something worth having |
| 19:37.26 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 19:37.28 | *** join/#asterisk garymc (n=gar@host86-162-158-214.range86-162.btcentralplus.com) |
| 19:37.55 | garymc | Hi I got a question about Astersisk |
| 19:38.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:38.03 | garymc | I got 3 rack servers |
| 19:38.23 | garymc | one running LTSP FEDORA 10 for my client comps to acces databases and internet file sharing etc |
| 19:38.42 | maskas | manxpower: are you around? |
| 19:38.47 | garymc | Im wondering if my second server could have astlinux installed and can I link it to my first server? |
| 19:38.59 | garymc | Will the end user think there is only one server in operation? |
| 19:39.24 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 19:39.27 | garymc | and is this the best way to use asterisk? |
| 19:41.12 | garymc | No one about ? |
| 19:41.44 | ariel_ | garymc: I just got here and yes there are folks around. I just missed your question |
| 19:41.58 | garymc | ok |
| 19:42.04 | garymc | ill repeat ;) |
| 19:42.12 | *** join/#asterisk old_monk (n=borgirc-@78-20-232-172.access.telenet.be) |
| 19:42.12 | garymc | Ive just bought 3 rack servers |
| 19:42.26 | garymc | I got Linux Fedora installed on one machine |
| 19:42.29 | brah | My fax machine hangs up so it kills my extension. Any workarounds? |
| 19:42.35 | garymc | I link my thin clients to it |
| 19:42.44 | *** join/#asterisk cryptanthus (n=newview@wsip-72-214-233-12.om.om.cox.net) |
| 19:42.47 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 19:42.50 | garymc | I want to use a second server to run asterisk |
| 19:42.58 | garymc | and link it ot my first server |
| 19:43.07 | garymc | is this possible |
| 19:43.23 | garymc | and can I link it so the thin clients think it is one machine? if you know what i mean? |
| 19:43.53 | ariel_ | you want to create a server farm? |
| 19:43.57 | garymc | yes |
| 19:43.59 | garymc | I think |
| 19:44.03 | garymc | Im new to all this stuff |
| 19:44.10 | garymc | I bought 3 HP G3 servers |
| 19:44.18 | garymc | lots of ram etc dual processors |
| 19:44.21 | afink | why do your thin clients need to think it is one machine? why do yo need two? |
| 19:44.40 | afink | just your phones/email server need to be able to talk to * |
| 19:44.43 | garymc | Well I have been told to run asterisk on another server is better |
| 19:45.10 | afink | ok you have 3 servers so run it on a different one |
| 19:45.16 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 19:45.16 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:45.17 | ariel_ | garymc: first you need to list what you really want ot do, 3 servers can do just about anything |
| 19:45.22 | cryptanthus | I want to download and compile Asterisk. On the Asterisk.org website there are three versions. 1.4.26, 1.6.0.10 and 1.6.1.2. Which one do I download? |
| 19:45.41 | garymc | Thats what im saying how do I link the servers together? |
| 19:46.10 | garymc | So when my clients loginto Linux server 1 they can also access astrisk? is that how it works? |
| 19:46.26 | ariel_ | garymc: most people draw up a plan on what they want to do, then buy the hardware that fits it |
| 19:46.30 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:46.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:46.32 | *** join/#asterisk propellerhead (n=yogurt2u@host144.190-30-188.telecom.net.ar) |
| 19:46.33 | afink | garymc: do you have phones? |
| 19:46.42 | garymc | We have an isdn2e line |
| 19:46.57 | [TK]D-Fender | cryptanthus: good suggestion these days : 1.6.0.10 |
| 19:47.05 | ariel_ | you have 2 e1'1 |
| 19:47.11 | afink | there really is no "logging in" to the asterisk server. Especially with a thin client |
| 19:47.25 | ariel_ | ssh |
| 19:47.31 | cryptanthus | D-Fender: Thanks. Why are there three versions? |
| 19:47.45 | afink | ariel_: as far the clients/thin clients are concerned |
| 19:48.08 | afink | cryptanthus: evolution? |
| 19:48.13 | garymc | and a BT versatilty point |
| 19:48.28 | garymc | we want to install a good few phones for a mini call centre |
| 19:48.51 | ariel_ | oh now we get more info, mini call center? how many users? |
| 19:48.54 | ariel_ | phones types |
| 19:49.00 | afink | garymc: I think you should read the book first |
| 19:49.04 | afink | ~book |
| 19:49.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:49.05 | ariel_ | location, and channels expected |
| 19:49.06 | garymc | which book |
| 19:49.10 | afink | ^^ |
| 19:49.11 | garymc | ahh |
| 19:49.23 | *** join/#asterisk amdtech (n=adaniel@nat/digium/x-xoxhbydwdksxqmpv) |
| 19:49.30 | afink | set one up first, then worry about clusterf**king |
| 19:49.38 | garymc | I need to record all incoming and outgoing calls |
| 19:49.39 | thehar | Hellos Aaron! |
| 19:49.47 | amdtech | waves at thehar |
| 19:49.50 | afink | garymc: read the book |
| 19:50.03 | thehar | :D |
| 19:50.04 | afink | garymc: it is extremely easy to do call monitoring |
| 19:50.11 | garymc | ok |
| 19:51.10 | cryptanthus | slaps forehead. Of course. Is the 1.6.1.2 version a development version where things could go haywire at anytime. I would like to avoid "haywire" if at all possible. |
| 19:51.41 | garymc | im downloading the pdf format |
| 19:51.51 | garymc | thanks ill give it a good read and get back in here at another time |
| 19:51.53 | garymc | thanks |
| 19:52.32 | cryptanthus | garymc: Make sure when you are reading the book to replace the references to zaptel with Dahdi. It was confusing to me. |
| 19:53.07 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:54.14 | garymc | ok cheers |
| 19:57.11 | [TK]D-Fender | cryptanthus: 1.6.1 isn't a development version. 1.6.2 while in beta is. 1.6.1 is still a little new and I'd give it another sub-ver or two to be sure its stable, and then I'd consider it. |
| 19:57.52 | [TK]D-Fender | cryptanthus: 1.6.0 is rather mature now. If you're just learning or for non-business use you could just jump in on 1.6.1, but it'll more likely have more issues that 1.6.0 |
