| 00:28.05 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 00:39.37 | *** join/#asterisk Mango (~iMango@d154-20-89-230.bchsia.telus.net) |
| 00:39.55 | Mango | Does anyone have a CallWithUs account, and five minutes to help me test something? |
| 00:40.42 | carrar | I have a unlimited zoo pass |
| 00:40.48 | Mango | lol |
| 00:40.55 | Mango | can you register a softphone to it? :) |
| 00:41.01 | carrar | I think so |
| 00:41.10 | Mango | Ok, thanks. |
| 00:41.26 | carrar | They can play the monkey sound bite |
| 00:41.39 | carrar | or maybe that was actual real monkeys |
| 00:41.57 | Mango | :) |
| 00:46.35 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 00:47.56 | TJNII | ~pastebin |
| 00:47.56 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:49.59 | cmendes0101 | Mango: do you need Callwithus account specifically |
| 00:50.27 | Mango | Yes. I'm trying to confirm an exploit I discovered. |
| 00:50.33 | Mango | Their tech support doesn't believe me and won't test it. |
| 00:51.45 | cmendes0101 | oh nvm then lol |
| 00:51.51 | Mango | And they are being really fucking pretentious. |
| 00:53.14 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 00:53.50 | *** join/#asterisk mathslinux (~mathslinu@74.86.0.138) |
| 00:54.06 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 01:00.31 | *** join/#asterisk Whtsup (~sssi@WimaxUser376-208.wateen.net) |
| 01:00.34 | Whtsup | hello |
| 01:00.35 | Whtsup | how r u |
| 01:01.11 | Whtsup | it is better to connect to asterisk boxes with sip to sip or iax to iax |
| 01:01.26 | Whtsup | ? |
| 01:01.27 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 01:01.32 | Whtsup | any suggestion |
| 01:01.40 | ChannelZ | it depends |
| 01:02.14 | ChannelZ | if you are routing the media from one to the other, IAX could save you bandwidth and grief with its own trunking |
| 01:02.15 | Whtsup | well i m getting some jerk in voice quality |
| 01:02.36 | Whtsup | i m using g723 codec |
| 01:02.52 | Whtsup | and delay b/w to asterisk boxes is 200 ms |
| 01:03.09 | Whtsup | i have connected with iax |
| 01:03.17 | Whtsup | any suggestion to improve sound quality |
| 01:03.21 | ChannelZ | Well if your network connection is crap there's not a lot you can do about it either way |
| 01:03.34 | Whtsup | no my connection is good |
| 01:03.50 | Whtsup | delay is 200 ms constant |
| 01:04.03 | Whtsup | even when call is active |
| 01:04.51 | ChannelZ | if you're getting dropouts it suggests something is going on |
| 01:05.25 | Whtsup | so what should i do now |
| 01:06.53 | ChannelZ | You could increase the jitter buffer I guess |
| 01:07.16 | Whtsup | well jitter buffer is off |
| 01:07.21 | Whtsup | i m new to asterisk |
| 01:07.36 | Whtsup | how can i on this jitter buffer |
| 01:08.00 | ChannelZ | see iax.conf (you said you're using IAX) |
| 01:08.16 | Whtsup | yes |
| 01:08.19 | devmod | there was a list somewhere of recommended places to get dids from... ? |
| 01:08.21 | Whtsup | i m using iax.conf |
| 01:08.21 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 01:08.26 | ChannelZ | ~itsplist-us |
| 01:08.27 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 01:10.43 | devmod | thx |
| 01:10.54 | devmod | is that only for did in the US? |
| 01:10.57 | devmod | yeah |
| 01:10.59 | xheliox | Nah. |
| 01:11.08 | xheliox | VoIP MS, Junction, and some of the others have international DIDs. |
| 01:11.16 | devmod | I need france and germany.. |
| 01:11.30 | xheliox | I can't say which ones they have without looking. |
| 01:11.55 | devmod | tried didww and they asked me for a bunch of documentation to get a did in france (passport copy, utility bill, etc) |
| 01:12.06 | ChannelZ | Oh, the French. |
| 01:12.26 | devmod | Is that something expected when getting DID in france? |
| 01:13.30 | ChannelZ | Dunno.. it just sounds like a very french thing to do |
| 01:13.33 | WIMPy | In Germany you won't get geonumbers without a corresponding postal address. |
| 01:13.37 | devmod | lol it sure does |
| 01:17.24 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 01:22.55 | *** join/#asterisk coppice (~chatzilla@m121-203-233-5.smartone-vodafone.com) |
| 01:27.36 | hhoffman | jaytee: sorry, got pulled away... I see the extension being dialed on the console with sip debug on... but on the sip client I don't hear what asterisk is playing |
| 01:36.37 | Mango | How can we add http://voip.ms to itsplist-ca? |
| 01:40.11 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 01:40.49 | hhoffman | anyone know of a provider that also offers SMS? |
| 01:41.07 | Mango | CallWithUs has it in beta for the US. |
| 01:41.49 | hhoffman | oh, nice! thanks, I'll check them out :-) |
| 01:41.51 | Mango | actually, that's not right |
| 01:42.10 | Mango | Looks like it's more than the US. However, you can only send, and not receive. |
| 01:42.21 | hhoffman | ah, ok |
| 01:42.35 | hhoffman | I'd like to replicate what google voice is doing |
| 01:42.40 | Mango | Google Voice allows you to send and receive. But they don't support SIP of course. |
| 01:42.43 | Mango | ...ah, you know that :P |
| 01:44.23 | hhoffman | yeah, their service is great... but as a company they have too much info already |
| 01:44.45 | Mango | aye. |
| 01:44.59 | Mango | Wouldn't trust them as far as I could throw them. |
| 01:45.27 | hhoffman | hehe, yeah |
| 01:46.56 | coppice | Wouldn't trust them any more that I really need to, just like I trust anyone else |
| 01:47.22 | hhoffman | so, I've got a asterisk service with a public ip and sip clients behind nat. sometimes voice works for the client but other times not. Usually I'll hear asterisk on the first extension called but on subsequent I get no audio even through the console shows audio being played |
| 01:47.58 | Mango | does it work if you set a stun server on the sip clients? |
| 01:48.00 | Mango | stunserver.org |
| 01:48.33 | Mango | byuckekgfjg |
| 01:48.42 | hhoffman | hmm, I don't know. I'll have to read up on using STUN |
| 01:49.07 | Mango | What hardware are the SIP clients? |
| 01:49.44 | hhoffman | software... twinkle on linux (at the moment) |
| 01:49.52 | Mango | hrm, not sure how to configure that |
| 01:50.07 | Mango | another way to diagnose is to do |
| 01:50.09 | Mango | rtp set debug on |
| 01:50.27 | Mango | in the Asterisk console. If one of the IPs is the public IP of the client, then a STUN server won't help. |
| 01:50.41 | hhoffman | oh, ok... let me start there |
| 01:50.52 | WIMPy | ~sipnat |
| 01:50.53 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:51.10 | hhoffman | ah, yeah... it's sending it to the rfc1918 address space |
| 01:51.23 | *** join/#asterisk coppice (~chatzilla@m121-202-6-57.smartone-vodafone.com) |
| 01:51.25 | Mango | ok. you can try setting nat=yes for that peer. |
| 01:52.12 | hhoffman | yeah, I have that... one sec, I'll pastebin sip.conf |
| 01:53.02 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 01:53.03 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:53.31 | hhoffman | http://pastebin.com/5i5c5gaB |
| 01:54.23 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:54.52 | hhoffman | so, my setup is a little different then the aocomputing.net link as my asterisk box has a public ip addr... but the rest of the setup seems to match |
| 01:55.30 | Mango | squints at it |
| 01:55.33 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 01:56.04 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:58.13 | hhoffman | hmm, do I need the high numbered ports open in the firewall on the asterisk server? |
