| 00:00.32 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 00:08.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:11.48 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 00:17.02 | *** join/#asterisk boolean12 (n=random@c-98-199-172-11.hsd1.tx.comcast.net) |
| 00:19.37 | hardwire | anbody have a tried/true method of generating g729 prompts? |
| 00:19.41 | hardwire | and legal |
| 00:19.43 | *** join/#asterisk savage1 (n=savageon@static-66-212-194-134.cpe.metrocast.net) |
| 00:19.45 | savage1 | howdie folks |
| 00:20.54 | giovani | hardwire: sorry, generating? as in what exactly? |
| 00:20.56 | savage1 | I'm having an issue on a bunch of systems where the calls take 1-2 rings to pickup |
| 00:21.15 | savage1 | so if I'm demoing the system for a client and I call in on my cell or any line, it's 2 rings I hear and then their phones ring |
| 00:21.20 | savage1 | is there a tweak to make it pick up quicker? |
| 00:21.37 | hardwire | giovani: transcoding from wav to g729 |
| 00:21.46 | giovani | hardwire: use sox, it's very easy |
| 00:21.55 | giovani | as long as you have the codec license |
| 00:22.03 | hardwire | it supports g729? I just plain didn't know that. |
| 00:22.06 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 00:22.10 | Primer | Anyone know why in asterisk 1.4.22, calls from asterisk to a polycom phone registered via SIP would show calls as: sip:8885551212@1.2.3.4 where 8885551212 would be the actual callerid of a call that came in via POTS, and 1.2.3.4 is the IP of the asterisk in question? The phones and asterisk are most definitely on the same subnet |
| 00:22.13 | giovani | I thought it did? |
| 00:22.21 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:22.52 | giovani | hardwire: it';s possible I'm wrong, google turned up these two free options: http://www.germanixsoft.de/ and http://www.asteriskguru.com/tools/audio_conversion.php |
| 00:24.34 | *** join/#asterisk LiCI (n=lici@195.56.58.67) |
| 00:24.40 | LiCI | hello |
| 00:25.15 | jsmith-teaching | hardwire: If you've got a g729 license on your Asterisk box, you can convert prompts to g729 from the Asterisk CLI |
| 00:26.27 | [TK]D-Fender | hardwire: http://www.digium.com/en/products/ivr/audio-converter.php |
| 00:26.58 | jsmith-teaching | [TK]D-Fender: Ah, there's that too! |
| 00:29.16 | LiCI | I'm newbie to asterisk/voip, I would like to use asterisk in a cisco system for call recording all conversations (security issues). Could this work? Cisco 28xx <--> sip <--> asterisk <--> CM6 and asterisk is mixmon all conversations...? |
| 00:29.21 | [TK]D-Fender | <- chock full of useful knowledge and lots of fiber |
| 00:29.52 | [TK]D-Fender | LiCI: If the protocols are compatible, sure. |
| 00:30.10 | *** join/#asterisk CrazyTux (n=brandon@adsl-75-4-22-105.dsl.irvnca.sbcglobal.net) |
| 00:30.46 | LiCI | currently there wont be exts on asterisk, only just recording voice to wav/mp3 |
| 00:31.39 | LiCI | hopefully afterwards there will be also a sugarcrm integration with this asterisk, but first I must do voice recording somehow |
| 00:31.50 | LiCI | [TK]D-Fender thank you |
| 00:32.47 | LiCI | [TK]D-Fender how can I make sure? I think cisco cm and routers are all sip compatibles |
| 00:33.17 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 00:34.06 | [TK]D-Fender | LiCI: if they are then you should make sure the RTP is in G.711 and then you should be fine. |
| 00:34.50 | LiCI | ok, thx |
| 00:35.07 | LiCI | actually this would be the design truely: cisco voice gw (29xx) - sip trunk1 - asterisk - siptrunk2 - cm6 |
| 00:39.53 | [TK]D-Fender | LiCI: Guess you could say that. |
| 00:40.09 | [TK]D-Fender | LiCI: * will just take in any target # and dial it right back out the other end. |
| 00:41.24 | LiCI | [TK]D-Fender : tell you the truth if it can work, asterisk can be a live-saver |
| 00:41.36 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:43.21 | LiCI | an old siemens hicom putting out cdr to rs232, cisco cannot put anything to rs232, an old sound recording system capture this cdr data on rs232, from this data it can identify b channels |
| 00:43.32 | *** part/#asterisk `paul (n=admin@125.252.70.126) |
| 00:44.43 | [TK]D-Fender | LiCI: Unless your CM is super picky this should be no issue |
| 00:45.24 | LiCI | what does it mean super picky (terible english of mine) |
| 00:46.49 | [TK]D-Fender | LiCI: "real pain in the ass" |
| 00:46.55 | LiCI | :D |
| 00:47.48 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-382ea9dc8faa6a83) |
| 00:48.29 | LiCI | actually, in this installation project as i know right now an absolutly new solution is needed for sound recording, and there is no more budgets, i haven't used asterisk before, right now I love it :-) |
| 00:49.12 | *** join/#asterisk pids (n=pids@196.sub-70-212-159.myvzw.com) |
| 00:49.24 | LiCI | [TK]D-Fender : thank you |
| 00:49.31 | LiCI | goodbye |
| 00:51.17 | *** part/#asterisk LiCI (n=lici@195.56.58.67) |
| 01:00.43 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 01:01.41 | hardwire | when passing TDM channels (monitored) through asterisk back out the same TDM card only a different port and provisioning type on the PRI, is echo cancel needed? |
| 01:02.05 | hardwire | asterisk stays in the media path so that causes a good delay (according to CPU load) which may not filter out @ the PSTN providers edge |
| 01:02.09 | hardwire | I'm guessing thats the case |
| 01:02.25 | hardwire | so I'd need echocan? |
| 01:02.45 | hardwire | I hate thinking about need it |
| 01:02.52 | *** join/#asterisk eit (n=eit@64.122.178.15) |
| 01:03.11 | *** part/#asterisk CrazyTux (n=brandon@adsl-75-4-22-105.dsl.irvnca.sbcglobal.net) |
| 01:05.08 | eit | Greetings, has anyone seen this? we have an asterisk PBX that is working rather well but whenever a phone dials a few specific external numbers and it goes to VM on the external VM system our call disconnects and does not allow us to leave a message. It seems to happen to all phones on our system that try to dial the specific external numbers. |
| 01:06.00 | hardwire | eit: you should call the owner of those numbers and tell them their PBX is probably having issues. |
| 01:06.31 | hardwire | eit: this is a PSTN call right? |
| 01:07.02 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:07.25 | hardwire | eit: like.. if it's a mobile provider in your area that is having voicemail system issues, that could be one reason.. esp if the numbers aren't very similar |
| 01:07.37 | hardwire | are they mobile? the same company/person? |
| 01:08.14 | hardwire | is it reproducible when you call from some other system other than your PBX? |
| 01:08.24 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:10.57 | hardwire | eit: herro? |
| 01:11.43 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 01:17.03 | *** join/#asterisk johann8384 (n=jonathan@75-120-52-110.dyn.centurytel.net) |
| 01:29.42 | *** join/#asterisk CrazyTux (n=brandon@user-vcauloa.dsl.mindspring.com) |
| 01:32.28 | *** join/#asterisk ild2002 (n=ild2002@212.118.140.12) |
| 01:32.38 | ild2002 | hi all |
| 01:34.27 | ild2002 | r there is any one can help me how to use teleon driver to use voice modem with skype to can receive call by useing my ladline ?? |
| 01:34.53 | ild2002 | landline |
| 01:36.23 | ild2002 | hello |
| 01:37.26 | ild2002 | any help |
| 01:38.11 | [TK]D-Fender | ild2002: voice modems are not supported in *. |
| 01:40.06 | drmessano | ZOMG |
| 01:40.16 | drmessano | Voice modem with skype for landline call? |
| 01:40.20 | drmessano | Thats.. a soup sandwich |
| 01:40.34 | drmessano | Thats like a wirenut |
| 01:40.36 | [TK]D-Fender | ild2002: and... |
| 01:40.38 | [TK]D-Fender | ~skype |
| 01:40.39 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 01:40.40 | [TK]D-Fender | ^^^ |
| 01:40.43 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6515dc0880cb01ba) |
| 01:41.01 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 01:48.24 | giovani | can anyone sign up for that beta? |
| 01:48.42 | giovani | when I looked at the application, it seemed geared towards ITSPs (the skype asterisk beta) |
| 01:50.16 | `Sean | has anyone used spa 3000 before? |
| 01:51.55 | *** join/#asterisk Carlos_PHX (n=Carlos@166.sub-70-210-139.myvzw.com) |
| 01:52.08 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-89db74c81e4247f1) |
| 01:55.42 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 01:57.18 | Carlos_PHX | makes note to never buy steak at Super Wal-mart |
| 01:57.29 | *** join/#asterisk prodyan (n=ian@124.104.71.66) |
| 01:57.33 | prodyan | hello all |
| 01:58.15 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 01:58.45 | *** join/#asterisk Firass-z0r (n=asadf@juicebox.vikcomm.wwu.edu) |
| 01:58.47 | codefreeze-lap | giovani: better to have loved, and lost, than.... |
| 01:59.02 | jblack | Ohhhh, va software is in delisting territory |
| 01:59.08 | giovani | codefreeze-lap: what? |
| 01:59.36 | codefreeze-lap | giovani: than to never have loved at all. Wouldn't hurt to try for the beta, eh? |
| 02:00.14 | giovani | codefreeze-lap: they ask for all sorts of developer-specific info -- I'd rather know ahead of time if other people have applied, or have inside info |
| 02:01.31 | codefreeze-lap | giovani: unknown to me. I've referred other folks to the application. From the sounds of it, they'll close the selection here pretty soon. |
| 02:01.51 | giovani | codefreeze-lap: have you, yourself, applied? |
| 02:02.16 | codefreeze-lap | no. I'm tempted, tho, but I'm pretty busy as-is. |
| 02:04.06 | codefreeze-lap | And, maybe, I haven't asked, since I work at Digium, they'd let me try it out; but I don't know if I have the free time... |
| 02:06.11 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 02:06.28 | Akiyuki | Anyone use sipdiscount with asterisk? My SIP device can login , but my register line in sip.conf keeps timing out. |
| 02:07.01 | *** join/#asterisk andresmujica (n=andresmu@190.25.97.120) |
| 02:07.12 | giovani | Akiyuki: contact them? |
| 02:07.21 | Akiyuki | It works fine from a softphone though. |
| 02:07.41 | giovani | well the softphone may not report every time the register expires |
| 02:07.51 | giovani | asterisk is verbose, devices often aren't |
| 02:08.04 | Akiyuki | oh |
| 02:08.21 | giovani | how often is there a timeout message? |
| 02:08.28 | Akiyuki | Because I can make calls from the softphone all day though without any disconnects. Is there a way to diagnose if asterisk is not connecting? |
| 02:08.35 | Akiyuki | Every few seconds it retries to register and never does. |
| 02:08.47 | giovani | so it's not registered at all? |
| 02:08.52 | Akiyuki | no |
| 02:08.56 | Akiyuki | About every 20 seconds. |
| 02:09.01 | giovani | well, then there's a misconfiguration somewhere |
| 02:09.04 | giovani | contact them about this |
| 02:09.29 | Akiyuki | It just says Registration for ..... timed out, trying again #18 etc |
| 02:09.50 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:10.36 | giovani | Akiyuki: yeah, that's gonna be a misconfiguration of some kind or a network issue -- assuming your hardphone works just fine |
| 02:10.40 | giovani | hence ... talk to them |
| 02:12.04 | Akiyuki | Its a free account, they wont support it until I purchase it. I thought maybe my register line was wrong or something |
| 02:12.16 | Akiyuki | Do you need the sip: portion? or just register=>foo:pass@foo.com ? |
| 02:13.00 | *** join/#asterisk mosty (n=mosty@130.41.233.220.exetel.com.au) |
| 02:14.14 | Akiyuki | Retransmitting #5 (no NAT) to 194.221.62.198:5060: |
| 02:14.33 | giovani | what sip: portion? |
| 02:15.09 | Akiyuki | like when you make a call sip:// does that need to be in the register line? |
| 02:15.15 | giovani | no |
| 02:15.20 | giovani | if you read the docs/sample |
| 02:15.25 | giovani | it'll show you |
| 02:15.32 | giovani | and yes, the way you wrote it is correct |
| 02:15.43 | Akiyuki | ok |
| 02:17.30 | mosty | the dahdi kernel module creates /dev/dahdictl but dahdi_cfg is looking for /dev/dahdi/ctl - is this a known bug? both packages are version 2.0.0 |
| 02:35.32 | `Sean | hey can someone help me configure a spa 3000 please? |
| 02:36.29 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 02:37.39 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 02:37.55 | Akiyuki | Found the problem. Was/Is a nat error, behind the firewall, the sipserver cant connect back and complete the register |
| 02:44.40 | giovani | heh |
| 02:48.23 | *** join/#asterisk itguru (n=p@5ac10652.bb.sky.com) |
| 02:59.28 | *** join/#asterisk johann8384 (n=johann83@75-120-52-110.dyn.centurytel.net) |
| 02:59.46 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:00.41 | jaytee | my brain's gonna just splode!!! |
| 03:03.31 | [TK]D-Fender | grabs a dustpan |
| 03:03.43 | [TK]D-Fender | offers up some more deadpan |
| 03:04.04 | [TK]D-Fender | jaytee: So the super-duper training getting to you? |
| 03:07.25 | *** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com) |
| 03:07.49 | mankash | I am not able to connect to asterisk console |
| 03:08.07 | jblack | make sure asterisk is running |
| 03:08.27 | mankash | asterisk is running my sip phone is registered with it |
| 03:08.39 | jblack | then you can connect to the console. :) |
| 03:10.30 | mankash | if I fo asterisk -c it just brings me back to linux console prompt |
| 03:11.54 | jaytee | [TK]D-Fender, the class if really good. It's filling in gaps in stuff I missed and the labs are good reinforcement. Jared's a great instructor |
| 03:12.38 | jaytee | had dinner with a couple guys from class tonight, one's from Montreal |
| 03:14.09 | jblack | asterisk -r, not asterisk -c |
| 03:16.57 | `Sean | hey [TK]D-Fender you ever use spa-3000? |
| 03:17.07 | `Sean | or anyone else for that matter? |
| 03:17.10 | drmessano | What do you want to know about it |
| 03:17.22 | `Sean | dial plan having problems crafting a simple dial plan horrible isn't it? |
| 03:17.36 | drmessano | (*x.|x.} |
| 03:17.38 | drmessano | err |
| 03:17.38 | jaytee | sits back and let's the SPA master do his magic |
| 03:17.40 | drmessano | (*x.|x.) |
| 03:17.49 | drmessano | lol |
| 03:17.55 | drmessano | Better than being the PAPmaster |
| 03:18.00 | jaytee | lol |
| 03:18.12 | drmessano | That sounds like a horrible machine women would hate me for.. |
| 03:18.15 | `Sean | heh lets see if it works tho |
| 03:18.21 | drmessano | Oh it will work |
| 03:18.54 | `Sean | nopes, got busy tone like literaly right away! |
| 03:19.03 | drmessano | What did you dial? |
| 03:19.29 | drmessano | (*x.|x.) is a valid SPA dialplan |
| 03:19.29 | `Sean | 6474301024 |
| 03:19.41 | drmessano | Sure you typed it correcty? |
| 03:19.44 | drmessano | correctly.. |
| 03:19.45 | seanbright | dude |
| 03:19.49 | seanbright | don't give out my number |
| 03:19.54 | drmessano | HAw |
| 03:20.03 | `Sean | lol |
| 03:20.22 | `Sean | umm anyway drmessano didn't work |
| 03:20.30 | `Sean | wanna ahve a a look at the interface? |
| 03:20.38 | drmessano | Sure |
| 03:20.42 | `Sean | pm |
| 03:21.00 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 03:21.18 | jaytee | wow, so far it's been at least a week without jeev. either he's been banned for being annoying or he's in Africa setting up Asterisk for the Congolese rebels. |
| 03:21.25 | `Sean | lol |
| 03:22.10 | seanbright | this him? *!*@87.236.144.38 |
| 03:23.12 | jaytee | seanbright, dunno, never really checked his ip and I'm not at home to check the logs on my home system. I'm in Huntsville taking a class at Digium |
| 03:23.19 | `Sean | :O |
| 03:23.30 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 03:24.00 | seanbright | he's got a cloak anyway |
| 03:24.54 | Carlos_PHX | What exactly is the chemistry that makes pretzels and beer so damn perfect together? |
| 03:24.57 | jaytee | well, an entire week of not having to hear about his bowel movements is a relief. |
| 03:25.15 | seanbright | i think it's been that long since i've had a bowel movement |
| 03:25.23 | jaytee | Carlos_PHX, the pretzels have two things going for them, carbohydrates and salt. |
| 03:25.51 | Carlos_PHX | Maybe it's mostly the salt, which begs for more beer, and... |
| 03:26.25 | jaytee | I just find people that volunteer that info without being solicited for it usually have deep seated issues and should be treated with suspicion and kept at a distance. |
| 03:27.37 | rob0 | Or, they're very old, but the suspicion and distance maintenance are warranted all the same. |
| 03:29.51 | mankash | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 03:30.08 | mankash | why I am getting this errro above |
| 03:30.29 | jaytee | mankash, does asterisk start if you do asterisk -c ? |
| 03:30.33 | drmessano | A week without Jeev is like.. a week without jeev |
| 03:30.48 | mankash | it is already started |
| 03:30.58 | [TK]D-Fender | seanbright: No, that was not jeev , just a troublemaker I dispatched earlier today |
| 03:31.00 | mankash | I started as asterisk start |
| 03:31.04 | jaytee | are you sure? do a ps aux | grep asterisk |
| 03:31.21 | [TK]D-Fender | mankash: What user are you running as? |
| 03:31.51 | mankash | root |
| 03:32.59 | mankash | but as root also I am gettign same error |
| 03:40.21 | mankash | anybody pls help me |
| 03:40.54 | jaytee | did you do a ps aux | grep asterisk |
| 03:41.04 | mankash | yes |
| 03:41.10 | jaytee | and? |