| 19:58.10 | [TK]D-Fender | cryptanthus: 1.4 is only for old throw-back deployments. |
| 19:58.55 | leifmadsen | I wouldn't necessarily describe 1.6.2 beta's as "development", but I agree with [TK]D-Fender's assessment of the current version numbers |
| 19:59.48 | [TK]D-Fender | leifmadsen: Well... lets say failure to hit the official .0 launch full release... you know bleeding edges shouldn't be allowed to be that long :p |
| 19:59.54 | [TK]D-Fender | goes to litigate |
| 20:00.12 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 20:00.55 | leifmadsen | [TK]D-Fender: sorry? I don't quite follow :) |
| 20:01.02 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 20:01.10 | ayeso | what do you guys do when you have say 100 active calls, and you want to run a command at the CLI but you dont get much of a chance to see the output because of all the calls? |
| 20:01.33 | ayeso | i could use script, but is there a way to limit whats on the screen? |
| 20:01.35 | leifmadsen | 1.6.2.0-beta4 was basically the last beta because it had some big T.38 changes in it. An RC should be out for 1.6.2.0 in the new few weeks, and 1.6.2.0 full not long after that |
| 20:01.52 | [TK]D-Fender | leifmadsen: Just saying so long as the branch hasn't hit 1.6.2.0 full release (non RC) I'd say "consider for entertainment use only" :p |
| 20:01.52 | leifmadsen | ayeso: core set verbose 0 ? |
| 20:02.02 | leifmadsen | [TK]D-Fender: I don't disagree with that statement :) |
| 20:02.04 | angryuser_laptop | ayeso: you can run it from shell asterisk -x and save output to a file |
| 20:02.09 | [TK]D-Fender | leifmadsen: EXCELLENT |
| 20:02.16 | ayeso | leifmadsen: never tried that lets see. |
| 20:02.31 | leifmadsen | runs off to clean the condo for the g/f before she gets back from the cottage |
| 20:02.48 | Kobaz | ayeso: edit logger.conf, and add: full => notice,warning,error,verbose |
| 20:02.57 | ayeso | what is the highest level of verbosity? it seems to let you set it as high as you want |
| 20:03.08 | Kobaz | ayeso: and then you can look at your log in a file, and filter out what you want |
| 20:03.23 | Kobaz | ayeso: not much exciting happens after verbosity 3 |
| 20:03.40 | ayeso | Kobaz: yeah I have that going as well, im just wanting a quick thing while im on the console... thx |
| 20:03.52 | Kobaz | filtering is pretty much non-existant on the console |
| 20:04.11 | Kobaz | you can set some sip debugs by ip or peer, and things like that |
| 20:04.16 | Kobaz | but it's very limited |
| 20:04.42 | ayeso | Kobaz: leifmadsen solution is working for me |
| 20:07.24 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 20:07.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:08.19 | *** join/#asterisk marga (n=nmarga@186.18.92.38) |
| 20:08.22 | lowtek | Hey guys, I have some Cisco 7940's. They are using a remote TFTP server but a local Asterisk server. The phones, for some reason, are registering with both servers. There's no indication of the remote (tftp) server anywhere in the config and a factory defaults doesn't help. It's somehow assuming the TFTP server is a proxy or registration point, any ideas? (SIP 8.2) |
| 20:09.19 | ayeso | lowtek: well the only way the phone knows who to register to is via those files, so best advise I can give is look harder. |
| 20:09.33 | marga | Hi! I'm running an asterisk server here, in Argentina, with localized messages. I'm having a problem with voicemail, because in Spanish we say "15 of June" instead of "June 15th" and I find no way of changing nor the order of the string nor the ordinal for a cardinal number... What am I missing? |
| 20:09.58 | ayeso | marga: you need to take a look at app_voicemail.c |
| 20:10.21 | marga | ayeso: is there no way of localizing this that does not imply recompiling? |
| 20:10.48 | marga | I'm running the official Debian package, and I would prefer not to divert because of thise. |
| 20:11.02 | lowtek | Here's the config: http://pastebin.com/m1177d22e |
| 20:11.11 | ayeso | marga: Not that i have seen, i just rewrote a ton of the voicemail app for my company to replace their existing platform, so its very fresh in my mind |
| 20:11.18 | lowtek | It's assuming option 66 via dhcp is also a registration point. |
| 20:11.28 | lowtek | .. I bet, checking now |
| 20:11.46 | marga | ayeso: but was it related to localizing? |
| 20:11.55 | marga | I'm looking for the code |
| 20:12.28 | ayeso | marga: alot of the localization format for the voicemail platform is hardcoded into the vm app itself. |
| 20:13.39 | ayeso | marga: you will definitly have to do some coding to change what you want to change. |
| 20:14.08 | ayeso | marga: are you sure its pulling this file on boot? |
| 20:14.21 | marga | ayeso: the coding is not the problem, what I don't want is to have to patch the code each time I upgrade asterisk. |
| 20:14.23 | marga | ayeso: sorry? |
| 20:14.28 | afink | this might sound stupid, but you could set envelope=no if you don't want to hear the date |
| 20:14.32 | ayeso | lowtek: are you sure its pulling this file on boot? |
| 20:14.52 | afink | ^^ might not be a big deal if your voicemails are being emailed anyways |
| 20:14.53 | ayeso | marga: I know that feeling... |
| 20:14.55 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:14.58 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 20:15.26 | lowtek | Yep, checking dhcp option 66 possibility |
| 20:15.29 | ayeso | afink: even if envelop=no the user still has the option to play the envelope after the message |
| 20:16.31 | ayeso | lowtek: well if it thinks the IP it gets from option 66 is not only the TFTP server but also the * server id try and load a different version of SIP on the cisco phones. |
| 20:16.35 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:18.25 | ayeso | lowtek: or you could say screw it and run tftp on your * box |
| 20:23.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:23.52 | marga | ayeso: so, apparently the problem is in the ast_say_date_with_format... And it might be related to my locale being es_AR instead of simply es. I'll keep digging |