| 01:58.53 | Mango | if Asterisk is replying to a 192 address, that won't help |
| 01:59.05 | hhoffman | yeah, good point |
| 01:59.34 | Mango | is there any configuration in Twinkle to set an external IP? |
| 01:59.38 | hhoffman | so, do I just plug in a public stun server to the client and that's it? |
| 01:59.47 | *** join/#asterisk mrchrisadams (~Adium@CPE-58-168-30-199.lns5.cht.bigpond.net.au) |
| 01:59.58 | Mango | I'm afraid I don't know Twinkle, but that should be pretty much it. |
| 02:00.03 | Mango | Let's try it and see if it works. |
| 02:00.22 | hhoffman | ok, and I don't have to "sign up" or anything for the stun service to work? |
| 02:00.27 | hhoffman | knows nothing about stun |
| 02:00.29 | Mango | no, it's free to use |
| 02:01.36 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 02:03.01 | hhoffman | hmm, ok... so now it's sending to my public ip addr... I guess I've got to forward from the nat/router back to the sip client? |
| 02:03.51 | Mango | That would work, but I'm not sure why it's not doing that on its own. |
| 02:04.16 | hhoffman | unsure... just seeing packets via rtp debug |
| 02:04.24 | Mango | and still no audio? |
| 02:05.24 | hhoffman | sometimes :-( voicemailboxmain just worked and then hangup and redial that extension doesn't provide audio now |
| 02:06.37 | *** join/#asterisk sourcode (~code@ppp-58-8-95-72.revip2.asianet.co.th) |
| 02:07.03 | hhoffman | oh, well... I think I'll get some sleep... thanks for the pointers! |
| 02:08.11 | Mango | Good |
| 02:08.13 | Mango | ...luck |
| 02:12.32 | Mango | ~action Mango |
| 02:12.32 | infobot | bonks Mango over the head |
| 02:12.37 | Mango | This little guy is amusing! |
| 02:22.16 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:34.39 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.130) |
| 02:39.05 | Mango | Does anyone have a CallWithUs account, and five minutes to help me test something? |
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| 04:35.52 | EmleyMoor | Is there information anywhere on what RTP ports softphones use? |
| 04:40.35 | *** join/#asterisk Benwa (~Benwa@ip-62-235-215-176.dsl.scarlet.be) |
| 04:40.55 | xEBIx | EmleyMoor, thats not specified mostly you can set the port in the configuration of the software. |
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| 05:18.12 | Alton35 | leifmadsen, I remember you after all this time |
| 05:18.25 | Alton35 | several others, actually, I guess the regulars |
| 05:25.39 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
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| 06:42.23 | angryuser | Good morning |
| 06:42.55 | angryuser | What is the best faxdetect module for asterisk ? |
| 06:43.35 | Godfather_ | hi |
| 06:54.29 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 06:54.30 | *** join/#asterisk af_ (~getsmart@78.134.21.122) |
| 06:58.44 | ChannelZ | angryuser: fax detection is built in |
| 06:59.52 | *** join/#asterisk eye-scuzzy (~light@sun28.ipfw.su) |
| 07:00.40 | angryuser | ChannelZ, i know, the question is there are different modules for the same thing ? For example more performant |
| 07:01.40 | *** join/#asterisk Flametail (48174409@gateway/web/freenode/ip.72.23.68.9) |
| 07:01.45 | angryuser | ChannelZ, i need as fast as possible detection. |
| 07:01.46 | Flametail | hello? |
| 07:02.28 | Flametail | what softphones do you guys recommend? |
| 07:06.38 | ChannelZ | angryuser: It can only detect as fast as the calling fax machine sends the tone |
| 07:06.46 | ChannelZ | I've found it to be instantaneous once it hears it |
| 07:07.05 | ChannelZ | Flametail: I like Zoiper (Classic even) |
| 07:07.24 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:07.48 | Flametail | and im kinda new to sip......say the guy I want to call is extension 123.....how do I dial him? |
| 07:07.50 | angryuser | ChannelZ, can you show me your conf ? I remember that there were a delay setting |
| 07:09.13 | mort_gib | Flametail: exten => 123,1,Dial(SIP/123) |
| 07:11.17 | ChannelZ | in my case, in chan_dahdi.conf all I have is "faxdetect=incoming" and then in my extensions.conf I have an extension called "fax" in the 'incoming' context which sends the call to my fax machine |
| 07:11.48 | ChannelZ | there is no delay setting I"m aware of |
| 07:12.00 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 07:12.11 | Flametail | @ mort_gib :S ok.....say server is callserv.net.....a little better explanation? |
| 07:12.45 | mort_gib | Flametail: how about a better explanation of your problem |
| 07:13.14 | Flametail | I dont really know how to dial an SIP from a softphone.... |
| 07:14.00 | mort_gib | Flametail: well, your softphone needs to register with some kind of SIP server, like asterisk |
| 07:14.40 | Flametail | yeah..I am trying to go through with that with the zoiper account wizard |
| 07:15.00 | mort_gib | Flametail: If your Softphone is on the same SIP server, and in the same context, my reply above is correct, although I would add exten => 123.1.Dial(SIP/123,20,tT) |
| 07:15.15 | ChannelZ | not necessarily, Zoiper can 'direct dial' SIP URIs |
| 07:15.30 | ChannelZ | but presumably this whole exercise is to play with Asterisk, so.. |
| 07:15.34 | Flametail | error code 102 when trying to register just now |
| 07:15.55 | mort_gib | ChannelZ: and he stated that "and im kinda new to sip" |
| 07:16.17 | mort_gib | So go ahead and explain SIP URI's to him |
| 07:16.58 | ChannelZ | well you're telling him to Dial peers which probably don't exist either, my guess is basically nothing is configured. We have no idea what he's doing |
| 07:17.15 | mort_gib | Flametail: how about a better explanation of your problem |
| 07:17.28 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 07:17.36 | ChannelZ | ~book |
| 07:17.36 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 07:20.11 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 07:21.18 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 07:21.34 | Flametail | So you think theres a problem serverside since I couldnt connect? |
| 07:23.43 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:25.22 | ChannelZ | we have no idea, you've told us nothing about your configuration, what/who you're trying to connect to... |
| 07:25.56 | ChannelZ | You maybe typed in the wrong credentials, the wrong hostname, your firewall may be blocking traffic... |
| 07:27.03 | SiNGLer | is where a wait to play sound/announcement to callee, after Dial L() parameter's connect announcement to caller? |
| 07:27.32 | Flametail | I have install asterisk on a server....192.168.2.23......no firewalls...I added the extension and put in the right credentials....more or less out of the box config since I have little idea of what im doing, I just have to do it |
| 07:29.01 | ChannelZ | SiNGLer: ummmm... what? |
| 07:29.24 | ChannelZ | Flametail: is this FreePBX? |
| 07:29.38 | Flametail | yes |
| 07:29.50 | Flametail | I used the asteriskNOW iso image |
| 07:30.08 | ChannelZ | See #freepbx and/or #asterisknow |
| 07:30.37 | *** part/#asterisk Flametail (48174409@gateway/web/freenode/ip.72.23.68.9) |
| 07:30.53 | ChannelZ | I dunno if you have to add a sip peer in freepbx, or if adding an 'extension' does these things for you, or what... such is the way with a system which is designed to completely obscure normal Asterisk configuration |
| 07:31.38 | SiNGLer | ChannelZ: I play connect announcement to caller (I set LIMIT_CONNECT_FILE=audiofile and then use Dial(ABC/asd,,L(x)), caller hears audiofile, meanwhile callee hears ringing. after audiofile ends, caller and callee can talk, but before that I want to play beep to callee |