| 03:41.30 | mankash | it is showing many lines having asterisk in it bcoz I have a user called asterisk |
| 03:41.47 | mankash | but one line is important for us that is asterisk -start |
| 03:42.05 | jaytee | what distro of linux? |
| 03:42.12 | mankash | ubuntu |
| 03:42.56 | jaytee | is this a new install? |
| 03:43.03 | mankash | yes |
| 03:43.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:43.14 | mankash | sip phones are registed with the asterisk |
| 03:43.18 | jaytee | from packages? or compiled |
| 03:43.29 | mankash | compiled |
| 03:43.58 | mankash | I am sure it is somethng to do with permissions of that file |
| 03:44.53 | mankash | It was running I could see the astrisk ccli with all the actvities. It screwed up after restart |
| 03:45.20 | jaytee | ubuntu often deletes subdirectories under /var/run at boot |
| 03:45.51 | mankash | oh |
| 03:46.17 | jaytee | have you verified that /var/run/asterisk/asterisk.ctl exists? |
| 03:46.21 | mankash | yes |
| 03:47.05 | jaytee | if you've got asterisk setup to run as a service in one of your init scripts restart that. |
| 03:47.48 | mankash | let me check |
| 03:49.07 | jaytee | I much prefer typing service asterisk restart than /etc/init.d/asterisk start |
| 03:49.51 | jaytee | mankash, are you running 8.10? |
| 03:49.57 | mankash | 8.05 |
| 03:50.19 | jaytee | no such thing. either 8.04 Hardy or 8.10 Intrepid |
| 03:50.36 | jaytee | applied any updates lately? |
| 03:50.41 | mankash | yes |
| 03:50.48 | jaytee | kernel update? |
| 03:50.48 | mankash | no it is not running as service |
| 03:50.55 | mankash | nooe |
| 03:50.57 | mankash | nope |
| 03:51.05 | [TK]D-Fender | mankash: what do mean not running as a service? |
| 03:51.07 | jaytee | and if you type asterisk -c it won't load either? |
| 03:51.18 | [TK]D-Fender | mankash: you can't do "asterisk -r" unless its running as a SERVICE |
| 03:51.36 | mankash | there is nothing like asterisk in /etc/init.d |
| 03:51.54 | jaytee | that's why I never run * on ubuntu. it hoses things too easy |
| 03:52.26 | jblack | Hmm. I never have a problem with * on ubuntu. |
| 03:52.56 | tengulre | hi,all |
| 03:53.01 | jblack | mankash: Did you install the asterisk packages? |
| 03:53.17 | tengulre | it's time to have lanuch! Let's go! |
| 03:53.25 | mankash | nope I downloaded thgzip and then untar and compiled it |
| 03:53.50 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 03:54.08 | jblack | Well, that's probably why you don't have /etc/init.d/asterisk. See if the tarball install has an init.d example for you to use. |
| 03:54.36 | mankash | ok |
| 03:55.19 | jaytee | CentOS rules!!!! |
| 03:56.12 | mankash | you mean centos is better' |
| 03:56.13 | jblack | jaytee: I'm not in the mood to argue tonight. Can we argue tomorrow instead? |
| 03:56.42 | jaytee | jblack, just kidding. I just prefer RHEL based distros for *. I run buntu as a desktop though. |
| 03:57.14 | jblack | mankash: It's all up to preference. What you want done for you, what you don't. |
| 03:57.15 | jaytee | I'm distro agnostic |
| 03:57.24 | mankash | ok |
| 03:57.36 | jsmith | mankash: If you compiled it, do a "make config" to install the initscript |
| 03:57.42 | jaytee | jblack, we could argue next thursday. That's a good night for me :-) |
| 03:57.54 | jblack | Ok. It's a hate date, then. =) |
| 03:57.59 | jaytee | hahaha |
| 03:58.19 | jaytee | jblack, should we invite a gentoo fan for some additional balance? |
| 03:58.20 | jblack | It'll be my first date in about... 15 years. |
| 03:58.26 | mankash | cd /usr/src/ |
| 03:58.27 | mankash | ls |
| 03:58.32 | mankash | sudo ls -al |
| 03:58.38 | mankash | oh sorry |
| 03:58.46 | jblack | I'd rather skip the football helmet wearing types, if you don't mind. |
| 03:58.52 | jaytee | lol |
| 03:59.18 | jblack | how about a slackware type? |
| 03:59.25 | jaytee | "um, I recompiled my kernel for low-latency but now it keeps segfaulting. Can someone help?" |
| 04:00.33 | lmadsen | jsmith: hi! |
| 04:00.34 | mankash | ok I did jsmith |
| 04:01.26 | lmadsen | anyone know why a box would, on one call, (in iax2) respond to a NEW with an AUTHREQ, but then on the next call not respond at all? |
| 04:01.55 | jblack | You did a module load hate_even_calls ? |
| 04:02.02 | jaytee | lol |
| 04:02.29 | lmadsen | I wish :) |
| 04:02.56 | tzanger | lmadsen: is it receiving the packet? |
| 04:03.02 | tzanger | actually |
| 04:03.06 | tzanger | this is tickling a very very faint memory |
| 04:03.18 | tzanger | about delays with IAX2 |
| 04:03.19 | lmadsen | tzanger: yes... Asterisk_A is sending a call to Asterisk_B, and it sees the NEW, but does not respond |
| 04:03.27 | tzanger | uhm... what was it |
| 04:03.32 | lmadsen | it does on some calls... but not others |
| 04:04.06 | tzanger | authdebug? iaxcompat? |
| 04:04.14 | lmadsen | seems to be reasonably busy doing some registration traffic (asterisk registering to remote ITSPs) |
| 04:04.43 | lmadsen | hrmmm... wonder if I increased autokill |
| 04:05.04 | tzanger | it was delayreject |
| 04:05.08 | tzanger | but that isn't your problem I don't think |
| 04:05.20 | lmadsen | ya I don't think so... not using a switch |
| 04:05.24 | tzanger | iax2 debug shows it receives the new? |
| 04:05.28 | lmadsen | just a Dial(IAX2/remote/411) |
| 04:05.29 | tzanger | delayreject did nothing with switches |
| 04:05.33 | lmadsen | yes it does |
| 04:05.48 | lmadsen | ; For increased security against brute force password attacks |
| 04:05.48 | lmadsen | ; enable "delayreject" which will delay the sending of authentication |
| 04:05.48 | lmadsen | ; reject for REGREQ or AUTHREP if there is a password. |
| 04:05.51 | lmadsen | right... that was something else |
| 04:05.54 | tzanger | :-) |
| 04:06.08 | lmadsen | sorry... that was iaxcompat |
| 04:06.13 | tzanger | lmadsen: I'd start sprinkling the iax2 receive path with verbose messages |
| 04:06.15 | lmadsen | the thing about switches |
| 04:06.19 | tzanger | so you can see where it's blocking |
| 04:06.26 | lmadsen | eh? |
| 04:06.29 | lmadsen | you mean in the dialplan? |
| 04:06.44 | tzanger | no, I mean in chan_iax2.c |
| 04:06.54 | tzanger | c'mon, you gotta earn your C wings |
| 04:07.00 | lmadsen | oh... i can't do that on this box |
| 04:07.02 | lmadsen | production |
| 04:07.57 | tzanger | so what :-) |
| 04:08.08 | tzanger | you can't recreate it on a lab box? |
| 04:09.12 | lmadsen | haven't tried, but probably not |
| 04:09.20 | tzanger | then you have no choice |
| 04:09.31 | tzanger | module unload chan_iax2.so, make your change and load it |
| 04:09.33 | lmadsen | this is not my box |
| 04:09.40 | lmadsen | and this is running in production righ tnow |
| 04:09.52 | tzanger | then schedule some time to do testing |
| 04:09.59 | tzanger | if they aren't willing to do that, they aren't serious about getting the problem fixed |
| 04:10.25 | *** join/#asterisk Micc (n=dotirc@c-67-183-169-202.hsd1.wa.comcast.net) |
| 04:10.44 | Micc | Is there a max number of contexts you can have in asterisk? |
| 04:10.56 | Micc | In a dialplan. |
| 04:11.05 | lmadsen | tzanger: ok |
| 04:11.12 | tzanger | that's all I can think of |
| 04:11.17 | tzanger | *especially* if you can't recreate it |
| 04:12.53 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 04:13.00 | tzanger | that's also why i asked about iax2 debugging in the cli |
| 04:14.53 | *** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-d651f776c112d95b) |
| 04:16.26 | lmadsen | ya... well I see the NEW... it just doesn't respond all the time |
| 04:16.33 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 04:17.21 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 04:19.16 | lmadsen | just tries to convince the client to move to SIP |
| 04:19.19 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:19.21 | jameswf | Micc: how many do you need |
| 04:23.20 | Micc | jameswf a few thousand. |
| 04:24.10 | jameswf | why? |
| 04:24.15 | jaytee | insanity |
| 04:24.23 | jameswf | heard of a database? |
| 04:24.32 | lmadsen | if you need a few thousand contexts, I think you have a design issue... |
| 04:24.46 | Micc | Yeah I was going to use a database. |
| 04:24.55 | jameswf | lmadsen knows he wrote a book or something |
| 04:25.01 | lmadsen | I know nothing |
| 04:25.06 | jaytee | I'm so glad someone else who's an authority said that. makes me feel better for thinking it. |
| 04:25.06 | lmadsen | I just fake it real good |
| 04:25.07 | Micc | I want to support over a thousand customers on a cluster of asterisk boxes. |
| 04:25.18 | lmadsen | ok... |
| 04:25.20 | jameswf | thats what she said |
| 04:25.22 | jameswf | ohh |
| 04:25.30 | lmadsen | I support a hundred customers from a single context... |
| 04:25.50 | lmadsen | authentication and call control done via that context, pulling information from a DB |
| 04:25.56 | lmadsen | func_odbc ftw |
| 04:26.14 | lmadsen | wants a hamburger... |
| 04:26.34 | jaytee | I ate too much at the China Buffet across from the hotel |
| 04:26.39 | lmadsen | ouch |
| 04:26.43 | jameswf | pictures the pink panther scene whenever hamburger is mentioned |
| 04:26.46 | lmadsen | jaytee: how you enjoying the class? |
| 04:26.51 | jaytee | "you too fat!!! you go home now!!! you been here 4 hour!!! |
| 04:27.01 | jaytee | lmadsen, it's great |
| 04:27.07 | jameswf | you scare my wife |
| 04:27.09 | lmadsen | jaytee: glad to hear! jsmith is a great teacher |
| 04:27.14 | jaytee | I'm definitely going for the dCAP test |
| 04:27.24 | lmadsen | jaytee: good luck! be glad I'm not marking :) |
| 04:27.36 | jaytee | lmadsen, yeah he is. |
| 04:28.06 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 04:28.08 | jameswf | indians opening a new casino wooohoo |
| 04:28.49 | jaytee | hey, if I fail, I fail. If I pass, I pass. just means I'll have to take it again and eat the cost. |
| 04:29.12 | jameswf | jaytee: wanna know a secret |
| 04:29.18 | jaytee | sure |
| 04:29.20 | jameswf | the answers are all "C" |
| 04:29.34 | jameswf | :) |
| 04:29.54 | jaytee | whacks jameswf up side the head with a rotting trout |
| 04:31.21 | lmadsen | jaytee: well the thing is that, unless they have changed the policy, you get 2 tries (i.e. you don't pay again to take your re-make within a year) |
| 04:31.34 | lmadsen | and you only have to re-take the part you didn't fail |
| 04:31.54 | jaytee | lmadsen, didn't know that. cool! |
| 04:31.59 | lmadsen | you have to pass both the practical and written independently, but if you pass one of them, and not the other, you just re-take the other one |
| 04:32.08 | lmadsen | I'm sure you can verify with jsmith, but that is what the policy used to be |
| 04:32.23 | lmadsen | you can take it again at an astricon, or other training |
| 04:32.32 | lmadsen | just have to get yourself there :) |
| 04:32.33 | drmessano | jaytee is gonna be a made man |
| 04:32.37 | drmessano | You know what that means |
| 04:32.41 | jaytee | lol |
| 04:32.43 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:32.54 | drmessano | Ba-da-bing-ba-da-boom, you can't touch him |
| 04:32.57 | lmadsen | jaytee: in fact... why are you on here?! you need to be studying!! |
| 04:33.00 | jaytee | I drank the Kool Aid, it was orange |
| 04:33.39 | lmadsen | s/,/. |
| 04:34.02 | lmadsen | sheep enumeration... nice :) |
| 04:34.15 | jaytee | lmadsen, ya got a point, I should get busy reviewing. I did an hour after dinner but I should reread what we covered today. |
| 04:34.37 | jaytee | so good nite all! |
| 04:34.41 | lmadsen | don't forget to practice some practical stuff too |
| 04:34.43 | lmadsen | night! |
| 04:34.50 | jaytee | thanks! |
| 04:35.08 | lmadsen | NHL09 time! |
| 04:36.03 | jblack | kicks ld in the nuts |
| 04:37.10 | mankash | I have a simple setup with 2 sip phones internally connected internally to asterisk |
| 04:37.57 | drmessano | internally connected internally? |
| 04:38.00 | mankash | I am tryng simple dialplan where if you press 1 go to some extesnion and 2 to another extenion |
| 04:38.06 | drmessano | How do you use them if they're inside the PBX |
| 04:38.10 | mankash | means in one subnet |
| 04:39.37 | mankash | when I call from one sip phone it doesn't wait for user to press any key it just exits out after Awnser |
| 04:41.55 | mosty | mankash, pastebin your dialplan and a verbose log of a call being made |
| 04:41.59 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 04:44.33 | mankash | http://pastebin.com/m79580c4a |
| 04:45.20 | kb3ien | anyone used one hint to monitor multiple bindings? IE one or the other is inuse.... |
| 04:49.28 | mankash | mosty, have you checked the pastebin |
| 04:50.59 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 04:51.48 | mosty | mankash, where is the verbose log of an example call? |
| 04:52.12 | mankash | oh just wait |
| 04:52.41 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:52.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:54.43 | mankash | http://pastebin.com/m79ba12fd - verobose log |
| 05:03.54 | mankash | mosty, have you checked the pastebin |
| 05:05.40 | *** join/#asterisk ManxPower (n=manxpowe@151.sub-75-203-77.myvzw.com) |
| 05:05.51 | ManxPower | ~ecfo |
| 05:05.51 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 05:06.51 | tzanger | ManxPower: haha |
| 05:07.10 | wnspark | When I try to connect to my sip server it gives me the error code 410 (Gone) and says Permission Denied. Anyone have any ideas on how to fix it, or what could be causing it? |
| 05:07.11 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-15-241.tx.res.rr.com) |
| 05:07.39 | mankash | http://pastebin.com/m79580c4a please check this dialplan |
| 05:10.58 | wnspark | Does this look right for a basic sip.conf file to get calls from my number forwarded to my server? |
| 05:11.02 | wnspark | http://pastebin.com/m79bf2609 |
| 05:12.43 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d7fe44897b76dbdb) |
| 05:14.03 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 05:19.39 | mosty | mankash, what is the problem again? Answer doesn't wait for input from the user... |
| 05:21.49 | *** join/#asterisk emiller (n=ed@c-76-124-139-140.hsd1.pa.comcast.net) |
| 05:23.06 | emiller | having a brain fart on my dial plan and variables. could someone lend me a hand? My 'TRIES' variable is not assigning a value correctly. http://pastebin.com/d26f96180 |
| 05:23.44 | ManxPower | tzanger: someone had a similar symptom on asterisk-users |
| 05:25.39 | ManxPower | emiller: look at extensions.conf.sample and the channelvariables.txt variables are referenced as ${HAPPYVAR} not {HAPPYVAR} Also for math you need to use $[ ] around the expression, also an example in the aforementioned files. |
| 05:26.34 | jblack | I don't suppose there's anyone here that knows ld that wouldn't mind privately answering a question or two? |
| 05:26.35 | emiller | great! thanks for the input ManxPower. |
| 05:27.42 | ManxPower | jblack: "LD" is a big thing. |
| 05:28.22 | jblack | ld, as in the linker. And yeah, I suppose it's a bit like walking around in the sewars. |
| 05:28.35 | ManxPower | emiller: I may have some examples online, standby |
| 05:28.49 | emiller | ManxPower: that would be excellent. |
| 05:29.00 | ManxPower | jblack: This is a telecom channel, silly, LD means "long Distance" |
| 05:29.37 | ManxPower | I'll give links for all my examples, in case others are interested in them. |
| 05:29.43 | jblack | Your right. I should have been more specific out of the gate. |
| 05:31.33 | ManxPower | Generic Example: http://www.fnords.org/~eric/dialplan-example-v2.txt Fale Arrays: http://www.fnords.org/~eric/array-example.txt Fax-to-Email: http://www.fnords.org/~eric/fax2email.txt |
| 05:32.16 | wnspark | When I try to connect to my sip server it gives me the error code 410 (Gone) and says Permission Denied. Anyone have any ideas on how to fix it, or what could be causing it? |
| 05:32.27 | emiller | ManxPower: Thank you. |
| 05:32.48 | wnspark | i changed a setting and now it says error 409 Conflict |
| 05:33.57 | ManxPower | Call Routing: http://www.fnords.org/~eric/macro-std-exten-v2.inc Polycom Examples: http://www.fnords.org/~eric/polycom-config-examples/ Bunch of Telco Test Numbers: http://www.fnords.org/~eric/Testline.TestlineReport.xls |
| 05:34.05 | ManxPower | I think that is all of them |
| 05:34.51 | ManxPower | Cisco QoS config example: http://www.fnords.org/~eric/cisco-qos.txt |
| 05:35.46 | ManxPower | Yes, I've been using Asterisk for a long time. |
| 05:36.20 | emiller | ManxPower, do you use a polycom boot server |
| 05:36.32 | ManxPower | emiller: I use a Linux boot server. 8-) |
| 05:36.40 | emiller | touche. |
| 05:36.56 | emiller | one last question and im off to bed... |
| 05:36.57 | ManxPower | emiller: I used to manage almost 200 polycom phones, you want a boot server. 8-) |
| 05:37.02 | emiller | wow! |
| 05:37.12 | emiller | i thought 10 was a pain in the arse |
| 05:37.15 | emiller | haha |
| 05:37.25 | ManxPower | emiller: I think the biggest single location was about 60 phones. |
| 05:37.42 | emiller | all using 430s? |
| 05:37.50 | emiller | or a range of different models |
| 05:37.55 | [TK]D-Fender | emiller: I have only 2 at home... damn right you want a boot server... thats what gives you the power to configure them the way you want |
| 05:38.08 | ManxPower | emiller: 300, 500, 301, 501, 430, 600, 601 and like a 350 or something like that. |