| 20:24.17 | Nullslash | hi |
| 20:24.54 | Nullslash | I need good SIP provider |
| 20:25.04 | Nullslash | wtf xD |
| 20:25.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:25.24 | Nullslash | I need any good SIP provider for my asterisk server |
| 20:25.49 | Nullslash | Do you guys have any suggestion ? |
| 20:26.52 | *** join/#asterisk wilsonj (n=jeremy@unaffiliated/dethstar) |
| 20:27.22 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:27.33 | [TK]D-Fender | checkout time, BBL |
| 20:28.17 | Nullslash | !help |
| 20:28.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:31.14 | exsync | !research |
| 20:31.15 | exsync | !google |
| 20:31.22 | leifmadsen | !dance |
| 20:31.44 | leifmadsen | ~help |
| 20:31.52 | leifmadsen | Nullslash: try that |
| 20:32.11 | Nullslash | :/ |
| 20:32.15 | kaldemar | ~isp |
| 20:32.16 | infobot | from memory, isp is (Internet Service Provider) An ISP provides Internet access to people or corporations. ISPs generally have pools of modems awaiting dial-up connections. Smaller ISPs buy bandwidth from larger ISPs. There are two basic ways of connecting: one is to get a PPP or Slip account; the other is to get a UNIX shell account. |
| 20:32.30 | kaldemar | ~itsp |
| 20:32.30 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 20:32.33 | Nullslash | ok |
| 20:32.35 | Nullslash | thannks |
| 20:32.38 | Nullslash | thanks* |
| 20:33.09 | KazaLite | hi all |
| 20:34.31 | KazaLite | i have built and intalled asterisk 1.6 and i have started it with command "asterisk -cg" but in asterisk i cannot run commands like list codecs, show codecs, list modules, show modules. Are there some other commands in 1.6 than 1.4? |
| 20:34.32 | ayeso | Nullslash: voicepulse is pretty good |
| 20:34.41 | KazaLite | or how can i troubleshoot it? |
| 20:36.25 | kaldemar | KazaLite: "help" will give you the rights commands. those won't work on 1.6. |
| 20:36.28 | Nullslash | ayeso , Thanks I will call them |
| 20:36.59 | ayeso | Nullslash: you want the sippbx solutuion if you are going to run your own * server. |
| 20:37.43 | ayeso | Nullslash: also there really is no reason to call them, they do everything online. |
| 20:37.54 | Nullslash | Yeah , I know |
| 20:38.21 | Nullslash | But I still looking for the hardware requirement |
| 20:38.49 | Nullslash | like the VOIP phones , 1800 number ..etc |
| 20:40.13 | ayeso | Nullslash: well they wont provide you with hardware, but they can get you 800 #s |
| 20:40.15 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 20:41.27 | doolittlework | hi all i know its not the right channel seeing that i am using freepbx to learn more dialplan basics, but i need to know if one can run a application after someone leave a message? |
| 20:42.06 | ayeso | doolittlework: yes you can |
| 20:42.25 | ayeso | doolittlework: look in voicemail.conf around where it talks about SMDI |
| 20:42.40 | manxpower | doolittlework: why not ask on the #freepbx channel? |
| 20:42.50 | doolittlework | channel dead |
| 20:42.51 | ayeso | doolittlework: there is a line you can uncomment and enter the path of the app you want to run after a message is left |
| 20:43.10 | manxpower | IIRC, freepbs overwrites your voicemail.conf. customizations need to be done in a different file. |
| 20:43.11 | doolittlework | ths will have a look |
| 20:43.44 | doolittlework | what is SMDI? ayeso? |
| 20:44.50 | ayeso | doolittlework: service message desk or something like that, its for turning on/off message waiting indicatior lights, i only mentioned it becuase that seciton is near what you are looking for in voicemial.conf |
| 20:45.11 | manxpower | SMDI is a protocol used for PBXs to communicated with a standalone voicmail system over a serial port. |
| 20:45.24 | *** join/#asterisk aaaa (n=ayoub@cpe-76-94-127-241.socal.res.rr.com) |
| 20:46.28 | ayeso | ^^^ there you go, better answer |
| 20:47.14 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 20:47.58 | citywok | i'm testing an upgrade to asterisk 1.6 from 1.4, and i've got a script that originates a call from a sip phone, which works in 1.4 but not in 1.6. http://pastebin.com/d7a9d7fd0 |
| 20:48.14 | manxpower | In the few times SMDI was considered for one of my customers the PBX vendor wanted large amounts of money to enable SMDI on their PBX. |
| 20:50.09 | manxpower | city in the Asterisk console what is the output of "module unload chan_sip.so" and "module load chan_sip.c" |
| 20:50.17 | manxpower | ..er.. .so not .c |
| 20:50.35 | manxpower | use pastebin.ca if there's more than a couple of lines of output. |
| 20:51.54 | citywok | lol... module unload... disconnected from server |
| 20:52.19 | manxpower | then you have a major problem. |
| 20:52.34 | citywok | yea, no kidding heh. i'm pruning my entire sip.conf to see if i can find it |
| 20:53.06 | manxpower | citywok: I recommend moving all the modules (/usr/lib/asterisk/modules or /var/lib/asterisk/modules) then do a make install to reinstall the modules. chances are you have a 1.4 specific module laying around. |
| 20:53.22 | citywok | it's a seperate box that never saw 1.4 |
| 20:53.50 | citywok | just a quick test box so i dont break the real system |
| 20:53.56 | manxpower | so much for that idea. LOL |
| 20:54.01 | ayeso | citywok: does it even attempt to dial anything? |
| 20:54.13 | manxpower | Asterisk should NEVER crash when you unload chan_sip |
| 20:54.17 | citywok | i copied sip.conf & extensions.conf from the production system, so i trimmed sip.conf to a single test extension and it doesnt crash |
| 20:54.32 | citywok | ayeso: it doesn't do anything other than give the invalid channel |
| 20:54.49 | ayeso | citywok: oh yeah, didnt see that |
| 20:54.59 | citywok | no worries :) |
| 20:55.43 | manxpower | citywok: SIP/number@sipconfentry is what I would expect to need as the Channel |
| 20:56.22 | citywok | I was letting the dialplan handle the provider, so would it be sip/number@context? |