| 07:36.22 | ChannelZ | there is the A() option but I'm not sure what order things occur in |
| 07:36.35 | SiNGLer | A() is played before |
| 07:36.56 | SiNGLer | and M() is executed before and it blocks |
| 07:41.57 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 07:43.26 | ChannelZ | dunno sorry |
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| 09:02.26 | slin | re |
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| 09:59.40 | Gary_B | quick question, anyone know if an old avaya ip office 403 supports routing all mobile calls down 1 of its 2 PRI cards |
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| 11:11.22 | Han | Hi, the on hold music of our asterisk is gone. We didn't change anything. Does anyone have a bright idea? |
| 11:12.01 | Chainsaw | Han: Please don't lie to me. |
| 11:12.08 | Han | ok |
| 11:12.12 | Chainsaw | Han: If nothing was changed then the behaviour would still be the same. |
| 11:12.25 | Chainsaw | Han: You don't know what has been changed, but the hold music has ceased. |
| 11:12.27 | Han | I didn't say nothing changed. |
| 11:12.40 | Chainsaw | Han: So, could you try "core set verbose 10" and "core set debug 10" |
| 11:12.49 | Han | ok |
| 11:12.49 | Chainsaw | Han: Then put a call on hold whilst watching the console. Any errors/warnings? |
| 11:12.56 | Han | letseee |
| 11:13.13 | DND | guys is the pri line really have to do some sort of "restarting" once in a while? |
| 11:13.18 | Chainsaw | Han: An arbitrary group of people said "it wasn't me". That's neither helpful nor relevant. Let's concentrate on the problem at hand. |
| 11:13.24 | DND | we have a E1 line over dsl |
| 11:13.43 | Han | oh, it's already at 10/10 and it only says: started music on hold, class default, on channel xxxxxx |
| 11:14.51 | Chainsaw | Han: moh reload |
| 11:15.00 | Chainsaw | Han: Check for errors. Call again and put on hold afterwards. |
| 11:21.28 | *** join/#asterisk voxter (~voxter@76.77.73.130) |
| 11:22.27 | *** join/#asterisk ccesario (~ccesario@189-29-61-213-ac.cpe.vivax.com.br) |
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| 11:43.52 | *** join/#asterisk gadams999 (gadams999@173-165-184-25-atlanta.hfc.comcastbusiness.net) |
| 11:45.31 | tzafrir_laptop | Han, start with: moh show classes and: moh show files |
| 11:45.41 | tzafrir_laptop | That should give some useful information |
| 11:50.13 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-rpaeikyfoettqedw) |
| 11:52.08 | gadams999 | has anyone played around with dahdi hardware under ESX (PCI passthrough mode)? |
| 11:59.56 | *** join/#asterisk jetlag (jetlag@pool-173-61-209-63.cmdnnj.east.verizon.net) |
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| 12:07.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 12:16.43 | Godfather_ | how do i dial my mobile configured with chan_mobile? Now i'm able to dial through the mobile with exten => 30,1,Dial(Mobile/nokia/${NUM_TO_DIAL},45), but, how can i dial to the mobile with another extension? i mean, exten=> 31,1,Dial(Mobile/nokia) or something like that? |
| 12:21.42 | [TK]D-Fender | Godfather_: http://svnview.digium.com/svn/asterisk-addons/branches/1.6.2/doc/chan_mobile.txt?revision=828&view=markup |
| 12:23.54 | Godfather_ | [TK]D-Fender, Dial a headset using Dial(Mobile/device) in the dialplan. |
| 12:24.00 | Godfather_ | i tried that |
| 12:24.06 | *** join/#asterisk sourcode (~code@ppp-58-8-239-141.revip2.asianet.co.th) |
| 12:24.11 | Godfather_ | -- Executing [31@internal:1] Dial("SIP/104-00000031", "Mobile/nokia") in new stack |
| 12:24.11 | Godfather_ | [Jun 30 14:17:16] WARNING[12807]: chan_mobile.c:601 mbl_request: Cant determine destination number |
| 12:24.36 | Godfather_ | But seems chan_mobile expects a 3rd parameter |
| 12:25.22 | [TK]D-Fender | Godfather_: it does. |
| 12:25.28 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-dfvzdxxnctyizfxf) |
| 12:26.08 | Godfather_ | [TK]D-Fender, i just want to dial this device with another SIP extension, for example, with my softphone dial to the Mobile |
| 12:26.31 | [TK]D-Fender | Godfather_: And did you read that doc about 20 more times? |
| 12:26.42 | [Outcast] | normallly how long does it take for a license to be approved? |
| 12:29.16 | [TK]D-Fender | Godfather_: Phone = fxo, headset=fxs. The End. |
| 12:30.47 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:31.40 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
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| 12:45.01 | evilbit | OT: does anyone have a recommendation for a provider that does both SMS (receive/send) and IAX or SIP? |
| 12:45.30 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:45.58 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 12:58.19 | [TK]D-Fender | evilbit: * doesn't do SMS over SIP/IAX |
| 12:58.24 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 12:59.23 | evilbit | nod |
| 13:02.33 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:06.51 | Godfather_ | [TK]D-Fender, then, should i configure a headset for my mobile? the same MAC address? |
| 13:07.04 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 13:07.41 | [TK]D-Fender | Godfather_: No, it is SEPARATE. |
| 13:08.15 | [TK]D-Fender | Godfather_: * talks DIRECTLY to a headset, NOT through a phone <---n |
| 13:09.03 | Godfather_ | [TK]D-Fender, and then... |
| 13:09.28 | [TK]D-Fender | Godfather_: there is no "then" |
| 13:09.37 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:10.14 | Godfather_ | [TK]D-Fender, its posibly to dial to the phone using mobile_chan yes or no |
| 13:10.35 | [TK]D-Fender | [08:29]<[TK]D-Fender>Godfather_: Phone = fxo, headset=fxs. The End. |
| 13:10.37 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 13:10.53 | [TK]D-Fender | Godfather_: Phone = FXO ONLY |
| 13:11.24 | Godfather_ | ok |
| 13:16.03 | *** join/#asterisk Benwa (~Benwa@cable-78.29.254.179.coditel.net) |
| 13:18.19 | evilbit | is there a good place to read up on setting up T.38 Fax support over IP? |
| 13:22.06 | *** join/#asterisk eye-scuzzy (~light@sun28.ipfw.su) |
| 13:22.40 | xheliox | Opposed to t.38 over some other protocol? :P |
| 13:23.01 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 13:24.34 | evilbit | just trying to be as specific as possible ;-) |
| 13:25.56 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 13:26.03 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 13:26.38 | fullstop | Hi all. I'm having some troubles with DTMF and RFC2833. It works for some time, but then stops, leaving the caller stuck. |
| 13:27.49 | fullstop | I see the RFC2833 RTP DTMF in both wireshark and RTP debug in asterisk, but they are never forwarded to the sip channel. |
| 13:28.19 | fullstop | (forward might be a bad word -- rtp.c sees them, chan_sip.c does not after a certain point) |
| 13:28.50 | *** join/#asterisk youngproguru (~youngprog@74.10.229.58) |
| 13:29.30 | fullstop | I ran a monitor on the channel, and listened to the inbound audio, and it appears as if there is a very short "real" DTMF whenever a button is pressed, but I am not set to "auto" so I don't believe this should make a difference. |
| 13:30.07 | fullstop | Is it possible that one of the inband DTMF turds is tricking asterisk into going to inband mode? |
| 13:30.39 | fullstop | Is there any way to identify what DTMF mode is being used on a particular channel from the console? |