| 05:38.12 | emiller | ahh. |
| 05:38.40 | emiller | 3 of our physical therapy centers use asterisk. |
| 05:38.45 | ManxPower | I was not involved in the day to day management after the newer phones came out. |
| 05:38.45 | emiller | all have boot servers |
| 05:39.28 | ManxPower | emiller: I set up Asterisk for a medical clinic about 5 years ago, 4 locations, 1 PRI, QoS'd Frac T-1s between offices. |
| 05:40.02 | emiller | ManxPower: is this what you do for a living? |
| 05:40.33 | ManxPower | emiller: Actually I mostly do networking. Asterisk is only a small part of what I do. I am accepting new clients. |
| 05:40.36 | ManxPower | ~manxpower |
| 05:40.37 | jbot | methinks manxpower is NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org http://www.fnords.org/skillslist.html |
| 05:41.25 | emiller | I've seen you and TKD-Fender in here a lot. thought you may have been a digium employee. |
| 05:41.40 | ManxPower | I'm OCD and have too much time on my hands 8-) |
| 05:41.52 | emiller | is a software engineer that got thrown in on an existing Asterisk project that was half way done :( |
| 05:41.55 | emiller | ha |
| 05:42.11 | ManxPower | I talked to Digium about becoming an employee but we could not agree on terms. |
| 05:42.27 | emiller | unfortunate. |
| 05:42.33 | emiller | they have a really nice facility. |
| 05:42.43 | emiller | i went to their bootcamp a few months ago |
| 05:42.47 | wnspark | When I try to connect to my sip server it gives me the error code 409 (Conflict) and says Permission Denied. Anyone have any ideas on how to fix it, or what could be causing it? |
| 05:43.34 | ManxPower | emiller: I'm talking to Digium regarding other projects -- who knows what might happen in the future. |
| 05:43.35 | emiller | ok ok, last one... |
| 05:43.42 | emiller | good luck with that! |
| 05:43.54 | emiller | one of my centers is saying that their caller id is not coming through |
| 05:44.05 | emiller | it came through before on their old phone system. |
| 05:44.35 | emiller | http://pastebin.com/d3be343e4 please excuse the gui generated extensions. |
| 05:44.42 | emiller | previous consultant did that |
| 05:45.25 | ManxPower | emiller: PRI? |
| 05:45.38 | emiller | ManxPower: im sorry, PRI? |
| 05:45.43 | emiller | unfamiliar with the acronym |
| 05:45.53 | ManxPower | The calls are coming in on a telco ISDN PRI line? |
| 05:46.07 | emiller | analog lines |
| 05:46.48 | [TK]D-Fender | emiller: pastebin your zapata.conf |
| 05:47.06 | ManxPower | emiller: Ah. |
| 05:47.09 | emiller | will chan_dahdi do? |
| 05:48.08 | emiller | http://pastebin.com/d1b7a808e |
| 05:48.44 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:49.48 | [TK]D-Fender | emithings you should have in there : usercallerid=yes , callerid=asreceived |
| 05:49.56 | [TK]D-Fender | usecallerid* |
| 05:53.07 | emiller | ill give it a whirl and check in the AM. Thanks for your help [TK]D-Fender |
| 05:55.02 | emiller | night all. |
| 05:55.04 | *** part/#asterisk emiller (n=ed@c-76-124-139-140.hsd1.pa.comcast.net) |
| 06:02.38 | phpboy | :D |
| 06:07.56 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 06:11.58 | [TK]D-Fender | Bed time... later all |
| 06:16.38 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 06:23.56 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:24.32 | *** join/#asterisk [_miT_] (n=mit@75.5.8.242) |
| 06:25.03 | [_miT_] | Can asterisk take a SIP call and direct it outbound on a analog pci modem? |
| 06:26.49 | denon | yes, but don't use a pci modem, get a real fxo device |
| 06:27.29 | [_miT_] | its for a small project using existing hardware |
| 06:27.38 | [_miT_] | just need to route one single channel |
| 06:27.41 | [_miT_] | nothing crazy |
| 06:30.32 | denon | get a real fxo device |
| 06:30.41 | denon | I'm not just being picky .. |
| 06:30.44 | denon | it's a necessity |
| 06:30.54 | denon | ~x100 |
| 06:30.58 | denon | ~x100p |
| 06:30.59 | jbot | i guess x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 06:49.40 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 06:52.30 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d31b346391666aef) |
| 06:58.26 | Katty | Qwell: :> |
| 06:58.35 | Katty | Qwell: 39% |
| 06:58.58 | ManxPower | I have a genuine digium X100P around here somewhere |
| 06:59.28 | ManxPower | coming to an ebay auction near you! |
| 06:59.42 | Qwell | 100% :p |
| 06:59.52 | Katty | :< |
| 07:00.41 | Katty | Qwell: how was the post-install patching? |
| 07:01.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:01.58 | Qwell | already had them downloaded |
| 07:02.10 | Katty | oh ah |
| 07:05.42 | Katty | twiddles thumbs. 70% |
| 07:05.50 | Katty | i could go get pizza, at this rate. |
| 07:06.10 | frozty_sa | now that ain't a bad idea |
| 07:06.27 | frozty_sa | it's a rainy day here...ordering pizza for breakfast at work...mmmm |
| 07:06.30 | Katty | at 1am? |
| 07:06.37 | Katty | i doubt the pizza place is open |
| 07:06.58 | frozty_sa | 09:06 < Katty> i doubt the pizza place is open |
| 07:07.02 | frozty_sa | ;-) |
| 07:08.31 | *** part/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 07:09.23 | Katty | ;) |
| 07:11.20 | Katty | Qwell: 90 :> |
| 07:11.22 | Katty | boingboing |
| 07:13.37 | frozty_sa | Katty: what are you downloading, new version of *? |
| 07:14.58 | Katty | frozty_sa: something far more entertaining. |
| 07:32.17 | Katty | Qwell: northrend here i come! |
| 07:36.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:44.55 | *** join/#asterisk Segnale007 (n=Pietro@host191-254-dynamic.18-79-r.retail.telecomitalia.it) |
| 07:48.56 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 07:51.11 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:51.25 | frozty_sa | Katty: ah, wotlk? |
| 07:53.00 | *** join/#asterisk the1_ (n=x@58.69.137.14) |
| 07:56.06 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 07:57.28 | *** join/#asterisk LiCi (n=lici@195.56.58.67) |
| 07:57.53 | LiCi | hello |
| 08:01.10 | SwK | frozty_sa, yes worlk |
| 08:01.15 | SwK | wotlk |
| 08:01.39 | tzafrir_laptop | Does chan_mobile support more than one phone per bluetooth adapter? |
| 08:06.20 | LiCi | I'd like to use asterisk for only for mix monitor in this system : cisco voice gw - sip trunk1 - asterisk - sip trunk2 - cisco cm6. How can I identify a recorded call in asterisk if there is a call forward/follow me/call transfer etc in cisco cm? |
| 08:10.35 | Katty | frozty_sa: indeed. |
| 08:12.18 | LiCi | I'm a newbie... is there some kind of call id exists which can be used in cm and asterisk as a key for the recorded calls? Actually as I see only the time is the only key |
| 08:26.02 | hi365 | does jabber/gtalk need to be specified in make menuconfig, or is it automatical included in 1.4? |
| 08:28.07 | *** join/#asterisk sheri_rao (n=rao@203.99.190.35) |
| 08:29.31 | sheri_rao | Quoting prices to my client, can anyone advise , how much to charge for SugarCRM with Asterisk integration on 6-trunks & 10-extensions ? |
| 08:34.22 | sheri_rao | Quoting prices to my client, can anyone advise , how much to charge for SugarCRM with Asterisk integration on 6-trunks & 10-extensions ? |
| 08:35.19 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 08:38.00 | *** join/#asterisk LiCi (n=lici@81.183.235.8) |
| 08:38.02 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 08:38.14 | dandre | Hello, |
| 08:38.26 | LiCi | hi |
| 08:38.50 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:38.52 | dandre | is there any way to get the callerid string for a channel from the manager interface? |
| 08:39.04 | synthetiq | sheri_roa : free |
| 08:41.25 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:42.06 | nicox | good morning |
| 08:42.28 | nicox | can someone give me a hint how to debug a problem? |
| 08:42.39 | nicox | <PROTECTED> |
| 08:49.36 | *** join/#asterisk LiCi- (n=lici@195.56.58.67) |
| 08:50.30 | *** join/#asterisk krdian (i=krdian@killer.radom.net) |
| 08:54.37 | *** join/#asterisk stoffell (n=kristof@165.199-136-217.adsl-dyn.isp.belgacom.be) |
| 08:54.55 | the1_ | can someone explain to me what is dahdi? |
| 08:56.02 | mort_gib | the1_: Zaptel |
| 08:56.34 | LiCi- | Hello |
| 08:57.30 | the1_ | it replaces zaptel? |
| 08:57.48 | orkid | yes |
| 08:58.09 | the1_ | :) cool. ive been reading outdated docs |
| 08:58.51 | dandre | is there any way to get the callerid string for a channel from the manager interface? |
| 08:58.54 | the1_ | if i dont have any digium hardware, i dont need to install dahdi right? |
| 08:59.14 | *** join/#asterisk iobug (n=gustav@93.123.102.99) |
| 09:01.29 | krdian | the1_: correct, but if u like to use iax2 trunking you;re should use it for timming purpose |
| 09:01.38 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:02.20 | kaldemar | dandre: be more specific |
| 09:02.32 | LiCi | cisco voice gw - sip trunk1 - asterisk - sip trunk2 - cisco call manager 6. I'd like use asterisk only for record all calls to wav. How can I identify a call if it is forwarded to another ext in CM? Is there a call id exitst in cdr which is exactly a same in cm and asterisk? |
| 09:02.39 | the1_ | krdian, thanks |
| 09:03.28 | the1_ | im tryin to have asterisk 1.6.0.1 and freepbx 2.5.1 working together, are these two compatible? |
| 09:04.13 | dandre | I am using asterisk manager insterface to know extenstion states (ringing, idle, in use, ...). I want to get the name string for that extension. I don't know how to do |
| 09:04.42 | dandre | I have tried status action with no success |
| 09:05.44 | Katty | Qwell: ding? |
| 09:05.56 | Qwell | just now |
| 09:06.32 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:08.32 | Katty | Qwell: :> |
| 09:09.55 | Katty | Qwell: doing bomb teh ship questabul. |
| 09:26.24 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:30.06 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 09:30.06 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 09:39.34 | *** join/#asterisk r0land (n=r0land@212.28.233.21) |
| 09:39.38 | r0land | hello all |
| 09:42.08 | r0land | can any1 guide me to adding CHANISAVAIL() to my current dialplan! wht im trying to do is simple.. exten =>_02,1,Dial(SIP/300) i wanna add chanisavail() to it so it actualy checks if the line is busy before the dial () |
| 09:48.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 09:52.33 | cousin_luigi | morn |
| 10:00.45 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 10:00.46 | cousin_luigi | is there a free skype connector for asterisk? |
| 10:00.57 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
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| 10:19.40 | *** join/#asterisk tompaw (n=tompaw@slave12.tesserakt.eu) |
| 10:19.42 | tompaw | hi |
| 10:26.27 | tompaw | can asterisk 1.6 perform HTTP queries? |
| 10:38.59 | kaldemar | tompaw: no, but it could be easily done with an agi, for example. |
| 10:41.44 | tompaw | kaldemar: right now I'm doing it by launching a bash script that performs a wget and then stores the value in asterisk's DB |
| 10:42.04 | tompaw | but I thought I've seen something during building of 1.6 that would suggest it's been implemented |
| 10:42.17 | tompaw | however I cannot find it now |
| 10:43.08 | *** join/#asterisk atlas95 (n=cyril@unaffilated/atlas95) |
| 10:43.50 | tompaw | got it! * Added a new module, res_config_curl, which permits using a HTTP POST url |
| 10:45.06 | tompaw | I cannot find any documentation regarding it ;/ |
| 10:52.54 | kaldemar | it's a resource module anyway, source says "plugin for portable configuration engine" |
| 10:53.29 | SwK | theres a curl dialplan function too |
| 10:53.42 | SwK | so you can curl a result directly into a channel variable |
| 10:53.53 | SwK | (or a DB key if you were so inclinded) |
| 10:54.50 | SwK | exten => _X.,n,Set(CALLERID(name)=${CURL(http://cnam1.secretsauce.net/getcnam?q=C&f=S&dn=${CALLERID(num)})}) |
| 10:55.14 | SwK | thats actually one thing I use now to get callerid name for calls coming in from sources that dont give me name info |
| 10:56.04 | kaldemar | hmm.. i've not compiled it for some reason. shame on me. |
| 10:58.30 | kaldemar | a missing libcurl-devel is a good enough reason. |
| 10:59.03 | SwK | yum it :P |
| 10:59.28 | SwK | (assuming of course you are on a yum enabled system) |
| 10:59.33 | kaldemar | i zyppered it. ;) |
| 11:00.15 | SwK | Suse/ |
| 11:00.17 | SwK | ? |
| 11:00.26 | kaldemar | yes. |
| 11:00.42 | kaldemar | so yum basically. |
| 11:00.49 | SwK | novell didnt kill it too? |
| 11:01.26 | kaldemar | zypper? no only the horrifying zmd. zypper has become quite nice with the 11 release. |
| 11:01.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:01.33 | SwK | i meant suse |
| 11:01.36 | SwK | :P |
| 11:01.47 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 11:01.49 | kaldemar | haha, opensuse is alive and kicking. |
| 11:01.54 | SwK | (it was a comment as to novells marketing ability not their engineering tallent) |
| 11:02.21 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-072-009.pools.arcor-ip.net) |
| 11:02.46 | Blackvel | hi all. is there any way to put also a comment when calling "database put whitelist <number> 1"? |
| 11:03.09 | tompaw | SwK: I will be using it for number portability lookup |
| 11:03.14 | tompaw | SwK: thanks for the xample! |
| 11:03.52 | Blackvel | for snom 370: is there any detail view? I navigate thru missed calls and I only see the callerid(name) but not the number? |
| 11:04.25 | SwK | tompaw, who's got LNP lookups via the web? |
| 11:04.35 | tompaw | SwK: I retrieve the GSM network ID using http and then basing on the result I choose appropriate gateway |
| 11:04.46 | SwK | nice |
| 11:04.48 | tompaw | SwK: I wrote my own script witch caching, stats etc |
| 11:04.59 | tompaw | we're using end2end's service |
| 11:06.49 | SwK | tompaw, we're actually in the process of writting a SS7 program that will allow you to query CNAM, LIDB, and LNP data via http queries like that... CNAM is done, LNP is working to return the LRN (ANSI style not ITU style yet) and LIDB is pretty close just figuring all out all the message types since none of the carriers lists are all the same |
| 11:07.10 | SwK | if I could get my hands on a ITU ss7 link we could do the ITU style stuff heh |
| 11:07.16 | *** part/#asterisk LiCi (n=lici@195.56.58.67) |
| 11:07.55 | tompaw | SwK: that's it, we don't have direct SS7 links (yet) ;> |
| 11:08.20 | SwK | tompaw, yeah... we're actually offering US CNAM via that interface now |
| 11:08.36 | SwK | my.metrostat.net for anyone interested.... |
| 11:09.08 | tompaw | Us.. how about EU? |
| 11:19.12 | *** part/#asterisk cousin_luigi (n=luigi@unaffiliated/cousinluigi/x-395723) |
| 11:23.08 | SwK | dont have EU unfortunately |
| 11:32.03 | tompaw | damn ;> |
| 11:37.23 | Katty | Qwell: :> |
| 11:37.29 | Katty | Qwell: 6 bars in! |
| 11:40.03 | Katty | Qwell: must nap soon tho. am running off breakfast pizza and caffeine. |
| 11:42.46 | *** join/#asterisk DarkRift (n=dark@65.92.166.80) |
| 11:44.01 | stix_ | Is it possible to reset the queue stats which is showed when using the command: queue show <queue> ? |
| 11:44.10 | stix_ | I have static members in my queues |
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| 12:29.27 | psykx-out | waves hello |
| 12:30.02 | psykx-out | can anybody recomend a good VOIP provider in the uk that uses IAX2? |
| 12:33.31 | *** join/#asterisk ultrav1olet (n=telnet@90.151.210.26) |
| 12:34.51 | ultrav1olet | is it possible to set up asterisk so that it allows to talk to a man who's calling you then transfer the call to another person? What IP softphones support that feature? |
| 12:35.36 | *** part/#asterisk knobo (n=user@cartman.nextra.no) |
| 12:36.14 | kaldemar | ultrav1olet: dtmf support on phones is enough. |
| 12:36.43 | kaldemar | see "core show application dial" and parameter t. |
| 12:37.00 | ultrav1olet | kaldemar: We have all Zoiper standard free edition installed - it surely supports DTMF |
| 12:37.34 | *** join/#asterisk Carlos_PHX (n=Carlos@166.sub-70-210-139.myvzw.com) |
| 12:37.41 | ultrav1olet | kaldemar: can you gave a bit more details? I'm very newbie to asterisk :-( |
| 12:38.52 | kaldemar | use the Dial command in your dialplan with parameter t, which allows the called person to transfer the call by sending the DTMF sequence defined in features.conf (# by default). |
| 12:38.57 | kaldemar | for the rest: |
| 12:39.01 | kaldemar | ~book |
| 12:39.02 | jbot | from memory, 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 |
| 12:39.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 12:40.16 | ultrav1olet | kaldemar: it seems like I have to make changes to features.conf - I have a default asterisk 1.4.X running |
| 12:40.33 | ultrav1olet | ah, yes, you have just mentioned that ;-) |
| 12:41.09 | kaldemar | you don't have to modify features.conf. |
| 12:41.45 | ultrav1olet | but how these changes are going to stay persistent between server reboots? |
| 12:43.52 | kaldemar | they stay in your config files, unless you're using some GUI to screw up your configurations. |
| 12:44.39 | ultrav1olet | kaldemar: I'm only using vi |
| 12:45.02 | ultrav1olet | and midnight commander, my server runs CentOS |
| 12:45.35 | kaldemar | so, just edit your configuration files. they are read by asterisk when it's started. |
