| 00:00.47 | *** join/#asterisk BBHoss (n=bbhoss@68.159.148.181) |
| 00:04.55 | watchy2 | i think its gonna be easy . i just need to check if it was a transfer or a normal call |
| 00:08.22 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 00:09.00 | dwery | I upgraded from 1.4 to 1.6 . Using a SIP phone, while Playtones seems to work nicely, Backgrpund seems not. Did I missed something? |
| 00:09.15 | Qwell | dwayne: 1.6 what? |
| 00:09.29 | dwery | 1.6.0.3 |
| 00:09.37 | Qwell | upgrade |
| 00:09.46 | dwery | :( |
| 00:09.48 | voxter | Qwell: so i mirror packages.asterisk.org onto my own box here, but its a pain using wget --mirror and then removing all the crap html i get. Do you guys offer an rsync or anything? |
| 00:09.59 | dwery | Qwell: to which version? |
| 00:10.00 | [TK]D-Fender | dwery: And then don't jsut spout off stuff like that. Show us the broken call. |
| 00:10.03 | Qwell | voxter: hmm, we don't. |
| 00:10.11 | voxter | Qwell: boooo. |
| 00:10.23 | Qwell | voxter: how about using reposync from yum-utils? |
| 00:10.30 | dwery | [TK]D-Fender: there's nothing to show, just an extension wich calls Background(.... |
| 00:10.36 | Qwell | I think that's what it's called.. I know there's something in there to do it. |
| 00:10.49 | voxter | Qwell: yeah i might do that. im blending it with a mix of my own packages and building my own repo out of the sum of the parts, i wonder if that will still do the trick |
| 00:10.56 | Qwell | should |
| 00:11.00 | Qwell | http://yum.baseurl.org/download/yum-utils/ |
| 00:11.11 | [TK]D-Fender | dwery: Sounds like something to show |
| 00:11.14 | voxter | nod. thanks, ill check it out. |
| 00:11.48 | voxter | doh. of course, the repo is on a debian box.. ill deal with it later :) |
| 00:11.49 | dwery | [TK]D-Fender: http://www.privatepaste.com/fa1AMBKQVN I have an extension which plays invalid_code.sln |
| 00:11.49 | voxter | thanks for the tip though |
| 00:11.52 | bmoraca | does anyone know of a good utility or database that would allow me to find NPA-NXX by OCN and LATA? |
| 00:11.57 | Qwell | voxter: you can install yum on debian |
| 00:11.58 | dwery | [TK]D-Fender: for audio test purposes |
| 00:12.17 | [TK]D-Fender | dwery: ANSWER first |
| 00:14.03 | dwery | [TK]D-Fender: tried, same result |
| 00:14.28 | [TK]D-Fender | dwery: And use Playback |
| 00:15.45 | dwery | [TK]D-Fender: ok, but is Background supposed to work? the same extension worked fine in 1.4 |
| 00:15.59 | dwery | even the built-in demo does not work |
| 00:16.23 | [TK]D-Fender | dwery: Did it just work now? |
| 00:16.27 | dwery | no |
| 00:16.31 | dwery | not tried yet |
| 00:16.49 | [TK]D-Fender | dwery: Go try. |
| 00:16.56 | dwery | a seco, reloading :) |
| 00:17.06 | Qwell | <Qwell> upgrade |
| 00:17.32 | voxter | if qwell's telling you to upgrade, its for good reason dude. |
| 00:17.36 | dwery | [TK]D-Fender: it didn't worked |
| 00:17.47 | dwery | I'd upgrade immediately if someone has an 1.6 deb.. |
| 00:18.08 | [TK]D-Fender | dwery: Show me |
| 00:18.32 | dwery | [TK]D-Fender: http://www.privatepaste.com/ab1bdvHEX4 |
| 00:19.17 | [TK]D-Fender | dwery: What is that SIP device? Where is it located relative to *? |
| 00:19.51 | dwery | [TK]D-Fender: budgetone 101 , same subnet, the machine is multihomed |
| 00:20.04 | *** part/#asterisk xhoy (n=xhoy@80-69-92-244.colo.transip.net) |
| 00:20.12 | [TK]D-Fender | EW |
| 00:20.27 | [TK]D-Fender | dwery: Multi-homed is a PITA |
| 00:20.42 | dwery | I know, but still it used to work :D |
| 00:20.45 | [TK]D-Fender | dwery: comes in 1 port/interface out another... |
| 00:20.55 | [TK]D-Fender | dwery: I'm so not getting into this tonight |
| 00:21.05 | dwery | I'm tired too.. it's late here |
| 00:21.52 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 00:23.08 | dwery | I checed all the interfaces, there's no udp packet going out |
| 00:23.25 | dwery | oh well, I'll handle that tomorrow.. |
| 00:25.13 | dwery | see you all and thanks for the help |
| 00:26.26 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 00:27.44 | markit | hi, I'm upgrading italian sounds. What is the exact meaning of conf-lockednow, "the conference is now locked"? that is closed (destroyed) or that no one can enter anymore or whatelse? |
| 00:28.23 | watchy2 | hmm, i still can't get asterisk to work with my features in features.conf |
| 00:28.54 | watchy2 | when i hide # for a blind transfer it just ignores it |
| 00:29.47 | markit | watchy2: maybe your sip phone is eating it and does not pass it to asterisk? just guessing... I had similar problems with an ATA |
| 00:30.10 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 00:30.19 | watchy2 | what did you do to fix it? i'm using a polycom, i enabled ttmonkeys thing which is #9 and it works fine |
| 00:31.00 | markit | watchy2: well, my ata had a configurable sort of dialplan, and was set to "capture" #, I disabled that feature |
| 00:31.02 | watchy2 | but anything under [featuremap] doesn't work |
| 00:31.18 | watchy2 | i don't think thats my problems since the ttmonkeys ap works |
| 00:36.42 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:49.54 | watchy2 | i figured it out |
| 00:51.17 | Rabenklaue | I have to ask again as I'm not able to accomplish installing asterisk-1.6 with support for my HFC (Cologne BRI) card. |
| 00:51.43 | Rabenklaue | dahdi_hardware tells me: pci:0000:02:09.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card |
| 00:52.21 | tzafrir_laptop | Rabenklaue, I have already pointed you to the driver |
| 00:52.24 | Rabenklaue | I thought dahdi is the replacement of zaphfc, but why shows the dahdi tool that I should use zaphfc, or am I wrong? |
| 00:52.32 | tzafrir_laptop | I forgot a tiny patch |
| 00:52.44 | Rabenklaue | You pointed me to the zaphfc.[ch] |
| 00:52.51 | tzafrir_laptop | svn export svn://svn.debian.org/pkg-voip/dahdi-linux/trunk/debian/patches/bri_dchan bri_dchan.diff |
| 00:52.59 | Rabenklaue | But I don't know what to do with 'em |
| 00:53.21 | tzafrir_laptop | apply that patch with -p1 |
| 00:53.32 | tzafrir_laptop | copy those files to drivers/dahdi/ |
| 00:53.34 | Rabenklaue | to dahdi-linux? |
| 00:53.38 | tzafrir_laptop | yes |
| 00:54.13 | tzafrir_laptop | and edit drivers/dahdi/Kbuild to also build zaphfc , like it e.g. build wctdm |
| 00:54.45 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 00:55.01 | tzafrir_laptop | BTW: there's a small fix in SVN for detection of the resulting driver in dahdi-perl |
| 00:55.13 | tzafrir_laptop | (svn of dahdi-tools, that is) |
| 00:55.28 | Rabenklaue | tzafrir_laptop: Ok, then I'll have a try patching dahdi-linux. Thanks again |
| 00:56.57 | kerframil | 7/lastlog Rabenklaue |
| 00:57.00 | kerframil | oops |
| 00:57.06 | kerframil | Rabenklaue: wanted to see what was said before :) |
| 00:58.27 | kerframil | Rabenklaue: please let me know how you get on with that. I can't test it but I'm interested in getting the packages updated regardless. |
| 01:01.29 | tzafrir_laptop | kerframil, on what distro / platform? |
| 01:07.24 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:09.26 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279393977.dsl.bell.ca) |
| 01:12.52 | Rabenklaue | tzafrir_laptop: Thanks a lot - it worked. I know can load the zaphfc driver and dahdi_cfg also seems satified |
| 01:14.01 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 01:14.14 | Rabenklaue | kerframil: Tomorrow I'll post an updated dahdi ebuild for HFC cards including the patches tzafrir_laptop gave me. I hope I can help with it other (Gentoo) users with this problem |
| 01:15.12 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 01:15.42 | brunner | I need to alert a user when he/she is being unmuted in a conference. what's the best way to do that while the rest of the prompts to turned off? |
| 01:17.05 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-67d43c9760e07cb2) |
| 01:18.15 | [TK]D-Fender | brunner: "core show application chanspy" ,_ |
| 01:19.56 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:21.44 | *** join/#asterisk jameswf-home (n=james@unaffiliated/jameswf-home) |
| 01:28.43 | *** join/#asterisk Badrobot- (i=BadRobot@cpe-76-173-233-75.socal.res.rr.com) |
| 01:28.43 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 01:31.49 | Rabenklaue | hmpf, now I get this: http://dpaste.com/1632/ when calling asterisk -vvvvv... and it ends with a segmentation fault. |
| 01:32.12 | Rabenklaue | I think the problem is: ERROR[28481]: codec_dahdi.c:399 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory |
| 01:33.25 | Rabenklaue | But when loading dahdi_transcode , I see the following in dmesg: http://rafb.net/p/eNXfbF56.html |
| 01:34.22 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:36.08 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 01:42.14 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 01:44.42 | voxter | Ugh. I put an old T100p card in a box for a temporary solution while we do a T1 cutover and dahdi is really not happy with it. |
| 01:45.58 | brunner | [TK]D-Fender: I'm not clear on how I could use the whisper feature of ChanSpy to pipe audio to a user in the conference |
| 01:47.35 | [TK]D-Fender | brunner: Read its instructions a few more times till the one that clearly says this stands out |
| 01:48.15 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 01:53.04 | *** join/#asterisk minotaur01 (n=test@d36-87-247.home1.cgocable.net) |
| 01:53.34 | brunner | [TK]D-Fender: maybe you're having a bit of trouble understanding what I said. I didn't say, "I can't find the option to use." I said that it's not clear to me how to use it to accomplish my objective of piping audio to a user in a conference when he is being unmuted. |
| 01:53.45 | brunner | [TK]D-Fender: go back and re-read what I said until it's clear to you what I meant |
| 01:54.12 | [TK]D-Fender | bruYes you want to play a single announcement to him when he's unmuted. |
| 01:54.22 | [TK]D-Fender | brunner: And Chanspy can help you do that. |
| 01:54.31 | [TK]D-Fender | brunner: He is a channel in a conference. |
| 01:54.43 | stabler | ~nat |
| 01:54.43 | jbot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 01:54.50 | stabler | ~sipnat |
| 01:54.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:54.55 | [TK]D-Fender | brunner: Go look what chanspy can do to BARGE IN and play something to him |
| 01:55.01 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:55.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:55.02 | [TK]D-Fender | brunits all there. |
| 01:55.10 | voxter | Hey so i know the Digium T100p is a dinosaur and no longer officiallyt supported, but Im trying to get it to work with dahdi here and having a hell of a time. Anyone have any indepth experience with dahdi/t100p that might be able to help get this to function even in a minimal state? |
| 01:55.11 | [TK]D-Fender | brunner: its all there. |
| 01:55.23 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 01:55.36 | [TK]D-Fender | voxter: pastebin what you've done and where it looks to have gone wrong. |
| 01:56.16 | voxter | [TK]D-Fender: sure, I'll give it a try. its not a config thing, it seems to be a driver issue now that dahdi has evolved, its causing irq thrashing that results in the kernel disabling the IRQ entirely, simply by loading the driver. |
| 01:56.46 | [TK]D-Fender | voxter: Those cards had IRQ issues from the start |
| 01:57.09 | [TK]D-Fender | voxter: key reason they were so God-awfully finnicky with motherboards, and interrupt sharing |
| 01:57.32 | voxter | [TK]D-Fender: yeah i know. Im not intending on using it for more than 2 hours, i was hoping to use it while i moved a PRI, just to keep the old circuit up during the move. |
| 01:57.56 | voxter | [TK]D-Fender: it was used in this exact machine for at least 2 years wiht minimal issues, but now on the newer kernel/dahdi, it is cranky. |
| 01:58.11 | pfn | so back down to the older version |
| 01:58.47 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3b4e25c77b718fc1) |
| 01:58.56 | voxter | I think the older version was asterisk 1.2.1 era. I could i suppose, during the cutover at least. |
| 01:59.11 | voxter | just introduces a crapton more work :P |
| 01:59.43 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-43.hsd1.fl.comcast.net) |
| 01:59.48 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.128.72) |
| 02:00.04 | voxter | What was the last version of 1.4 that worked with zaptel, anyone remember? |
| 02:01.38 | voxter | This could also be a problem with the particular linux kernel im running with as well. |
| 02:02.26 | tuxd00d | my head is not in the game today, any of you know what I should look for when diagnosing "WARNING[4388]: chan_iax2.c:2212 __attempt_transmit: Max retries exceeded to host X.X.X.X on IAX2/XXXX-inbound-8345 (type = 6, subclass = 2, ts=7982329, seqno=171) |
| 02:02.26 | tuxd00d | My head is not in the game today, where should I start when diagnosing this: "WARNING[4388]: chan_iax2.c:2212 __attempt_transmit: Max retries exceeded to host X.X.X.X on IAX2/XXXX-inbound-8345 (type = 6, subclass = 2, ts=7982329, seqno=171)" |
| 02:02.30 | tuxd00d | . |
| 02:02.40 | tuxd00d | sorry, I didn't see that my messages where going through |
| 02:03.04 | mazpe | enable iax debug? |
| 02:03.22 | tuxd00d | mazpe: boy, I'm really not in the game |
| 02:07.08 | *** join/#asterisk LemensTS (n=customgt@adsl-75-42-150-213.dsl.stlsmo.sbcglobal.net) |
| 02:07.49 | tuxd00d | mazpe: thanks, I'll take a look |
| 02:07.56 | mazpe | np |
| 02:10.54 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 02:11.50 | joobie | i have 5 phones located at one site. Had an issue where all 5 phones kept coming up "unreachable" on the sip_poke_noanswer query very regularly. Looked into it, and it turned out to be an issue with the Internet line they were running through. I had that fixed up, now only 1 of the phones is experiecning this issue........ any idea how to troubleshoot this? |
| 02:11.58 | joobie | it basically comes up "unreachable" .. then it comes up "reachable" about 30s later |