| 13:32.45 | fullstop | sip show settings shows global settings, but they could be different for a channel. |
| 13:33.59 | pabelanger | fullstop: If you are using rfc2833, you should not hear any tones. |
| 13:34.23 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:35.07 | fullstop | pabelanger: should is the key word! |
| 13:35.30 | pabelanger | fullstop: Asterisk is not smart enough to do dynamic DTMF. If you set rfc2833, it will not do any inband |
| 13:35.34 | [TK]D-Fender | fuuYou should be using rfc2833 not auto as your mode then |
| 13:35.49 | pabelanger | <PROTECTED> |
| 13:35.51 | fullstop | pabelanger: we are using a PaeTec SIP connection, maybe I need to talk to them. |
| 13:36.01 | fullstop | I am using rfc2833, not auto. |
| 13:36.23 | fullstop | http://pastebin.com/BvisA5TB |
| 13:37.04 | fullstop | rtp correctly identifies the rfc2833 packets, but never gives them to the channel. |
| 13:37.30 | fullstop | It doesn't happen all the time, but I have confirmed that it happens occasionally. |
| 13:38.34 | pabelanger | fullstop: I see the problem.... |
| 13:38.37 | pabelanger | fullstop: Toronto_Ontario |
| 13:38.42 | pabelanger | j/k |
| 13:39.18 | fullstop | pabelanger: Clearly I've run into the elusive metric RTP vs imperial RTP problem. |
| 13:39.20 | *** join/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 13:40.15 | pabelanger | fullstop: Are you able to reproduce the problem; even tho it is intermittent? |
| 13:40.28 | fullstop | pabelanger: yes, but sometimes it takes 30 minutes or more. |
| 13:40.48 | jtrimmer | is there any way to make a call out through asterisk and have it loop it back into a specific channel into the phone system. trying to test out some stuff like a customer calling in. |
| 13:42.33 | pabelanger | fullstop: You'll need to capture a full debug log, with RTP enabled showing the problem then. Then trace, where asterisk is loosing the dtmf |
| 13:42.47 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 13:43.02 | pabelanger | jtrimmer: Yes, Transfer() |
| 13:43.43 | pabelanger | jtrimmer: or Dial the specific channel |
| 13:43.59 | fullstop | pabelanger: I don't have the asterisk log where I lose the DTMF (I have it right after it lost it), but I do have a wireshark log from when it lost the DTMF. |
| 13:44.46 | pabelanger | fullstop: If you if see the problem in wireshark, then the problem is outside the control of Asterisk |
| 13:44.54 | structz | Hi i'm having a problem when seding a fax using a D-LINK ata i get this message on asterisk CLI and my asterisk justs stops working - channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/mediagateway-1755 (asterisk 1.6.2.9 on debian) |
| 13:45.14 | pabelanger | fullstop: Assuming the trace is inbound to Asterisk |
| 13:45.16 | fullstop | pabelanger: No. I see the DTMF in both wireshark and in asterisk |
| 13:45.34 | fullstop | Asterisk sees the DTMF and prints them out in the RTP log |
| 13:45.51 | fullstop | It just stops sending them to the channel at some point. |
| 13:45.52 | pabelanger | fullstop: but I do have a wireshark log from when it lost the DTMF. <-- typo? |
| 13:46.27 | fullstop | Sorry.. by "lost" I mean that asterisk no longer presents the DTMF to my dialplan. |
| 13:47.17 | fullstop | It most certainly receives the rfc2833 DTMF -- this has never stopped coming through. |
| 13:47.58 | pabelanger | fullstop: Well, you have the right idea. Capture both debug and wireshark log, we'll have to trace why Asterisk is dropping them. Is this a new problem? Have you changed Asterisk recently? |
| 13:48.36 | fullstop | pabelanger: This is a brand new setup, and we are testing it out. Interestingly enough, I can recreate it in both 1.4.32 and 1.6.2.9. |
| 13:49.02 | fullstop | I've not changed asterisk, except from 1.4 -> 1.6 to see if it was present in both. |
| 13:49.42 | pabelanger | fullstop: Can you test with another ITSP? Is this problem specific to them? |
| 13:49.49 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 13:49.58 | fullstop | pabelanger: Unfortunately, I do not have another ITSP at present. |
| 13:50.06 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:50.24 | fullstop | However, PaeTec is listed under supported providers. |
| 13:50.46 | pabelanger | fullstop: Time to debug, and if you believe an Asterisk bug open an issue on the tracker. |
| 13:51.36 | fullstop | pabelanger: to get started, do you have any idea how data from the RTP stream makes it to the channel? |
| 13:52.57 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 13:53.08 | pabelanger | fullstop: not will out looking |
| 13:53.15 | pabelanger | s/will/with/ |
| 13:53.32 | fullstop | :D |
| 13:53.47 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 13:53.52 | fullstop | s/D/( |
| 13:53.58 | fullstop | s/D/(/ |
| 13:54.03 | fullstop | hahaha |
| 13:54.09 | fullstop | i'll do better next time |
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| 14:02.49 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-137.dhcp.embarqhsd.net) |
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| 14:06.18 | *** join/#asterisk imcdona (~Administr@173.160.189.74) |
| 14:06.42 | Winkie | hey chaps, i am trying to use my phone to forward a call to my mobile. when the call comes in, the phone sends a 301 or 302 (i forget) with the new number, and asterisk proceeds to dial Local/mobilenum@correctcontext |
| 14:06.44 | *** join/#asterisk apal0s (~shiny@net.gennetsa.ondsl.gr) |
| 14:06.53 | *** part/#asterisk imcdona (~Administr@173.160.189.74) |
| 14:07.00 | Winkie | however, it then immediately gets a: Everyone is busy/congested at this time |
| 14:07.01 | *** join/#asterisk imcdona (~Administr@173.160.189.74) |
| 14:07.11 | Winkie | i can make the calls manually on my phone, one in and one out through the provider |
| 14:07.17 | apal0s | Hello, quick question, is there a way to issue a tranfser via REFER? |
| 14:07.35 | [TK]D-Fender | Winkie: pastebin a complete failed call with your dialplan/ |
| 14:07.37 | [TK]D-Fender | ~pb |
| 14:07.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:07.39 | [TK]D-Fender | ^^^^^^^^^^ |
| 14:07.54 | Winkie | willdo |
| 14:07.56 | apal0s | full question is how can i issue a call transfer, vi a 480 REFER on a remote sip server |
| 14:08.08 | apal0s | with asterisk 1.4.26 acting as "client" |
| 14:08.11 | [TK]D-Fender | apal0s: Transfer() |
| 14:08.20 | apal0s | [TK]D-Fender: it doesnt seem to work |
| 14:08.28 | apal0s | and i am guessing 1.4.26 isn't buggy |
| 14:08.35 | [TK]D-Fender | apal0s: If reinvites have not been disabled it should. |
| 14:08.41 | [TK]D-Fender | 1.4.26 is old |
| 14:08.41 | imcdona | when using agent penalties, does the "ringall" strategy ring in order of penalty? |
| 14:08.59 | apal0s | lot's of things are backpoted to him, so it should be fine |
| 14:09.02 | [TK]D-Fender | imcdona: It will ring GROUPS of memebers by penalty |
| 14:09.10 | apal0s | by reinvites you mean the canreinvite option ? |
| 14:09.24 | imcdona | thnks tk |
| 14:09.39 | [TK]D-Fender | apal0s: no, we're at 1.4.33.1 now. I'm not talking another branch. You are several releases behind within 1.4 itself |
| 14:10.25 | [TK]D-Fender | apyes |
| 14:10.25 | Winkie | [TK]D-Fender: looks like i don't have the log, just generating one now so gimmie a few mins, i'm probably using a v old asterisk too |
| 14:10.25 | leifmadsen | 1.4.34-rc1 is out too |
| 14:10.25 | Winkie | 1.6.2.5-0ubuntu1 |
| 14:10.25 | Winkie | oh i guess that's not too old |
| 14:10.26 | [TK]D-Fender | Winkie: Shouldn't matter |