| 12:46.06 | mankash | I have a simple setup with 2 sip phones connected internally to asterisk |
| 12:46.11 | mankash | when I call from one sip phone it doesn't wait for user to press any key it just exits out after Awnser |
| 12:46.17 | mankash | http://pastebin.com/m79580c4a please check this dialplan |
| 12:46.31 | ultrav1olet | kaldemar: that's quite clear. I'd be glad to know what are exactly should I change in features.conf ;-) |
| 12:47.15 | kaldemar | ultrav1olet: you do not have to touch features.conf. only extensions.conf. |
| 12:48.07 | kaldemar | mankash: what number are you calling? |
| 12:48.17 | mankash | 600 |
| 12:48.37 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000374.dsl.bell.ca) |
| 12:48.54 | kaldemar | show a cli pastebin of a failed call |
| 12:49.08 | mankash | http://pastebin.com/m79ba12fd this is a verbose log. It doesn't wait for pressing the buttons |
| 12:49.56 | kaldemar | do you hear the message playing? |
| 12:51.29 | ultrav1olet | kaldemar: it seems like I have to add an extension for parked calls, right? |
| 12:51.58 | kaldemar | ultrav1olet: no, no. add option "t" for your Dial command. |
| 12:52.31 | ultrav1olet | hm, I don't get it ;-) Where should I add it? |
| 12:53.13 | *** join/#asterisk mark_csi (n=csiadmin@host81-149-93-47.in-addr.btopenworld.com) |
| 12:54.26 | kaldemar | ultrav1olet: to the line where you dial your phone. read up on dialplans. |
| 12:54.50 | mankash | kaldemar have you checked |
| 12:55.05 | mark_csi | hello - anybody know how to move a phone from one macro to another dynamically? |
| 12:57.58 | kaldemar | mankash: do you hear the message playing? |
| 12:59.01 | mankash | yes |
| 12:59.21 | mankash | but it doesn't wait for me to press keys |
| 12:59.49 | kaldemar | if you want it to wait after the message, you must tell it to. |
| 13:00.26 | kaldemar | core show application WaitExten |
| 13:01.06 | kaldemar | background reads keypresses as long as the recording is playing, but when the recording stops, it moves on in the dialplan. |
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| 13:06.36 | mankash | what to do with "core show application WaitExten" |
| 13:07.56 | mankash | if background message is is of very samll length then user will not be able to press the keys |
| 13:08.52 | kaldemar | core show application WaitExten is a CLI command to show you what that application is and how to use it. |
| 13:09.25 | kaldemar | you better learn to read application and function documentation that way if you're going to build dialplans. |
| 13:09.35 | mankash | ok thx |
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| 13:22.33 | *** join/#asterisk kernelsensei (n=boris@gentoo/developer/kernelsensei) |
| 13:22.38 | kernelsensei | hi |
| 13:23.03 | Dr-Linux|home | sip.conf has by default out-of-band DTMF (rfc2833) and what about zapata.conf? |
| 13:24.36 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 13:24.39 | Telemac | Hello |
| 13:24.40 | kernelsensei | when I do a 'restart gracefully' or 'restart when convenient' with the AMI (telnet), asterisk doesn't seem to restart |
| 13:24.58 | jsmith | Dr-Linux|home: DTMF is typically inband on analog and channel-associated signalling T1s, and out-of-band (on the D channel) on ISDN BRI and PRI connections. |
| 13:25.11 | kernelsensei | in the CLI I get a lot of 'Unregistered application ' messages, but the CLI doesnt get killed |
| 13:25.27 | jsmith | kernelsensei: Which version of Asterisk? |
| 13:25.32 | WimpMan | Better ditch the thing about ISDN. |
| 13:25.33 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 13:25.46 | kernelsensei | jsmith: 1.4.18 |
| 13:26.20 | kernelsensei | jsmith: restart now works well |
| 13:27.14 | Dr-Linux|home | jsmith: sometime i face dtmf detection probolem on my analog zaptel channels, should i changed in zapata.conf dtmfmode=rfc2833 ? or analog only support inband dtmf? |
| 13:27.22 | jsmith | kernelsensei: That's strange... so if you later do a "core show uptime", what does the uptime show? |
| 13:27.48 | jsmith | Dr-Linux|home: Yes, analog channels don't speak RFC2833 or any other out-of-band signalling for that matter. You could try setting "relaxdtmf=yes" |
| 13:28.33 | Telemac | Anyone know if its possible to use SIP presence with realtime? I've tried several configurations but my test hint is still displayed as idle, even if I was talking with that peer |
| 13:28.46 | kernelsensei | jsmith: the uptime has not been reseted |
| 13:29.33 | kernelsensei | jsmith: In the AMI console I don't get a Response: Success, but Response: Follows |
| 13:29.46 | kernelsensei | jsmith: http://rafb.net/p/IkJmMk54.html |
| 13:30.10 | jsmith | kernelsensei: (You do understand the meaning of "restart gracefully" and "restart when convenient", correct?) |
| 13:30.26 | jsmith | kernelsensei: Assuming you do, I'd say that's probably a bug worth reporting on the bug tracker |
| 13:30.50 | *** join/#asterisk chazz (n=chazz@nat/digium/x-9c185d3b1faa22f9) |
| 13:31.03 | jsmith | Telemac: It should be... do you have call limits enabled so that the SIP channel driver actually keeps track of device state? |
| 13:31.44 | kernelsensei | jsmith: gracefully waits for empty call volume and does not accept new calls, convenient does the same but accepts new calls, right ? |
| 13:32.04 | Telemac | jsmith: Yes, limitonpeers=yes, call-limit=2, busylimit=1 |
| 13:32.16 | jsmith | kernelsensei: Correct |
| 13:32.38 | kernelsensei | jsmith: there isn't any call now on the test server |
| 13:32.54 | jsmith | Telemac: rtcachepeers might need to be set as well... (but I'm not expert on Realtime) |
| 13:33.03 | kernelsensei | jsmith: and the weird thing is that convenient/gracefully work well with the CLI |
| 13:33.18 | kernelsensei | but not with the AMI |
| 13:34.02 | jsmith | kernelsensei: You're right... I just tried it here. That's definitely a bug! |
| 13:34.18 | WimpMan | kernelsensei: Do they work from CLI when you are connected on AMI? i.e. could it be an AMI connection is regarded as activity? |
| 13:34.30 | Dr-Linux|home | jsmith: what about this unanswered question: |
| 13:34.31 | Dr-Linux|home | http://www.syednetworks.com/asteriskforums?forumaction=showposts&forum=5&thread=140&start=0 |
| 13:34.40 | Telemac | jsmith: I'll check it |
| 13:34.49 | jsmith | WimpMan: Yes, the CLI commands work even when AMI connections are present |
| 13:35.06 | kernelsensei | WimpMan: yes it works |
| 13:35.12 | kernelsensei | WimpMan: the ami connection gets killed |
| 13:35.30 | jsmith | Dr-Linux|home: Which question... that seems to be an entire forum! |
| 13:36.44 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
| 13:36.49 | Dr-Linux|work | jsmith: there is only one question about inband dtmf and out-of-band dtmf :S can you refresh the page |
| 13:37.03 | psykx-out | I have all the voip calls (not pstn) dropping out on my asterisk box in the first 20 seconds |
| 13:37.14 | *** part/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 13:37.31 | psykx-out | has any body seen this before? I'm sure it's not a bandwidth issue |
| 13:37.45 | jsmith | Dr-Linux|home: Easy answer: in-band DTMF means that there are actually the DTMF frequencies as audio tones right in the audio itself. |
| 13:38.20 | jsmith | Dr-Linux|work: Out-of-band DTMF means that the DTMF information isn't relayed in the audio itself, but is sent outside of the audio in another signalling stream |
| 13:39.00 | jsmith | Dr-Linux|work: The one out-of-band method you mention (rfc2833) is only valid for the SIP protocol |
| 13:39.13 | jsmith | has to run teach an Asterisk class... back online later! |
| 13:40.53 | anonymouz666 | has to run outside because Brianna Love is waiting |
| 13:41.44 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:43.24 | Telemac | jsmith: I get message "handle_request_subscribe: Received SIP subscribe for peer without mailbox: imoison", could that cause trouble to SIP presence ? |
| 13:46.40 | jsmith-away | Telemac: No, that means that a SIP device is trying to subscribe to a mailbox on a SIP peer for MWI, but that peer doesn't have a mailbox. |
| 13:47.01 | Telemac | jsmith-away: ok so not from it, thanx |
| 13:55.06 | Telemac | jsmith-away: You were right, it seems to need rtcachefriends=yes to work |
| 13:57.08 | *** join/#asterisk Segnale007 (n=Pietro@host191-254-dynamic.18-79-r.retail.telecomitalia.it) |
| 13:57.16 | Blackvel | for snom 370: is there any detail view on the screen? I navigate thru missed calls screen and I can only see the callerid(name) but not the callerid(number)? If I hit the O.K button the party gets called instead of moving to the detail view (I used to check the webinterface with my computer, but like to have all informations directly on the snom phone) |
| 13:58.03 | Blackvel | so it looks like the display is too small (the callerid(name) is too long) AND the caller information is not scrollable |
| 14:00.09 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:04.46 | tzanger | blah. the power supply on this adit 600 is screwed, that's why I am not getting ringing voltage |
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| 14:41.05 | kernelsensei | pickupmark doesn't seem to work correctly. If I've 3 phones at the office (A B C), A receives a incoming call (extern), B is calling an external number (ringing). If C tries to pick up the incoming call (A), it's the outgoing call (B) wich is picked up |
| 14:43.14 | *** join/#asterisk r0land (n=r0land@193.227.191.91) |
| 14:43.17 | r0land | hi all |
| 14:44.01 | r0land | i'm having trouble with my box/spa where my box doesnt detect the congestion signal from SPA. so it bind the 2 channels together.. causing an existing opened channel to merge with the caller attendant voice... |
| 14:44.40 | r0land | any advice on how to prevent that! i figure that chanavail() would do the trick.. but i fail to see how to set it up properly as its a live system and i wanna do it right from the first time |
| 14:44.59 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 14:54.47 | *** join/#asterisk evangelion (n=manzy_ze@ufficio.sonic.it) |
| 14:55.00 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 14:55.03 | sosperec | Hello |
| 14:55.09 | sosperec | How can I get a second dialtone for a call going out? |
| 14:55.13 | sosperec | I want to authenticate a user, and then allow to dial out |
| 14:57.29 | *** join/#asterisk seaq (n=seaq@190.144.113.26) |
| 14:59.35 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
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| 15:01.29 | [TK]D-Fender | sosperec: "core show application disa" |
| 15:04.56 | sosperec | Thank You, [TK]D-Fender! |
| 15:04.59 | sosperec | It works now |
| 15:05.05 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 15:05.17 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:06.26 | *** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 15:07.35 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 15:07.39 | evangelion | maybe is a stupid question, btw what is the difference between exten => 100,1,Dial(SIP/200) and exten => 100,1,Transfer(SIP/200) ? |
| 15:08.09 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 15:08.28 | [TK]D-Fender | evangelion: If the originating channel is a SIP channel it will tell them to completely reinvite over to that other device and * will no longer be involved with it in any way. |
| 15:08.33 | E-bola | Anybody here looked into integrating asterisk and ms live communication server? |
| 15:10.40 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
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| 15:19.06 | Akiyuki | Does anyone know if the Nortel 6812 will work with Asterisk? |
| 15:20.04 | jameswf | <troll> look another dahdi is an evil plot for digium to sell more cards post: http://pbxinaflash.com/forum/showthread.php?t=2612 </troll> |
| 15:21.04 | jameswf | Akiyuki: It could... the question is what do you need to make it happen |
| 15:21.47 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:22.06 | awk_r | jameswf, I followed the link and got "The server is too busy at the moment." |
| 15:22.28 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 15:22.29 | Akiyuki | How can assign the username/password in the phone? Or, assign it by mac address in sip.conf? |
| 15:22.39 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 15:23.14 | jameswf | awk_r: sadly that happens alot.... probably running on the $199 Everex wallmart special.... |
| 15:23.31 | awk_r | facepalms. |
| 15:23.43 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 15:23.55 | jameswf | hey I know we should make it a phone system :) |
| 15:24.06 | awk_r | heh |
| 15:24.09 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:24.45 | [TK]D-Fender | Akiyuki: http://products.nortel.com/go/product_assoc.jsp?segId=0&parId=0&catId=null&rend_id=23603&contOid=100217902&prod_id=61660&locale=en-US |
| 15:24.53 | jameswf | These atom processors are likr crack though... still finding cool things to do with em |
| 15:25.10 | [TK]D-Fender | Akiyuki: Says it talks SIP and is compatible with other major SIP servers. Probably should work fine with *. |
| 15:25.20 | r0land | hi [TK]D-Fender |
| 15:25.28 | [TK]D-Fender | aiksa[LV]: If you already own it, ok, if not, save yourself potential trouble and buy something else. |
| 15:25.37 | Akiyuki | Well, it only allows registration by IP address or MAC address, and i dont know if Mac address is sent across SIP? Or a way to define it in sip.conf? |
| 15:25.46 | r0land | i'm having trouble with my box/spa where my box doesnt detect the congestion signal from SPA. so it bind the 2 channels together.. causing an existing opened channel to merge with the caller attendant voice... any advice on how to prevent that! i figure that chanavail() would do the trick.. but i fail to see how to set it up properly as its a live system and i wanna do it right from the first time |
| 15:26.37 | [TK]D-Fender | Akiyuki: "registration by IP address" <- thats how EVERY device works. ... its not VoIP for nothing. |
| 15:26.53 | awk_r | VoMAC! |
| 15:27.21 | [TK]D-Fender | r0land: Sounds like you configured your SPA wrong |
| 15:28.57 | Akiyuki | Well, when I try to register the sip device over the network, I get this error in the CLI [Nov 13 10:28:10] NOTICE[1186]: chan_mgcp.c:1726 find_subchannel_and_lock: Gateway '00405a1468c5' (and thus its endpoint 'd001') does not exist |
| 15:29.16 | [TK]D-Fender | Akiyuki: Go set it for SIP, not MGCP |
| 15:29.37 | Akiyuki | In the device itself? Or in sip.conf ? |
| 15:30.13 | [TK]D-Fender | Akiyuki: You seem to be missing the big print : chan_mgcp.c: <-- * is getting **MGCP** communications from it. |
| 15:30.43 | [TK]D-Fender | aiksa[LV]: * can't make your device change the protocol it uses or how its configured |
| 15:30.55 | *** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
| 15:30.57 | Akiyuki | MGCP is the only option in this phone. |
| 15:31.16 | [TK]D-Fender | aiksa[LV]: The tech sheet from Nortel seems to say it also supports SIP. |
| 15:31.21 | r0land | [TK]D-Fender yes i know, ive looked into the sipura's forum and they mentioned this is a common issue with asterisk and it should b solved with chanavail() on asterisk's side instead of immediatly going to dial() |
| 15:31.21 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 15:31.29 | [TK]D-Fender | aiksa[LV]: Perhaps it needs a firmware flash to do so, but its highly recommended. |
| 15:31.44 | [TK]D-Fender | aiksa[LV]: and sip.conf has NOTHING to do with that phone if its in MGCP mode. |
| 15:31.48 | Un1x | [TK]D-Fender, in your opinion was 1.2 more stable then recent branches? |
| 15:32.07 | [TK]D-Fender | Un1x: Procrastination : the art of keeping up with yesterday. |
| 15:32.43 | [TK]D-Fender | Un1x: 1.2 is not even supported any longer and thus I do not recommend. 1.6 just left beta so I'd let the dust settle on this for a little bit. so latest 1.4 it is... |
| 15:32.49 | Un1x | [TK]D-Fender, i'm not procasinating i was just told by alot of people that 1.2 is alot more stable then recent branches |
| 15:33.05 | Akiyuki | damn, it appears the config file for these fones has them hard coded as MGCP |
| 15:33.14 | Un1x | so 1.4 |
| 15:33.20 | [TK]D-Fender | Un1x: that is a rather generic statement given 1.6 is out... |
| 15:33.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:33.44 | [TK]D-Fender | aiksa[LV]: Go see about reflashing them |
| 15:33.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:34.27 | *** join/#asterisk Nasra (n=maxshipp@190.166.71.124) |
| 15:36.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:36.21 | Akiyuki | Or, I guess I can make it work with mgcp.conf |
| 15:38.51 | *** join/#asterisk Segnale007 (n=Pietro@host234-249-dynamic.27-79-r.retail.telecomitalia.it) |
| 15:38.56 | awk_r | Un1x, that would have been the case a yr or longer ago, but not now. 1.4 is more stable than 1.2 |
| 15:39.23 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f565beee5e44b2cb) |
| 15:39.23 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:39.44 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 15:41.13 | *** join/#asterisk eit (n=eit@64.122.178.15) |
| 15:42.02 | [TK]D-Fender | aiksa[LV]: If that's the case then I hope you stop looking at sip.conf |
| 15:42.44 | Akiyuki | I added the device to mgcp.conf and reloaded and am getting the same error |