| 02:12.04 | joobie | does it maybe once or twice an hour.. |
| 02:15.22 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 02:16.28 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 02:19.03 | riddlebox | joobie, what kind of phone? |
| 02:20.26 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 02:21.50 | mazpe | Anyone know how to access a Cisco 7960G menu, when the phone boots up its stock on Ethernet disconnected |
| 02:22.18 | obnauticus | Anyone here know how to make a realtime stream of a meetme conference? |
| 02:22.22 | stabler | mazpe: the settings button doesnt do anything? |
| 02:22.23 | mazpe | i tried holding #, headset, mute, speaker.. and no flashing lites |
| 02:22.28 | obnauticus | without having people connect, like a real time audio stream |
| 02:22.33 | obnauticus | one that i can put on a webpage |
| 02:22.34 | obnauticus | :\ |
| 02:22.42 | mazpe | stabler: no |
| 02:22.53 | tengulre | why not got any thing in txt file, vmstat 1 | awk '{print strftime("%H:%M:%S") $0 }' >> a.txt |
| 02:23.06 | stabler | mazpe: are you just trying to access the main menu |
| 02:23.33 | mazpe | Just got the phone trying to enable it for sip |
| 02:24.41 | voxter | hm. so this t100p issue appears to be kernel related. zaptel 1.4 does the same thing. :| |
| 02:25.01 | stabler | mazpe: good luck with that |
| 02:25.09 | mazpe | why? |
| 02:25.13 | obnauticus | mazpe i have a 7690 |
| 02:25.15 | stabler | mazpe: i still havent been successful converting my 7940 |
| 02:25.16 | obnauticus | on SIP |
| 02:25.16 | obnauticus | :P |
| 02:25.34 | obnauticus | I did it pretty easily, I figure i was RELLY lucky because I did it the first time without any problems. |
| 02:26.30 | stabler | ill probably give mine another whirl tonight |
| 02:28.05 | kerframil | Rabenklaue: great stuff, thanks |
| 02:29.07 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 02:41.44 | Rabenklaue | Now it seems my card is working properly: http://rafb.net/p/eI9bBY63.html |
| 02:42.12 | Rabenklaue | But I have no idea how to configure chan_dahdi.conf: http://dpaste.com/1644/ |
| 02:42.35 | Rabenklaue | dahdi show status: HFC-S PCI A ISDN card 0 [TE] layer 1 AC OK 0 0 0 |
| 02:43.05 | Rabenklaue | But: dahdi show channels: pseudo default de default |
| 02:44.02 | mazpe | do i need to have the cisco 7960 connected the network to do a reset to factory |
| 02:44.03 | Rabenklaue | So I think * PBX does not recognize the channels, on which the calls incoming and outgoing should be handled. |
| 02:44.07 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:44.27 | Rabenklaue | [Feb 26 03:44:10] WARNING[5611]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 02:44.44 | kerframil | Rabenklaue: yes, the sample file can be somewhat overwhelming |
| 02:45.40 | *** join/#asterisk RB2 (n=RB2@pool-72-88-225-136.nwrknj.east.verizon.net) |
| 02:46.35 | stabler | mazpe: no |
| 02:46.58 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 02:48.36 | kerframil | Rabenklaue: I would suggest putting the sample chan_dahdi.conf file aside and building a new one from scratch, taking it step by step |
| 02:48.39 | *** join/#asterisk andalou (n=chatzill@190.233.223.193) |
| 02:49.19 | Rabenklaue | kerframil: Ok, but that I'll tomorrow. Thanks a lot. |
| 02:49.39 | Rabenklaue | s/I'll tom/I'll do tom/ |
| 02:50.29 | *** part/#asterisk andalou (n=chatzill@190.233.223.193) |
| 02:50.30 | kerframil | Rabenklaue: very well. I'm not sure what the signalling method would be in your case, but you will probably be able to benefit from seeing my configuration file |
| 02:50.45 | joobie | riddlebox, polycom 320 |
| 02:50.54 | joobie | riddlebox, all the phones are polycom 320's.. |
| 02:51.11 | joobie | just one of them keeps dropping out though |
| 02:52.02 | *** part/#asterisk norgs (n=ken@WoWUIDev/Norganna/Administrator/Norgs) |
| 02:56.28 | *** join/#asterisk raasdnil (n=mikel@mx2.scientology.net.au) |
| 03:05.02 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 03:05.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:06.13 | *** join/#asterisk double_cheesburg (n=chatzill@ip70-162-238-155.ph.ph.cox.net) |
| 03:11.41 | double_cheesburg | I just installed my FXO port on asterisk. Does anyone know of an asterisk command to test it? |
| 03:11.52 | *** join/#asterisk Magicblaze007 (n=sony@fl-67-233-192-118.dhcp.embarqhsd.net) |
| 03:19.03 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 03:19.56 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 03:23.28 | Magicblaze007 | I just installed asterisk 1.6, anyone knows a good website to help me configure it/test it? |
| 03:23.35 | keith4_ | ~book |
| 03:23.35 | jbot | [book] probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:23.51 | Magicblaze007 | it says the book covers 1.4? |
| 03:23.56 | keith4_ | double_cheesburg: zttest? |
| 03:24.23 | keith4_ | Magicblaze007: http://www.voip-info.org/wiki-Asterisk ? |
| 03:24.40 | double_cheesburg | keith4: thanks. what does zttest do? |
| 03:25.01 | keith4_ | man zttest |
| 03:25.18 | keith4_ | or zt<tab> |
| 03:26.38 | double_cheesburg | keith4: zttest does nothign for me. |
| 03:26.41 | double_cheesburg | nothing |
| 03:26.51 | keith4_ | ztcfg helps, first |
| 03:26.55 | keith4_ | try ztcfg -vvv |
| 03:27.06 | keith4_ | then, there are a bunch of zt tools |
| 03:27.35 | keith4_ | like, ztmonitor, ztdiag, zttool, ztest, etc. |
| 03:27.38 | keith4_ | and they all have man pages |
| 03:28.39 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
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| 03:29.31 | double_cheesburg | keith4: thanks. |
| 03:31.46 | *** part/#asterisk woonix (i=woot@c-24-20-125-121.hsd1.or.comcast.net) |
| 03:37.10 | *** join/#asterisk geek (n=Yamato@unaffiliated/geek) |
| 03:40.42 | k-man | i have this billion ADSL modem with an ATA built into it - whenever i call someone on that, they say they hear a very bad echo of themselves. I also have a linksys SPA942, and i have no problems with that. do you think its just cos the Billion ADSL modem's ATA is crap? |
| 03:43.21 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 03:44.52 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-405ed66df69c7d66) |
| 03:52.35 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 03:52.57 | eric2 | conference calling... its only limited by the hardware of the server correct? |
| 03:53.32 | drmessano | As is everything |
| 03:55.40 | stabler | <PROTECTED> |
| 03:57.39 | *** part/#asterisk Magicblaze007 (n=sony@fl-67-233-192-118.dhcp.embarqhsd.net) |
| 03:57.56 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 03:59.01 | *** join/#asterisk maddog01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 04:00.58 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
| 04:06.47 | obnauticus | Does anyone here know how to make a live stream of a conference so I can embed it with a conventional ....streaming client on a webpage... |
| 04:06.54 | *** join/#asterisk SlicerDicer (n=SlicerDi@69-92-107-4.cpe.cableone.net) |
| 04:06.56 | obnauticus | one that's in realtime too...not buffered.. |
| 04:08.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:09.19 | kb3ien | my voicemail stopped working : http://pastebin.com/m33f02b51 anyone seen that before? |
| 04:12.31 | keith4_ | kb3ien: asterisk is unable to create /var/spool/asterisk/voicemail/pstn/13475553400/tmp/9apueY |
| 04:12.39 | keith4_ | probably a full volume |
| 04:12.42 | keith4_ | or a permissions problem |
| 04:13.27 | *** join/#asterisk geek (n=Yamato@unaffiliated/geek) |
| 04:13.43 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 04:15.33 | kb3ien | not that i see. asterisk is having the same problem even as root. and the volume is 0.5%full |
| 04:16.55 | kb3ien | the dir was created 644 root:root withou problem. |
| 04:20.29 | keith4_ | are you running asterisk as root? |
| 04:20.49 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.252) |
| 04:21.35 | keith4_ | also, that might not be right... because my voicemail dir is 750 asterisk:asterisk |
| 04:22.15 | thehar | mine is asterisk:asterisk u+wrx g+xr o+x |
| 04:24.03 | Carlos_PHX | Anyone ever deal with an Avaya One-X QE phone? Supposed to be SIP, but Avaya SIP != real SIP. |
| 04:29.50 | kb3ien | no its making the dir just fine, though which is weird. |
| 04:30.38 | keith4_ | that is weird |
| 04:33.09 | kb3ien | ah seems wav49 isnt a supported codec, removing it in vociemail.conf gets this: |
| 04:33.35 | kb3ien | [Feb 25 23:32:09] WARNING[17045]: translate.c:280 ast_translator_build_path: No translator path from unknown to g723 |
| 04:33.35 | kb3ien | [Feb 25 23:32:09] WARNING[17045]: file.c:183 ast_writestream: Unable to translate to format gsm, source format slin |
| 04:33.42 | kb3ien | over and over.... |
| 04:33.51 | kb3ien | but it recorded the message. |
| 04:34.59 | kb3ien | im happy keeping it in one format only. .wav is okay by me. i think. |
| 04:35.05 | kb3ien | any reason not to? |
| 04:35.18 | kb3ien | the gsm file is 0 bytes... |
| 04:39.54 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 04:42.14 | *** join/#asterisk mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:42.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:54.12 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 04:58.21 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-dbb85eeb2cc7c3c6) |
| 05:00.25 | keith4_ | kb3ien: are you missing audio libraries? |
| 05:00.47 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:00.51 | thehar | haha |
| 05:20.04 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 05:21.27 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 05:25.41 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 05:37.42 | double_cheesburg | how do you check to see what version of asterisk you are running? |
| 05:38.28 | stabler | double_cheesburg: what version did you install |
| 05:38.29 | stabler | lol |
| 05:38.44 | double_cheesburg | stable: hah hah ur so funny |
| 05:39.03 | double_cheesburg | nevermind. found it with ftp |
| 05:40.09 | stabler | core show version |
| 05:41.43 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:41.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:42.53 | mazpe | im out |
| 05:43.43 | joobie | guys im calling voicemail() to transfer a call to the voicemail system.. when it transfers, the lady on the ivr says "please leave your message after the tone.. press # after you've finished or hangup" .. is ther ea way i can disable that "please leave your message after the tone" part? |
| 06:02.11 | double_cheesburg | stabler: thanks |
| 06:04.13 | stabler | double_cheesburg: np |
| 06:18.12 | [TK]D-Fender | joobie: "core show application voicemail" |
| 06:25.54 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 07:00.12 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 07:35.06 | *** join/#asterisk bullium (n=will@216.68.250.30) |
| 07:35.16 | bullium | good morning all' |
| 07:38.19 | *** join/#asterisk xrmx__ (n=rm@host221-177-dynamic.25-79-r.retail.telecomitalia.it) |
| 07:40.13 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 07:46.10 | stabler | bullium: i havent been to bed yet |
| 07:46.13 | stabler | ;) |
| 07:46.31 | bullium | yeah me either and I've been up since 4AM :) |
| 07:46.45 | bullium | and I've been on the clock since 7:45AM ;) |
| 07:51.55 | stabler | heh |
| 07:53.54 | stabler | finals week for me =/ |
| 07:56.58 | *** join/#asterisk chi6IT41 (n=chigital@noc.mivitec.net) |
| 07:58.00 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 08:01.22 | *** join/#asterisk contactramu (n=contactr@123.201.69.115) |
| 08:05.59 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:12.05 | stabler | does anyone know what would cause an outbound call to go into a hold status until the other end picked up and then when they do pick up there is just dead air on both ends |
| 08:12.35 | stabler | all of this doesnt happen if the use has made a call into my *box |
| 08:12.40 | stabler | *user |
| 08:13.05 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 08:13.21 | stabler | on a prior occasion |
| 08:13.52 | stabler | also |
| 08:14.17 | stabler | if someone make an incoming call the first time they do both users get dead air |
| 08:14.34 | stabler | but when they call back a second time everthing is fine |
| 08:15.25 | stabler | its a very odd issues but i think it has to do with my *box |
| 08:21.30 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:25.00 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 08:34.43 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 08:34.53 | scruz | good day everyone |
| 08:35.14 | scruz | i'd really appreciate some help with building DAHDI |
| 08:41.00 | scruz | every time i try building dahdi-linux, i get an error that there's no rule to make |
| 08:42.05 | scruz | target /packages/dahdi-linux-2.1.0.4/drivers/dahdi |
| 08:42.54 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 08:47.27 | *** part/#asterisk contactramu (n=contactr@123.201.69.115) |
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| 08:50.31 | *** join/#asterisk stoked (n=df@S01060016b62857a6.vc.shawcable.net) |
| 08:51.24 | stoked | anyone around? Trying to use the stdexten macro, but seems to be failing |
| 08:52.34 | stabler | stoked: i was having issues with it too... :( |
| 08:54.25 | stoked | any luck? |
| 08:55.01 | kaldemar | the one in extesions.conf.sample? show a cli output of a failed call and we'll take a look. |
| 08:55.07 | tamiel | scruz: do you have linux kernel headers ? |
| 08:55.28 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:55.55 | stoked | kaldemar: I just copied the [stdexten] to my extensions.conf file from the sample |
| 08:55.58 | stabler | stoked: no |
| 08:56.04 | stoked | WARNING[16500]: app_macro.c:239 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten' |
| 08:56.05 | stabler | stoked: i didnt mess with it for long |
| 08:56.05 | kaldemar | how are you trying to use it? |
| 08:56.22 | kaldemar | it's not a macro, you call it with Gosub. |
| 08:56.45 | stoked | exten => 101,1,Macro(stdexten,100,SIP/aastra) |