| 20:56.58 | citywok | i'm basically forcing the phone into the normal outbound dial context, without hte agent dialing it. so the context handles SIP provider failure, etc. |
| 20:57.30 | manxpower | the Local/ channel is to make a call that would normally take a tech/etc to jump to the dialplan |
| 20:58.20 | citywok | okay, cleaned my sip.conf http://pastebin.com/m57660ec0 |
| 20:58.24 | *** join/#asterisk Cuban0 (n=cubano@190.70.234.233) |
| 20:58.36 | Cuban0 | hello |
| 20:58.37 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:58.39 | Cuban0 | i´m new here |
| 20:58.47 | Cuban0 | i´m seeking for help |
| 20:59.00 | Cuban0 | some advice on using codecs on a specific scenario |
| 20:59.05 | Cuban0 | anyone available ? |
| 20:59.08 | citywok | manxpower: i read the example from voip-info when i originally wrote it, which just says channel: SIP/extension |
| 20:59.12 | Cuban0 | so i can ask? |
| 20:59.22 | cryptanthus | <[TK]D-Fender> Thanks for the version information. I had stepped away for a moment. This project is business related. I will use 1.6.0.10 |
| 20:59.41 | beek | infobot: tell Cuban0 about ask |
| 21:00.28 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 21:00.38 | Cuban0 | i´m routing calls on a very lagged and jittery route, DirecWay |
| 21:00.53 | citywok | manxpower: thanks, i'm reading up on asterisk local channels right now, i'll figure it out this way heh |
| 21:01.01 | manxpower | citywok: SIP/extension has never been valid. |
| 21:01.24 | Cuban0 | round trip is about 1500 ms there antd Jitter is about 400 ms |
| 21:01.30 | *** join/#asterisk geocar (n=user@proxy.dmvnoc.com) |
| 21:01.33 | manxpower | SIP/devicenameinsipconf is what you would expect |
| 21:01.39 | Cuban0 | what codec will you recommend GSM, G729 or SPEEX ? |
| 21:01.49 | manxpower | Cuban0: no codec will help you much. |
| 21:02.07 | Cuban0 | i´m experimenting with jitter buffer values and is going well |
| 21:02.17 | citywok | I just switched to Local/1593 (my sip phone name) and it still says channel not specified |
| 21:02.23 | Cuban0 | but i´m guessing what codec is less affected by jitter and packet loss ? |
| 21:02.29 | manxpower | you would need Local/extension@context |
| 21:02.36 | citywok | yea, i tried it both ways heh |
| 21:02.37 | manxpower | Cuban0: Speex, IIRC |
| 21:03.14 | Cuban0 | IIRC ? |
| 21:03.20 | manxpower | If I recall correctly. |
| 21:03.28 | citywok | 1.5second delay? satellite? wow thats gonna be a brutal call |
| 21:03.46 | ayeso | manxpower: does that work? Local/ext@context? never tried that |
| 21:03.47 | citywok | we work with 300ms delay and it's not that much fun |
| 21:04.13 | manxpower | ayeso: that is the only valid way to use Local/ |
| 21:04.29 | manxpower | there should be some file named something like localchannel.txt in the Asterisk doc/ directory in the asterisk source. |
| 21:04.35 | Cuban0 | satellite link is thatway and is the only way i can reach my destination ;) |
| 21:04.43 | ayeso | manxpower: I see, what Im gettting at is that i have never used 'local' before |
| 21:04.44 | angryuser_laptop | citywok: i heard 300 is the limit |
| 21:04.53 | Cuban0 | so i must addapt the best i can to the scenario |
| 21:04.54 | citywok | theres also a good page on voipinfo i'm reading up, but doesnt seem to want to work |
| 21:05.05 | angryuser_laptop | citywok: if more human can hear lag..... |
| 21:05.27 | citywok | 3 tenths of a second can be tough, it takes getting used to, you have to wait for calls to clear or you talk ovr each other a lot |
| 21:05.29 | angryuser_laptop | i mean notice not hear |
| 21:05.59 | manxpower | lag + jitter is what the minimum delay would be. |
| 21:06.06 | Cuban0 | if you have no otherway of talking or too expensive |
| 21:06.11 | Cuban0 | it is tolerable |
| 21:06.18 | Cuban0 | SPEEX is not buggy ?? |
| 21:06.25 | Cuban0 | i´m now using GSM |
| 21:06.35 | Cuban0 | but i will test SPEEX then |
| 21:06.43 | Cuban0 | many tweaking options for SPEEX |
| 21:06.58 | geocar | Does anyone here know much about budgetone 101s? |
| 21:07.07 | manxpower | ~phones |
| 21:07.07 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 21:07.10 | Cuban0 | main thing i´m looking for is a good behaviour of codec when some packets get lost |
| 21:07.41 | manxpower | Cuban0: GSM, ULAW, ALAW, and G729 don't conceal packet loss. |
| 21:08.19 | Cuban0 | ULAW and ALAW to much bandwidth |
| 21:08.26 | Cuban0 | SPEEX not recommended ? |
| 21:08.42 | manxpower | I did not say that. |
| 21:08.49 | Cuban0 | yes about G729 i have noticed it turns completely robotic when packet miss |
| 21:15.25 | *** part/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 21:15.30 | angryuser_laptop | any codec with a packet loss sound like crap |
| 21:17.06 | citywok | yes, missing voice is very difficult to make NOT sound horrible |
| 21:18.31 | angryuser_laptop | an btw latest grandstream phones are not so bad, we are using some, sound is not so bad |
| 21:18.51 | angryuser_laptop | but they still look ugly a bit |
| 21:18.57 | *** part/#asterisk geocar (n=user@proxy.dmvnoc.com) |
| 21:19.06 | angryuser_laptop | too huge... |
| 21:19.08 | citywok | we have some grandstreams' that suck. not sure what model but they are only a few months old. the speakerphone is worthless |
| 21:19.31 | citywok | GXP2000 |
| 21:19.36 | *** part/#asterisk h00k (n=anthonyr@unaffiliated/h00k) |
| 21:19.42 | Cuban0 | another question to the experts, there is any noticeable difference between G729 commercial and the Free version ? |
| 21:19.44 | ariel_ | Cuban0: I work in setting up systems for voice calls to ships on the ocean via sat links. G729 works best and you should stay away from iax2 use sip. |
| 21:20.40 | Cuban0 | thats cool info i always thought IAX2 would be better ....so i will set up some SIP channels and see behaviour |
| 21:20.46 | angryuser_laptop | citywok: we have some gpx2010 |
| 21:21.24 | Cuban0 | ariel_: are you using the free G729 or paid one ? |
| 21:21.57 | ariel_ | paid only |
| 21:22.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:22.23 | Qwell | Cuban0: There is no free one |