| 14:10.49 | leifmadsen | 1.6.2.5 was released in February 2010 |
| 14:11.03 | apal0s | [TK]D-Fender: for X reasons oprting everything back on an embedded device, doesn't seem too appealing to me atm, but i don;t think i've seen any bug reports for the Transfer() for a while |
| 14:11.07 | apal0s | i might be wrong |
| 14:11.15 | apal0s | and 1.6 is way too weird atm |
| 14:11.18 | apal0s | at least for me |
| 14:11.29 | apal0s | oprting = porting :) |
| 14:11.36 | leifmadsen | 1.6.2.5 is actually based on 1.6.2.1 |
| 14:11.40 | leifmadsen | so yes... it is old |
| 14:11.51 | leifmadsen | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.5 |
| 14:11.52 | [TK]D-Fender | apal0s: Well perhaps at a minimum you should provide actual debug for us to look at |
| 14:12.17 | apal0s | ok, suppose we are using AEL |
| 14:12.21 | apal0s | the entry is like this |
| 14:12.39 | apal0s | 115 => { transfer(SIP/XXXXXXXX }; |
| 14:12.42 | apal0s | where XXXX is a number |
| 14:12.47 | [TK]D-Fender | apal0s: Show the CALL DEBUG. |
| 14:12.51 | apal0s | only thing i see is a re-invite going there |
| 14:12.58 | [TK]D-Fender | apal0s: Your dialplan will show istelf there |
| 14:13.13 | leifmadsen | heh: 1.6.2.5, revision 238499. Current 1.6.2 branch revision: 273145. You do the math :) |
| 14:13.16 | *** join/#asterisk neurosys (~neurosys@adsl-77-76-226.mia.bellsouth.net) |
| 14:14.52 | Winkie | [TK]D-Fender: http://pastebin.com/F8wWUZzK is the dialplan and log, i've replaced sensitive numbers with AAA, BBB and CCC |
| 14:15.04 | apal0s | [TK]D-Fender: meaning ? core set verbose = xxxx, sip set debug ? |
| 14:15.16 | apal0s | which party of the call actually interests you ? |
| 14:15.19 | Winkie | doing this manually from the phone works fine, the call completes outbound and can be transferred |
| 14:15.40 | [TK]D-Fender | apal0s: I see no Transfer there <------- |
| 14:15.57 | [TK]D-Fender | oops |
| 14:16.01 | Winkie | [TK]D-Fender: sorry for the terrible hacky bits in the dialplan, they're mostly unused and just from me messing around |
| 14:16.04 | [TK]D-Fender | uncrosses some wires |
| 14:16.05 | Winkie | i don't think they're affecting it |
| 14:17.13 | [TK]D-Fender | Winkie: Now do that again and stop masking numbers. this is precisely what is screwing up and you are killing the most important evidence |
| 14:17.35 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:17.55 | Winkie | [TK]D-Fender: i'll have to put it somewhere private then |
| 14:17.59 | Winkie | gimmie a minute |
| 14:18.31 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:19.16 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.7.73) |
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| 14:24.53 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:25.01 | *** part/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 14:27.52 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:27.56 | Yoda_1204 | Has anyone been able to get a Vodavi/IPECS IP 7024D h.323 hard phone working with asterisk 1.6? |
| 14:28.51 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:28.56 | *** join/#asterisk Winkie (~urmom@ur.fa.gs) |
| 14:29.04 | Winkie | [TK]D-Fender: well trying to paste that massive log into screen killed it :D |
| 14:29.07 | Winkie | i'll get it for you momentarily |
| 14:31.15 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.131.160.mtnl.net.in) |
| 14:33.29 | Winkie | [TK]D-Fender: thanks to debugging, i found the problem i think |
| 14:33.33 | Winkie | no chan_local or w/e for some reason |
| 14:33.36 | Winkie | my own stupid fault |
| 14:35.05 | Winkie | cheers for the help |
| 14:40.53 | *** join/#asterisk Yoda_1204_ (~chatzilla@12.111.169.98) |
| 14:51.11 | [TK]D-Fender | Yoda_1204: Noone uses those phones here |
| 14:51.41 | [TK]D-Fender | Yoda_1204: And as far as I'm aware FreePBX doesn't support H.3232 devices |
| 14:52.18 | *** join/#asterisk brycebaril (~bbaril@sea02-v612-nat.marchex.com) |
| 14:54.47 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:54.49 | wcselby | o/ |
| 14:57.01 | fullstop | pabelanger: Are you there? |
| 14:58.53 | Yoda_1204 | Ok thanks for checking, just trying to use the 80 phones we have already. bummer. |
| 15:01.26 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:02.02 | Yoda_1204 | [TK]D-Fender Thanks again for you help |
| 15:05.02 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 15:06.18 | Mango | Looks a bit like a Grandstream |
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| 15:44.11 | evilbit | for manager.conf and http.conf is there already a manager app for asterisk 1.6 or do I need to enable in both those files and then get a manager app in addition? |
| 15:46.34 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:47.26 | roni | i know that this is the wrong to do that but is working .. somebody can help me to resume this ... |
| 15:47.45 | roni | exten => _2XXXXXX,1,Dial(SIP/022${EXTEN:1}@IPCOMUNICACIONES) |
| 15:47.45 | roni | exten => _3XXXXXX,1,Dial(SIP/023${EXTEN:1}@IPCOMUNICACIONES) |
| 15:47.46 | roni | exten => _4XXXXXX,1,Dial(SIP/024${EXTEN:1}@IPCOMUNICACIONES) |
| 15:47.46 | roni | exten => _5XXXXXX,1,Dial(SIP/025${EXTEN:1}@IPCOMUNICACIONES) |
| 15:52.18 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 15:52.41 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 15:52.50 | wcselby | roni - what is your question? |
| 15:52.51 | Lantizia | Hey is there a way I can ask asterisk which codecs it supports? |
| 15:53.14 | wcselby | Lantizia - sip show settings |
| 15:53.20 | Lantizia | cool thanks |
| 15:53.32 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:53.33 | Qwell | Lantizia: you can just say "Asterisk, which codecs do you support?" |
| 15:53.40 | Qwell | asking isn't the issue |
| 15:54.14 | p3nguin | What's wrong with using core show translations? |
| 15:54.17 | wcselby | evilbit - manager is basically an API, you can either write your own app for it, download one of many out there, or simply telnet to the box on the proper port and enter the commands in the order required |
| 15:54.27 | wcselby | p3nguin - that'll probably work just as well |
| 15:54.41 | wcselby | if not better |
| 15:54.49 | Lantizia | Qwell, wcselby understood what I was asking, it's lovely you have to pick at the question _after_ an answer has been given |
| 15:54.59 | p3nguin | If there's no time in the field, it's not supported at thist ime. |
| 15:55.16 | p3nguin | s/at thist ime/now/ |
| 15:55.43 | Qwell | Lantizia: If his question answered the problem, then you didn't ask the right question. :) |
| 15:56.14 | Lantizia | Qwell, his question answered a question? wow you're just as good as saying what you mean as I am |
| 15:56.43 | Qwell | It's still before noon. I'm allowed to use a wrong word here and there. |
| 15:56.44 | p3nguin | snickers |
| 15:57.30 | Qwell | The correct answer to the question asked, however, was given by p3nguin. |
| 15:57.51 | evilbit | wcselby: ok, thanks.... is Asterisk Flash Operator Panel still a good choice in 1.6.x? |
| 16:01.13 | wcselby | evilbit - i don't use it, so I can't give you a good answer |
| 16:01.27 | evilbit | nod, thanks anyways :-) |
| 16:03.32 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 16:03.43 | spiceycurry | ~ |
| 16:03.48 | spiceycurry | ~book |
| 16:03.48 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:04.02 | spiceycurry | infobot, who is spiceycurry |
| 16:04.03 | infobot | somebody said spiceycurry was a bot molester. He touches my no-no area! |
| 16:04.32 | fullstop | bad spicycurry |