| 08:56.49 | kaldemar | fail |
| 08:57.06 | stoked | oh... hmm |
| 08:57.35 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-73-96.w86-215.abo.wanadoo.fr) |
| 08:57.45 | mvanbaak | stoked: a macro should be called [macro-<name>] |
| 08:57.47 | kaldemar | erm. what version of asterisk do you have? |
| 08:57.56 | stoked | 1.6.0.6 |
| 08:58.05 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-73-96.w86-215.abo.wanadoo.fr) |
| 08:58.13 | mvanbaak | stoked: so in your situation change [stdexten] to [macro-stdexten] |
| 08:59.00 | stoked | mvanbaak: you mean for the macro's heading? or for the Macro( part |
| 08:59.28 | mvanbaak | stoked: the heading |
| 09:00.10 | stoked | ok thanks, I now get: Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1 |
| 09:00.15 | kaldemar | Gosub(stdexten,${EXTEN},stdexten[,ARGS]) |
| 09:01.03 | kaldemar | stoked: DON't rename it to a macro, because is is not a macro. use it with Gosub. |
| 09:01.24 | stoked | kaldemar: oh ok |
| 09:01.52 | mvanbaak | that's an option as well yeah |
| 09:02.41 | kaldemar | you'll end up rewriting most of it if you make it a macro. |
| 09:04.23 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 09:04.27 | kaldemar | my earlier gosub was wrong, you need () around arguments. see core show application gosub and the examples in default extensions.conf for more. |
| 09:04.46 | stoked | ok thanks kaldemar, I'll take a look |
| 09:07.50 | *** join/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch) |
| 09:09.09 | stoked | success |
| 09:09.12 | stoked | thanks kaldemar |
| 09:13.24 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 09:14.47 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) |
| 09:20.52 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 09:32.16 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 09:33.37 | *** join/#asterisk jbjapan (n=jbjapan@mail.biosjp.com) |
| 09:34.59 | jbjapan | Hi ! |
| 09:35.04 | *** join/#asterisk KU0N (n=kuon@217.144.50.4) |
| 09:35.24 | stabler | hi |
| 09:35.48 | dwery | Qwell: I upgraded :) now the situation is inverted: Background() works but Playtones() does not :D |
| 09:36.30 | stabler | lol |
| 09:36.49 | dwery | (that was from 1.6.0.3 to 1.6.0.6) |
| 09:38.33 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:39.08 | dwery | ok, Playtones requires Answer() while Background() does not |
| 09:48.35 | *** join/#asterisk jad_jay (n=chatzill@public.axolys.fr) |
| 09:50.15 | k-man | is this the correct way to get 2 extensions to ring on incomming call? exten => 02xxxxxxxx,1,Dial(SIP/1000,30&SIP/1003,30) |
| 09:51.22 | mvanbaak | k-man: nope |
| 09:51.34 | mvanbaak | k-man: Dial(SIP/1000&SIP/1003,30) |
| 09:52.04 | Rico29 | hello |
| 09:52.52 | Rico29 | is there a way to automatically delete auto-created peers (with autocreatepeers=yes) when their status goes "offline" ? |
| 09:54.58 | k-man | thanks mvanbaak |
| 09:56.24 | k-man | where can i read about the dial() syntax? |
| 09:57.23 | mvanbaak | core show application Dial |
| 09:57.38 | *** join/#asterisk jcollie (n=jcollie@fedora/jcollie) |
| 09:58.31 | k-man | what happens when you reach the timout in Dial()? |
| 09:59.24 | mvanbaak | goes to next prio |
| 10:00.13 | k-man | mvanbaak: thanks |
| 10:01.02 | Rico29 | anybody knows for my autocreatepeer problem |
| 10:01.03 | Rico29 | ? |
| 10:01.45 | *** join/#asterisk buzukelis (n=buzukeli@ip-195-14-171-9.bnk.lt) |
| 10:02.07 | k-man | whats the difference betwee Congestion and Busy? |
| 10:02.46 | buzukelis | how to escape quotation marks in func_odbc query? |
| 10:05.35 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 10:08.33 | *** join/#asterisk ultrav1olet (n=ultrav1o@94.180.49.133) |
| 10:09.07 | lanning | k-man, congestion is no available channels between end points. busy is end point is unavailable. |
| 10:09.32 | ultrav1olet | How can I Play() only if Dial returned CONGESTION? |
| 10:09.32 | k-man | lanning: i see.. |
| 10:10.45 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 10:11.21 | ultrav1olet | How can I Play() only if Dial returned CONGESTION? |
| 10:12.01 | *** part/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 10:12.10 | stoffell | can you see just by looking at a backtrace what is causing the segfault ? -> http://pastebin.com/d17cdc423 |
| 10:12.14 | Gido-E | ultrav1olet with a if? |
| 10:13.31 | *** join/#asterisk angryuser_ (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 10:13.58 | angryuser_ | hello |
| 10:15.21 | angryuser_ | what is the quantity of frames per packet for the g711 used by defaul in asterisk ? |
| 10:16.28 | angryuser | double login .. |
| 10:16.40 | angryuser | my question still stands |
| 10:18.07 | ultrav1olet | Gido-E: will try to look how to use that command |
| 10:19.05 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
| 10:19.22 | mvanbaak | angryuser: 20ms per packet I think |
| 10:20.09 | ultrav1olet | Gido-E: can you please post an example? I cannot find If usage on the net |
| 10:21.03 | angryuser | mvanbaak: ok, so g711 will take 87.2 Kbps per channel all signalling and headers included |
| 10:21.22 | mvanbaak | yup |
| 10:22.31 | Gido-E | ultrav1olet maybe? http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS |
| 10:23.53 | angryuser | i have one person which has 1 mbit sdsl guaranteed 2 mbit bursted when he has 5 communication in g711 the quality is crappy, i have tested the sdsl speed it caps 250 kb/s bursted so bandwith is fine, he had to use g729 to make this actually work, is there any way to mesure the packet loss for the sip ? |
| 10:24.37 | lanning | not sip, rtp |
| 10:24.48 | angryuser | his cpu is ok, so it is not a transcoding problem |
| 10:25.00 | lanning | you need to look at jitter during communications. |
| 10:25.23 | angryuser | lanning: how to ? (never used actually) |
| 10:25.53 | lanning | I have seen a lot of DSL that the latency spikes with higher traffic. the changing latency (jitter) is the death of VoIP. |
| 10:26.00 | mvanbaak | 5*87 = over 400 |
| 10:26.23 | Gido-E | angryuser 1mbit is only ~100KBytes/sec |
| 10:26.33 | lanning | the simplest is to ping the next hop, and run multiple channels out |
| 10:26.43 | mvanbaak | Gido-E: read again. 1mbit guaranteed, 2mbit burstable |
| 10:27.53 | angryuser | lanning: i think you got the reason, i will go read how to monitor jitter |
| 10:29.15 | angryuser | and during my test it was 250KBytes/sek even more than 2 mbits |
| 10:37.07 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 10:41.45 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:43.15 | ultrav1olet | Why PlayTones("IAX2/birdie-3225", "busy") in new stack |
| 10:43.22 | ultrav1olet | doesn't work for me? |
| 10:43.59 | ultrav1olet | It's a message from the asterisk console but I don't hear "busy" sound/signal |
| 10:46.20 | kaldemar | ultrav1olet: do you have a defined busy signal in indications.conf for the country you're using? |
| 10:47.10 | ultrav1olet | wait a minute, I'll check it out |
| 10:47.32 | ultrav1olet | Yes I have it |
| 10:47.45 | ultrav1olet | busy = 425/350,0/350 |
| 10:50.00 | kaldemar | hmm. maybe playtones needs an answer. |
| 10:50.27 | ultrav1olet | kaldemar: sure? |
| 10:51.13 | kaldemar | may depend on the current state of the call and the channel type. |
| 10:51.57 | ultrav1olet | Answer didn't help - no I've got silence ;) |
| 10:52.22 | kaldemar | sounds like you have some other issue then. |
| 10:52.39 | ultrav1olet | (before I had the indefinite answer tone - even though a remote side was busy) |
| 10:53.48 | kaldemar | how about using the Busy app? |
| 10:56.13 | ultrav1olet | I will try it right now. |
| 10:57.39 | ultrav1olet | <PROTECTED> |
| 10:57.44 | ultrav1olet | and nothing |
| 10:57.59 | ultrav1olet | I.e. I don't hear the busy signal |
| 10:58.41 | ultrav1olet | And according to Wiki: "Note that this command does not actually play a busy tone to the user." |
| 10:59.08 | ultrav1olet | kaldemar: are you sure this command should play the busy signal? |
| 10:59.53 | dwery | I compiled * 1.6.0.6 but chan_dahdi was not compiled . I think I'm missing some depends but dahdi includes are installed. anybody has an hint on what to check next? |
| 11:00.28 | *** join/#asterisk Dj-Neo (n=Jarrings@104.237-243-81.adsl-dyn.isp.belgacom.be) |
| 11:00.43 | kaldemar | ultrav1olet: i didn't say it would. it sends a busy indication, not necessarily by playing inband busy. |
| 11:00.45 | tzafrir_laptop | dwery, got an error? |
| 11:01.20 | dwery | tzafrir_laptop: none, it compiled and built the .deb correctly |
| 11:01.30 | dwery | I used your packages as a base |
| 11:01.40 | tzafrir_laptop | grep -i chan_dahdi menuselect.makedeps |
| 11:01.45 | tzafrir_laptop | err... |
| 11:01.47 | tzafrir_laptop | grep -i chan_dahdi menuselect.makeopts |
| 11:02.02 | dwery | MENUSELECT_DEPENDS_chan_dahdi=RES_SMDI DAHDI TONEZONE PRI SS7 |
| 11:02.05 | ultrav1olet | kaldemar: so how can I play a busy tone? |
| 11:02.12 | Dj-Neo | who know trixbox ? trixbox channel is sleeping :p |
| 11:02.23 | angryuser | Dj-Neo: well depends |
| 11:02.35 | dwery | MENUSELECT_CHANNELS=chan_console chan_dahdi chan_features chan_misdn chan_nbs chan_usbradio |
| 11:02.35 | dwery | MENUSELECT_DEPSFAILED=MENUSELECT_CHANNELS=chan_dahdi |
| 11:02.55 | Dj-Neo | i've find a howto install trixbox remotly to dedicated server but normaly I need to copy"cp trixbox.tar.gz /var/trixbox_load |
| 11:02.55 | Dj-Neo | " |
| 11:03.07 | Dj-Neo | I do'nt see trixbox.tar.gz on the iso file |
| 11:03.19 | Dj-Neo | this howto is for trixbox 2.2.4 |
| 11:03.19 | kaldemar | ultrav1olet: just like you have. do you have some other issue with audio? |
| 11:03.31 | ultrav1olet | no, everything works just fine |
| 11:03.44 | angryuser | Dj-Neo: dead end for me, search for the file in trixbox repos |
| 11:04.14 | dwery | tzafrir_laptop: it seems it failed ome depends. I have all the -dev installed for the libraries |
| 11:06.38 | ultrav1olet | kaldemar: I should have used Answer() instead of Answer |
| 11:06.51 | kaldemar | ultrav1olet: that makes no difference |
| 11:06.54 | ultrav1olet | Asterisk 1.6 ignores Answer without parameters |
| 11:07.45 | ultrav1olet | I mean without "()" |
| 11:10.06 | kaldemar | no it doesn't. |
| 11:11.36 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 11:11.45 | Assid | wassup peeps |
| 11:12.09 | Assid | if anyone knows anyone in asterlink please let me know .. their services are dead again |
| 11:14.17 | ultrav1olet | kaldemar: very strange! The first time I get busy - I hear no sound, the second time I call and get busy - I hear the sound, and so forth and so on |
| 11:16.19 | ultrav1olet | well, that's my own PC problem |
| 11:16.32 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 11:20.35 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 11:21.39 | tamiel | dwery: it seems you need libtonezone, libss7 and libpri dev packages. |
| 11:21.54 | dwery | tamiel: already installed. |
| 11:22.08 | dwery | tamiel: I'll try to re-configure again and check the log |
| 11:22.34 | *** part/#asterisk ultrav1olet (n=ultrav1o@94.180.49.133) |
| 11:22.42 | tamiel | dwery: ss7 is needed in 1.6 ? |
| 11:23.44 | dwery | tamiel: dunno, but it's there :) |
| 11:26.44 | dwery | seems fine, trying to compile without creating the .deb |
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| 11:34.28 | *** join/#asterisk ndalis (n=dalison@160.55-201-80.adsl-dyn.isp.belgacom.be) |
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| 11:35.02 | ndalis | hi. newbie here. may i ask a few questions ? |
| 11:35.32 | dwery | ~ask |
| 11:35.32 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:35.52 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:36.00 | ndalis | thanks jbot |
| 11:36.54 | *** join/#asterisk vyom_at_prg (n=chatzill@121.246.233.222) |
| 11:36.55 | ndalis | Question : Where can i find a list of all packages included in the asterisknow.iso distribution ? |
| 11:37.09 | vyom_at_prg | i need help on making outbound calls on Zap channels |
| 11:37.48 | vyom_at_prg | when i place a call on zap channel .. asterisk says "connected" even before the user actually picks up the phone |
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| 11:40.54 | vyom_at_prg | any one? |
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| 11:46.24 | *** join/#asterisk ndalis_ (n=dalison@213.26-240-81.adsl-dyn.isp.belgacom.be) |
| 11:46.39 | ndalis_ | Where can i find a list of all packages included in the asterisknow.iso distribution ? |
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| 11:57.07 | stoffell | can anyone point me to a direction as to why i would get these segfaults: http://pastebin.com/d17cdc423 |
| 11:58.00 | kaldemar | ndalis_: #asterisknow is a better place to ask, if there are any people there. |
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| 12:22.04 | dwery | found a compilation problem: in app_dahdiras.c (1.6.0.6) the include <usr/dahdi.h> must happen before sys/capability . |
| 12:22.47 | dwery | dahdi/user.h * |
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| 12:54.19 | Dj-Neo | Error: Missing Dependency: libnewt.so.0.51 is needed by package asterisk |
| 12:54.30 | Dj-Neo | How I can install this ? |
| 12:55.20 | *** part/#asterisk yahya (n=yahya@80.227.107.194) |
| 12:55.49 | tamiel | Dj-Neo: depends on your distribution packages system |
| 12:57.48 | Dj-Neo | centos |
| 12:58.01 | Dj-Neo | 5 |
| 12:58.04 | tzafrir_laptop | Dj-Neo, install using apt/yum rather than directly using dpkg/rpm |
| 12:58.16 | Dj-Neo | I use yum |
| 12:58.24 | angryuser | Dj-Neo: you search newt |
| 12:58.32 | Dj-Neo | yes |
| 12:58.34 | angryuser | yum* |
| 12:59.06 | tzafrir_laptop | Is that package intended for your distribution? |
| 13:00.39 | angryuser | Dj-Neo: type "yum install newt-devel.i386 newt" if you are using i386 |
| 13:01.35 | angryuser | or just newt-devel should work also |
| 13:02.27 | Dj-Neo | all is installed but always the error |
| 13:03.16 | tzafrir_laptop | rpm -ql news | grep libnewt.so |
| 13:03.24 | tzafrir_laptop | what version is it? |