| 21:22.27 | ariel_ | in fact most of our main server have the g729 card that digium sells |
| 21:23.01 | Cuban0 | wich card is that ? i´m using digium TDM410 |
| 21:23.56 | Cuban0 | tell me in your opinion why SIP over IAX2 please ? |
| 21:26.36 | Cuban0 | @Qwell check this out http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 21:26.48 | Qwell | Cuban0: That isn't free. |
| 21:27.06 | Qwell | You cannot legally use it in any circumstances. |
| 21:27.23 | Qwell | no more discussion about it here. |
| 21:27.41 | Cuban0 | ok |
| 21:27.44 | Cuban0 | sorry |
| 21:29.35 | ariel_ | Cuban0: we found over the Sat links IAX2 is ok but it uses same port for rtp sound. You will loose less if you use sip that has sends it's rtp sound via different ports for each call |
| 21:30.16 | Cuban0 | interesting |
| 21:30.21 | Cuban0 | i will test and see results |
| 21:30.23 | ariel_ | less jitter and if you loose packets it's for one side of the call or just for one call not all of them if your using more then one call |
| 21:30.35 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:30.51 | Cuban0 | i´m loosing packets un upload to satellite |
| 21:30.54 | Cuban0 | download is Ok |
| 21:31.30 | Cuban0 | just decided for IAX because the Trunk feature |
| 21:31.42 | Cuban0 | but i will test on your experienced opinion |
| 21:32.32 | sumasuma | Can askterisk hand's off two PRI channels to provider while monitoring the signalling part of it |
| 21:32.32 | sumasuma | ? |
| 21:32.50 | citywok | okay in 1.6 if i set my sip friend as context=from-sip, when i dial it says call from <sipphone> to extenssion rejected because extension not found. it looks like it's failing because of no context, because if i do a _. context to catch all in from-sip it doesnt hit it. |
| 21:33.06 | Qwell | citywok: then it isn't matching your peer |
| 21:33.33 | citywok | any suggestions on what to try? |
| 21:34.51 | buttons840 | when issuing an originate through the ami, how can i determine if the number i am calling is disconnected or in any other unusual state? i notice on the pstn that the line will ring, and then a robot answered and reports and disconnected number, this is on a fxo line. will i have the same behavior on a line with signalling? |
| 21:35.11 | citywok | hrmm. qwell shouldn't it use the default context in the top of my [general] then? which is also set to from-sip |
| 21:35.46 | Cuban0 | anyone here using Elastix ? |
| 21:36.37 | citywok | "Found user '1593' for '1593'" if i crank up debugging -- so i think it's matching me |
| 21:37.00 | *** join/#asterisk keyp (n=diomedes@66.184.128.98) |
| 21:38.21 | *** part/#asterisk aaaa (n=ayoub@cpe-76-94-127-241.socal.res.rr.com) |
| 21:38.34 | ariel_ | Cuban0: yes but there is an #elastix channel |
| 21:39.07 | manxpower | citywok: pastebin the output of a failed call w/the debug |
| 21:39.16 | citywok | doing that now, heh |
| 21:39.32 | manxpower | buttons840: no matter how many times you ask, the answer will be the same. |
| 21:41.07 | buttons840 | i'm not asking anything about fxo, i'm asking about channels with signalling, like pri. do they signal for disconnected numbers? |
| 21:42.03 | citywok | manxpower: http://pastebin.com/d3b841223 |
| 21:42.12 | buttons840 | i'm sorry if i'm annoying you by asking the same question over and over, but it's kind of hard to avoid when you consider any question, about any subject, to be the same question |
| 21:42.59 | citywok | i see Looking for 1253495XXXX in from-sip (domain 192.168.11.1) |
| 21:43.12 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:43.22 | citywok | but then it says not found, i pasted in the extensions.conf relevant section as well, where you can see it. i have recounted the length like 10 times to make sure it's right and it is |
| 21:47.06 | Qwell | citywok: dialplan show from-sip |
| 21:47.50 | citywok | oh wtf There is no existence of 'from-sip' context |
| 21:48.26 | citywok | dialplan show doesnt show anything in my extensions.conf |
| 21:48.29 | Qwell | now you get to figure out why that is (tip: try to reload the dialplan) |
| 21:49.35 | citywok | when i type reoad i see it do some demo contexts but it appears as though it doesnt touch extensions.conf -- did something change in 1.6 the way that works? |
| 21:50.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:50.10 | citywok | lol extensions.ael eh? dohhhh rtfm citywok |
| 21:51.06 | Qwell | are you loading pbx_config.so? |
| 21:52.47 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:53.43 | *** join/#asterisk s14ck (n=s14ck@190.77.92.226) |
| 21:54.05 | citywok | hey look at that, load it and it errors on an invalid include (doesnt exist on demo system). fix that, and afterl oading the module it dials properly |
| 21:54.45 | citywok | one step closer! now to make that module autoload |
| 22:01.11 | *** join/#asterisk Cynagen (n=Cynagen@ip72-222-183-236.ph.ph.cox.net) |
| 22:03.06 | b14ck | sup guys |
| 22:04.33 | *** join/#asterisk speedy (n=speedy@91.140.192.154) |
| 22:07.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:09.21 | citywok | alright, so now i'm back to where i was before, when i originate a command from the telnet interface, it says invalid channel. i rewrote it to use the local/extension@context syntax and it still doesnt work |
| 22:09.30 | speedy | what does it mean when i have lots of Starting simple switch on 'DAHDI/1-1 and Hungup 'DAHDI/1-1' repeated every second? |
| 22:09.48 | manxpower | speedy: that means "your line is screwed up" |
| 22:10.26 | manxpower | citywok: did you compile from source or install a package? |
| 22:10.36 | Cuban0 | i´m using TDM410p from DIGIUM, why call start counting as answered instantly when DAHDI dials the number ???? |
| 22:10.43 | citywok | debian package |
| 22:10.49 | speedy | manxpower: the line is ok, is this something i can fix by configuration? |
| 22:10.51 | manxpower | speedy: Is that port an FXO port or an FXS package? |
| 22:10.57 | citywok | i wonder if it's not loading the modules i need, lol |