| 15:43.17 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:43.25 | eit | Greetings, has anyone seen this? we have an asterisk PBX that is working rather well but whenever a phone dials a few specific external numbers and it goes to VM on the external VM system our call disconnects and does not allow us to leave a message. It seems to happen to all phones on our system that try to dial the specific external numbers. |
| 15:44.01 | awk_r | eit, are you certain it isn't the external system? |
| 15:44.16 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.13) |
| 15:44.29 | eit | The problem happens each time you call the given numbers and doesn't happen with those same numbers from a different system or cell phone. It's really strange. |
| 15:45.02 | awk_r | eit, version of Ast? and type of trunk? |
| 15:46.02 | eit | it's a PRI trunk and asterisk is.... |
| 15:50.22 | eit | asterisk is version 1.2.26.1 svn rev 79171 |
| 15:50.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:51.07 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 15:55.11 | awk_r | eit, disconnects on a PRI could be in a few places. I would start with a pri debug and see whats going on there. |
| 15:55.48 | *** join/#asterisk ManxPower (n=manxpowe@87.sub-70-222-233.myvzw.com) |
| 15:56.48 | *** part/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 15:57.31 | aliver | Is there any way to have a meetme conf originate a call without associating it specifically with a registered client? |
| 15:57.42 | aliver | Is that what n-way calling is? |
| 15:59.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:59.28 | ManxPower | aliver: The best way is to use .call files. That is one of the standard ways to make automated calls. |
| 16:01.02 | *** join/#asterisk LiCi (n=lici@195.56.58.67) |
| 16:01.05 | LiCi | hu |
| 16:01.07 | LiCi | i |
| 16:01.10 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 16:02.26 | iratik | Extension 002 is ringing too soon ... even though i've provided a timeout argument to dial. I want extension 002 to ring only after 000 doesn't answer for 10 seconds ... I've got the documentation up... but i'm still doing something wrong http://www.pastie.org/313945 |
| 16:02.39 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:02.47 | iratik | (I increased the timeout to 12, just to see if it would make a difference) |
| 16:02.55 | Assid | [TK]D-Fender: i contacted polycom, they recommended me the HDX7000 |
| 16:03.17 | Assid | besides polycom, doesnt nortel and other vendors deal with this kinda equipment |
| 16:03.36 | WimpMan | assid |
| 16:03.42 | Assid | hey WimpMan |
| 16:03.44 | WimpMan | ups |
| 16:03.53 | Assid | ups? |
| 16:04.08 | WimpMan | actually wanted to search back for the question. |
| 16:04.22 | [TK]D-Fender | iratik: Dial(SIP/000,12) <- well you're ringing it for 12 seconds. theres your problem :p |
| 16:04.34 | [TK]D-Fender | Assid: Cisco. |
| 16:04.40 | Assid | WimpMan: video conferencing |
| 16:05.09 | eit | Your thinking that it's on the circuit and not an internal issue in asterisk? I was assuming it was in asterisk but will look more at the PRI. |
| 16:05.52 | iratik | [TK]D-Fender: The 000 extension used to get at least 5 rings in... now it only gets 2... then the 002 extension gets 2 rings... then the gray phone starts ringing again... doesn't make much sense |
| 16:06.05 | [TK]D-Fender | aliver: n-way calling is a sparse term that need boundaries attached to it. |
| 16:06.14 | [TK]D-Fender | aliver: as in which side is the one doing the work. |
| 16:06.24 | WimpMan | eiy: You shouldtry to find out who's disconnecting those calls and if it's not * who and why (cause and location). |
| 16:06.28 | iratik | Feels like the 000 extension .. is only ringing for 5-6 seconds |
| 16:06.55 | [TK]D-Fender | iratik: Well your dial is in seconds, not rings. watch CLI and I'm sure * is doing exactly what you've told it to. |
| 16:07.22 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:07.22 | iratik | Thanks |
| 16:07.40 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:09.49 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 16:10.32 | LiCi | There is something like this: cisco voice gw - sip trunk - asterisk - sip trunk - cisco cm6 system. Asterisk is for call recording, asterisk cdr is in a rdbms. If in CM6 there is a call transfer, how can be the call identified? Is there something like CALL ID, which is the same key in asterisk and CM? |
| 16:12.34 | LiCi | I mean is there something in SIP that I use for identify a conversation recorded? |
| 16:13.42 | LiCi | Or just two ntp syncronized server and the time? It can be complicated if there are many calls in the same minutes |
| 16:14.20 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:14.39 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 16:14.51 | WimpMan | LiCi: Have you thought about using a packetsniffer instead of *? |
| 16:15.09 | LiCi | WimpMan: not yet |
| 16:16.03 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-e5ffeb622ef98594) |
| 16:16.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:16.06 | LiCi | WimpMan: should I sniff all voice vlans? |
| 16:17.08 | [TK]D-Fender | LiCi: You should look at your * SIP dialog. |
| 16:17.49 | LiCi | [TK]D-Fender: is it in a log file? |
| 16:18.17 | *** join/#asterisk itguru (n=p@host81-134-10-140.in-addr.btopenworld.com) |
| 16:18.29 | *** join/#asterisk sdaniels (n=chatzill@216.65.195.52) |
| 16:20.13 | LiCi | I'm fairly new to VOIP and therefore dangerous... hope this isn't a waste of your space, thanks for guidelines |
| 16:20.59 | [TK]D-Fender | LiCi: You look in * CLI as the calls are being placed,e tc |
| 16:21.50 | anonymouz666 | [TK]D-Fender: did you see the blitzrage presentation about clustering ? |
| 16:21.57 | jameswf | Application for bailout money.... everyone should do this http://www.ustreas.gov/press/releases/reports/applicationguidelines.pdf |
| 16:22.01 | [TK]D-Fender | anonymouz666: nope |
| 16:22.39 | ManxPower | iratik: pastebin the CLI output of a problem call |
| 16:23.08 | ManxPower | iratik: A standard USA ring cycle is 6 seconds. |
| 16:23.15 | itguru | one of my extensions refuses to make calls, even though the others work just fine - but nothing comes up in the asterisk CLI in debug mode. The SIP settings are correct, and the extension can recieve calls, just not make them |
| 16:23.16 | iratik | really? |
| 16:23.18 | anonymouz666 | My english sucks and the volume is very low. Almost impossible to understand |
| 16:23.22 | ManxPower | So a 12 second timeout would be 2 rings. |
| 16:23.36 | *** join/#asterisk getsql08 (n=getsql05@68.244.39.102) |
| 16:23.43 | iratik | 24 seconds for 4 rings |
| 16:23.51 | [TK]D-Fender | anonymouz666: No... thats how he talks in real life ;) |
| 16:23.52 | LiCi | [TK]D-Fender: it sounds like that could be quite different to find some conversations recorded in some cases |
| 16:24.04 | [TK]D-Fender | LiCi: huh? |
| 16:24.17 | LiCi | :) |
| 16:24.26 | shazaum | anonymouz666, ur good, my is sux |
| 16:24.36 | [TK]D-Fender | itguru: Enable SIP debug and see. |
| 16:26.35 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 16:29.16 | LiCi | [TK]D-Fender : I mean If I have to listen to a conversation ASAP, that's could be different to find it, if there was a call transfer in CM, because as I know * don't know anything what happens in CM |
| 16:29.17 | itguru | i've just enabled SIP debug - and still for reasons unknown to me - it still won't connect |
| 16:31.03 | ManxPower | it passtebin.ca |
| 16:31.23 | ManxPower | itguru: pastebin the CLI output of a problem call. |
| 16:31.45 | ManxPower | LiCi: I think you need to read the Asterisk book. |
| 16:31.50 | ManxPower | ~book |
| 16:31.51 | jbot | i heard 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 |
| 16:32.02 | LiCi | definitly, downloaded last night |
| 16:32.25 | ManxPower | LiCi: Then go read it. What is your native language? |
| 16:33.02 | LiCi | I started to read, simply I just would like to know, is it possible to use * in this situation |
| 16:33.06 | LiCi | ManxPower: hungarian |
| 16:34.04 | ManxPower | LiCi: Maybe posting on the mailing list would be better. Your lack of english skills make it difficult to help you. There is nothing wrong with that. |
| 16:34.14 | *** join/#asterisk klictel (n=klictel@nat/digium/x-4ed9b459c83cc8f0) |
| 16:34.17 | LiCi | I wrote :) |
| 16:34.18 | Katty | Qwell: 3 hour nap, and ready to go at it again! |
| 16:34.49 | *** join/#asterisk ecm (n=ae@ns.stalfort.ro) |
| 16:35.02 | Deeewayne | offers Katty catnip |
| 16:35.41 | Akiyuki | Anyone have experience configuring the Nortel 6812/30 with Asterisk MGCP? Or, enabling SIP in the device? |
| 16:35.46 | ecm | Hello, all. Could someone please point me to an application I could use so as to transfer a call to another phone, if the currently interogated devices doesn't pickup within X seconds? |
| 16:36.14 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 16:36.19 | WimpMan | ecm: Dial |
| 16:36.24 | ManxPower | ecm: Put a timeout on the Dial line, then do a 2nd Dial in your dialplan. |
| 16:36.43 | ecm | ManxPower: great! thanks alot |
| 16:36.44 | ManxPower | See "core show application dial" and extensions.conf.sample (included in the Asterisk source code) |
| 16:37.17 | [TK]D-Fender | LiCi: If the call is transferred INSIDE of your CCM, then * will not be making another recording for it. |
| 16:37.23 | ecm | also, has anyone made spa-932 console to work with asterisk so as the leds show if any currennt subscribed phone is in the 'unavailable' state?(asterisk gets all the hints right) |
| 16:37.44 | itguru | ManxPower I get no info in CLI at all :( |
| 16:37.52 | ManxPower | ecm: google search for: spa-932 BLF |
| 16:38.05 | ManxPower | itguru: if you get no CLI output then Asterisk is NOT calling the phone. |
| 16:38.20 | ecm | ManxPower: I managed to make BLF work, but SPA-932 only responds to Ringing and other hints like that, not UNAVAILABLE |
| 16:39.17 | itguru | ManxPower: I'm trying to make a call from the extension in question. At first, I thought it was the sip account not working, but I put in the info of an extension that I know is working, and it still would not make calls |
| 16:39.26 | [TK]D-Fender | ecm: then the SPA cannot indicate that state |
| 16:39.45 | [TK]D-Fender | ecm: Polycom does not report "ringing". It flags it as "busy". |
| 16:39.52 | [TK]D-Fender | ecm: Aastra does everything right |
| 16:39.57 | [TK]D-Fender | ecm: YMMV <-- |
| 16:40.29 | WimpMan | Hmmm. Didn't I read that you can tell the SPAs what pattern to light on what state? |
| 16:40.46 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 16:40.50 | b11d` | hello everyone |
| 16:40.51 | ecm | [TK]D-Fender: here's what I get when I unplug the phone "31": 34@stalfort-intern : SIP/34 State:Unavailable Watchers 0 |
| 16:41.08 | LiCi | [TK]D-Fender: thx |
| 16:41.34 | [TK]D-Fender | ecm: Yes, I already understand that... |
| 16:41.43 | [TK]D-Fender | b11d`: Long time no see... |
| 16:41.44 | ecm | WimpMan: nope, you can only tell spa-962 tthe pattern of the lights associated with it's internal leds, not the ones at the spa-932 console |
| 16:41.52 | b11d` | question.. i have setup the system to use .call files for every extension on my system in the event of an emergency.. there are about 200 .call files -- i have a pretty high end system, but asterisk can not or does not handle receiving all of those .call files at the same time.. any suggestions? I really need all extensions to be called.. |
| 16:42.03 | b11d` | yeah, been busy TK.. you know how it is |
| 16:42.30 | WimpMan | ok |
| 16:42.42 | [TK]D-Fender | b11d`: Make your script mv them in 1s increments. |
| 16:42.55 | ecm | ok, thanks all for your help, I'll get back to doing dumb things |
| 16:42.58 | b11d` | you think that would make a big enough difference? I was thinking the exact same thign actually |
| 16:43.02 | [TK]D-Fender | b11d`: and tighten up as you see it surviving |
| 16:43.05 | *** join/#asterisk bkruse (n=bkruse@nat/digium/x-8203be67d0df8742) |
| 16:43.05 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 16:43.05 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) [NETSPLIT VICTIM] |
| 16:43.05 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 16:43.05 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) [NETSPLIT VICTIM] |
| 16:43.05 | *** join/#asterisk Cybertoy (n=Cybertoy@adsl-ecom-4-c15-p038.vtx.ch) [NETSPLIT VICTIM] |
| 16:43.05 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) [NETSPLIT VICTIM] |
| 16:43.05 | *** join/#asterisk markcw (n=markc@beigebox.liquidev.com) [NETSPLIT VICTIM] |
| 16:43.05 | *** mode/#asterisk [+o bkruse] by irc.freenode.net |
| 16:43.16 | b11d` | sure.. |
| 16:43.25 | [TK]D-Fender | b11d`: 200s = just over 3 mins for all calls. |
| 16:43.41 | b11d` | maybe I could move two of them every 1 second instead |
| 16:43.43 | [TK]D-Fender | b11d`: I'm sure its plenty of time. You could trying 500ms to be safer. |
| 16:43.48 | [TK]D-Fender | b11d`: exactly. |
| 16:43.55 | b11d` | hrm.. worth a shot! :) |
| 16:44.02 | [TK]D-Fender | b11d`: test it out and see. Just sit around on TTOP & CLI to watch |
| 16:44.08 | *** join/#asterisk Ccomp5950 (n=Ccomp595@66.190.102.236) |
| 16:44.14 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:44.29 | b11d` | aye.. i'd prefer to dump all log data out to a file via syslog so I can review it carefully later.. |
| 16:45.03 | *** join/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
| 16:45.39 | *** join/#asterisk dash (n=washort@ampere.divmod.com) |
| 16:45.53 | dash | howdy |
| 16:46.44 | b11d` | high |
| 16:46.53 | dash | anybody know if asterisk's sip implementation supports internal bridging? |
| 16:46.59 | LiCi | [TK]D-Fender: just one thing: it means that, if I want to lookup a recorded call from days ago, I can find it by it's time stamp, or by incoming phone number, or by called extension in Cisco CM. But in a special case, when I only know a time range and a transferred extension how can i find the record? |
| 16:47.12 | dash | I have an external 3PCC SIP program and i'm wondering if I can tell one sip stream to bridge to another just be sending a new SDP to it |
| 16:47.22 | dash | s/be/by/ |
| 16:47.36 | dash | jbot: thanks for telling me what I meant. |
| 16:47.36 | jbot | dash: my pleasure |
| 16:47.54 | rob0 | ~botsnack |
| 16:47.54 | jbot | rob0: aw, gee |
| 16:48.41 | LiCi | [TK]D-Fender: i mean lookup from *, using the extension (cdr) from cisco cm... |
| 16:48.47 | [TK]D-Fender | LiCi: You might not be able to pin this down. * is just sitting in between.... |
| 16:49.09 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 16:49.10 | dash | so maybe what I actually mean is: does asterisk deal with receiving SIP reinvites in a reasonable way? |
| 16:49.14 | [TK]D-Fender | LiCi: If you can tie your CCM's transfer records back to the original then you might be OK |
| 16:49.27 | [TK]D-Fender | dash: Reinvites = trouble |
| 16:49.30 | Dr-Linux|home | anybody is using SRV record with Asterisk? |
| 16:49.45 | dash | [TK]D-Fender: Do tell. |
| 16:49.59 | [TK]D-Fender | dash: Well... esp with NAT anyways... |
| 16:50.20 | dash | [TK]D-Fender: Everything's trouble with NATs though |
| 16:50.35 | dash | [TK]D-Fender: so that's not surprising or interesting |
| 16:50.40 | [TK]D-Fender | dash: true, but this above all else. |
| 16:50.42 | LiCi | [TK]D-Fender: this is what i was trying to ask wheter there is any info (for ex. in SIP) that can be used for this "reverse lookup" |
| 16:51.04 | [TK]D-Fender | LiCi: So far * may not even be aware of this transfer... |
| 16:51.34 | dash | [TK]D-Fender: At the moment I'm not concerned if it works with NAT or not |
| 16:51.34 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:51.51 | dash | since none of the peers i'm dealing with are behind it |
| 16:59.46 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:59.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:00.32 | Katty | lmadsen: :> |
| 17:00.53 | LiCi | [TK]D-Fender: okey, this means the lookup keys can be in *: timestamp, incoming number, destination ext., and in CCM, i have also the timestamp, the ext. But If there was redirection in ccm, I must use the previous keys in ccm, or use the data from * and narrow down the needed call from several others |
| 17:02.07 | WimpMan | LiCi: The calls are transferred internally? |
| 17:02.17 | LiCi | WimpMan: sure |
| 17:03.04 | LiCi | WimpMan: I can work from * cdr and all i have in ccm |
| 17:05.10 | LiCi | WimpMan: actually I was trying to find out wheter is there a "set of fields" that can be found in both dbs for unique call identification |
| 17:05.57 | *** part/#asterisk dash (n=washort@ampere.divmod.com) |
| 17:06.42 | LiCi | WimpMan: currently my only guess is the timestamp of the calls (at least ntp syncronized...) |
| 17:06.55 | WimpMan | I have no clue what you can do on CM, but on * it should be possible to read the SIP ID and place it into an CDR field. |
| 17:06.56 | Qwell | Katty: 3am |
| 17:07.08 | Qwell | DK > 56 |
| 17:09.31 | WimpMan | LiCi: ${SIPCALLID} should be the one uniquely identifying calls form end to end, I think. |
| 17:10.37 | LiCi | thank you! |
| 17:12.00 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:12.03 | LiCi | [TK]D-Fender:thx u too |