| 13:03.44 | Dj-Neo | rpm -ql news | grep libnewt.so ==> No return |
| 13:04.26 | Dj-Neo | Package newt-devel - 0.52.2-10.el5.i386 is already installed |
| 13:04.46 | Dj-Neo | I need 0.51 |
| 13:04.54 | Dj-Neo | and I think it's 0.52 installed |
| 13:10.03 | *** join/#asterisk contactramu (n=contactr@123.201.69.115) |
| 13:10.52 | Rabenklaue | Hello. As posted tonigth, I need help configuring my chan_dahdi.conf - probably I misconfigured a thing concerning the signalling. Here are all (I think) necessary information: http://dpaste.com/1830/ |
| 13:11.26 | *** part/#asterisk contactramu (n=contactr@123.201.69.115) |
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| 13:16.41 | Dj-Neo | the problem is I try to install trixbox remotly to my dedicated serve r:p |
| 13:16.58 | *** join/#asterisk elbuit (n=toni@alabulie.ci.uv.es) |
| 13:18.32 | elbuit | hi all |
| 13:18.48 | elbuit | i've a issue with sip port on asterisk |
| 13:19.35 | elbuit | in sip.conf I put port=5070; but asterisk listen 5060!! |
| 13:19.54 | elbuit | I restart asterisk |
| 13:20.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:20.15 | elbuit | but also listen 5060 |
| 13:20.39 | elbuit | I use debian lenny |
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| 13:28.58 | kaldemar | elbuit: the correct parameter is bindport, not port. |
| 13:29.20 | elbuit | thanks |
| 13:29.27 | stoffell | Rabenklaue, are you connecting to a regular telco ? |
| 13:30.32 | elbuit | now work fine. |
| 13:30.54 | Rabenklaue | Telco? telephone company? Yes, my card is connected to the ISDN at home. It worked before with zaptel. |
| 13:31.19 | Rabenklaue | Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" (MASTER) AMI/CCS |
| 13:31.38 | Rabenklaue | The activated indicates, that it's correctly connected, or? |
| 13:32.01 | stoffell | yes, looks good.. you probably used similar config as before? (with zaptel) |
| 13:33.12 | stoffell | just a hint, if you pastebin your chan_dahdi.conf, you might awnt to leave out the comments (;) ;) |
| 13:33.29 | kaldemar | Rabenklaue: your signalling is now fxs_ks, it should be something else. and you don't even have any channels configured in chan_dahdi.conf. |
| 13:33.35 | Rabenklaue | Yes. My my /etc/dahdi/system.conf looks mostly the same as my zaptel.conf. Only the ec stuff is added |
| 13:34.17 | Rabenklaue | kaldemar: Do you've got an example how to configure channels? |
| 13:34.38 | Rabenklaue | The comments are not that easy to understand - for me |
| 13:35.32 | kaldemar | well, without uncommented channel => lines you have no channels, for starters. |
| 13:35.37 | tzafrir_laptop | Rabenklaue, signalling = bri_cpe ; for ptp, or bri_cpe_ptmp |
| 13:35.47 | stoffell | can't you take your old zaptel config, and use that ? |
| 13:35.58 | tzafrir_laptop | don't forget to set: switchtype = euroisdn |
| 13:36.13 | kaldemar | stoffell: that's not zaptel config, that's equivalent to zapata.conf. |
| 13:36.35 | Rabenklaue | bri_cpe_ptmp was the signalling from my zapata.conf - but as it was not shown within the comments, I changed it. |
| 13:36.45 | stoffell | ow, sorry, correct.. sorry for my ignorance ;) |
| 13:37.49 | Rabenklaue | channel => 1-2 ? That is taken from my old zapata.conf |
| 13:38.41 | Rabenklaue | module reload chan_dahdi.so : [Feb 26 14:38:19] ERROR[5983]: chan_dahdi.c:10955 build_channels: Unable to reconfigure channel '1-2' |
| 13:39.30 | kaldemar | IIRC you need a restart for asterisk if you change channel lines. |
| 13:40.04 | kaldemar | only (some?) parameters for channels can be changed with a reload. |
| 13:40.26 | Rabenklaue | [Feb 26 14:39:56] ERROR[18691]: chan_dahdi.c:11337 process_dahdi: Unknown signalling method 'bri_cpe_ptmp' |
| 13:40.40 | Rabenklaue | And dahdi is not available in CLI. |
| 13:40.52 | tzafrir_laptop | Rabenklaue, do you use 'dahdi restart' ? |
| 13:40.53 | stoffell | did you also update libpri? |
| 13:41.22 | tzafrir_laptop | ah, right: no pri support |
| 13:44.47 | *** part/#asterisk elbuit (n=toni@alabulie.ci.uv.es) |
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| 13:46.35 | Rabenklaue | With signalling=pri_cpe it seems to work now. Thanks a lot |
| 13:47.17 | tzafrir_laptop | Rabenklaue, works? calls flow? |
| 13:47.26 | tzafrir_laptop | What version of libpri do you have? |
| 13:47.29 | *** join/#asterisk mortum (n=fjbf@TELEYECLA-89-29-236-3.teleyecla.net) |
| 13:47.36 | tzafrir_laptop | What version of asterisk? |
| 13:48.05 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 13:48.54 | Assid | asterlink just killed my business :( |
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| 13:52.35 | Rabenklaue | tzafrir_laptop: asterisk shows me the channels, I've configured. But when trying to call I get the following: [Feb 26 14:48:48] WARNING[5987]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/101-081ec830' status is 'CONGESTION' |
| 13:52.50 | Rabenklaue | net-libs/libpri-1.4.7 and net-misc/asterisk-1.4.22.1 |
| 13:53.44 | tzafrir_laptop | ok. that's because of asterisk 1.4.22 |
| 13:53.50 | Rabenklaue | And in dmesg I see all the time: zaphfc: bchan rx fifo not enough bytes to receive! (z1=7215, z2=7208, wanted 8 got 7), probably a buffer overrun. |
| 13:53.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:54.14 | Rabenklaue | What version of * do you prefer? |
| 13:54.30 | tzafrir_laptop | Rabenklaue, maybe it will work for you with PtP . surely not with PtMP |
| 13:55.45 | Rabenklaue | How can I see, whether I'm using PtP or PtMP and is there a way to change it? |
| 13:58.22 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 13:58.33 | scruz | good day everyone |
| 13:58.51 | scruz | what's the difference between a sip peer and a sip client? |
| 14:00.23 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 14:03.09 | jad_jay | scruz: not sure but a peer is for example a sip operator, and a sip client (maybe friend) is a device |
| 14:04.03 | jad_jay | friend can call and receive, maybe client could only receive... |
| 14:04.37 | jad_jay | scruz: does this match for you? |
| 14:04.46 | jad_jay | scruz: ~book |
| 14:04.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:04.54 | jad_jay | ~book |
| 14:04.54 | jbot | methinks book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:06.30 | Assid | err anyone played with jabber module? |
| 14:07.21 | *** join/#asterisk syere (n=syere@204.10.20.30) |
| 14:07.53 | Assid | i added the information in the jabber.conf file.. but i dont see that user online |
| 14:10.08 | Rabenklaue | tzafrir_laptop: But incoming calls are handled correctly. Better than nothing to start with. Thank you very much. |
| 14:11.06 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 14:12.56 | Rabenklaue | Also the connection is aborted after 3 seconds and than the voice tells me: "the person you've called is temp..." |
| 14:13.45 | Rabenklaue | I'll have a look on * 1.6 and hope the DAHDI implementation is better there. |
| 14:13.50 | *** join/#asterisk Dj-Neo (n=Jarrings@140.228-244-81.adsl-dyn.isp.belgacom.be) |
| 14:14.15 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 14:18.08 | syere | hello all. I'm looking through the asterisk docs and such but I am having a hard time finding exactly where the config file for adding a voip carrier is |
| 14:19.28 | [TK]D-Fender | syere: sip.conf , iax.conf depending |
| 14:19.56 | syere | [TK]D-Fender, im new to voip, is there a page that will define the difference for me? |
| 14:20.10 | [TK]D-Fender | syere: in the BOOK |
| 14:20.12 | [TK]D-Fender | ~book |
| 14:20.12 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:20.17 | [TK]D-Fender | syere: they are jsut different protocols. |
| 14:20.22 | aksyn | anyone know what MS OCS R2 uses for key negotation on SRTP? |
| 14:20.27 | syere | [TK]D-Fender, thank you |
| 14:22.36 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 14:22.37 | *** mode/#asterisk [+o file] by ChanServ |
| 14:26.37 | dwery | tzafrir_laptop: rebuilt asterisk and chan_dahid, it now runs. I'm now trying to define the channels. pri_cpe_ptmp is Unknown signalling method 'pri_cpe_ptmp' |
| 14:27.13 | tzafrir_laptop | what version of libpri? |
| 14:27.30 | Assid | anyone know asterlink personally |
| 14:27.42 | Assid | i seriously need to switch vendors quick |
| 14:27.50 | Assid | these guys are causing downtimes icant afford |
| 14:27.57 | dwery | tzafrir_laptop: 1.4.7-0.6459 |
| 14:28.05 | Assid | and no decency to have a emergencycontact number |
| 14:28.36 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 14:29.33 | tzafrir_laptop | dwery, bri_cpe_ptmp |
| 14:29.38 | tzafrir_laptop | (s/p/b/ |
| 14:29.40 | tzafrir_laptop | ) |
| 14:29.58 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 14:30.43 | dwery | ok. seems fine with that. I get a warning there's no D channel available |
| 14:31.00 | mort_gib | dwery: Sorry, what hardware and what Asterisk version?? |
| 14:31.28 | dwery | mort_gib: 1.6.0.6, zaphfc driver dahdi with ISDN PCI card by asus |
| 14:32.08 | dwery | how should the D channel be defined in chan_dahdi.conf ? |
| 14:32.09 | mort_gib | dwery: okay, I tried Asterisk 1.4. but it will only work in 1.6. so you should be fine |
| 14:32.23 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:32.34 | dwery | mort_gib: hope so :) |
| 14:32.47 | mort_gib | dwery: You have an issues with libpri though, that's why the signalling in unavail |
| 14:33.13 | dwery | I think I'm configuring the 3rd channel in a wong way |
| 14:33.37 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 14:33.44 | dwery | chan_dahdi.c: Signalling requested on channel 3 is ISDN BRI Point to MultiPoint but line is in Unknown signalling 896 signalling |
| 14:34.26 | fcois93 | hello all |
| 14:34.52 | fcois93 | is it possible to limit the number of members in a conference room (meetme app) ? |
| 14:35.00 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
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| 14:35.37 | mort_gib | dwery: You'r in UK?? |
| 14:35.45 | dwery | mort_gib: IT |
| 14:35.56 | mort_gib | dwery: ?? |
| 14:36.02 | dwery | mort_gib: italy |
| 14:36.49 | coppice | fcois93: easy. discuss the drying of paint |
| 14:36.55 | mort_gib | dwery: sorry :-) point to multipoint sounds right, but if that signalling is not available, then you have not included libpri at the right point in time |
| 14:37.15 | mort_gib | dwery: Strange card though! |
| 14:37.25 | dwery | ptmp is available, is that 896 that it isn't I included it before compiling asterisk |
| 14:37.38 | dwery | mort_gib: why strange? usual single BRI |
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| 14:39.07 | mapmagic | can asterisk be made to do all its communications using one port (like 5060?) |
| 14:39.16 | mapmagic | I dont want to open a lot of ports on my machine. |
| 14:41.02 | jaytee | mapmagic, nope |
| 14:41.15 | jaytee | mapmagic, you still need ports for rtp |
| 14:41.28 | jaytee | and if you want to remote manage the server a port for ssh |
| 14:41.35 | mort_gib | dwery: Asus card. -Right??? |
| 14:41.56 | mapmagic | my ssh ports are already open. I just didn't want to open too many ports. |
| 14:42.05 | dwery | mort_gib: yes. it's the usual HCF rebranded |
| 14:42.21 | mort_gib | dwery: You get HWEC?? |
| 14:42.51 | dwery | mort_gib: please define HWEC? hw echo cancel? I guess not |
| 14:43.01 | dwery | it's apassive one |
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| 14:43.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:44.01 | fcois93 | coppice: what ? |
| 14:44.09 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 14:44.46 | Vec | Do Grandstream phones tag the voice data and/or the PC data when using 802.q VLAN tagging ? |
| 14:45.10 | Vec | pretty sure its or but not sure which one voice or data port ? |
| 14:45.59 | mort_gib | dwery: Hardware Echo Cancellation |
| 14:47.25 | [TK]D-Fender | mapmagic: This has been said before so I'll say it again now : it doesn't matter how many RTP ports you leave open, its all the same DAEMON LISTENING TO THEM. |
| 14:47.26 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 14:47.31 | dwery | I tried to remove the definition for the third channel from the .conf and I get no more errors. let's try if I can reeive a call |
| 14:48.05 | [TK]D-Fender | fcois93: it was a joke from coppice |
| 14:48.26 | mapmagic | [TK]D-Fender: Thanks. |
| 14:48.31 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 14:48.48 | fcois93 | [TK]D-Fender: I dont understand english joke ... |
| 14:49.09 | fcois93 | my question again, is it possible to limit the number of members in a conference room (meetme app) ? |
| 14:49.18 | Mr_BOnD_007 | if i want to use only GUI dont Want to Dial The Number is it possible ? in asterisk ? just it will show the Number Name Street Disposition ETC. (VICIDIAL ) |
| 14:49.19 | [TK]D-Fender | fcois93: Comment limiter le nombre de gens dans un conference? Change le sujet pour discuter la peinture se seche :) |
| 14:49.40 | [TK]D-Fender | fcois93: Limit them yourself in the dialplan with something like GROUP_COUNT |
| 14:53.14 | dwery | ok, receving a call seems to work nicely |
| 14:53.25 | dwery | MOH works fine too. |
| 14:53.36 | dwery | I got zaphfc: dropped audio (z1=7797, z2=7780, wanted 8 got 17, dropped 9). once |
| 14:53.55 | dwery | and a bunch of WARNING[8219]: chan_dahdi.c:3347 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! |
| 14:56.25 | dwery | now it's time to try an outbound call |
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| 15:01.06 | dwery | something worked, but I got a voice from the telco saying nknown number.. |
| 15:02.54 | dwery | curious.. I can call my llandline but not my cell phone |
| 15:04.55 | dwery | it also seems that dahdi cannot detect when one of the B channels is in use by another TE |
| 15:06.03 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:08.33 | *** join/#asterisk minotaur01 (n=test@mail.upperjamestoyota.ca) |
| 15:09.11 | jaytee | fcois93, you could use MeetMeCount() in your dialplan for MeetMe to get the current count of the number of people in a conference and use a GotoIf to jump to a labeled priority that plays a recording if the max number of members has already been reached. |