| 22:10.59 | manxpower | citywok: remove the package, install from source. |
| 22:11.05 | citywok | since it didnt load pbx_config automatically |
| 22:11.10 | speedy | manxpower: port 1 is FXS |
| 22:11.22 | manxpower | speedy: what brand of phone do you have plugged into that port? |
| 22:12.03 | manxpower | citywok: the fact it did not load pbx_config automatically means the package is REALLY BADLY broken. |
| 22:12.04 | speedy | manxpower: panasonic |
| 22:12.16 | citywok | alright, ./configure right now |
| 22:12.24 | manxpower | speedy: do you know the model number? |
| 22:12.36 | manxpower | Panisonic has like a zillion different phones. |
| 22:12.51 | speedy | 1 min |
| 22:13.05 | [TK]D-Fender | 1 min? Is that like 3com? |
| 22:13.18 | manxpower | I'll be trying to look up the specs for the phone, so accuracy is important. |
| 22:13.36 | speedy | KX-TG2340JXS |
| 22:14.13 | Cuban0 | it is supposed to retunr status ringing or busy first...and then when remote side picks up the phone |
| 22:14.16 | Cuban0 | says answered |
| 22:14.50 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 22:14.59 | [TK]D-Fender | Cuban0: By default analog channels do not have call progress and ar thus considered answered as soon as they are placed. |
| 22:15.29 | [TK]D-Fender | Cuban0: Call progress on analog can be unreliable. You can enable it with "callprogress=yes", but you risk random call drops. |
| 22:15.30 | Cuban0 | noway to process the signal and know when remote party answer ? |
| 22:15.49 | Cuban0 | ohhh thats a cool option i will try it...thanks |
| 22:16.13 | Cuban0 | callprogress is monitored by hardware on the TDM card ??? or by software ? |
| 22:16.28 | manxpower | Cuban0: If you use callprogress=yes you should expect calls to be randomly disconnected. |
| 22:17.23 | manxpower | speedy: what country are you in? |
| 22:17.32 | speedy | manxpower: Kuwait |
| 22:18.16 | Cuban0 | how call progress works on analog lines then ?? its some king of signal processing by software or DIGIUM hardware has some features for that ? |
| 22:18.25 | manxpower | speedy: those DAHDI messages indicate Asterisk thinks the phone went off hook, then went onhook again. I do not know if that phone is different from what Asterisk/DAHDI expects. |
| 22:19.07 | [TK]D-Fender | Cuban0: Software |
| 22:19.24 | manxpower | speedy: can you try a different phone? |
| 22:19.30 | [TK]D-Fender | Cuban0: this is audio processing, that is why its so random |
| 22:19.41 | citywok | manxpower: compiled from source, and it works just fine |
| 22:19.47 | Cuban0 | ooopssss i have bought TDM410p card to replace X100p thinking it will make the difference |
| 22:19.50 | citywok | i love not having any T1's anymore, 100% sip makes things so much easier |
| 22:19.50 | [TK]D-Fender | manxpower: maybe his phone is set to PULSE DIAL :p |
| 22:19.57 | manxpower | citywok: now you understand why we hate packages. |
| 22:20.03 | manxpower | [TK]D-Fender: excelent idea! |
| 22:20.07 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-24.tricom.net) |
| 22:20.24 | [TK]D-Fender | <_ SMRT |
| 22:20.24 | speedy | manxpower: I dont have one right now, if i unplug the phone, the messages will go away? |
| 22:20.39 | Cuban0 | i agree with the pulse dialing possibilty it switch the line constantly :p |
| 22:20.43 | manxpower | speedy: they should. If they don't then you may have a problem with the card |
| 22:22.06 | manxpower | speedy: Are you sure you have the power plugged into the card inside the PC? |
| 22:22.16 | citywok | manxpower: point taken, i'll stick to source for asterisk from now on lol |
| 22:22.31 | manxpower | citywok: How many hours did you waste? |
| 22:22.39 | speedy | manxpower: yes i've plugged the D style cable to the card |
| 22:22.48 | manxpower | speedy: Good. |
| 22:22.53 | citywok | about an hour and a half i'm guessing |
| 22:23.03 | speedy | manxpower: but i'm getting alarms about the power |
| 22:23.04 | manxpower | AHA! speedy: how often do you see those messages? |
| 22:23.05 | citywok | 5 minutes to compile it from source, including the time to download the tarball |
| 22:23.17 | speedy | Power alarm on module 1, resetting! |
| 22:23.20 | manxpower | citywok: that's not so bad. I've seen people waste days on precompiled packages. |
| 22:23.53 | speedy | manxpower: first few minutes after booting the machine |
| 22:24.09 | citywok | okay now i get originate failed still. back to the original issue hahaha |
| 22:24.19 | manxpower | Many phones check for a stutter dialtone so they can light up something on the phone if you have voicemail. |
| 22:24.33 | manxpower | they do this by going offhook and trying to listen for the stutter tone. |
| 22:25.33 | speedy | manxpower: I've unplugged the phone and the messages still showing (starting/hangup) |
| 22:26.10 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:26.46 | manxpower | speedy: you may have a bad card. Can you contact Digium support? |
| 22:27.46 | speedy | manxpower: I have OpenVox card |
| 22:28.08 | citywok | manxpower: http://pastebin.com/d4ca6071a |
| 22:28.09 | speedy | thats fine, i'll send call them to check whats wrong with the FXO module |
| 22:28.19 | speedy | s/FXO/FXS/ |
| 22:28.30 | buttons840 | manxpower, do you know anything about the possible values for response in an OriginateResponse event? http://lists.digium.com/pipermail/asterisk-users/2007-April/186038.html This is the extent of the information I've been able to find, however, unlike this guy I seem to gets lots of 3's and 5's for responses.? |
| 22:29.38 | manxpower | buttons840: No idea. I know the proper format of channels, not much AMI |
| 22:30.05 | buttons840 | k thx |
| 22:30.07 | speedy | I'm not sure if changing the place of the module will solve the problem |
| 22:30.39 | speedy | is there any tools to help for debugging? |
| 22:31.32 | manxpower | speedy: I've never seen or used an OpenVox card. |
| 22:31.42 | Cuban0 | i use it |
| 22:31.45 | Cuban0 | A400p |
| 22:31.51 | Cuban0 | never gave problems |
| 22:32.00 | Cuban0 | but check the POWER Source |