| 16:04.37 | evilbit | the asteriskdocs.org links don't work :-( timeouts |
| 16:04.59 | spiceycurry | infobot, what is sex |
| 16:05.00 | infobot | [~sex] updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; emerge --oneshot condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; emerge -C condom; make clean; sleep; |
| 16:05.17 | spiceycurry | infobot, who is your daddy |
| 16:05.17 | infobot | I think you lost me on that one, spiceycurry |
| 16:05.32 | spiceycurry | infobot, your daddy is dahdi |
| 16:05.33 | infobot | ...but your daddy is already something else... |
| 16:05.40 | spiceycurry | infobot, what is daddy |
| 16:05.41 | infobot | from memory, daddy is copperd |
| 16:06.11 | spiceycurry | infobot what is t.38 |
| 16:06.12 | infobot | spiceycurry: I think you lost me on that one |
| 16:06.33 | spiceycurry | infobot, t.38 seems to be a question that gets you fisted here. |
| 16:06.39 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 16:06.58 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:07.13 | Qwell | leifmadsen: asteriskdocs site b0rked? |
| 16:07.23 | Qwell | (I dunno who actually manages that) |
| 16:08.24 | wcselby | does someone need a copy of the pdf? |
| 16:08.36 | Qwell | wcselby: evilbit I assume |
| 16:09.11 | evilbit | I got a copy of the pdf via google... but giving a heads up b/c lots of ppl here reference those links via the bot |
| 16:09.12 | wcselby | evilbit - if you need the pdf, PM your email and I'll send it over |
| 16:09.18 | wcselby | gotcah |
| 16:09.23 | wcselby | gotcha even |
| 16:09.26 | evilbit | :-) |
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| 16:13.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 16:49.04 | nix8n82 | Does anyone know or is Nir Simionovich here? |
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| 16:52.49 | CaT[t3] | has anyone compiled the wanpipe sangoma drivers? i'm having issues with unresolved symbols wrt it and dahdi and whilst I resolved this once before I cannot, for the life of me, remember how. anyone able to help? :/ |
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| 17:00.38 | leifmadsen | Qwell: huh... that is hosted by file I believe |
| 17:00.52 | leifmadsen | Qwell: oh I know the problem |
| 17:00.57 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:00.58 | leifmadsen | jsmith needs to update the DNS |
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| 17:13.07 | joe_k | anyone know of a softphone that will pop up a web page when an incoming call occurs |
| 17:13.16 | joe_k | with an url parameter of the caller id |
| 17:13.21 | joe_k | (to do a CRM query) |
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| 17:21.13 | leifmadsen | joe_k: that's the first useful idea for sending a phone a URL that I've ever heard of |
| 17:21.26 | leifmadsen | joe_k: which of course helps you none -- don't know of any phones that will do that |
| 17:22.14 | [TK]D-Fender | joe_k: Plenty of other little scripts that will let you push things like this |
| 17:22.26 | [TK]D-Fender | joe_k: Doesn't have to be part of the phone itself |
| 17:25.02 | wcselby | queuemetrics has a section that will do that, but it may be more than what you're looking for |
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| 17:29.26 | markfeatherston_ | On the StartMonitor event, how would I get the extensions in that channel? GetVar doesn't seem to give me any extensions in the channel it returns. |
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| 17:38.53 | ManxPower | Can anyone tell me that number is "1 dB" on zt_monitor/dahdi_monitor? |
| 17:39.35 | Kobaz | anyone ever configure lldp on a dell powerconnect switch |
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| 17:47.08 | evilbit | can anyone point me to docs for setting up faxing via t.38 in asterisk-1.6 |
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| 17:52.58 | wcselby | evilbit - using FFA? |
| 17:53.29 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:53.30 | wcselby | that would be here ---> https://www.digium.com/en/supportcenter/documentation/viewdocs/FAX |
| 17:54.10 | evilbit | that's the proprietary one? I see modules and entries in sip.conf is there a free one to mess around with? |
| 17:55.55 | *** join/#asterisk Z_God (~julius@wlan226192.mobiel.utwente.nl) |
| 17:57.17 | wcselby | you can lookup spandsp |
| 17:57.28 | *** join/#asterisk Tili (~faizan@115.167.124.72) |
| 17:58.14 | evilbit | ah, is that the other way? I'd read about it but thought it was for analog lines as opposed to IP |
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| 18:01.53 | citywok | I was told that the reason i'm having issues with one of my carriers is they are sending a mulit-part mime message, and * doesn't support that. I'm running 1.6.1.18, is that something that has been fixed in a future version? |
| 18:02.43 | *** join/#asterisk sarah_p (~sarah_p@108.1.169.2) |
| 18:04.15 | pabelanger | citywok: Depends, what messages are they sending to you? |
| 18:04.58 | WIMPy | evilbit: There is also the version with iaxmodem and hylafax. |
| 18:05.56 | citywok | pabelanger: just building the call, but when building the call through to microsoft office communicatinos server it fails, not when going through to an aastra handset |
| 18:06.21 | *** join/#asterisk Alii (~alii@nat/digium/x-yvkarlaempygkyub) |
| 18:06.35 | *** part/#asterisk Alii (~alii@nat/digium/x-yvkarlaempygkyub) |
| 18:06.36 | citywok | a guy on the OCS team looked at my packet dumps and said "Looking closer at the Invite, I believe I see the issue. The SIP Provider is sending a multipart MIME body and Asterisk does not support multipart SDP. There is a bug on multipart " |
| 18:07.22 | pabelanger | citywok: I have OCS 2007 Speech Server and asterisk 1.6.0 working with no problem. Recently upgraded to 1.6.2. |
| 18:07.43 | citywok | yea, mine works great with flowroute and bandwidth.com, but not with qwest |
| 18:08.04 | pabelanger | citywok: I would upgrade to latest 1.6.2, and retry. And if a problem open an issue on the tracker. We'll need SIP debugs tho |
| 18:08.13 | *** join/#asterisk githogori (~githogori@168.61.10.135) |
| 18:08.26 | *** join/#asterisk SomethingISODD (~Dan@d75-152-168-97.abhsia.telus.net) |
| 18:08.29 | SomethingISODD | Hello all |
| 18:08.44 | SomethingISODD | question can anyne tell me how to convert from GSM to a g729 audio file? |
| 18:09.06 | pabelanger | citywok: FYI: https://issues.asterisk.org/view.php?id=17179 |
| 18:09.08 | citywok | Ok. I'm planning on goign to 1.6.2 next week in production, so when i do my next test upgrade i'll test out this issue and see what happens. |
| 18:09.20 | pabelanger | SomethingISODD: sox |
| 18:09.39 | SomethingISODD | sox can convert to g729 ok thank you |
| 18:09.40 | tzafrir_laptop | pabelanger, can sox decode / decode g729? |
| 18:09.55 | pabelanger | actually not sure |
| 18:10.46 | citywok | pabelanger: oh okay, so this may have been fixed in 1.6.2.9 |
| 18:10.46 | pabelanger | figure it could |
| 18:11.07 | SomethingISODD | the reason i ask, i am having an issue with asterisk, it will not play numbers because its not in G729 format |
| 18:11.32 | SomethingISODD | and i cant find any references, to convert from the gsm to a g729 and it doesnt seem like sox will handle it |
| 18:12.40 | leifmadsen | SomethingISODD: use the 'convert' option on the Asterisk CLI when you have codec_g729a.so loaded (and working) |
| 18:13.24 | SomethingISODD | oh i didnt know asterisk had a convert option.. thank you |
| 18:15.29 | [TK]D-Fender | SomethingISODD: what "numbers"? |