| 15:09.37 | *** join/#asterisk freckle (n=chatzill@195.74.96.119) |
| 15:09.44 | scruz | jad_jay: thanks |
| 15:10.37 | fcois93 | jaytee: thank you, it is the first time I use meetme for that specific use :) |
| 15:14.10 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.70) |
| 15:14.59 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 15:15.01 | Greek-Boy | is there any way in asterisk to convert a number to decimal spacing? For eg, 123567 should be 1,234,567 |
| 15:16.21 | *** join/#asterisk minotaur01 (n=test@mail.upperjamestoyota.ca) |
| 15:16.50 | manxpower | Greek-Boy: see channelvariables.conf, pay special attention to the syntax for extracting ranges out of a variable |
| 15:17.35 | Greek-Boy | manxpower: will do, thanks |
| 15:19.46 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 15:20.43 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 15:20.43 | *** mode/#asterisk [+o Mog] by ChanServ |
| 15:22.56 | *** join/#asterisk chi6IT41 (n=chigital@noc.mivitec.net) |
| 15:24.41 | *** join/#asterisk stmaher (n=stephen@mateus.province5.tv) |
| 15:24.43 | stmaher | hello everyone.. |
| 15:24.57 | stmaher | I just installed 1.6 fresh on a machine.. |
| 15:25.03 | stmaher | I have two hard sip phones.. |
| 15:25.10 | stmaher | regsitered and working.. but cannot call between them.. |
| 15:25.16 | stmaher | saying extension does not exist |
| 15:25.22 | stmaher | yet I have it hard coded in sip.conf |
| 15:25.23 | stmaher | any ideas? |
| 15:25.30 | stmaher | Thanks |
| 15:25.36 | manxpower | stmaher: you don't put extensions in sip.conf |
| 15:25.39 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 15:25.47 | stmaher | sorry.. extensions.conf |
| 15:26.00 | stmaher | the sip phones are hard coded in sip.conf and extensions is simply |
| 15:26.07 | [TK]D-Fender | stmaher: Then * isn't looking where you think it should be looking. |
| 15:26.16 | manxpower | stmaher: then you have a context issue. put the output of a failed call on pastebin.ca |
| 15:26.18 | [TK]D-Fender | stmaher: pastebin is your friend. |
| 15:26.19 | stmaher | exten => 7101,1,Dial(SIP/7101) |
| 15:26.19 | stmaher | exten => 7301,1,Dial(SIP/7301) |
| 15:26.28 | [TK]D-Fender | stmaher: PASTEBIN the entire thing |
| 15:26.36 | [TK]D-Fender | stincluding the failed call with SIP DEBUG enabled |
| 15:26.37 | [TK]D-Fender | ~pb |
| 15:26.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:26.41 | stmaher | the context is default |
| 15:26.49 | [TK]D-Fender | stmaher: Means little. PASTBIN IT |
| 15:26.57 | manxpower | stmaher: less talk, more pastebining |
| 15:27.08 | jaytee | TRABAJO! |
| 15:27.09 | [TK]D-Fender | tags manxpower into the ring |
| 15:27.12 | stmaher | thanks guys.. effecient as ever ;-) |
| 15:27.31 | [TK]D-Fender | jaytee: lol... has been a little quiet without him, hasn't it? |
| 15:27.53 | jaytee | real wrestlers do not wear eyemakeup or use folding chairs to beat people....but it is entertaining |
| 15:28.08 | jaytee | [TK]D-Fender, yeah! where is eppigy lately? |
| 15:28.09 | manxpower | At least UFC is more real |
| 15:28.35 | [TK]D-Fender | manxpower: "More real" (tm) |
| 15:28.51 | [TK]D-Fender | manxpower: Most of them are &#^$ing ass-clowns |
| 15:29.00 | jaytee | [TK]D-Fender, "Almost Authentic" (tm) |
| 15:29.12 | [TK]D-Fender | manxpower: Wired psycho idiot who make wrestling look good. |
| 15:29.38 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-95afeddcdaf2db97) |
| 15:29.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:29.44 | stmaher | http://www.pastebin.ca/1347840 thanks guys |
| 15:31.09 | [TK]D-Fender | stmaher: NONE of your contexts specified in sip.conf point to [default] |
| 15:31.36 | stmaher | ah poo! |
| 15:31.38 | stmaher | thanks |
| 15:31.44 | manxpower | see how important pastebin is? |
| 15:31.48 | [TK]D-Fender | flattens stmaher with a folding chair and tags manxpower back in |
| 15:31.55 | stmaher | LOL |
| 15:32.03 | manxpower | We could have spent HOURS trying to figure out what was wrong. |
| 15:32.06 | Greek-Boy | manxpower: If I a |
| 15:32.24 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 15:32.34 | [TK]D-Fender | stmaher: If it wasn't as obvious what things were landing on I'd have had to blast you for not providing the failed call with SIP DEBUG like I asked :) |
| 15:32.59 | Greek-Boy | manxpower: If I'm not mistaken, I have to set a variable for every three characted starting from the back and then set those three variable back into one variable containing the decimals? |
| 15:33.53 | dwery | ok, I think I have a mostly working configuration. In order to not let te memory fade away, where's the best place to contribute it? |
| 15:33.59 | [TK]D-Fender | Greek-Boy: I would make a Macro to do this so its reusable. |
| 15:34.21 | manxpower | or something like Set(NEWVAR=${EXTEN:x:x},${EXTEN:x:x},${EXTEN:x:x}) I would have to look up what you need to put in for x |
| 15:34.50 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) |
| 15:35.00 | manxpower | I once made a macro to add natural sounding pauses when reading numbers. |
| 15:36.07 | [TK]D-Fender | manxpower: Your method there works if the length is fairly limited. |
| 15:36.26 | [TK]D-Fender | manxpower: A decent start :) |
| 15:41.57 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 15:42.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:42.35 | beek | [TK]D-Fender: morning |
| 15:42.50 | [TK]D-Fender | beek: Mornin' |
| 15:47.33 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:49.17 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-164-14.lns10.mel4.internode.on.net) |
| 15:51.13 | manxpower | [TK]D-Fender: it was only intended for phone numbers |
| 15:51.54 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 15:51.54 | Yourname` | Hi, is the Digium FTP down? |
| 15:51.54 | [TK]D-Fender | manxpower: Yeah... not quite what he asked for, but a similar theme : [10:14]<Greek-Boy>is there any way in asterisk to convert a number to decimal spacing? For eg, 123567 should be 1,234,567 |
| 15:51.54 | scruz | i just built asterisk (again) and it's not loading chan_sip.so even thought it was selected using make menuselect |
| 15:52.08 | [TK]D-Fender | Yourname`: Digium doesn't do FTP :) |
| 15:52.20 | [TK]D-Fender | scruz: Check your modules.conf |
| 15:52.23 | tzafrir_laptop | Yourname`, ftp2.digium.com? yes. http://downloads.digium.com/ ? |
| 15:52.25 | Yourname` | [TK]D-Fender: Yes it did. ftp.digium.com, I used it many times :( |
| 15:52.36 | [TK]D-Fender | Yourname`: Look again. |
| 15:52.36 | Yourname` | tzafrir_laptop: ftp2 it is now? and it's down? |
| 15:52.56 | Yourname` | tzafrir_laptop: You know when it's going to be back up? |
| 15:52.58 | tzafrir_laptop | no. it's http://downloads.digium.com/ |
| 15:53.06 | scruz | silly me. i don't have a modules.conf |
| 15:53.12 | [TK]D-Fender | Yourname`: Like I said, no FTP |
| 15:53.13 | scruz | :( |
| 15:53.23 | scruz | [TK]D-Fender: thanks |
| 15:53.31 | [TK]D-Fender | scruz: Start with the sample one and noload specific things you don't want |
| 15:53.43 | Greek-Boy | manxpower: asterisk doesn't like the , - it things i am trying to set multiple variables |
| 15:54.13 | scruz | [TK]D-Fender: i didn't do "make samples" |
| 15:54.17 | Yourname` | thanks guys |
| 15:54.25 | [TK]D-Fender | Greek-Boy: \o/ for lack of typed values in the dialplan! |
| 15:54.34 | [TK]D-Fender | Greek-Boy: Do it in an AGI then |
| 15:55.00 | *** join/#asterisk SparFux (n=raoul@e182019097.adsl.alicedsl.de) |
| 15:55.13 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 15:59.12 | manxpower | you should be able to escape it with \, |
| 16:02.10 | tzafrir_laptop | only now realized that above we have an escaped o and a /. What's this subdirectory o? |
| 16:02.19 | *** join/#asterisk watchy2 (n=watchy@76.196.98.139) |
| 16:10.18 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 16:12.31 | *** join/#asterisk docelmo (n=vircuser@pool-141-152-221-248.lyn.east.verizon.net) |
| 16:12.36 | stmaher | does anyone know what KM... displayed on a PolyCom IP430 phone mean? |
| 16:13.05 | [TK]D-Fender | stmaher: Displayed where? |
| 16:13.15 | stmaher | On the hard phone itself? |
| 16:13.48 | [TK]D-Fender | stmaher: is that a QUESTION? |
| 16:13.57 | [TK]D-Fender | stmaher: I asked YOU where you see this "KM" |
| 16:14.02 | *** join/#asterisk supjigator (n=shanebur@64.147.216.10) |
| 16:14.16 | stmaher | [TK]D-Fender Yes ON THE PHONE ITSELF |
| 16:14.34 | supjigator | Anyone have any issues binding asterisk to two nics? |
| 16:14.35 | [TK]D-Fender | stmaher: where on the phone? |
| 16:14.49 | stmaher | LCD display right hand side under the extension number |
| 16:14.56 | dwery | supjigator: works on 4 nics here, no special setup |
| 16:15.04 | [TK]D-Fender | stmaher: Under or next to? |
| 16:15.09 | stmaher | under |
| 16:15.12 | stmaher | in caps |
| 16:15.20 | stmaher | with ... to the right of it.. |
| 16:15.22 | stmaher | KM... |
| 16:15.42 | manxpower | stmaher: sounds like you figured the line appearance that way in the config file. |
| 16:15.44 | [TK]D-Fender | stmaher: Since there are 4 lines of "text" on the screen, you're telling me you see this in the soft-key bar? |
| 16:16.12 | [TK]D-Fender | stabler: is it on a LINE KEY, or somewhere else ont he screen? |
| 16:16.20 | [TK]D-Fender | stmaher: is it on a LINE KEY, or somewhere else ont he screen? |
| 16:17.00 | supjigator | dwery: I'm getting one-way audio during.some calls. Sometimes it comes back. |
| 16:17.20 | stmaher | [TK]D-Fender one sec.. taking pic |
| 16:17.33 | dwery | supjigator: you can see the rtp stream with the debug options . it will tell if packets are received or transmitted |
| 16:17.44 | manxpower | "Timmy, show me on the picture where the phone touched you" |
| 16:17.59 | supjigator | dwery: I tried that but its too much data to sift thru. |
| 16:18.10 | supjigator | Best I can tell internal phone nic is good. |
| 16:18.31 | supjigator | I'm loosing it on the external or within asterisk. |
| 16:18.34 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:18.45 | dwery | supjigator: aterisk version? |
| 16:18.50 | supjigator | its always tx |
| 16:20.46 | supjigator | dwery: Asterisk 1.4.22-rc5 |
| 16:21.14 | dwery | supjigator: I had .21 for a while and worke nicely, and I'm now testing 1.6 . I think I cannot help further :( |
| 16:22.13 | stmaher | [TK]D-Fender http://www.lavalan.net//wtfiskm.jpg |
| 16:23.42 | [TK]D-Fender | stmaher: You have allocated 1 line key to a reg, and the other has fallen to an entry in your DIRECTORY as a speed dial |
| 16:24.12 | [TK]D-Fender | stmaher: Go read the User's Guide for your phone. |
| 16:24.28 | stmaher | cheers thanks |
| 16:25.21 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:26.23 | *** join/#asterisk boomboom99 (n=boomboom@c-71-229-40-177.hsd1.ga.comcast.net) |
| 16:27.04 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:27.17 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:28.51 | boomboom99 | I have 3 offices with an asterisk server in each, all linked to each other by IAX. anyone have suggestions on how to have a call queue with static members in each of the offices? |
| 16:29.04 | *** join/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 16:29.06 | martyn-dev | Hi |
| 16:29.38 | boomboom99 | for example: call comes into office1 and joins a queue. the queue rings one person in office2 and another in office3 using "ringall" |
| 16:30.19 | boomboom99 | those 2 users are connected to their local asterisk servers using SIP |
| 16:31.10 | martyn-dev | If i have an asterisk runnning and the memory % on top is on aument (+), so, when all calls is hangup the memory still on x% wihout change to minimize :( is asterisk leave open sub process that load on cpu memory ? |
| 16:31.22 | martyn-dev | why is that? |
| 16:33.53 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 16:35.28 | *** join/#asterisk jiiv (n=jay@static-71-244-11-226.dllstx.fios.verizon.net) |
| 16:35.40 | *** part/#asterisk jiiv (n=jay@static-71-244-11-226.dllstx.fios.verizon.net) |
| 16:36.10 | boomboom99 | figured out my own problem... I have to create dummy extensions that dial the SIP lines without voicemail timeouts and have those extension numbers be the static agents. |
| 16:37.36 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 16:45.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:48.10 | manxpower | [TK]D-Fender: he could have just pressed the KM button and see what it does 8-| |
| 16:49.36 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 16:50.02 | *** join/#asterisk moy (n=moy@74.12.124.158) |
| 16:51.16 | [TK]D-Fender | manxpower: SHHHHHH!!!!!!!! |
| 16:52.57 | *** join/#asterisk loproc (i=d40a3735@gateway/web/ajax/mibbit.com/x-88da83814e282f24) |
| 16:53.25 | loproc | Hi there folks. Anyone who knows some good resources for free mp3 moh? |
| 16:55.39 | Qwell | loproc: the ones included in Asterisk are from freeplaymusic.com - you might try there. I'm not sure what the licensing is.. |
| 16:55.54 | loproc | Thank you - i'll try that! |
| 16:56.00 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 16:56.41 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 16:56.54 | *** join/#asterisk propellerhead (n=yogurt2u@host72.190-138-95.telecom.net.ar) |
| 16:58.13 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-3552028d45eea296) |
| 16:58.13 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:00.24 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 17:08.15 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 17:08.17 | eppigy | hello |
| 17:08.20 | eppigy | i am dave |
| 17:10.19 | [TK]D-Fender | Speak of the Devil... |
| 17:10.33 | [TK]D-Fender | eppigy: We've noticed how quiet it's been with you lately :) |
| 17:11.01 | eppigy | i know dude |
| 17:11.09 | eppigy | I always switch shells |
| 17:11.17 | eppigy | and forget to log on to freenode |
| 17:11.34 | eppigy | then Im like |
| 17:11.41 | eppigy | man |