| 22:32.00 | speedy | Cuban0: I have the same model |
| 22:32.06 | Cuban0 | for low voltages |
| 22:32.28 | Cuban0 | you know for FXS it has an independent power conector directly from power source |
| 22:32.37 | speedy | thats right |
| 22:32.46 | Cuban0 | use a good ans stable power source on the PC |
| 22:33.14 | speedy | Cuban0: will try that |
| 22:33.17 | Cuban0 | cheap 400 W poower sources can run low on 12 volts line |
| 22:37.11 | *** join/#asterisk sjobeck (n=Adium@70.103.112.19) |
| 22:37.36 | box2 | yea, i always use a power conditioned UPS in my server room |
| 22:37.44 | *** part/#asterisk sjobeck (n=Adium@70.103.112.19) |
| 22:37.51 | box2 | a bit more expensive in general, but definately worth it |
| 22:38.16 | box2 | clean energy is safe energy! |
| 22:38.41 | box2 | reduces the shit out of random line noise too |
| 22:39.06 | Kobaz | crappy power supplies can wreck havok on components |
| 22:39.14 | Kobaz | i've had hard drive failures from bad psu's |
| 22:39.38 | xa0z | Anyone had any problems from PicoPSUs ? |
| 22:39.51 | box2 | never used one |
| 22:40.07 | xa0z | I bought one for my tiny itx box I built. |
| 22:46.41 | s14ck | is there a max. number for sip peers in Asterisk? |
| 22:47.27 | [TK]D-Fender | s14ck: No practical limit, though your system will simply choke after a point |
| 22:48.34 | carrar | 5 |
| 22:48.44 | carrar | depends what state your in |
| 22:48.52 | carrar | or country |
| 22:49.01 | s14ck | lol |
| 22:49.21 | s14ck | [TK]D-Fender, and SIP sessions? |
| 22:49.40 | Qwell | s14ck: same answer |
| 22:49.56 | Alfio | s14ck in venezuela you only will have as many as chavez want |
| 22:50.16 | s14ck | lol |
| 22:50.23 | s14ck | Alfio, whaz up man! |
| 22:50.28 | Alfio | fine and you |
| 22:50.52 | manxpower | In my experience once you hit about 300 channels you can expect to start having issues. That is assuming an ideal setup and design. |
| 22:50.54 | s14ck | Alfio, me fine too |
| 22:52.43 | s14ck | manxpower, thank you |
| 22:53.41 | *** part/#asterisk Cynagen (n=Cynagen@ip72-222-183-236.ph.ph.cox.net) |
| 22:54.59 | manxpower | I would not push it past 150 at the most. |
| 22:55.24 | speedy | manxpower: the problem seem to be in FXO module not the FXS, whenever I connect the line i get starting/hangup, btw the line is behind a DSL filter is that ok? |
| 22:55.26 | [TK]D-Fender | manxpower: More than that..... |
| 22:55.33 | [TK]D-Fender | manxpower: think Quad PRI... |
| 22:56.17 | leifmadsen | manxpower: version also plays an important part too |
| 22:56.20 | [TK]D-Fender | 120 channels in 1 card |
| 22:56.29 | leifmadsen | 1.6.x versions should handle more channels on the same hardware |
| 22:56.39 | [TK]D-Fender | leifmadsen: Application, transcoding, and recording, moreso :) |
| 22:56.45 | manxpower | leifmadsen: good point. My experience was with 1.4 |
| 22:56.47 | leifmadsen | [TK]D-Fender: also true |
| 22:57.04 | [TK]D-Fender | G.729 = insta-chop! |
| 22:57.08 | leifmadsen | there are additional core improvements that are too invasive for 1.4 |
| 22:57.19 | [TK]D-Fender | Lets add SWEC to the mix for flavour. |
| 22:57.25 | leifmadsen | but ya -- the things [TK]D-Fender outlines certainly have the heaviest effect |
| 22:57.44 | [TK]D-Fender | likes the blunderbuss "This WILL kill you" approach |
| 22:57.48 | manxpower | My experience was with SIP, no transcoding, high number of call setups/teardowns |
| 23:00.54 | Cuban0 | what are the main dvantages of Asterisk 1.6 over 1.4 ? |
| 23:01.04 | Cuban0 | never used 1.6 :S |
| 23:09.41 | _ShrikE | [TK]D-Fender: I have seen you recommend a POE switch in the past. Wasn't it a dlink? |
| 23:17.36 | s14ck | [TK]D-Fender, do you know if g729 was included into some Asterisk branch ? |
| 23:17.41 | s14ck | Alfio, |
| 23:17.54 | Alfio | s14ck yes |
| 23:18.11 | Alfio | s14ck i mean what |
| 23:18.11 | *** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30) |
| 23:18.14 | Alfio | ? |
| 23:18.54 | Alfio | s14ck sorry i was writing something, i dont said yes to your question |
| 23:19.06 | s14ck | well |
| 23:19.16 | s14ck | [TK]D-Fender, ? |
| 23:19.18 | Alfio | s14ck but you mean like free |
| 23:19.19 | Alfio | ? |
| 23:19.21 | manxpower | s14ck: G729 is a codec under patent protection. You can purchase a G729 license from Digium. You cannot legally get one for free. |
| 23:19.23 | s14ck | Alfio, yes |
| 23:19.27 | Alfio | nope |
| 23:19.37 | *** join/#asterisk skirmisha (n=asd@79-100-33-119.btc-net.bg) |
| 23:19.49 | skirmisha | hello guys |
| 23:20.04 | manxpower | This topic is the only one I have EVER seen Digium remove from the mailing lists. If they allow the conversation they could have their G729 license revoked. |
| 23:20.17 | skirmisha | any ideas how to get timestamp using agi->get_variable? |
| 23:20.27 | [TK]D-Fender | s14ck: G.729 is a SEPARATE download, always has been |
| 23:20.47 | skirmisha | will this work for me? |
| 23:20.51 | skirmisha | my $timestamp = $AGI->get_variable("STRFTIME(${EPOCH},,%F%T)"); |
| 23:21.13 | xa0z | If my softphone is connected to asterisk, and asterisk is connected to my provider, how do I ensure that calls to/from my extension are setup properly? |
| 23:21.18 | s14ck | manxpower, [TK]D-Fender and G729A? |
| 23:21.37 | skirmisha | same |
| 23:21.44 | skirmisha | g729 is not a free codec |
| 23:21.49 | manxpower | skirmisha: the Asterisk time will be the same as the system time. Use whatever you would use outside of Asterisk to get the time. |
| 23:21.53 | s14ck | manxpower, [TK]D-Fender *CLI> core show codecs |
| 23:21.55 | s14ck | <PROTECTED> |
| 23:22.23 | skirmisha | manxpower ok great |
| 23:22.25 | manxpower | s14ck: You can argue or you can accept what we say. "core show formats" will show you what Asterisk supports as codec. |
| 23:22.26 | skirmisha | another issue |
| 23:22.39 | s14ck | ahh |
| 23:22.40 | skirmisha | i am trying to insert data into mysql |
| 23:22.41 | s14ck | ok |
| 23:22.42 | manxpower | Sorry, "core show translations" |
| 23:22.47 | skirmisha | but not luck so far |