| 18:16.50 | SomethingISODD | actually any number. |
| 18:17.06 | [TK]D-Fender | SomethingISODD: How? What recording? |
| 18:17.56 | SomethingISODD | ok this is from the last test |
| 18:17.58 | SomethingISODD | [2010-06-30 10:52:41] WARNING[30327]: file.c:664 ast_openstream_full: File digits/1 does not exist in any format |
| 18:17.58 | SomethingISODD | [2010-06-30 10:52:41] WARNING[30327]: file.c:991 ast_streamfile: Unable to open digits/1 (format 0x100 (g729)): No such file or directory |
| 18:19.01 | [TK]D-Fender | SomethingISODD: You shouldn't be converting those |
| 18:19.02 | xheliox | sighs |
| 18:19.16 | [TK]D-Fender | SomethingISODD: You should have installed the entire recording pack in G.729 |
| 18:19.32 | SomethingISODD | actually i just checked in G729 directly and its showing the digits in there. |
| 18:19.34 | [TK]D-Fender | SomethingISODD: this was an option when you installed * |
| 18:21.42 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 18:22.39 | SomethingISODD | [TK]D-Fender if the sound files are in the g729 directory any idea why they would not be working |
| 18:23.13 | [TK]D-Fender | SomethingISODD: If you upgraded from older versions your folders may not be in the right state |
| 18:23.29 | citywok | i just had a user in a meetme conference, all of a sudden there call was no longer going, the phone was offline. he dialed the conference again, and it put him back in the conference, as well as had the conference greeting overlayed on TOP of it. wtf? |
| 18:23.49 | citywok | their*. 5 seconds later it dropped him out again |
| 18:33.07 | citywok | any ideas what that is about? It's happened to me quite a few times actually. |
| 18:33.38 | citywok | and it only seems to happen when a user is in a meetme conference, so i'm thinking it's pretty coincidental / likely an issue with meetme. |
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| 18:43.54 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 18:45.46 | grinder13 | hello! after a reboot all of a sudden nothing works in both of my asterisk machines!!!! "sip/iax2/dialplan reload" gives me an error of no such command. "module show" says that no (0) modules are loaded!! what the heck? |
| 18:46.57 | [TK]D-Fender | grinder13: pERHAPS YOU SHOULD LOOK AT WHO * IS RUNNING AS AND HOW OWNS THE CONFIGS |
| 18:47.05 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:47.53 | rossand | Need some advice compiling 1.6.2.9 on F13. Details here: http://pastebin.com/zjzXrGde Thanks. |
| 18:47.57 | grinder13 | i think i 've found it and it has nothing to do with the ownership of the files. let me check it |
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| 18:48.32 | [TK]D-Fender | rossand: configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
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| 18:48.43 | [TK]D-Fender | rossand: go install the pre-reqs for *. These haven't changed in years |
| 18:49.16 | evilbit | I asked about this yday but still haven't figured it out. Is there a way to automate playing a IVR to a SIP client that has just registered as opposed to making them dial a extension to get to it? |
| 18:49.25 | markfeatherston_ | rossand: run "yum groupinstall "Development Libraries" "Development Tools" |
| 18:50.18 | [TK]D-Fender | evilbit: Set something like regexten" and trap the AMI event the dialplan addition "should" make (not certain). Or track teh status yourself and originate a call to it |
| 18:50.25 | *** join/#asterisk saftsack (~oliver@137.226.140.21) |
| 18:51.36 | rossand | [TK]D-Fender: markfeatherston_ Thanks. I'll install those groups and try again. Thought I had added the dependencies listed but must have missed something. Thanks again. |
| 18:51.45 | evilbit | oh, thx! |
| 18:52.29 | grinder13 | just as I suspected. i am a complete idiot! i moved modules.conf in a different folder in both machines... what kind of an a.....e am I? |
| 18:52.43 | grinder13 | facepalm |
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| 18:58.25 | markfeatherston_ | using AMI, how can I get a list of all local extensions in a channel? I'm trying to send a sip notify to my phones so we can have an indicator light to show when call on demand (MonitorStart/MonitorStop) is active. |
| 19:00.53 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 19:01.09 | evilbit | hmm, well I can see it add the extension with a priority of 1 but nothing happens |
| 19:05.10 | [TK]D-Fender | evilbit: Of course not. YOU haev to trap it and fire off the Originate |
| 19:06.53 | evilbit | ok, thanks. i'll read up on how to do that |
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| 19:33.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:33.30 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 19:33.35 | drfreeze | Hi |
| 19:33.52 | drfreeze | Yesterday we changed from an analog setup to a PRI |
| 19:34.06 | drfreeze | We are using the same dialplan. |
| 19:34.33 | drfreeze | When an inbound call comes in it rings phones A and B |
| 19:34.42 | drfreeze | A answers the call and all is good |
| 19:34.53 | drfreeze | When another inbound call comes in, B answers the call |
| 19:35.11 | drfreeze | When B answers, A loses inbound audio for 30 seconds, then it returns |
| 19:36.03 | drfreeze | Any ideas on where to start looking for problems? |
| 19:38.30 | [TK]D-Fender | * cli |
| 19:39.47 | drfreeze | sip debug maybe? |
| 19:40.53 | *** join/#asterisk REdOG (~REdOG@gentoo/user/redog) |
| 19:41.29 | REdOG | Has anyone here taken the asterisk advanced course from digium? |
| 19:41.35 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:42.24 | pabelanger | REdOG: Yes |
| 19:43.28 | REdOG | was it worth it? |
| 19:43.29 | [TK]D-Fender | drfreeze: and PRI debug |
| 19:44.17 | *** join/#asterisk Entulho (~foo@sol.sj.ifsc.edu.br) |
| 19:45.24 | REdOG | Im a linux admin/consultant not much of a phone guy. But our company is looking at getting a new system so...I natuarlly began reading up on asterisk |
| 19:45.26 | raden_work | is there a way to dial my phone from a web browser ? |
| 19:45.37 | drfreeze | I'm using the latest version of dahdi. First time for that. 2.3.0.1+2.3.0 |
| 19:45.46 | drfreeze | Wonder if new bugs have been introduced |
| 19:46.12 | pabelanger | REdOG: For me it was a way to visit Digium and write the dCAP. I already had a strong understanding of Asterisk and VoIP in general |
| 19:46.15 | [TK]D-Fender | raden_work: Several web-based softphones out there... |
| 19:47.03 | pabelanger | REdOG: If your boss is paying for the trip, then go for it :) |
| 19:47.13 | REdOG | lol, I'd have to convince him but yea |
| 19:47.26 | REdOG | so Im trying to get myself confident enough to take that route |
| 19:47.27 | [TK]D-Fender | REdOG: What kind of deployment are you looking at making? |
| 19:47.39 | REdOG | our company is moving to a new facility |
| 19:47.47 | REdOG | so an entirely new network basically |
| 19:48.04 | raden_work | [TK]D-Fender, Sorry, should have been more specific. I would like to dial via clicking a number in a web browser and having asterisk dial my number and pickup my phone at my desk a aastra 9133i |
| 19:48.11 | [TK]D-Fender | REdOG: I meant in terms of call handling requirements, etc/ |
| 19:48.30 | [TK]D-Fender | raden_work: Originate <_ AMI, CLI, etc |
| 19:48.56 | raden_work | Ill look into it, much appreciated . |
| 19:48.57 | REdOG | to begin aprox 20 phones which would expand to about 75 within a year or 2 |