| 17:12.05 | eppigy | I have not seen anyone asking about hooking a telrad up to and avaya predictive dialer then in to asterisk |
| 17:12.08 | eppigy | in a couple days |
| 17:12.16 | eppigy | *an |
| 17:14.05 | *** join/#asterisk oserr (n=oserr@c-a174e255.016-134-73746f32.cust.bredbandsbolaget.se) |
| 17:14.47 | manxpower | I can't imagine why. |
| 17:14.57 | oserr | hello, how can I from the dialplan check if an extension exists without using Dial()? |
| 17:14.58 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 17:15.23 | manxpower | oserr: what are you trying to accomplish |
| 17:15.40 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 17:15.53 | oserr | I have to determine which peer to dial |
| 17:16.42 | carrar | ChanIsAvail |
| 17:17.14 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 17:17.23 | KavanS | does anyone here use ekiga 3.x w/h264 w/asterisk? |
| 17:17.28 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:17.31 | oserr | carrar, does that work for extensions or just phones? |
| 17:18.09 | carrar | sip/iax |
| 17:18.56 | *** join/#asterisk eric2 (n=ejc@dsl-66-186-95-181.vianet.ca) |
| 17:19.20 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 17:19.41 | carrar | probably not what you want |
| 17:20.02 | oserr | carrar,My latter question wasn't very clear. What I meant to ask was does ChanIsAvail() test if a dialplan extension exists or just the availability of a registered phone? |
| 17:20.29 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:20.36 | carrar | technically it does test to see if a "Extension" exists |
| 17:20.40 | carrar | err does not |
| 17:20.51 | carrar | only a test a device |
| 17:21.26 | carrar | core show application ChanIsAvail |
| 17:22.17 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 17:22.43 | [TK]D-Fender | Yes, ChanIsAvail can test if an extension is valid |
| 17:23.39 | oserr | [TK]D-Fender, carrar: Thanks... I guess ChanIsAvail(Local/FOO) should work |
| 17:24.18 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 17:26.14 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:27.24 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 17:29.08 | Kobaz | when i use static-realtime for sip the 'status' column in 'sip show peers' is always unmonitored.... even if a peer is reachable.... when i use static-realtime for iax, the status column is updated.... any ideas? |
| 17:31.29 | Yourname` | So after installation of a 1.4, does it automatically install the init scripts too? |
| 17:31.40 | Yourname` | It always seemed to, just not today. |
| 17:31.46 | eppigy | make config |
| 17:31.48 | eppigy | does |
| 17:32.09 | Yourname` | Lemme give that a try |
| 17:35.00 | Yourname` | Thanks buddy, that worked |
| 17:37.37 | eppigy | true that |
| 17:44.29 | KavanS | how do I disconnect an SIP call in progress? |
| 17:44.38 | KavanS | i.e. I have something hitting a menu on loop, how can I kill said SIP connection |
| 17:45.06 | *** join/#asterisk linuxmaniac (n=maniac@19.Red-83-32-138.dynamicIP.rima-tde.net) |
| 17:46.22 | eric2 | you could implement a counter and then kill the connection with a GotoIf if the value hits true |
| 17:46.23 | jaytee | KavanS, soft hangup "channel" |
| 17:49.29 | *** join/#asterisk seanmh (n=johndoe@209-193-76-148.mammothnetworks.com) |
| 17:52.19 | Kobaz | anyone have any idea about the sip status not being updated in the peers list, when using asterisk realtime-static? |
| 17:59.19 | *** join/#asterisk clintc (n=clintc@n128-227-159-41.xlate.ufl.edu) |
| 17:59.39 | *** join/#asterisk clintc (n=clintc@n128-227-159-41.xlate.ufl.edu) |
| 18:04.11 | *** join/#asterisk DeFiAnCeNL (n=jefrecha@cp1337400-a.venlo1.lb.home.nl) |
| 18:04.59 | DeFiAnCeNL | Hi all, im running asteriskNOW for the first time, ive been able to call to the box (isdn2) so incoming works, im now trying to setup calling rules, however in the GUI, i cant save them? |
| 18:05.13 | DeFiAnCeNL | ive tried in FF3 , IE8, and Chrome, all a no go |
| 18:05.35 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 18:06.32 | Dovid | hello ev1 |
| 18:06.46 | DeFiAnCeNL | hello |
| 18:06.55 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 18:08.15 | DeFiAnCeNL | Anyone know wich browser will be able to save them ? strange thing is that the cancel button DOES work , the save button doesnt |
| 18:10.18 | DeFiAnCeNL | strange |
| 18:10.27 | DeFiAnCeNL | it does save it when i choose a voip |
| 18:12.16 | kaldemar | DeFiAnCeNL: have you tried in #asterisknow ? |
| 18:12.26 | DeFiAnCeNL | no sorry |
| 18:12.30 | DeFiAnCeNL | didnt know it existed |
| 18:14.26 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 18:17.34 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 18:19.14 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 18:20.16 | manxpower | DeFiAnCeNL: since none of us use AsteriskGUI, perhaps somewhere else would be a good place to get help. |
| 18:20.38 | DeFiAnCeNL | im already trying @ asterisknow ;) |
| 18:23.42 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 18:26.12 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 18:29.09 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 18:33.29 | *** join/#asterisk Micho123 (n=mcho123@94.187.72.191) |
| 18:35.29 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 18:36.00 | *** join/#asterisk NotForResale-US (n=nrds@24.102.131.63.res-cmts.sm.ptd.net) |
| 18:36.22 | NotForResale-US | im trying to use a SM56 Motorola Modem as a FXO Device how can i get configured? |
| 18:37.03 | manxpower | NotForResale-US: what in the world makes you think that will work? |
| 18:37.15 | docid | hrmm, so , anybody have experence useing asterisk as a switch/voicemail server? i want to pass most calls through that come in on one t1 out the second t1 port with the same caller id / and everything, but catch a certain range of numbers and pass them to voicemail on the asterisk box, any ideas on this one? google been leaveing me dry |
| 18:37.27 | NotForResale-US | the wiki |
| 18:37.31 | Micho123 | Dear All, |
| 18:37.31 | Micho123 | I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion here?Thanks |
| 18:37.33 | manxpower | the only driver for single port analog is wcfxo If that driver does not find the card then it won't work with Asterisk |
| 18:37.50 | NotForResale-US | how od i know if it found it |
| 18:38.03 | manxpower | dmesg is a good place to start |
| 18:38.18 | NotForResale-US | i see sm56 modem |
| 18:39.07 | manxpower | do you see wcfxo loading? |
| 18:39.42 | *** join/#asterisk jcollie (n=jcollie@fedora/jcollie) |
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| 18:50.06 | *** part/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 18:51.49 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 18:52.32 | ikevin | bonjour / hi |
| 18:55.59 | *** join/#asterisk stoffell (n=stoffell@d51A4D629.access.telenet.be) |
| 19:07.47 | manxpower | Micho123: As I understand it, what you are seeing is normal. |
| 19:09.38 | *** part/#asterisk syere (n=syere@204.10.20.30) |
| 19:12.38 | *** join/#asterisk mellow-yellow (n=mellow-y@mycomp.norris-stevens.com) |
| 19:16.31 | Micho123 | manxpower, I'm getting now [Feb 26 21:03:23] ERROR[2503]: chan_sip.c:12457 handle_response_invite: Got error on T.38 initial invite. Bailing out. |
| 19:19.22 | Kobaz | when i use static-realtime for sip the 'status' column in 'sip show peers' is always unmonitored.... even if a peer is reachable.... when i use static-realtime for iax, the status column is updated.... any ideas? |
| 19:21.36 | manxpower | Micho123: what version of Asterisk |
| 19:23.09 | Micho123 | manxpower, 1.4.22.1 |
| 19:24.01 | manxpower | Micho123: T.38 is only supported in passthru mode in 1.4 |
| 19:24.51 | jaytee | I need to change over one of my PRI spans from pri_net to pri_cpe and set it to take timing from the telco. Is a restart of zaptel sufficient after editing the zaptel.conf and zapata.conf or will I need to do a complete system restart? |
| 19:24.57 | manxpower | and even then I've not heard of many people getting it working |
| 19:24.59 | file | Micho123: http://bugs.digium.com/view.php?id=12437 |
| 19:25.20 | manxpower | jaytee: ztcfg Will drop all calls. |
| 19:25.52 | jaytee | manxpower, I know I'll be offline for a short span but that's already considered and acceptable. |
| 19:26.19 | Micho123 | manxpower, what is the solution?Upgrade to 1.6? |
| 19:28.26 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 19:29.22 | manxpower | jaytee: I doubt setting the span to pri_net will work connected to a telco PRI |
| 19:29.55 | manxpower | jaytee: you'll also have to module unload/module load for chan_zap.so to apply the PRI changes. ztcfg will apply the timing source changes |
| 19:30.57 | jaytee | manxpower, sorry if I didn't state that clearly enough. It is currently set to pri_net and connected to a t1 card on my Nortel Option 11C using a T-1 crossover cable. I will be swapping out the cable and connecting that span to the Adtran DSU that is currently connected directly to the Nortel on another T1 span connection. |
| 19:31.25 | manxpower | jaytee: Ah, OK |
| 19:33.18 | eppigy | ADTRAN |
| 19:33.21 | jaytee | manxpower, I've got the config part down pat (I had good teachers, hint! hint! ) but wasn't sure if I needed a full shutdown and restart of the hardware. Sometimes when I've taken the system offline in the past I've had to do that to get the TE212P card to come up right. |
| 19:33.37 | [TK]D-Fender | TRABAJO |
| 19:33.47 | eppigy | NO NECESITO |
| 19:33.55 | manxpower | jaytee: you should not need to do that in this case, but plan on it just in case 8-| |
| 19:34.17 | Micho123 | manxpower, it seems that the 1.6 version has the same problem |
| 19:34.18 | jaytee | manxpower, yeah it's an after hours job |
| 19:34.34 | manxpower | Micho123: see the bug report file gave you |
| 19:34.51 | jaytee | fortunately our telco runs their NOC 24/7 365 and they're good people to deal with if I need them to reset at their end. |
| 19:35.29 | manxpower | I threw a hissy fit at AT&T last night. |
| 19:36.17 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 19:36.25 | manxpower | My line doesn't have VOLTAGE on it and they missed their appt yesterday |
| 19:36.35 | Micho123 | manxpower, i saw it...but did not find the solution |
| 19:36.37 | eppigy | all telcos must dioe |
| 19:36.40 | eppigy | die |
| 19:36.47 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:37.26 | KavanS | can anyone point me to information in regards to a menu loop....I have a main menu that loops indefinitely, and I'd prefer to set a reasonable loop limitation (i.e. loop 10 times, then disconnect) |
| 19:37.26 | manxpower | Micho123: some problems don't have solutions |
| 19:37.33 | KavanS | can anyone suggest an article/howto for me to read? |
| 19:37.49 | manxpower | KavanS: standby |
| 19:38.00 | Micho123 | manxpower, so there is no way to make incoming call working while T.38 passthru is enabled? |
| 19:38.06 | Micho123 | manxpower, Please confirm |
| 19:39.44 | KavanS | manxpower: i googled this, but came across high-availability for asterisk instead |
| 19:40.27 | manxpower | Micho123: I cannot confirm, I can only read the same things you read |
| 19:41.12 | manxpower | KavanS: sorry, I can't find any examples |
| 19:42.01 | manxpower | do a search for IVR LOOP |
| 19:42.07 | KavanS | manxpower: I found a partial example that increments a menu loop counter, I'll look into how to disconnect after such loop |
| 19:42.20 | KavanS | I had someone hit our toll-free...about 30 mins of looping |
| 19:42.21 | manxpower | or better yet, google search: site:lists.digium.com IVR LOOP |
| 19:42.25 | KavanS | I feel like an idiot :P |
| 19:42.30 | KavanS | ok, will do |
| 19:43.28 | manxpower | I have posted examples before and LOOP or INDEX is what I used as a counter in the example |
| 19:44.10 | KavanS | ok, thank you for the direction....I found an article on mailing list which appears to be a good example, going to run some tests on this end |
| 19:44.17 | KavanS | definitely appreciated! |
| 19:45.53 | [TK]D-Fender | KavanS: "Hangup()" |
| 19:46.49 | manxpower | KavanS: set the counter, do the ivr, in the timeout extension check the count, goto hangup if it exceeds the max loops, if not the add 1 to the loop number and go back to the start of the ivr |
| 19:47.15 | eppigy | BOOYA |
| 19:48.58 | file | Micho123: As I mentioned in a note at the bottom of the issue I have a branch that I am trying to solve the issue in and am currently looking for testing |
| 19:49.25 | file | Micho123: the issue only comes up if you have an endpoint that provides T38 in the SDP in the initial invite, most solutions don't do that - they send a reinvite |
| 19:50.29 | Micho123 | file, if endpoint does not support T.38 then incoming call will work while T.38 pass through is enabled? |
| 19:50.37 | file | Micho123: yes. |
| 19:51.10 | Micho123 | file, But how if I need to send a fax using this endpoint? |
| 19:52.00 | file | see if you can disable T38 on the initial invite with your endpoint, see if it can be set to do a reinvite... it will work fine then |
| 19:52.00 | file | otherwise try my branch (it is against 1.4) |
| 19:52.09 | file | or change to a different endpoint that does use reinvites, like the Sipura |
| 19:56.01 | KavanS | doh! bad syntax on my loop |
| 19:57.09 | KavanS | ManxPower, [TK]D-Fender: could you look at my loop? http://www.pastebin.ca/1348076 btw: using asterisk 1.4 and it's complaining about my syntax |
| 19:57.40 | Micho123 | file, Your branche contains a new version of asterisk? |
| 19:57.58 | file | Micho123: it is Asterisk 1.4 with changes to try to fix the issue |
| 19:58.05 | ManxPowerAsteria | KavanS: 1.4 or 1.6? |
| 19:58.12 | KavanS | ManxPowerAsteria: 1.4 sir |
| 19:58.14 | Micho123 | file, did you test it? |
| 19:58.20 | ManxPowerAsteria | KavanS: paste the cli error |
| 19:59.00 | KavanS | ManxPowerAsteria: now updated: http://www.pastebin.ca/1348078 |
| 19:59.00 | ManxPowerAsteria | KavanS: note, in 1.6 | has been replaced by , in 1.2 and 1.4 either will work |