| 17:12.24 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 17:13.26 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:15.37 | itguru | Has anyone here had problems with international calls from BT ISDN 30 ? |
| 17:20.02 | tzafrir_laptop | BT has lots of them |
| 17:20.11 | tzafrir_laptop | Could you please be more specific? |
| 17:21.17 | *** join/#asterisk hi365_m (n=hi365@213.151.41.7) |
| 17:22.25 | itguru | ISDN 30e |
| 17:22.36 | itguru | I can't get international calls working |
| 17:23.20 | WimpMan | itguru: That's the part you already told us. |
| 17:23.27 | itguru | aww dammit - battery on laptop... |
| 17:23.33 | itguru | will be back soon :) |
| 17:25.09 | *** join/#asterisk Uatec (i=57c24701@gateway/web/ajax/mibbit.com/x-a8aaa4d66cc4c0e2) |
| 17:25.13 | Uatec | hello |
| 17:25.20 | Uatec | iax2 is TCP right? on port 4569? |
| 17:25.30 | WimpMan | No. UDP. |
| 17:25.33 | Uatec | so when i telnet to my asterisk server on port 4569 i should get SOME... |
| 17:25.34 | Uatec | oh |
| 17:25.49 | WimpMan | For rather obvious reasons. |
| 17:26.13 | Uatec | aahhhh |
| 17:26.18 | *** join/#asterisk ZB2 (n=ZB@189.102.227.29) |
| 17:26.24 | Uatec | i THOUGHT it was UDP, but for some reason my firewal was configured for TCP |
| 17:26.29 | [TK]D-Fender | Somebody need to read the big print :) |
| 17:26.43 | Uatec | :) |
| 17:27.56 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:29.09 | Uatec | well my firewall now allows iax2 through, infact both firewalls allow UDP:4569 through and send it to the right place |
| 17:29.17 | Uatec | but my servers still aren't connection |
| 17:29.20 | Uatec | they're listed as unreachable |
| 17:29.48 | [TK]D-Fender | Uatec: then go look at your * configs & traffic |
| 17:30.05 | Uatec | i am, tis interesting |
| 17:32.46 | lmadsen | hugs katty |
| 17:35.04 | Uatec | dammit, my firewall isn't lettting traffic out |
| 17:36.01 | c4t3l | speaking of 4569... I remember when Time Warner Cable (in Houston) blocked traffic on that port |
| 17:36.09 | anonymouz666 | lmadsen: I was looking at your presentation about clustering... not sure if it's a good idea by using LVS to share the registrations between the boxes that both will share a centralized DB with the peers |
| 17:36.25 | lmadsen | LVS? |
| 17:36.29 | anonymouz666 | so the phones will register with the virtual IP |
| 17:36.43 | anonymouz666 | heartbeat ldirectord package |
| 17:36.52 | lmadsen | in that configuration the registration point was OpenSER |
| 17:37.41 | lmadsen | I haven't thought of a reason why it couldn't work in asterisk though |
| 17:38.18 | anonymouz666 | correct I understood the presentation. The idea I wrote is the one I had after reading your presentation |
| 17:40.12 | c4t3l | katty: How's your dog? |
| 17:40.13 | lmadsen | please elaborate |
| 17:42.24 | anonymouz666 | lmadsen: phones register to a "Virtual IP", let's say 10.0.0.1. The LVS system will do a load balance between 10.0.0.2/3 asterisk boxes. these boxes will also use the same centralized peers configuration in a database. that's the idea |
| 17:43.08 | anonymouz666 | the main disadvantage is the database (centralized), that can become let's say... also a cluster in the future. |
| 17:43.27 | lmadsen | right... |
| 17:43.35 | lmadsen | waits for the gotcha |
| 17:44.29 | anonymouz666 | just wondering if that sounds reasonable... |
| 17:44.52 | lmadsen | I had thought of that same scenario. Have no implemented it, but I don't bother with LVS anymore |
| 17:44.56 | lmadsen | I just let all systems run at the same time |
| 17:45.18 | lmadsen | you can register to any of the servers, and get access to the same data |
| 17:45.28 | lmadsen | same permissions, same dialplan, etc... |
| 17:45.34 | lmadsen | why have a hot standby when it could be doing something? |
| 17:45.56 | lmadsen | if it dies... then the phone just has to have a backup registration and failover to another server |
| 17:46.30 | lmadsen | and we have the registration information in the DB... so should still be able to send a call to the phone |
| 17:47.05 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 17:47.54 | c4t3l | lmadsen: Is your presentation online anywhere? I'd like to read it. |
| 17:48.05 | lmadsen | blog.leifmadsen.com I think |
| 17:48.16 | lmadsen | probably in the post about astricon |
| 17:48.23 | c4t3l | k |
| 17:48.30 | c4t3l | thx |
| 17:48.32 | lmadsen | you can actually view the presentation... I think it is also linked there |
| 17:48.42 | lmadsen | (like... the video where I spoke at astricon) |
| 17:48.53 | c4t3l | sweet |
| 17:49.14 | anonymouz666 | lmadsen: so the balance between the servers is kind of manual? |
| 17:49.26 | *** join/#asterisk ALAMI (n=ALAMI@ce3.net.hrz.tu-darmstadt.de) |
| 17:49.28 | c4t3l | Is this * clustering using OpenSER and some sort of shared data pool? |
| 17:49.29 | lmadsen | anonymouz666: sure |
| 17:49.31 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 17:49.40 | lmadsen | c4t3l: view the presentation then ask questions |
| 17:49.46 | c4t3l | werd |
| 17:50.10 | raasdnil | evening all. Anyone around who has gotten 1.6 to connect with dahdi drivers on a TE210P to a Telstra E1 (Onramp 30) in Australia? - having some issues here |
| 17:50.11 | lmadsen | anonymouz666: idea is to use DUNDi to transport information across the clusters before delivering calls to other servers |
| 17:50.36 | lmadsen | you could have a "router" in the front that intelligently distributes calls based on various information passed back in the DUNDi dialog |
| 17:50.43 | anonymouz666 | lmadsen: I was thinking about LVS to do this job... So I wouldn't care about the "fair distribution" between the servers... another disadvantage will be much harder to control the calls |
| 17:50.59 | lmadsen | raasdnil: wow... that's pretty specific :) if you have digium hardware though, you should be able to call to digium support |
| 17:51.17 | lmadsen | anonymouz666: there are various ways to solve the same problem, yes |
| 17:51.31 | lmadsen | depends on your goals and needs |
| 17:51.31 | raasdnil | lmadsen: I like providing the info I can :) |
| 17:51.39 | lmadsen | raasdnil: :) |
| 17:51.59 | *** part/#asterisk LiCi (n=lici@195.56.58.67) |
| 17:52.58 | anonymouz666 | c4t3l: go to the astricon.net website, there are all the presentations to download |
| 17:53.13 | lmadsen | well... all the presentations that were recorded :) |
| 17:53.38 | anonymouz666 | lmadsen: I saw yours. It's very low |
| 17:53.45 | anonymouz666 | My english sucks I didn't understand much :) |
| 17:54.09 | WimpMan | download? I only see a blak rectagle, probably of some plugin I removed for security reasons. |
| 17:54.18 | anonymouz666 | http://www.astricon.net/2008/glendale/web/confTracks.php#t152 |
| 17:54.25 | anonymouz666 | here is the link |
| 17:54.31 | lmadsen | anonymouz666: yes... unfortunately wasn't the best quality |
| 17:55.04 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 17:55.21 | Qwell | Down Load? O.o |
| 17:55.24 | Qwell | how silly |
| 17:55.38 | anonymouz666 | haha |
| 17:55.41 | anonymouz666 | it's beautiful |
| 17:58.51 | WimpMan | anonymouz666: Thanks a lot |
| 17:58.51 | hardwire | Polycom 501 is pretty beautiful right? |
| 17:58.52 | hardwire | I shouldn't think about anything more complex than a 501? |
| 17:58.52 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 17:58.52 | hardwire | cause I just need to do simple and remote phones.. and the SPA-94x series is a bit .. wonky |
| 17:58.52 | lmadsen | I like polycom phones |
| 17:58.52 | lmadsen | but use the 430 or 450 |
| 17:58.52 | hardwire | I know you do lil' guy |
| 17:58.52 | hardwire | I know. |
| 17:58.52 | hardwire | lmadsen: orly? |
| 17:58.52 | awk_r | likes his 430. |
| 17:58.52 | lmadsen | 501 is kinda old and outdated. Still has firmware... but the 430 and 450 is cheaper I'm pretty sure |
| 17:58.52 | hardwire | I need something with sidecar compat.. I'm gonna research those |
| 17:58.52 | lmadsen | pretty sure any of the polycom phones will do sidecar... |
| 17:58.52 | hardwire | and I need to research the Jabbra EHS support |
| 17:58.52 | lmadsen | isn't positive |
| 17:59.08 | hardwire | you're quite positive.. thats's why we love you. |
| 17:59.45 | tzanger | that reminds me of a terrible joke |
| 18:00.30 | *** join/#asterisk km2 (n=x@32.178.20.165) |
| 18:00.30 | tzanger | two atoms are walking down the street when one stops suddenly and exclaims "Wait! Wait, I think I've lost an electron!" "Are you sure?" asks the other atom. "Sure, yeah I'm sure," the first one says, "I'm positive!" |
| 18:00.30 | hardwire | the 450 is athe conference phone? |
| 18:00.44 | hardwire | tzanger: boo |
| 18:01.02 | hardwire | lols anyways |
| 18:01.39 | awk_r | nice |
| 18:01.41 | hardwire | beats voipsupply.com's new website |
| 18:02.34 | hardwire | you know.. for the price.. 430's are pretty awesome |
| 18:02.45 | hardwire | not sure I'm down with the 2 line part tho |
| 18:03.05 | hardwire | I never know when phones advertise "lines" if they can handle concurrent calls per sip account |
| 18:03.21 | Primer | Anyone know why in asterisk 1.4.22, calls from asterisk to a polycom phone that is registered to it via SIP would show calls as: sip:8885551212@1.2.3.4 where 8885551212 would be the actual callerid of a call that came in via POTS, or an extension on the same asterisk, and 1.2.3.4 is the IP of the asterisk in question? The phones and asterisk are most definitely on the same subnet |
| 18:03.27 | Primer | In 1.4.20 this doesn't happen |
| 18:04.35 | hardwire | Primer: because that's how it works. |
| 18:05.27 | *** join/#asterisk ACK_NAK (n=Miranda@home.chicagoventure.com) |
| 18:05.28 | sdaniels | thats normal behavior |
| 18:05.39 | [TK]D-Fender | hardwire: unless you need RJ9 headset the IP 330 is a better deal. |
| 18:05.44 | ACK_NAK | Morning gentlemen |
| 18:05.52 | [TK]D-Fender | hardwire: IP 450 is new and effectively replaces the IP 50X |
| 18:05.54 | Primer | hardwire: how can I make it NOT do that? |
| 18:05.59 | sdaniels | you cant |
| 18:06.12 | hardwire | [TK]D-Fender: ok.. I'll keep searching for 450 issues |
| 18:06.21 | hardwire | and I need rj9 for the EHS GN9330 |
| 18:06.23 | [TK]D-Fender | hardwire: What "issues"? |
| 18:06.28 | hardwire | er 9350 |
| 18:06.31 | Primer | setting ${CALLERID(num)} doesn't do it |
| 18:06.47 | hardwire | err |
| 18:06.51 | hardwire | s/issues/reviews/ |
| 18:06.53 | beek | Primer: You can change that via the polycom configuration |
| 18:06.53 | Primer | there MUST be a way |
| 18:06.58 | hardwire | I can't seem to find 450 info |
| 18:07.06 | Primer | beek: meaning, I have to parse it out? |
| 18:07.09 | hardwire | nm |
| 18:07.10 | hardwire | found it |
| 18:07.12 | ACK_NAK | hardwire: that's a good headset |
| 18:07.16 | sdaniels | what are you asking exactly? you dont want the calls to look like this: sip:##########@#.#.#.#? |
| 18:07.21 | hardwire | ACK_NAK: I'd rather just use a softphone and the USB |
| 18:07.23 | hardwire | joo know! |
| 18:07.24 | beek | Primer: no, it's a polycom configuration file issue. |
| 18:07.30 | beek | Primer: I'm looking for it now.. |
| 18:07.44 | Primer | beek: excellent, thanks |
| 18:07.55 | [TK]D-Fender | hardwire: http://www.polycom.com/usa/en/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip450.html |
| 18:08.02 | ACK_NAK | hardwire: Not a big fan of softphones |
| 18:08.08 | awk_r | Primer, beek is right. Asterisk is most likely sending the correct CID info (you can check the sip packets). Its most likely the phone config |
| 18:08.13 | [TK]D-Fender | hardwire: You don't seem to have tried very hard at all. |
| 18:08.32 | hardwire | [TK]D-Fender: feeling prickly today? |
| 18:08.33 | Primer | awk_r: all I'm saying is that 1.4.20 doesn't do this, and 1.4.22 does |
| 18:08.39 | ACK_NAK | hardwire: give me dedicated devices. Compartmentalization. |
| 18:08.40 | Primer | so...it's still the phone? |
| 18:08.42 | hardwire | I found that info.. I just couldn't find resale for it |
| 18:08.43 | Primer | think not... |
| 18:09.08 | [TK]D-Fender | hardwire: When you look in the most obvious place and you don't even have to dig in the slightest, its right there in front of you... well... what do YOU thik? :p |
| 18:09.16 | hardwire | [TK]D-Fender: stfu? |
| 18:09.17 | beek | Primer: I'm trying to find it. If you have the configuration guide it's in there. |
| 18:09.25 | Primer | beek: what should I look for? |
| 18:09.29 | [TK]D-Fender | hardwire: Its not even buried under some obscure "upcoming products" page or anything. |
| 18:09.30 | hardwire | not feeling all that pick-on-able today. |
| 18:09.39 | Primer | beek: any sort of hint would be appreciated |
| 18:09.46 | hardwire | calls voipsupply.com |
| 18:09.47 | [TK]D-Fender | hardwire: You're entitled... |
| 18:09.56 | [TK]D-Fender | hardwire: we all have those days |
| 18:10.08 | beek | Primer: I did a google search on polycom callerid and found the param. I'm still looking through my configs to see what I changed. |
| 18:10.17 | [TK]D-Fender | hardwire: Unless you've got a good rep with them TD's pricing is pretty much always better |
| 18:10.53 | Primer | I'm dealing with this issue with my co-worker, and he tells me "This will work for people with polycom phones, but what about people with softphones? It destroys their call history, making it imposible to simply 'redial' a caller" |
| 18:11.14 | hardwire | TD? |
| 18:11.15 | Primer | I'm guessing that's not intended behavior |
| 18:11.16 | hardwire | techdata? |
| 18:11.19 | beek | Primer: Its in the <feature> </feature> section. I *think* that its: feature.9.name="url-dialing" feature.9.enabled="0 |
| 18:11.40 | beek | Primer: check to see if yours is enabled by default. |
| 18:11.45 | Primer | beek: checking |
| 18:11.50 | beek | Primer: This took care of the problem for me. |
| 18:12.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:12.22 | hardwire | [TK]D-Fender: techdata? |
| 18:12.36 | [TK]D-Fender | hardwire: www.telephonydepot.com |
| 18:13.03 | [TK]D-Fender | hardwire: Looking at VS's pricing, the IP 550 is $10 more than the 450. I can't see how the 450 can be worth it... |
| 18:13.35 | Primer | co-worker is telling me now that if we enable that feature, url dialing will stop working |
| 18:13.40 | Primer | and I use url dialing |
| 18:13.51 | Primer | beek: so I'm guessing that's not the right one |
| 18:14.03 | beek | Primer: Hmmm... |
| 18:14.08 | [TK]D-Fender | hardwire: at that price point there is no point to the 450 |
| 18:14.41 | [TK]D-Fender | hardwire: IP 550 has a bigger screen, higher res, more line-keys..... total win-win comparison so far |
| 18:14.46 | hardwire | yeh.. I didn't realize they sold them for just over the minimum resale price |
| 18:14.59 | beek | Primer: I may be wrong about that so just google your problem and you may find the same link that I did. I'll keep looking. |
| 18:15.24 | Primer | yeah, we're looking too |
| 18:15.35 | [TK]D-Fender | hardwire: Frankly the IP 550 is a mistake... bigger than a 50X, but below a 6XX and they are trying to squeez stuff under it and its juswt getting stupid. |
| 18:15.50 | [TK]D-Fender | hardwire: Poor product placement. |
| 18:16.09 | raasdnil | lmadsen: hey, thanks, called Digium, it half works now :) |
| 18:16.27 | [TK]D-Fender | raasdnil: call them again! |
| 18:16.35 | hardwire | [TK]D-Fender: didn't you just say the 550 was a great buy over the 450? |
| 18:16.36 | [TK]D-Fender | raasdnil: And it'll be 100% |
| 18:16.41 | raasdnil | now i just have to figure out how to take the calls arriving in from the NEC PABX on the 1st e1 span and forward them out the second e1 span |
| 18:16.45 | raasdnil | [TK]D-Fender: heh :) |
| 18:16.46 | hardwire | or was it a mistake for polycom to make them? because they undercut their new products? |
| 18:17.30 | [TK]D-Fender | hardwire: Yeah in a way... just that this point in the segment is too crowded. They shouldn't have come out with the 550. the 450 was the right spot between 3XX and 6XX, and should replace the 430 as well. |
| 18:17.53 | [TK]D-Fender | hardwire: Yes, theya re flooing their lineup now. |
| 18:18.01 | hardwire | I can't afford the 6xx's thats for sure |
| 18:18.15 | [TK]D-Fender | hardwire: if you can affor 550's, the 6XX are like a few bucks more :) |
| 18:18.27 | hardwire | [TK]D-Fender: where do you work? |
| 18:18.46 | hardwire | looks up the telephonedepot.com employee contact information page |
| 18:19.07 | [TK]D-Fender | hardwire: Non-tech company. |
| 18:19.31 | hardwire | my sales dude is trying to put together a package to sell companies that just need 5 x lines or so |
| 18:19.37 | [TK]D-Fender | hardwire: I jsut promote companies and products that I feel are worthwhile for others to benefit from. |
| 18:19.50 | hardwire | and $117 for IP501's is soo much easier to wrap into 3-years than $221 |
| 18:19.50 | Primer | beek: just not having much luck with google terms for this issue, I guess |
| 18:19.54 | Primer | still can't find anything |
| 18:19.56 | hardwire | he's gonna poop when I suggest the 550s |
| 18:20.01 | [TK]D-Fender | hardwire: thats a great price for 501's |
| 18:20.08 | hardwire | err |
| 18:20.09 | hardwire | 177 |
| 18:20.15 | hardwire | baka |
| 18:20.19 | [TK]D-Fender | hardwire: thats a LESS great price for 501's ;) |
| 18:20.19 | hardwire | "You save $117.00!!" |