| 19:49.16 | pabelanger | REdOG: Read up the book, download, compile asterisk and start playing with it. |
| 19:49.22 | [TK]D-Fender | REdOG: Again, not quantity of phones.. or line. What kind of CALL PROCESSING are you looking at? |
| 19:49.28 | REdOG | I've compiled asterisk plenty |
| 19:49.30 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 19:49.47 | [TK]D-Fender | REdOG: Who will administer your system? |
| 19:49.48 | REdOG | Got in on my desktop actually |
| 19:49.51 | REdOG | me |
| 19:50.13 | REdOG | unless I just get a phone company installed deal |
| 19:50.25 | [TK]D-Fender | REdOG: So far you probably don't need any special coruse to get you started. You just need.... to get started |
| 19:50.35 | REdOG | ya kind of |
| 19:50.41 | REdOG | in the planning faze |
| 19:50.48 | REdOG | about 4 months from construction |
| 19:50.54 | REdOG | 1 yr till move |
| 19:51.16 | [TK]D-Fender | REdOG: tons of time |
| 19:52.15 | REdOG | ya, so ive got asterisk 1.6 installed on my desktop now, and am working on messing with broadvoice to play for now |
| 19:53.16 | pabelanger | REdOG: Playing with it is not too hard. The difficult part is troubleshooting problems :) |
| 19:54.06 | REdOG | problems? what problems? |
| 19:54.08 | REdOG | lol |
| 19:54.56 | *** part/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 19:56.20 | REdOG | ok, so what would I need to play with our PRI after hours? |
| 19:58.33 | [TK]D-Fender | REdOG: a PRI card in your server |
| 19:59.09 | REdOG | a problem free one? |
| 19:59.13 | REdOG | lol |
| 20:00.07 | [TK]D-Fender | REdOG: An even better idea. |
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| 20:17.56 | evilbit | is there a sendDMTF or similar that can be used when answering a incoming call? google voice has a annoying prompt to answer a call |
| 20:18.25 | [TK]D-Fender | evilbit: Yes.... SendDTMF() <---- |
| 20:18.32 | evilbit | thx :-) |
| 20:18.43 | [TK]D-Fender | evilbit: We hid it... in the big print. |
| 20:19.03 | evilbit | hahah, I mistyped DMTF that's why I didn't see it ;-) |
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| 20:25.08 | [TK]D-Fender | checkout time, BBIAB |
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| 20:35.00 | boodu | slt |
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| 21:36.12 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
| 21:36.49 | JerJer | i am attempting to setup a free fax channel but there is no 'register' app like the README says there should be |
| 21:37.09 | JerJer | http://downloads.digium.com/pub/telephony/fax/ |
| 21:39.19 | [TK]D-Fender | JerJer: "register" is for FFA which is not free |
| 21:40.01 | JerJer | i have the one 'free' key that starts with FFA |
| 21:40.13 | JerJer | but there is no register app in that directory structure |
| 21:43.23 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:46.50 | JerJer | ugh - i found it |
| 21:47.36 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:48.25 | JerJer | wtf - Server response: ERR - Invalid prefix, should be 'FAX' |
| 21:48.37 | JerJer | i was given a FFA-blah key |
| 21:49.46 | JerJer | grr |
| 21:50.32 | JerJer | don't even know why i bother |
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| 21:54.56 | joe_k | [TK]D-Fender: sorry about the late reply. If the calls aren't coming from my asterisk service, it's hard to "catch" them to take action (otherwise I would just send an IM to the recipient with the CRM link) |
| 21:55.09 | joe_k | [TK]D-Fender: but if the softphone did it, it would be damn simple and hard to break |
| 21:58.17 | [TK]D-Fender | joe_Pardon? |
| 21:58.45 | [TK]D-Fender | joe_k: What do you mean not coming from *? |
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| 22:29.01 | Paulius | What's the best way to disable Dahdi? I'm not using it, never will, and it's bugging the hell out of me. |
| 22:29.59 | WIMPy | So why did you install it? |
| 22:30.18 | Paulius | Came with the distro. Can I uninstall it easily? |
| 22:31.21 | WIMPy | I don't know if YOU can remove it easily. But be aware, that meetme and page need dahdi. |
| 22:31.39 | Paulius | Not using those features either. |
| 22:32.45 | fenrus | try unloading the dahdi modules |
| 22:32.53 | fenrus | trial and error o/ |
| 22:33.14 | Paulius | nice |
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| 22:48.05 | encinoman | Does anyone know of a provider that can give me a toll-free line with no connection charge from payphones? |
| 22:48.47 | [TK]D-Fender | encinoman: The ultimate free-ride. NO |
| 22:49.03 | encinoman | Ringcentral does. |
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| 23:11.29 | *** join/#asterisk twanny796 (~twanny@78.133.65.141) |
| 23:12.18 | twanny796 | in the asterisk console I get 'Remote UNIX connection/disconnection' ?? |
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| 23:14.23 | WIMPy | Some[one|thing] is opening/closing a *console. |
| 23:17.34 | twanny796 | WIMPy: how can I see who/where/how/why? |
| 23:17.59 | *** part/#asterisk sekil (~Ognjen@78.24.111.218) |
| 23:18.06 | Alton35 | netstat maybe? |
| 23:18.18 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:18.19 | *** mode/#asterisk [+o file] by ChanServ |
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| 23:26.15 | paulc | fdsa |
| 23:26.20 | paulc | uh.. ignore me.. |
| 23:27.05 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:27.30 | drfreeze | Hey, what is it called when a phone gives the tone that another call is incoming? |
| 23:28.12 | drfreeze | I have this problem with polycom phones when calling multiple phones via sip |
| 23:28.29 | drfreeze | The problem is this. I have phones A and B |
| 23:28.35 | drfreeze | A takes an inbound call |
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| 23:29.13 | drfreeze | When another call comes in and asterisk rings A and B, the audio from call A is lost for 10-12 seconds |
| 23:29.14 | ariel_ | call waiting |
| 23:29.31 | drfreeze | The audio returns without picking up the call |
| 23:30.01 | drfreeze | This just cropped up when switching from a TDM analog setup to PRI. We are using the same dialplan |
| 23:30.28 | drfreeze | So, I don't think the problem is a network problem, but a feature in asterisk or on the polycom phones |
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| 23:43.04 | drfreeze | Ok, have more data |
| 23:43.08 | drfreeze | What is happening is when the second call comes in and rings the current in-use phone, the audio is suppressed for 20 seconds and then returns |
| 23:43.30 | drfreeze | its like call waiting gone awry |
| 23:44.58 | drfreeze | Anyone have any clues on fixing this? |
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| 23:46.11 | Gugge | dont call the phone if its in use |
| 23:46.40 | boodu | hello |
| 23:49.37 | drfreeze | I'm wondering if this has anything to do with single line keys |
| 23:49.43 | drfreeze | Still, it shouldn't kill the audio |
| 23:52.24 | drfreeze | Gugge: rrrriiight...... |
| 23:53.47 | p3nguin | While I wasn't able to figure it out, I would disable call waiting and make the queues not ring phones that are in use. |
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| 23:55.31 | drfreeze | p3nguin: know how to disable call waiting? |
| 23:56.50 | p3nguin | That's phone-specific. |
| 23:58.00 | p3nguin | And what I mean by that is: I do not know how to disable it on your phone. |
| 23:59.06 | p3nguin | Ask me about a Cisco 7912/7940/7960, and I can probably walk you through it. Anything else, you can probably find out as quickly as I can. |