| 19:59.03 | file | Micho123: as best as I could, but I'm asking for people to test it |
| 19:59.45 | KavanS | ManxPowerAsteria: okay, we decided to use 1.4 because it was more stable at the time (or so the internet told us :P) |
| 19:59.59 | ManxPowerAsteria | KavanS: I would not use 1.6 in production yet on my servers. |
| 20:00.27 | ManxPowerAsteria | Unless they have upgraded recently not even Digium uses 1.6 on their production servers. File, has that changed? |
| 20:00.31 | KavanS | ok hehe yes...I try to keep things as stable as possible, I'm always "one step" behind the times |
| 20:01.02 | ManxPowerAsteria | KavanS: 1.6 is MUCH more stable than 1.4 was for the same spot in the release cycle --- still it's too new for me. |
| 20:02.06 | *** join/#asterisk mortum (n=fjbf@TELEYECLA-89-29-236-3.teleyecla.net) |
| 20:02.57 | russellb | digium is not yet using 1.6 on the main pbx |
| 20:02.59 | KavanS | yeah we are using it in our other office I hear....our european counterparts are more willing to try new releases |
| 20:03.06 | Micho123 | file, how I can download this versio?Give me thelink plz |
| 20:03.13 | russellb | but i know of a bunch of people using it |
| 20:03.38 | file | Micho123: it is in subversion, and the details are on the link I gave earlier |
| 20:04.05 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:04.10 | rwaite | damn you asterisk! |
| 20:04.24 | Wayhigh | another pwn'd user? |
| 20:04.48 | rwaite | so is 1.6 ready for prime time yet? |
| 20:04.57 | Corydon76-dig | Yep |
| 20:05.05 | KavanS | hrm, I can't find where my error is |
| 20:05.21 | KavanS | let's see if google can give me another example to work from |
| 20:05.47 | rwaite | i am switching to lenny in a month or two, considering giving 1.6 a go, hell it cant f up any more for me than 1.4 i'd guess |
| 20:06.37 | ManxPowerAsteria | KavanS: I'm a bit rusty, but it looks good to me. |
| 20:07.07 | KavanS | yes I see other examples with same syntax... ;\ |
| 20:11.25 | *** join/#asterisk korihor (n=korihor@200.44.218.45) |
| 20:18.28 | *** part/#asterisk korihor (n=korihor@200.44.218.45) |
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| 20:21.50 | watchy2 | is there a way to check if a call is a transfer or a normal call? |
| 20:22.05 | *** join/#asterisk cheGGo (n=cheGGo2@dslb-088-069-028-239.pools.arcor-ip.net) |
| 20:22.35 | ManxPowerAsteria | watchy2: you asked that yesterday. The answer has not changed. |
| 20:23.10 | watchy2 | i don't remember ever getting the answer to that except if it was read channelvariables? |
| 20:24.33 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 20:25.09 | docid | so need help setting up a context where any call that comes in on one t1 port automatically gets sent out of the other t1 dialing the number originally dialed, except for numbers that match a set defined for voicemail |
| 20:25.15 | ManxPowerAsteria | watchy2: That's still the answer. There are a couple of transfer related variables in that document. |
| 20:25.47 | watchy2 | would a transfer be any different wether you used the transfer button on the polycom or transfer using the asterisk featuremap? |
| 20:26.16 | ManxPowerAsteria | watchy2: I imagine it would be. |
| 20:26.34 | ManxPowerAsteria | But I suspect the difference is blind .vs. supervised transfer. |
| 20:26.37 | jjshoe | hrmm asterisk box with high load, only two channels up, strace shows http://pastebin.com/m3496779c any thoughts on which resource it might be? or a way to dig further? |
| 20:26.44 | ManxPowerAsteria | But if you don't want to read the file there's really not much more I can help with. |
| 20:26.55 | watchy2 | i'm reading it |
| 20:27.01 | Qwell | jjshoe: upgrade :D I think I saw something about that - hang on |
| 20:27.19 | watchy2 | i guess i need to have my customers start doing blind transferes. they all do supervised |
| 20:27.32 | jjshoe | Qwell probably :\ 1.2.30.2 |
| 20:27.58 | Qwell | oh...bleh |
| 20:28.37 | watchy2 | ${GOTO_ON_BLINDXFR} looks useful |
| 20:28.58 | ManxPowerAsteria | watchy2: maybe you need to stop thinking of "transfer" and start thinking "call gets sent BACK to the dialplan" |
| 20:29.11 | jjshoe | Qwell no? thinking of a bug for 1.6? |
| 20:29.30 | watchy2 | well what if someone is just dialing normally and not transfering? then it would call back their phone if no one answered |
| 20:29.47 | ManxPowerAsteria | you can set variables in the dialplan that you can read/check later |
| 20:30.41 | jjshoe | Qwell this anything worth worrying about? Blocking in: ast_waitfor_nandfds |
| 20:30.57 | watchy2 | i guess you could check callerid(num) and if the calling party is the same # and the # needing transfered back to, it would just end |
| 20:31.14 | watchy2 | that sound logical manx? |
| 20:31.18 | ManxPowerAsteria | watchy2: that is just one of several ways -- and it might be better than my idea |
| 20:31.57 | watchy2 | well i'm getting better and better with asterisk but i'm alweays afraid my ideas could be done better |
| 20:32.31 | watchy2 | and i hate writing a dialplan and then knowing it could be done in a better more logical way |
| 20:33.52 | watchy2 | SIPGetHeader(REFBY=Referred-By) also looks useful |
| 20:35.49 | *** join/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
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| 20:40.23 | Kobaz | watchy2: that's always the case, code can always be improved |
| 20:40.28 | watchy2 | well its deprecated but something exist to replace it |
| 20:40.58 | watchy2 | Kobaz: true. i just don't want to code something and it be functional but someone look at it and wonder if i'm retarded |
| 20:41.45 | Kobaz | it happens |
| 20:42.11 | Kobaz | i look at code, and i'm like man, this guy is retartded... and then i realize it's my code |
| 20:42.45 | NotForResale-US | how do i instal wxfxo |
| 20:42.48 | NotForResale-US | wcfxo |
| 20:43.00 | Kobaz | it's in the zaptel/dahdi drivers |
| 20:43.05 | watchy2 | heh. well i guess since i don't have anyone going behind me on any pbxs i don't have to worry about people looking at my code. |
| 20:46.13 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 20:46.25 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 20:46.34 | nullable_type | ~take-a-number |
| 20:47.35 | nullable_type | Hey guys I have sip.conf careinvite=yes. Then i dialed two numbers and connected. I was hoping voice data will not flow through the asterisk box. But it seems it flow through it. |
| 20:47.36 | *** join/#asterisk tompaw (n=tompaw@slave20.tesserakt.eu) |
| 20:47.54 | nullable_type | Can someone help me to achive this? |
| 20:50.45 | nullable_type | Can anyone help me? |
| 20:51.57 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 20:52.09 | outtolunc | something has to be keeping it going through the asterisk, like Tt or recording.. |
| 20:52.43 | nullable_type | i just had a fresh asterisk install |
| 20:52.56 | nullable_type | is there any config i have to turn off like audio logging |
| 20:53.41 | NotForResale-US | how do i install wcfxo |
| 20:59.04 | *** part/#asterisk DeFiAnCeNL (n=jefrecha@cp1337400-a.venlo1.lb.home.nl) |
| 21:00.53 | watchy2 | NotForResale-US: using dadhi |
| 21:01.27 | scooby2 | Anyone know what options are needed to go from gsm to wav with sox? quality loss is fine. Just need to show some people what the gsm files sound like. |
| 21:02.16 | Kobaz | NotForResale-US: i told you already.. install either zaptel or dahdi |
| 21:03.33 | ManxPowerAsteria | scooby2: get both versions of the standard asterisk sound files. They come in several formats |
| 21:06.34 | ManxPowerAsteria | I think watchy2 really means "how do I get someone to install wcfxo for me" |
| 21:07.27 | Kobaz | ManxPowerAsteria: not watchy |
| 21:07.44 | ManxPowerAsteria | Well, maybe I need more coffee. Sorry about that watchy2 |
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| 21:08.31 | *** join/#asterisk viq_ (n=viq@unaffiliated/viq) |
| 21:12.22 | watchy2 | manx: haha thats ok. i can setup an wcfxo myself :) |
| 21:12.35 | watchy2 | manx: but can you help me install the linux? |
| 21:15.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 21:17.40 | NotForResale-US | i have zaptel |
| 21:21.07 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:24.34 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
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| 21:30.29 | ManxPowerAsteria | NotForResale-US: any reason you can't follow the install instructions like the rest of us? |
| 21:33.21 | jaytee | other than 1-900-NXXXXXX what other NPA numbers are charge numbers for like phone sex lines or chat lines that I should block? are there any valid area codes that start with 9? |
| 21:33.40 | Qwell | jaytee: yes |
| 21:34.35 | rwaite | there is a whole group of area codes out of the country that will charge you, they look like normal numbers too |
| 21:34.38 | ManxPowerAsteria | NXX976-XXXX and all the Caribbean countries are expensive to call but are not "internationa" |
| 21:35.02 | Qwell | 976 isn't an NPA? |
| 21:35.27 | ManxPowerAsteria | nope, it's in many area codes, all "local" premium numbers |
| 21:35.32 | Qwell | huh |
| 21:36.06 | Qwell | jaytee: 909 is a valid areacode (though, why would you ever want to call there?) |
| 21:37.04 | jaytee | I had a dialplan example that had all the areacodes to block but now I can't find it. I looked on the WIKI but I don't think I got it from there in the first place. |
| 21:37.13 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 21:37.29 | eppigy | NOTHING SOUNDS LIKE AN 808 |
| 21:37.39 | ManxPowerAsteria | jaytee: your best bet is http://www.nanpa.net/ |
| 21:37.54 | jaytee | ManxPowerAsteria, thanks, I'll check that out. |
| 21:37.56 | nullable_type | Does anyone know how to bypass RTP Media? I have careinvite=yes and directrtp=yes but still RTP flowing through asterisk |
| 21:38.21 | ManxPowerAsteria | http://en.wikipedia.org/wiki/Premium-rate_telephone_number |
| 21:38.26 | ManxPowerAsteria | 976 is listed there as well |
| 21:38.45 | ManxPowerAsteria | nullable_type: what options are on your dial line? are the codecs the same for the two legs of the call? |
| 21:39.37 | nullable_type | I am jus testing with Originate call and then i have an extension.conf Context with a Dial command with no options |
| 21:40.29 | nullable_type | Both phone get calls, everything is great. except RTP still flows through asterisk |
| 21:41.16 | ManxPowerAsteria | nullable_type: so what is the ACTUAL dial line? |
| 21:42.08 | nullable_type | exten => s,n,DIAL(SIP/17789825555@gatewayName) |
| 21:42.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 21:42.45 | ManxPowerAsteria | nullable_type: where "gatewayname" is a [gatewayname] in sip.conf? |
| 21:43.01 | nullable_type | thats right |
| 21:43.43 | ManxPowerAsteria | when you have the two legs of the calls up do a "sip show channels" to confirm the two legs of the call have the same codec. Fortunatly you are not using w/W/r/T/t/whatever |
| 21:44.01 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:44.48 | nullable_type | Manx >> i am using same gateway for both channels so first go like phone1@gateway, second being phone2@gatewat |
| 21:45.06 | nullable_type | So that should be same codec since its same gateway rite |
| 21:45.25 | ManxPowerAsteria | nullable_type: assume nothing |
| 21:45.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:45.50 | ManxPowerAsteria | if everything was working the way you think it works you would not be here asking these questions. |
| 21:46.02 | nullable_type | ok i will test it quickly |
| 21:46.09 | ManxPowerAsteria | So let's not assume anything at this point. |
| 21:46.46 | nullable_type | agreed |
| 21:47.32 | nullable_type | both shows Format: ulaw |
| 21:48.01 | nullable_type | Peer = same ip, Format = same; Last Message: ACK |
| 21:49.39 | nullable_type | and i have careinvite = yes, nat = no, directrtpsetup = yes in sip.conf |
| 21:49.56 | keith4 | holy crap. i just realized that my CDR has been doing to cdr-csv (in addition to postgres) for *years* |
| 21:50.17 | ManxPowerAsteria | nullable_type: you don't have any options on the dial line that would cause Asterisk to stay in the RTP path, you have the same codec for both legs of the call, you have canreinvite=yes and directrtp=yes. Maybe the GATEWAY is rejecting the reinvite. If that's not the case then you'll have to turn on SIP debug and hope you can find a SIP guru to help |
| 21:51.34 | nullable_type | ok! that makes sense, I will see if i can find anything in the sip log. thank you very much for your help |
| 21:51.42 | docid | so need help setting up a context where any call that comes in on one t1 port automatically gets sent out of the other t1 dialing the number originally dialed transparently, im really trying to avoid learning how to use freeswitch right now since this is kinda time sensative, so can this be done with asterisk? |
| 21:51.54 | ManxPowerAsteria | nullable_type: I suspect your gateway isn't accepting or creating the reinvite |
| 21:52.16 | outtolunc | or the endpoints |
| 21:52.19 | ManxPowerAsteria | docid: 1) yes. 2) it's too complicated to explain easily. |
| 21:52.26 | nullable_type | ya may be the case, also i could try a different gateway. |
| 21:53.02 | docid | umm, possibly point me to a wiki or sit with some info relative to this? google been leaving me dry |
| 21:53.43 | ManxPowerAsteria | docid: well my employer can do it for a fee. Does a call that comes in channel 1 on T-1 #1 have to go out channel 1 on T-1 #2? |
| 21:54.14 | ManxPowerAsteria | docid: also the T-1s have to support DIDs |
| 21:54.33 | docid | no, shouldnt be nesscary, as long as it gets through to the box on the other side |
| 21:55.52 | docid | they do, |
| 21:56.31 | ManxPowerAsteria | exten => _patterntomatch,1,Dial(Zap/g2/${EXTEN}) where g2 is group=2 for the channels on T-1 #1 |
| 21:56.53 | ManxPowerAsteria | where pattern to match could be _XXXX if you get 4-digit DIDs |