| 18:20.47 | hardwire | TD doesn't sell the 450's dangit |
| 18:20.49 | hardwire | calls em |
| 18:20.58 | [TK]D-Fender | hardwire: 501 si a nicer phone, but IP 320/330 covers jsut about all basic use your typical user needs... |
| 18:21.17 | Un1x | [TK]D-Fender what else do i need to install asterisk, other then Asterisk 1.4.22 and zaptel, btw wich version of zaptel shall i install also? |
| 18:21.19 | hardwire | [TK]D-Fender: except people complain about the LCD having too little real information |
| 18:21.25 | hardwire | in the 320/330 |
| 18:21.28 | hardwire | I like the look of em tho |
| 18:21.40 | [TK]D-Fender | Un1x: You should get DAHDI as per the channel topic and *'s web-site |
| 18:21.56 | [TK]D-Fender | hardwire: OH, but the 320/330 are 2.5mm, not RJ9 |
| 18:22.02 | Un1x | also the new zaptel drivers will they support my card even tho my card is old its the TDM22B |
| 18:22.10 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 18:22.30 | [TK]D-Fender | Un1x: DAHDI replaced zaptel... its just a new name really... and split up a little |
| 18:22.40 | hardwire | [TK]D-Fender: indeed.. I've already prices out the nice headsets with the 320's in mind |
| 18:23.00 | hardwire | I just don't know where the 320/330 look/feel is going to go in the next year or so |
| 18:23.05 | beek | Primer: I just changed my setting an am rebooting my phone to confirm ... |
| 18:23.05 | Un1x | so i dont need to get zaptel?, because on asterisk.org zaptel is stil there |
| 18:23.13 | [TK]D-Fender | hardwire: I have NO idea how good headsets on a 320/330 can be... I've only experienced el-cheapo residential grade ones.... |
| 18:23.23 | Primer | beek: thanks, really appreciate it |
| 18:23.39 | Un1x | [TK]D-Fender do i also need to install LIBPRI ? |
| 18:23.46 | [TK]D-Fender | Un1x: That ver of * onwards supports DAHDI as well, and future versions will now support Zaptel at all going forward IIRC. |
| 18:23.57 | [TK]D-Fender | Un1x: If you're going to be doing PRI, yes |
| 18:24.22 | Un1x | no not going to be doing PRI so all i need is Asterisk 1.4 and Dahdi |
| 18:24.33 | [TK]D-Fender | hardwire: I use Plantronics M22 + H261 binaural polaris rigs here for my call center |
| 18:24.40 | [TK]D-Fender | Un1x: That should do |
| 18:24.44 | Un1x | okay |
| 18:25.14 | Primer | beek: yeah, it didn't work here |
| 18:26.05 | Un1x | [TK]D-Fender, does dahdi-linux-2.0.0.tar.gz support TDM22B, also quick question that is the right version for 1.4 correct? |
| 18:26.36 | beek | Primer: that was the setting. When I switch it to '1' I get the sip: callerid, otherwise i get just the phone number. |
| 18:26.47 | raasdnil | how do I forward calls from a PABX on one E1 span out through the PSTN on another E1 span? |
| 18:27.07 | raasdnil | anyone got a good newbie website for that? |
| 18:27.42 | raasdnil | I just can't quite figure out how to refer to the lines separately in the extensions.conf... both are DAHDI... |
| 18:27.43 | hardwire | wonders if an IP350 exists |
| 18:28.21 | ACK_NAK | DAHDI: "dmesg" gives me the following message "wctdm24xxp0: Host system set a cache size of 4 which is not supported. Disabling memory write line and memory read line.<6>Port 1: Installed -- AUTO FXO (FCC mode)" I don't see any reference to cache size in any of the /etc/dahdi files or chan_dahdi.conf. Is this something simple or should I talk to Digium support? |
| 18:28.51 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:29.09 | hardwire | 42 lines on a 650? |
| 18:29.13 | hardwire | the insanity is too much |
| 18:29.20 | Primer | beek: here, all that feature does is remove my ability to dial urls |
| 18:29.21 | hardwire | can hardly imagine port 5060->5070 |
| 18:29.23 | *** join/#asterisk ManxPower (n=manxpowe@128.sub-75-248-59.myvzw.com) |
| 18:30.26 | Un1x | [TK]D-Fender should i install dahdi first or Asterisk |
| 18:30.42 | ACK_NAK | dahdi |
| 18:30.48 | raasdnil | Un1x: libpri (if you need it) then dahdi, then asterisk |
| 18:30.59 | [TK]D-Fender | Un1x: DAHDI then *. |
| 18:31.09 | ACK_NAK | mommy, then DAHDI |
| 18:31.12 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 18:31.17 | Qwell | ACK_NAK: So original. |
| 18:31.20 | StephenF | if I call a toll free number with my caller id set as a toll free number is it normal to get a Sip 486 Busy signal? |
| 18:31.24 | ACK_NAK | couldn't help myself |
| 18:31.47 | Un1x | [TK]D-Fender, how do check if my card is detected i tried dmesg | grep wild card and | grep Wild nothing came up |
| 18:31.49 | ManxPower | StephenF: depends on the terminating switch, but usually not. |
| 18:31.53 | Un1x | Oh shit lol im stupid |
| 18:31.57 | Un1x | i forgot to plug in power |
| 18:31.58 | Un1x | HAHA |
| 18:32.05 | [TK]D-Fender | Un1x: 11 steps to go.... |
| 18:32.07 | StephenF | hmm, something else must be wrong then |
| 18:32.24 | ACK_NAK | # dahdi_cfg -vv |
| 18:32.48 | Un1x | [TK]D-Fender, i'm assuming i'll need dahdi tools |
| 18:32.51 | Un1x | heh |
| 18:34.04 | ACK_NAK | I meant to say lspci -n |grep d161 |
| 18:34.05 | [TK]D-Fender | Un1x: You might want to READ the docs you know... and the blog posts on their site, ets. |
| 18:34.32 | lmadsen | raasdnil: w00t! |
| 18:34.40 | raasdnil | lmadsen: :) |
| 18:34.41 | Un1x | where is the docs, other then reading the README in each package? |
| 18:34.47 | lmadsen | [TK]D-Fender: pfft.... that reeks of effort |
| 18:34.57 | ManxPower | Un1x: /path/to/src/asterisk/doc |
| 18:35.03 | ACK_NAK | anybody know what that cache size message is about? |
| 18:35.07 | ManxPower | read it, love it, hold it, give it flowers. |
| 18:35.12 | lmadsen | and chocolate! |
| 18:35.16 | raasdnil | lmadsen: now it is talking to the NEC system and the PSTN. Just got to figure out how to accept incomming calls on either span and route them to the other (act as a pass through gateway for now) |
| 18:35.17 | ManxPower | ACK_NAK: perhaps if you told us the ACTUAL message? |
| 18:35.19 | lmadsen | or better yet... give me the chocolate |
| 18:35.33 | ACK_NAK | That is the actual message. You want the whole thing in a pastebin? |
| 18:35.39 | ManxPower | any pates over about 2 or 3 lines should be put on pastebin or you could be automagically kickbanned |
| 18:36.19 | ManxPower | ACK_NAK: I strongly doubt Asterisk has a message that simply says "cache size" |
| 18:36.47 | ACK_NAK | "wctdm24xxp0: Host system set a cache size of 4 which is not supported. Disabling memory write line and memory read line.<6>Port 1: Installed -- AUTO FXO (FCC mode)" |
| 18:37.10 | ManxPower | There we go! That was not so hard! |
| 18:37.25 | ACK_NAK | I posted it a few lines back. |
| 18:37.26 | ACK_NAK | http://pastebin.com/m387b85da |
| 18:37.47 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
| 18:38.15 | raasdnil | ok all, how do I specify span 1 or span 2 on a te210p card? ie, how can I tell extensions.conf when receiving a call on a context to forward the call to the TE210 span 2 and when receiving on another context to forward to span 1? |
| 18:38.30 | ManxPower | ACK_NAK: You have both a 2400P and an 800P? |
| 18:38.49 | ACK_NAK | no, just an 800p which according to the doc uses the same driver |
| 18:39.18 | ManxPower | raasdnil: You don't. first span is channels 1-24, 2nd span is 25 - 48, etc use group= to set outgoing hunting tell your telco to set up incoming hunting. |
| 18:39.33 | ACK_NAK | I appreciate you taking the time to think about this |
| 18:39.40 | Madkiss | what is the codec that requires the least possible connection conditions? i.e. which codec to use for small connections? |
| 18:40.09 | ACK_NAK | Madkiss: lowest bandwidth or best concealment? |
| 18:40.12 | [TK]D-Fender | Madkiss: G.723 |
| 18:40.20 | Madkiss | ACK_NAK: the first |
| 18:40.40 | ACK_NAK | [TK]D-Fender: g.723 or ILBC? |
| 18:40.56 | [TK]D-Fender | ACK_NAK: Is there an echo in here? |
| 18:41.13 | kerx | where is the least cost routing documentation for AMI? |
| 18:41.14 | tzafrir_laptop | Madkiss, speex |
| 18:41.14 | ACK_NAK | [TK]D-Fender: that was a question for you, not a regurgitation :-) |
| 18:41.35 | [TK]D-Fender | ACK_NAK: You ask for confirmation on an answer I already gave. |
| 18:41.37 | tzafrir_laptop | tweak it a bit in codec.conf |
| 18:41.53 | ACK_NAK | no I'm asking you to render an opinion about ILBC |
| 18:42.07 | [TK]D-Fender | ACK_NAK: ILBC takes up more BW. |
| 18:42.07 | ACK_NAK | .[TK]D-Fender:, if you have one |
| 18:42.14 | raasdnil | ManxPower: OK, I already have the hunt group on and I have set span 1 to be group=0,11 and span 2 to be group=0,12. |
| 18:42.32 | awk_r | i think [TK]D-Fender was asking you to deduce his opinion of ILBC from his answer of g.723 |
| 18:42.35 | ACK_NAK | do you find ILBC its better at anything? for example concealment? |
| 18:42.45 | [TK]D-Fender | ACK_NAK: Now of course which is the best fit for codec in the lower end depends on what devices are involved, etc |
| 18:42.55 | raasdnil | ManxPower: so then I need to just in the incomming-pstn something like exten => MYPHONENUMBER,1,Dial(Group ... something? |
| 18:42.56 | [TK]D-Fender | ACK_NAK: Concealing what? |
| 18:43.23 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 18:43.58 | Un1x | line 0: Unable to open master device '/dev/dahdi/ctl' |
| 18:43.58 | Un1x | 1 error(s) detected |
| 18:44.08 | Un1x | umm in dmesg im not seeing my card |
| 18:44.08 | Un1x | :S |
| 18:44.13 | ACK_NAK | [TK]D-Fender: packet loss. I understand that g.726 (for example) is better at concealing psychoaccoustic artifacts created by packet loss than other codecs utilizing similar bandwidth |
| 18:44.33 | kerx | how to do Least Cost Routing w/ asterisk AMI? |
| 18:45.15 | ACK_NAK | [TK]D-Fender:...therefore my question was whether ILBC has any virtues that you know of in the concealment department. I'm completely ignorant on the subject of ILBC |
| 18:45.28 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 18:47.15 | Un1x | [TK]D-Fender can you help me detect my card please? iinstalled dahdi linux and dahdi tools |
| 18:48.17 | jameswf | dang it takes forever to zero a disk |
| 18:48.30 | ACK_NAK | Anybody have an option about that dmesg cache size complaing from a bit earlier? Should I take it up with Digium? http://pastebin.com/m387b85da |
| 18:48.41 | Corydon76-dig | jameswf: why zero it? Just toss it in an industrial grinder |
| 18:49.00 | awk_r | kerx, why AMI? I usually see LCR done with func_odbc and database lookups. |
| 18:49.04 | Un1x | Corydon76-dig could you help me please? |
| 18:49.26 | Madkiss | what could be the reason for throttled sound over an IAX2 connection? |
| 18:49.38 | ACK_NAK | Corydon76-dig: disk drives are SO expensive these days! |
| 18:49.47 | ACK_NAK | Corydon76-dig: not |
| 18:50.11 | kerx | I thought it would be w/ AMI, cause when I originate a call I have to give to it the SIP channel to use |
| 18:50.23 | kerx | How can LCR change the SIP channel based on the least cost route ? |
| 18:50.46 | jameswf | write all zeros then read all zeros for integrity check |
| 18:50.48 | kerx | Or am I setting up a LOCAL channel that Least Cost Routing is configured to answer and then forward to a SIP trunk based on the least cost route from the database ? |
| 18:50.50 | Corydon76-dig | kerx: I've done it with EVAL() |
| 18:51.03 | Un1x | root@darkstar:~# dmesg |grep -i tdm |
| 18:51.03 | Un1x | root@darkstar:~# |
| 18:51.10 | Un1x | and i did plug in the power too :( |
| 18:51.10 | jameswf | no smart support totsl arse |
| 18:51.19 | Corydon76-dig | grab the destination out of the database |
| 18:51.33 | kerx | Corydon76-dig, May I see an example of your context in the extensions on a pastebin like site? |
| 18:52.11 | Corydon76-dig | kerx: basically, it's something on the order of Dial(${EVAL(chan_from_db)}) |
| 18:52.20 | ACK_NAK | You guys seem to know stuff: Anybody know where I left my keys? |
| 18:52.30 | kerx | And how do you setup the chan_from_db variable? |
| 18:52.31 | Madkiss | look in the toilet |
| 18:52.35 | Corydon76-dig | kerx: since switching jobs, I don't have ready access to that dialplan anymore |
| 18:52.36 | awk_r | ACK_NAK, dresser drawer |
| 18:52.44 | kerx | Ok, I understand, thanks :) |
| 18:52.52 | Corydon76-dig | kerx: use func_odbc and the SORT() function |
| 18:53.13 | ACK_NAK | Oh, here they are, sitting on this operating PCB |
| 18:53.28 | kerx | I thought there was a builtin module that does all this work for you ? |
| 18:53.40 | kerx | I need LCR w/ heavy load :-( I'm not sure if this is the most effective method? |
| 18:53.41 | Un1x | Someone want to please help me to find out why my card is not detected?, like i put in the power thing trying to check on dmesg no luck |
| 18:53.42 | Corydon76-dig | All this work? |
| 18:54.12 | kerx | You have to read from the database yourself using func_odbc then sort |
| 18:54.14 | ACK_NAK | Un1x: how old is your machine. I'm guesing you don't have a PCI 2.2 compliant slot. |
| 18:54.18 | Corydon76-dig | kerx: the dialplan is the most efficient way of doing it. There isn't an end-all be-all module for that, but you can build one easily enough |
| 18:54.32 | kerx | Yes, I would imagine a C module much more efficient |
| 18:54.35 | Corydon76-dig | kerx: I just gave you the 3 hardest pieces |
| 18:54.36 | ACK_NAK | Un1x: I had the same problem. I had to use a new MOBoard |
| 18:54.39 | kerx | Using the dialplan might equate a lot of load |
| 18:54.50 | Un1x | ACK_NAK, how can i check if my board is compatible or not? |
| 18:54.51 | Corydon76-dig | kerx: it does NOT |
| 18:55.17 | kerx | k |
| 18:55.21 | Corydon76-dig | kerx: The dialplan is simply a way to take several building blocks and string them together |
| 18:55.25 | awk_r | pbx_lua would be good |
| 18:55.38 | Corydon76-dig | C isn't really any more efficient |
| 18:56.16 | Corydon76-dig | Or really, the boost it gives you is miniscule, compared to the trouble you might have debugging it |
| 18:56.17 | ACK_NAK | Un1x: Check the specs of the board online. You'll find what I just told you in the 'owners manual' of the Digium card itself. Yesterday I was completely ignorant on the subject. |
| 18:56.29 | Un1x | ACK_NAK lspci shows the card visible |
| 18:56.46 | kerx | I'm going to need to learn the Dialplan more and more |
| 18:56.55 | kerx | The more I look at it, the more I realize I don't know it |
| 18:57.01 | ACK_NAK | Un1x: I understand that's the right test for PCI compliance |
| 18:57.09 | Corydon76-dig | kerx: have you gotten the book yet? |
| 18:57.09 | Un1x | 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 18:57.14 | awk_r | ~book |
| 18:57.14 | 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 |
| 18:57.18 | kerx | Yea |
| 18:57.23 | kerx | for some reason the book's not enough :P |
| 18:57.23 | Akiyuki | Anyone have a Nortel MGCP phone working with asterisk? |
| 18:57.26 | kerx | I need some toy hardware |
| 18:57.29 | kerx | and some toy projects |
| 18:57.30 | kerx | to play w/ |
| 18:57.46 | Un1x | I own the Damn Book, in paper print like i bought it from the book store, im trying to figure out my pone |
| 18:57.58 | awk_r | s/ hardware(?<=toy)/s/ |
| 18:58.15 | Un1x | ACK_NAK any reason why lspci shows the card there but not dmesg? |
| 18:58.22 | Madkiss | is 320ms too much average latency to to use iax2 over that connection? |
| 18:58.44 | ACK_NAK | Un1x: I'm no expert, but do you see DAHDI starting when you boot? |
| 18:59.40 | Akiyuki | Does * support MGCP phones? |
| 18:59.41 | awk_r | Un1x, Digium TS is helpful :-) |
| 19:00.57 | Un1x | ack_nack can i check using dmesg? |
| 19:01.06 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:03.53 | Un1x | well i did /etc/init.d/dahdi start |
| 19:03.59 | Un1x | and everything started sucessfully |
| 19:08.48 | Un1x | and its finaly showing up in dmesg like dahdi the card etc :D |
| 19:10.24 | Spirits-Sight | Is there a service for toll free that can point to a local number, that won't either be a per min service or require a large depost? |
| 19:11.31 | Spirits-Sight | I have picked my outbound services and I think I have chosen my inbound for local numbers |
| 19:13.18 | *** join/#asterisk jetsnoc (n=brian@pdpc/supporter/active/jets) |
| 19:15.57 | [TK]D-Fender | Spirits-Sight: Just about any TF provider can point to a DID of your choosing |
| 19:16.27 | Spirits-Sight | Can you point me to a good one please. |
| 19:16.32 | Spirits-Sight | or two |
| 19:17.23 | Spirits-Sight | What do you think of Pulsevoice for incoming? |
| 19:17.24 | Spirits-Sight | local |
| 19:17.24 | [TK]D-Fender | Spirits-Sight: just shop around yourself, and they are ALWAYS per min or per channel, and the per channel is priced at a certain ratio. |
| 19:17.24 | Akiyuki | Can someone help set up an mgcp device? |
| 19:18.43 | Akiyuki | http://pastebin.ca/1255442 is the mgcp.conf and extensions.conf of what i am trying to do |
| 19:19.15 | Spirits-Sight | I got my incoming for local service, I am looking for a company to handle the 800 as the one that I am using for local DID is way to much to setup the 800 number |
| 19:22.57 | *** part/#asterisk mark_csi (n=csiadmin@host81-149-93-47.in-addr.btopenworld.com) |
| 19:24.06 | *** join/#asterisk emiller (n=ed@host-69-95-90-173.har.choiceone.net) |
| 19:25.15 | emiller | i just switched my card from a 400p to an 800p with echo cancellation modules (not HPEC), what is the syntax to enable echo cancel in chan_dahdi |