| 23:22.57 | skirmisha | shall i use dbh->do |
| 23:23.02 | skirmisha | or dbh->prepare |
| 23:24.08 | s14ck | manxpower, *CLI> core show translation |
| 23:24.10 | s14ck | <PROTECTED> |
| 23:25.09 | manxpower | there you go. |
| 23:27.04 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:27.22 | s14ck | manxpower, sorry but what do you mean? |
| 23:27.57 | manxpower | I mean that your Asterisk does not support G729 transcoding (required for anything useful with G729) |
| 23:28.03 | Cuban0 | that means G729 support is not present |
| 23:28.07 | Cuban0 | you need to put file |
| 23:28.22 | Cuban0 | codec_g729.so in your modules folder |
| 23:28.34 | Cuban0 | and then reload |
| 23:28.48 | manxpower | If you need G729 then call up Digium sales, pay a small license fee and get a legal G729 license. |
| 23:28.58 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 23:30.44 | manxpower | Remember g729 uses large amounts of CPU. You will not be able to handle many g729 calls if you have to do any transcoding. |
| 23:31.58 | Cuban0 | or have a XEON or Quad Core CPU :p |
| 23:33.30 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 23:34.37 | *** join/#asterisk lojine (n=loalw@75.33.45.65) |
| 23:35.03 | lojine | does anyone know if it's possible to connect a landline phone to a laptop with modem and get an interface |
| 23:35.05 | s14ck | manxpower, and G722? |
| 23:41.01 | *** join/#asterisk speedy (n=speedy@91.140.200.43) |
| 23:41.36 | manxpower | G722 should not be parented and is freely available. |
| 23:42.13 | manxpower | in the show translations output, and place two codecs intersect with a number means Asterisk can convert between those two codecs. |
| 23:42.59 | manxpower | The number is the complexity of the translation. It can give you an idea of the resource usage of translating between the two codecs |
| 23:42.59 | xa0z | Having a problem getting calls routed, can someone help me for a second? |
| 23:43.19 | manxpower | xa0z: put the output of a failed call on pastebin.ca |
| 23:43.51 | xa0z | It just says "Call from 'xxxxxxx' to extension 'xxxxxxx' rejected because extension not found. |
| 23:44.30 | exsync | has anyone dealt with phones that will register and work for a minute or two, then just stop incoming calls completely? but outgoing works? |
| 23:44.33 | manxpower | xa0z: 1) need to know the extension (the ACTUAL number) and a pastebin of the dialplan. I assume you are not using some sort of Asterisk GUI |
| 23:44.49 | xa0z | No GUI, it won't work. |
| 23:45.12 | s14ck | manxpower, you now if Asterisk have TAPI support? |
| 23:45.20 | manxpower | s14ck: It does not. |
| 23:45.36 | xa0z | Extension 123, Number is a CallCentric free Number, and the dialplan is just the from- and to- right? |
| 23:46.03 | manxpower | xa0z: dialplan in extensions.conf |
| 23:46.37 | manxpower | I suspect you think "s" extension is a "wildcard". A classic newbie mistake, but I won't know for sure until I see the dialplan and the actual error message |
| 23:46.38 | xa0z | this is all I have for now for starters... http://pastebin.ca/1517328 |
| 23:48.11 | s14ck | is there a max limit of conferences? |
| 23:48.51 | [TK]D-Fender | s14ck: Same kind of answer as for # of SIP devices |
| 23:48.57 | xa0z | Call from '17772876988' to extension '17772876988' rejected because extension not found. |
| 23:49.12 | xa0z | That is the error I got when calling that # from another callcentric account. |
| 23:49.12 | [TK]D-Fender | xa0z: And nothing in there can match that # |
| 23:49.19 | manxpower | xa0z: "s" does not mean "wildcard" it means "Asterisk did not receive the dialed number". Asterisk received the dialed number of "17772876988" so the call will never match "s" |
| 23:49.31 | xa0z | Oh. |
| 23:49.39 | speedy | if I connect the line to the box directly i get busy tone when i call the number, if i use ADSL splitter the phone rings then I get Dahdi Starting/Hangup, any hints? |
| 23:49.42 | manxpower | The only time "s" is used is on FXO ports, macros, and idiot voip providers |
| 23:49.55 | manxpower | xa0z: I recommend _1NXXNXXXXXX,1,Whatever |
| 23:50.13 | xa0z | erm |
| 23:51.02 | xa0z | on to, or from? |
| 23:51.31 | manxpower | whatever you have the context set to in sip.conf. I assume "from", but asterisk has NO defaults in that regard. |
| 23:52.19 | manxpower | speedy: if you plug a phone line into an FXS port and call the number you blow the port and it must be replaced. That is why I asked you what kind of port you have. |
| 23:52.29 | xa0z | Thanks. |
| 23:53.32 | speedy | manxpower: I'm connecting the phone line to FXO port not FXS port |
| 23:53.47 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 23:54.13 | xa0z | manxpower, okay that fixed the inbound calling, now my outbound calling issue if you have a moment. |
| 23:54.26 | speedy | according to dmesg: Module 3: Installed -- AUTO FXO (FCC mode) |
| 23:54.35 | manxpower | _XX will never match more than 2 gidits |
| 23:54.38 | manxpower | 'and digits too |
| 23:54.39 | speedy | which is the 4th port from left |
| 23:54.58 | xa0z | manxpower, ok. |
| 23:55.11 | manxpower | speedy: you don't have immediate=yes anywhere in your configs do you? |
| 23:55.48 | speedy | chan_dahdi.conf:immediate=no |
| 23:56.05 | manxpower | I have no idea of anything else to suggest, speedy. |
| 23:56.18 | xa0z | manxpower, the error is: Call from '123' to extension '17772034112' rejected because extension not found. |
| 23:56.26 | speedy | thats fine, thanks for your help manxpower |
| 23:57.02 | manxpower | xa0z: I guess you need _1NXXNXXXXXX in the outbound context to match the number you are dialing |
| 23:57.15 | xa0z | That is a 10 digit dialplan? |
| 23:57.40 | manxpower | that is 11 digits |
| 23:57.50 | manxpower | You need the number of digits your carrier expects |
| 23:58.02 | manxpower | xa0z: I think you should read the part of the Asterisk book about dialplans. |
| 23:58.03 | manxpower | ~book |
| 23:58.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 23:58.11 | xa0z | I will, thanks. |