| 21:58.02 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 21:59.01 | nullable_type | Max >> Do you think any sort of voice-logging into files causing the RTP to pass through asterisk during directrtpsetup? |
| 21:59.13 | ManxPowerAsteria | nullable_type: YES! |
| 21:59.14 | jplank | docid: what are you ultimately trying to accomplish? |
| 21:59.19 | ManxPowerAsteria | you mean like Monitor in Asterisk |
| 21:59.25 | nullable_type | yes |
| 21:59.32 | nullable_type | is it turned on by default? |
| 22:01.03 | jplank | if your just trying to patch two T1s together, you can do it with a xover T1 cable |
| 22:01.09 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 22:01.20 | Assid | anyonehere interfaced asterisk with jabber/gtalk? |
| 22:02.12 | docid | umm, need the system to pass calls through to the cell system while catching calls that are directed to voicemail, basically replaceing the current nortel meridian, ... the calls come in on the T1, want them to be passed out via another t1 port , to another box where if they are unanswered they will be passed back in for voicemail |
| 22:02.28 | jplank | oh |
| 22:03.00 | jplank | why not send the calls directly to the other box? |
| 22:03.33 | jplank | ohhh |
| 22:03.34 | docid | that i havent been able to get a solid answer on...they dont want to license the second t1 port on the box is as far as i can tell |
| 22:03.43 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 22:03.52 | docid | thanks manx, that should be a big help |
| 22:03.58 | Chuggs | ~book |
| 22:03.59 | jbot | rumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 22:04.17 | jplank | when the call leaves the first box into the second, and the second box doesn't answer and the first one times out, your going to push it to VM? |
| 22:04.40 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:04.41 | docid | and the vm is hosed by asterisk, but yes |
| 22:04.48 | jplank | I got ya |
| 22:05.27 | jplank | so box 1 is the VM server, box to houses the extensions, so when box 1 sends the calls to box 2, if box 2 doesn't answer, it falls back to BM |
| 22:05.28 | jplank | I get it |
| 22:05.40 | jplank | BM=VM |
| 22:06.20 | jplank | seems backwards though |
| 22:07.34 | docid | i thought so to..but i dont get to reconfigure it, they just want the asterisk box implimented like that, and it also has to handle the office phones etc, but that part easy |
| 22:14.53 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 22:16.17 | docid | thanks much manx, that seems to be on the right track |
| 22:18.08 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:18.13 | Assid | http://assid.pastebin.com/d6e2b88c4 |
| 22:19.43 | *** join/#asterisk ecl (n=felipe@unaffiliated/ecl) |
| 22:19.47 | *** part/#asterisk ecl (n=felipe@unaffiliated/ecl) |
| 22:24.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 22:25.23 | Assid | i cant get my asterisk to connect to ejabberd server |
| 22:25.28 | Assid | located elsewhere |
| 22:25.44 | *** join/#asterisk Shaun222 (n=shaun@ip68-5-154-128.oc.oc.cox.net) |
| 22:26.27 | Shaun222 | anyway to have a dummy voicemail? basically one that acted just like a real voicemail but didnt really record the message, or would discard the message after? |
| 22:27.07 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:29.02 | docid | hrmmm, i think i boo bood somewhere, can get the calls to pass through, but when they are hung up the trunks stay active....hrmzz |
| 22:30.28 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 22:33.57 | *** join/#asterisk Pazzo (n=ugelt@sadsl-246059.rol.raiffeisen.net) |
| 22:34.35 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 22:34.54 | ManxPowerAsteria | Shaun222: yes, just fake it using playback |
| 22:37.05 | Assid | anyone here connnected their asterisk box to gmail? |
| 22:38.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:40.29 | *** join/#asterisk BuSyAnToS (n=31749@81-208-83-253.fastres.net) |
| 22:41.50 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 22:46.48 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 22:52.56 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 22:53.06 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 22:53.20 | Micc | my asterisk 1.6.0.5 keeps crashing. I'm having problems with my res_odbc connections and cdr_tds |
| 22:53.49 | Micc | Its been fine for a week or two until the database started being intermittent. |
| 22:54.26 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 22:54.39 | timeshell_atwork | Micc : Get 1.6.0.6 |
| 22:54.43 | timeshell_atwork | 1.6.0.5 has issues |
| 22:55.23 | Assid | can someone helpme regarding asterisk+jabber module to speak to gtalk |
| 22:55.28 | Assid | it says the password is wrong |
| 22:55.50 | timeshell_atwork | Enter the right password. *There you go* |
| 22:55.57 | timeshell_atwork | :p |
| 22:56.29 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 22:56.50 | Micc | timeshell_atwork, does it address some crashing problems? |
| 22:58.11 | timeshell_atwork | Micc , can't answer you there. I know there are fixes in 1.6.0.6 that didn't make it to 1.6.0.5 |
| 22:58.19 | Micc | ok. |
| 22:58.34 | timeshell_atwork | Apparently 1.6.0.5 was a security fix to 1.6.0.3 |
| 23:00.37 | Micc | I should have not deployed 1.6.0.5 on my production server. |
| 23:00.40 | Micc | customers a pissed. |
| 23:01.08 | timeshell_atwork | Uh, should have tested first. |
| 23:01.19 | *** join/#asterisk double_cheesebur (n=chatzill@ip70-162-238-155.ph.ph.cox.net) |
| 23:05.49 | Micc | everything worked great for weeks. |
| 23:05.59 | timeshell_atwork | What version did you upgrade from |
| 23:06.09 | ManxPowerAsteria | and WHY did you upgrade?? |
| 23:06.33 | timeshell_atwork | heh, yah. I'm considering using the version you had before... so long as it's a 1.6 |
| 23:06.36 | Shaun222 | uhh, is timeout=15 in queues.conf in seconds or minutes. |
| 23:06.41 | Shaun222 | i think it must be minutes? |
| 23:06.59 | [TK]D-Fender | SECONDS |
| 23:07.15 | timeshell_atwork | lol |
| 23:07.17 | Shaun222 | what one defines how long to leave sombody in the queue |
| 23:07.29 | [TK]D-Fender | Shaun222: when you CALL voicemail. |
| 23:07.32 | Shaun222 | my test call has already been in the queue for 15 minutes |
| 23:07.38 | [TK]D-Fender | Shaun222: "core show application queue" |
| 23:07.57 | Shaun222 | oh you specify it in queueu()? not in the .conf? |
| 23:08.04 | [TK]D-Fender | Shaun222: Correct |
| 23:08.07 | Shaun222 | what happends if i didnt set anything... unlimited? |
| 23:08.17 | [TK]D-Fender | Shaun222: yup |
| 23:08.24 | Shaun222 | sweet...... |
| 23:11.34 | [TK]D-Fender | Shaun222: the "timeout" in the conf is to set how long to ring an agent for before moving on to the next in strategy |
| 23:12.28 | [TK]D-Fender | Off to martial arts, back later |
| 23:13.35 | Assid | hrmm.. i managed to get asterisk connect to my jabber server.. but cause timeout=100 it goes insane by logging in and out |
| 23:13.42 | Micc | ManxPowerAsteria, I upgraded from 1.4.22 I think. And I upgraded because I wanted multiple parking lots. |
| 23:14.25 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 23:16.29 | nullable_type | Hey guys have any of you had sip.conf careinvite=yes;directrtpsetup=yes and tested RTP debug succesfullly? |
| 23:19.23 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:21.47 | mapmagic | stintel: Have you ever configured asterisk ? I am trying to configure it so that I can access it from two soft phones. |
| 23:22.21 | mapmagic | sorry, to wrong user. |
| 23:22.36 | mapmagic | Anyone can help me configure an asterisk 1.6.1 with softphones? |
| 23:22.47 | timeshell_atwork | ? |
| 23:22.52 | timeshell_atwork | Is it any different? |
| 23:23.16 | mapmagic | timeshell_atwork: is it similar to setting up and running 1.4? |
| 23:23.56 | timeshell_atwork | mapmagic: WHen you said with softphones, I presumed you meant as opposed to hardware phones... |
| 23:24.20 | mapmagic | timeshell how do i setup asterisk and a softphone to connect to it? Any ideas? |
| 23:25.32 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:25.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:25.52 | timeshell_atwork | mapmagic, I think you need to do some reading on www.asterisk.org before you go any further. |
| 23:26.30 | timeshell_atwork | At any rate, I'm just packing up for the day, so maybe someone else may be inclined to assist at this point. |
| 23:27.36 | mapmagic | seems like 1.6 does not have ztcfg? |
| 23:28.00 | double_cheesebur | When I make changes to FreePBX I get the following error code: Could not reload FOP server. Anyone know how to debug this? |
| 23:28.27 | mapmagic | how do i get to asterisk CLI? |
| 23:28.32 | double_cheesebur | oops. wrong channel. sorry |
| 23:28.39 | mapmagic | got it |
| 23:28.42 | double_cheesebur | mapmagic: asterisk -r |
| 23:29.35 | mapmagic | got it, thanks. |
| 23:29.54 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 23:30.13 | mapmagic | double_cheesebur: how do i let a softphone contact my asterisk using a dialplan? |
| 23:30.16 | mapmagic | any ideas? |
| 23:30.38 | Chuggs | mapmagic: 1.6 doesnt work with zaptel you have to look at dahdi_linux , dahdi_tools |
| 23:30.40 | mapmagic | do i need to just worry about sip.conf? |
| 23:31.18 | mapmagic | Chuggs: I am just trying to setup one softphone to talk to my asterisk server. I have 1.6 running. Do I need to just concentrate on sip.conf? |
| 23:32.25 | Chuggs | mapmagic: that and a context in extensions.conf that is referenced in sip.conf |
| 23:33.09 | mapmagic | Chuggs: What is the default setting in 1.6? It looks like both sip.conf and extensions.conf are pretty complicated...for a newbie. |
| 23:35.34 | Chuggs | mapmagic: im not sure about a default setting. I guess it is whatever gets created with make samples in the asterisk src dir . I wrote mine from scratch |
| 23:35.59 | mapmagic | Chuggs: I'm just dreading to follow the 1.4 book to create 1.6 settings, will i get in trouble? |
| 23:37.27 | Chuggs | mapmagic: i never looked at 1.4 . i researched 1.6 only |
| 23:37.44 | mapmagic | do you have a book on 1.6 sip configuration? |
| 23:37.48 | mapmagic | or some other reference? |
| 23:39.48 | mapmagic | how do i get/set the Username/sip proxy/stun server/authentication name etc for my soft phone from the asterisk installation? |
| 23:40.07 | Chuggs | mapmagic: no . I just read everyting I could find on google. played with it till it started to work |
| 23:42.32 | docid | ok...so i have a setup that takes incoming calls that dont match any defined dids being sent out to a trunkgroup, the problem is that once these calls are connected and the user hangs up, th trunks stay open, how to i detect a hangup and close both trunks? i think too many hours of staring at exten => 's has frazzled my search skills, not finding anything usefull on google, |
| 23:42.44 | Micc | do I need an iax timing interface? |
| 23:43.14 | Chuggs | mapmagic: try this one. just ignore the OfficeServer stuff . http://tinyurl.com/dfbtow |
| 23:44.20 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 23:44.29 | mapmagic | Chuggs: Thanks. That helps a lot. |
| 23:45.42 | Micc | is the new ulaw algorithm that says its cleaner, does that mean it sounds clearer? |
| 23:46.32 | Chuggs | mapmagic: no problem |
| 23:46.42 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:46.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:46.55 | mapmagic | Chuggs: In that webpage, what is the [SIP_TRUNK] host parameter? Is that the static ip of the machine that asterisk is running on? |
| 23:47.00 | lmadsen | voxter: ping! |
| 23:47.07 | lmadsen | <---- in Vancouver |
| 23:47.45 | docid | fun... |
| 23:48.23 | *** join/#asterisk Corndawg_ (n=notreall@c-71-196-33-219.hsd1.fl.comcast.net) |
| 23:48.27 | *** part/#asterisk Corndawg_ (n=notreall@c-71-196-33-219.hsd1.fl.comcast.net) |
| 23:49.35 | docid | i miss vancouver, specially now that im up inside the artic circle |
| 23:50.18 | lmadsen | docid: ouch |
| 23:50.21 | lmadsen | lives in Toronto |
| 23:51.06 | docid | inuvik |
| 23:52.05 | lmadsen | what kind of internet do you get there.... |
| 23:52.10 | Qwell | lmadsen: he typed that a week ago, and it was delivered by dogs. |
| 23:52.22 | docid | 1 T1 serves the whole cable modem network, or ya can get dsl that sometimes bursts up to 100k, but usually sits around 20k for $100/mo+$10/gig |
| 23:52.40 | docid | its HARSH |
| 23:54.09 | Chuggs | mapmagic: i think the SIP_TRUNK is the Officeserver stuff. I would ignore it |
| 23:54.22 | mapmagic | How can i fix this?ERROR[26344]: tcptls.c:231 ast_tcptls_client_start: Unable to connect SIP socket to 129.186.121.178:5060: Connection refused |
| 23:54.32 | lmadsen | docid: eep! |
| 23:55.17 | mapmagic | I removed the office server stuff, and it went out |
| 23:55.32 | docid | ok...sooooo......... i have a setup that takes incoming calls that dont match any defined dids being sent out to a trunkgroup, the problem is that once these calls are connected and the user hangs up, th trunks stay open, how to i detect a hangup and close both trunks? i think too many hours of staring at exten => 's has frazzled my search skills, not finding anything usefull on google, |
| 23:55.53 | docid | useing sangoma a102 and a200 |
| 23:56.04 | docid | if thats relevant to nebody |
| 23:56.15 | nullable_type | Hey guys have any of you had sip.conf careinvite=yes;directrtpsetup=yes and tested RTP debug succesfullly? |
| 23:56.57 | Chuggs | docid: where did you source your hardware from ? Im in Alberta |
| 23:57.10 | lmadsen | docid: really you should be getting some sort of signal from the T1 that would cause the line to drop after. It should be hitting 'h' extension |
| 23:57.16 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 23:57.20 | mapmagic | Chuggs: My softphone says registration failed, service unavailable! |
| 23:57.53 | docid | gnet.ca it appears chuggs, i was jsut involved in finding what cards to use, not the ordering process, |
| 23:59.52 | Chuggs | mapmagic: your softphone either can't reach the server or the server isnt listening. use you asterisk IP as the softphones domain |