| 19:25.31 | [TK]D-Fender | aiksa[LV]: [192.138.1.84] <- dyslexic |
| 19:26.04 | [TK]D-Fender | emiller: "echocancel=yes" |
| 19:26.34 | emiller | easy enough. thanks for always helping [TK]D-Fender. |
| 19:27.23 | Akiyuki | What are the different line types for mgcp? i see things like d001, alns.. etc |
| 19:28.11 | jameswf | Qwell: apparently yum update update your kernel.... |
| 19:29.00 | Spirits-Sight | [TK]D-Fender: all the ones I am finding seem to have voice mail and what not, I just need one that point to a Local number and has a good rate (what is a good rate in the USA for TF number?) |
| 19:29.37 | Un1x | [TK]D-Fender where is the zapata.conf / zaptel.conf in asterisk 1.4 with dahdi is it renamed too? |
| 19:30.07 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 19:30.28 | Qwell | jameswf: yes |
| 19:30.49 | Akiyuki | What port(s) does asterisk need for inbound calls ? just 5090? |
| 19:30.50 | Un1x | Qwell, what are the files renamed to for dahdi? |
| 19:30.50 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 19:30.55 | Qwell | Un1x: "the files"? |
| 19:30.59 | krondorl | allo allo.. been a while... |
| 19:31.08 | Un1x | Qwell zapata.conf zaptel.conf |
| 19:31.20 | *** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
| 19:31.22 | Qwell | Un1x: You need to read the upgrade file.. |
| 19:31.41 | Qwell | Un1x: UPGRADE.txt |
| 19:31.55 | *** part/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
| 19:32.34 | Katty | Qwell: 2 barst ill 71! |
| 19:32.42 | Qwell | boo |
| 19:32.45 | Qwell | so far.. |
| 19:33.22 | Katty | how far did you get your DK? |
| 19:33.27 | *** join/#asterisk oej (n=olle@58.Red-80-38-98.staticIP.rima-tde.net) |
| 19:34.03 | *** join/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212) |
| 19:34.23 | Un1x | Qwell it doesn't tell you what zapata.conf has been moved to or renamed to |
| 19:34.52 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 19:34.58 | *** join/#asterisk galeras (n=galeras@190.159.51.107) |
| 19:35.28 | Un1x | someone just wanna tell me lol |
| 19:35.48 | krondorl | Anyone know if call parking can have different on hold music? |
| 19:36.05 | galeras | Hello. Which gateway is a good option to be used as h323 endpoint? |
| 19:37.14 | hardwire | you might want to check if "hell" ships to your country.. then I'll suggest a few. |
| 19:38.21 | Ariel_Calzada | hi i have a problem i have installed vmware version of RAGUI but the apache displays Application errorRails application failed to start properly any idea???? |
| 19:38.41 | Un1x | Qwell, I'm not getting a Dial tone on the phone i replaced all the configs... from my old installation to new installation |
| 19:39.44 | jameswf | no dahdi files for -18 |
| 19:39.47 | galeras | sorry, what i looking for is an T38 endpoint gateway. Could you suggestme any? |
| 19:40.32 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 19:40.46 | Un1x | hey sorry aboiut that can anyone tell me, the soloution tomy problem? |
| 19:40.50 | Un1x | im getting no dial tone |
| 19:40.51 | [8none1] | galeras: I've been told the gateways from AudioCodes and Dialogic are the only ones to have for t.38 |
| 19:41.09 | kaldemar | Un1x: chan_dahdi.conf |
| 19:41.40 | ACK_NAK | FYI: Spoke with support, and the message I asked about earlier (http://pastebin.com/m387b85da) can be safely ignored, and will not show in the next version of dahdi. |
| 19:41.50 | Un1x | kaldemar i did set it up want me to paste it on pastebin.ca |
| 19:42.03 | kaldemar | no. |
| 19:42.11 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-217-192.phlapa.east.verizon.net) |
| 19:43.01 | ACK_NAK | ...also support helped me to determine that my keys were in fact... under the sofa. Yes. The sofa. The answer was the sofa. |
| 19:43.18 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 19:43.24 | Spirits-Sight | Is .035 cent min for out state (incoming from 47 other states) and 4.9 cent from instate, is this a good deal or not with a 25 dollar porting fee? |
| 19:43.44 | Un1x | kaldemar, here is my chan_dahdi.conf http://pastebin.ca/1255468 |
| 19:43.52 | *** join/#asterisk bakermd (n=bakermd@38.104.0.102) |
| 19:44.11 | Spirits-Sight | 1.99 fee a month to host the 800 number |
| 19:44.16 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 19:44.25 | bakermd | My asterisk box registers to a provider. How do I tell it that this is to be a 1 hr registration? It is doing 120 sec right now |
| 19:44.45 | bkruse | bakermd: I am not sure why you would want that long of a delay |
| 19:44.53 | bkruse | But what is 1 hr in seconds? :/ |
| 19:45.00 | *** join/#asterisk seaq (n=seaq@60.198.60.190.host.ifxnetworks.com) |
| 19:45.17 | bkruse | bakermd: That is only to make sure that the peer is still available. If you set it to 1 hour, you wouldn't know it's down for an hour! |
| 19:47.35 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 19:47.41 | bakermd | Well, maybe I am chasing the wrong thing. I can register and receive calls. All of a sudden, without warning (maybe after an hour, maybe after five minutes) the inbound calls start failing. An ethereal trace shows that one of two things happens - either the asterisk box rejects the call as unauthorized, or it tries to re-register with the provider and the provider says Unauthorized (but Immediately doing sip show registry shows the connection as registe |
| 19:48.38 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 19:48.39 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:49.00 | krondorl | Ariel_Calzada: That does not sound like it has anything to do with Asterisk PBX |
| 19:50.38 | Ariel_Calzada | krondorl: where to find help? |
| 19:51.33 | krondorl | bakermd: That sounds like a communications failure tween you and provider. We've had that issue once and we talked with provider and they were able to trace it to something on their side.. Not sure what.. they didn't say. |
| 19:51.54 | krondorl | Ariel_Calzada: Donno. #apache maybe? |
| 19:52.02 | Un1x | here is my chan_dahdi.conf http://pastebin.ca/1255468 |
| 19:52.08 | Un1x | why am i not getting a tone? |
| 19:52.25 | harry_v | dial tone? |
| 19:52.57 | krondorl | Anyone know if call parking can have different on hold music depending on conditions? |
| 19:53.11 | bakermd | krondorl: The "provider" is down the hall, connected via Gig-E ;) FreeSwitch (from the ppl @ FreeSwitch - they built it for us to handle 3M subscribers) |
| 19:53.38 | krondorl | wow lucky you.. lol.. |
| 19:54.38 | Un1x | harry_v yes |
| 19:54.49 | Un1x | also, doesn't asterisk 1.4 support sip? |
| 19:54.58 | Un1x | im trying sip peers in cli and its not working either |
| 19:55.26 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:55.42 | bakermd | Un1x: sip show peers |
| 19:55.48 | *** join/#asterisk gabegundy (n=gabe@57.251.sfcn.org) |
| 19:55.51 | Un1x | bakermd doesn't work on 1.4 |
| 19:56.01 | Un1x | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
| 19:56.01 | Un1x | *CLI> |
| 19:56.08 | bakermd | Un1x: core sip show peers ? |
| 19:56.21 | Un1x | nopes |
| 19:56.22 | Un1x | same thing |
| 19:56.36 | l2trace99 | show modules |
| 19:56.52 | bakermd | Un1x: sip show peers works for me and I am on 1.4 |
| 19:56.55 | l2trace99 | sry module show |
| 19:56.59 | Un1x | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 19:57.00 | l2trace99 | me too |
| 19:57.15 | l2trace99 | i only get that when the sip module isn't loaded |
| 19:57.23 | gabegundy | Anyone know about the yum repos? I see lots of packages in there, but no asterisk server. |
| 19:57.30 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 19:57.37 | Un1x | how can i fix it? |
| 19:57.37 | bakermd | Agreed - if you cannot do sip show peers, you do not have sip loaded |
| 19:58.20 | Un1x | Well, can you please tell me how to load it and also why im not getting a call tone is there other modules i also have to load? |
| 19:59.03 | l2trace99 | chan_sip |
| 20:00.11 | [8none1] | Un1x: you can load by editing /etc/asterisk/modules.conf and add load => chan_sip.so |
| 20:00.13 | Un1x | yes i did module load chan_sip |
| 20:00.48 | [8none1] | Un1x: or turn on autoload |
| 20:01.02 | Un1x | how, can i turn on autoload? |
| 20:01.19 | Un1x | oh autoload is et as on in modules.conf |
| 20:01.24 | Un1x | but the problem comes from tones? |
| 20:02.04 | *** join/#asterisk WimpMan (n=wimpy@213.240.181.251) |
| 20:02.31 | Deeewayne | Un1x: compare the output of dahdi_scan with what you have configured |
| 20:03.41 | Un1x | Deeewayne, here it is http://pastebin.ca/1255484 |
| 20:03.54 | *** part/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212) |
| 20:05.00 | bakermd | So, no one knows why (out of the blue) Asterisk would not be able to register but tell you that it is registered? |
| 20:06.33 | Un1x | bakermd, im just not understanding i restarting asterisk after the module thing and it still now showing peers |
| 20:06.40 | Un1x | and no tone these are my old configurations that worked before |
| 20:07.18 | bakermd | Un1x: Configuration files from what version? A lot changes from one major release to another |
| 20:07.51 | Un1x | 1.2 i think bakermd |
| 20:08.21 | bakermd | Un1x: Thats 2 major versions - you have a Lot of release-note reading to do.. |
| 20:09.12 | [TK]D-Fender | 1 major version... |
| 20:09.36 | Deeewayne | Un1x: dahdi_cfg -vv |
| 20:09.52 | Un1x | 0 channels to configure. |
| 20:09.57 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 20:10.56 | Deeewayne | Un1x: are you trying to switch from zaptel -> dahdi without reading UPGRADE.txt like Qwell suggested earlier ? |
| 20:11.13 | Un1x | Deeewayne i read UPGRADE.txt |
| 20:11.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:11.32 | [8none1] | bakermd: what are you talking about nothing changed in config files between 1.2 and 1.4 ;) |
| 20:11.44 | harry_v | seems that asterisknow is not the path to go if anyone here is running it. mine is borked to a degree. bugs perhaps. |
| 20:11.55 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 20:12.24 | Deeewayne | Un1x: you configured /etc/dahdi/system.conf ? |
| 20:12.51 | *** join/#asterisk ManxPower (n=manxpowe@161.sub-75-200-31.myvzw.com) |
| 20:12.53 | Micc | I'm going to have to upgrade my 1.2 to 1.4 or 1.6 soon. I'm not looking forward to it. |
| 20:12.56 | Un1x | yes put all the same information from zaptel.conf |
| 20:13.51 | krondorl | Anyone know if call parking can have different on hold music depending on conditions? |
| 20:13.52 | *** join/#asterisk luke-jr_ (n=luke-jr@wsip-98-172-136-249.om.om.cox.net) |
| 20:14.40 | Deeewayne | Un1x: pastebin your /etc/dahdi/system.conf |
| 20:15.42 | Un1x | Deeewayne. http://pastebin.ca/1255496 |
| 20:16.43 | *** join/#asterisk zydoon (n=zydoon@41.225.159.36) |
| 20:16.53 | *** part/#asterisk zydoon (n=zydoon@41.225.159.36) |
| 20:17.01 | *** join/#asterisk cuco (n=elcuco@bzq-79-178-115-131.red.bezeqint.net) |
| 20:17.15 | harry_v | beining to wonder if asterisknow was released to prematurly. To many features that do not work including the important ones. |
| 20:17.18 | cuco | tzafrir_laptop: ping |
| 20:17.28 | tzafrir_laptop | cuco, pong |
| 20:17.32 | Un1x | Deeewayne? |
| 20:17.43 | cuco | tzafrir_laptop: do you know NX ? |
| 20:17.56 | tzafrir_laptop | I played with it a bit. Not much |
| 20:18.05 | cuco | ok, listen to this: |
| 20:18.09 | l2trace99 | anyone else having delay when using whisper |
| 20:18.10 | l2trace99 | ? |
| 20:18.40 | Deeewayne | don't think it matters, but can you remove the spacing from your loadzone and defaultzone settings, then run dahdi_cfg -vv |
| 20:19.01 | [TK]D-Fender | Un1x: "cat /proc/interrupts" |
| 20:19.06 | cuco | I am connected to KDE4 over NX and normal ssh. On KDE4/NX I run konsole. my local konsole3 and remote konsole4 are connected to the same "screen -x". are you following? |
| 20:19.21 | Katty | Qwell: 1 bar from 71! |
| 20:19.27 | Un1x | Deeewayne did that same thing |
| 20:19.42 | Un1x | [TK]D-Fender what do you want from that output? |
| 20:19.55 | gabegundy | BTW, the package name is asterisk14-core that is missing from the YUM repos. |
| 20:19.57 | [TK]D-Fender | Un1x: ALL of it |
| 20:20.17 | Un1x | [TK]D-Fender, http://pastebin.ca/1255499 |
| 20:20.42 | cuco | tzafrir_laptop: now on the graphical konsole4/NX I type "ls -l" on a dir with 70 files. WHY THE HELL AM I SEING THE FILE LIST *FIRST* ON NX AND THEN ON SSH!??!? |
| 20:20.46 | Katty | Qwell: DING |
| 20:20.49 | [TK]D-Fender | Un1x: re-run "dadhi_cfg -vvvv" |
| 20:21.17 | Deeewayne | Un1x: your dahdi_scan shows your alarms=unconfigured, which I see before I run dahdi_cfg |
| 20:21.26 | Un1x | http://pastebin.ca/1255500 |
| 20:21.50 | Un1x | Deeewayne, so alarms has to be configured? |
| 20:21.58 | Un1x | [TK]D-Fender, http://pastebin.ca/1255500 |
| 20:22.30 | Deeewayne | no, after dahdi_cfg is successful it would say alarms=OK |
| 20:22.58 | cuco | tzafrir_laptop: i heard of such things before.... but I am seeing this for the first time myself... it has to do with buffering, as NX sends the whole X "diffs" in a single compressed tcp packet (or very few) while ssh sends a separate packete for every key stroke. |
| 20:23.06 | Deeewayne | Un1x: ls /etc/dahdi |
| 20:23.23 | cuco | tzafrir_laptop: you see? it's faster to use screen on X11 then connecting directly ! |
| 20:23.35 | Un1x | init.conf modules system.conf systems.conf |
| 20:23.39 | Un1x | Deeewayne, init.conf modules system.conf systems.conf |
| 20:23.46 | Deeewayne | mv systems.conf system.conf |
| 20:23.52 | Deeewayne | that's your problem ^' |
| 20:24.16 | Deeewayne | mv systems.conf system.conf; dahdi_cfg -vv |
| 20:24.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:24.31 | tzafrir_laptop | for each keystroke (mostly) - due to the delay between them |
| 20:25.02 | Un1x | Deeewayne, still no dialtone or sip |
| 20:25.05 | cuco | tzafrir_laptop: key stroke, means for every char written to the terminal |
| 20:25.12 | [8none1] | krondorl: Not sure if it honors it but you can probably call SetMusicOnHold() to set the MOH class before parking the call. |
| 20:25.23 | *** part/#asterisk bakermd (n=bakermd@38.104.0.102) |
| 20:25.23 | Deeewayne | is dahdi_cfg -vv happier ? |
| 20:25.46 | cuco | is noticing the asterisk mambo-jambo did not change a lot in the last year and a half... only a new shinier version |
| 20:26.28 | tzafrir_laptop | yeah. dahdi and all |
| 20:26.30 | krondorl | [8none1]: I had thought the same, but it sdoesn't seem to work that way.. unless my testing was messed up.. I ended up with the same ohm every time... |
| 20:26.36 | Deeewayne | Un1x: O.O |
| 20:26.39 | Un1x | Deeewayne, http://pastebin.ca/1255505 |
| 20:26.48 | tzafrir_laptop | Un1x, what is the output of dahdi_test -v |
| 20:27.19 | tzafrir_laptop | if you run it for a while |
| 20:27.23 | Un1x | okay |
| 20:27.42 | harry_v | tzafrir_laptop who here works at digium? |
| 20:27.44 | cuco | dahdi... i almost confused it with dundy.... is this dahci_cfg related to what you did back then? (ztcfg?, zap_auto?) |
| 20:27.45 | Deeewayne | so now run asterisk or reload chan_dahdi.so and do "dahdi show status" |
| 20:28.10 | tzafrir_laptop | dahdi_cfg is the new name of ztcfg, yes |
| 20:28.38 | cuco | based on the same bash code? |
| 20:28.56 | *** join/#asterisk ManxPower (n=manxpowe@248.sub-70-220-190.myvzw.com) |
| 20:28.56 | Un1x | Deeewayne i got some errorrs |
| 20:29.09 | Deeewayne | pastebin |
| 20:29.12 | cuco | wait... yousaid ztcfg... not the zap autodetection script... |
| 20:29.14 | Un1x | Deeewayne, http://pastebin.ca/1255506 |
| 20:29.50 | Un1x | Deeewayne, dahdi show status output http://pastebin.ca/1255508 |
| 20:29.57 | ManxPower | Un1x: I see no errors in that pastebin |
| 20:30.07 | ManxPower | red alarms are not errors, they are telling you "no line connected" |
| 20:30.26 | Un1x | Ah :) |
| 20:31.05 | Un1x | ok i get dial tone now, sorry last problem |
| 20:31.06 | Un1x | http://pastebin.ca/1255510 |
| 20:31.25 | Un1x | when i do module load chan_sip it wont work i get errorrs but i copied the sip.con from BKW..... |
| 20:31.55 | *** part/#asterisk cuco (n=elcuco@bzq-79-178-115-131.red.bezeqint.net) |
| 20:32.13 | [8none1] | krondorl: I found this : http://osdir.com/ml/telephony.pbx.asterisk.devel/2004-03/msg00006.html |
| 20:32.37 | [8none1] | krondorl: I wonder if it never made it into the code. I've have to extract the source to see what it's doing. |
| 20:32.51 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 20:32.55 | iCEBrkr | http://www.submityoursip.com/wiki/Main_Page |
| 20:34.36 | Deeewayne | Un1x: is chan_sip.so in "module show" output ? |
| 20:34.39 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 20:35.03 | Un1x | one sec |
| 20:35.57 | Un1x | Deeewayne, yes |
| 20:36.56 | Un1x | Deeewayne, this is if i try to make a Call... http://pastebin.ca/1255510 |
| 20:38.15 | ManxPower | Un1x: you have /etc/asterisk/modules.conf screwed up |
| 20:38.45 | ManxPower | Un1x: That was not what Deeewayne asked for. |
| 20:38.52 | ManxPower | he asked for the module show output. |
| 20:38.59 | Un1x | ManxPower, http://pastebin.ca/1255517 |
| 20:39.34 | Un1x | Deeewayne, it shows 0 |
| 20:40.01 | ManxPower | Un1x: try "module load chan_sip.so" |
| 20:40.29 | Un1x | No category context for line 2 of /etc/asterisk/sip.conf |
| 20:40.30 | ManxPower | Un1x: If "module show" lists 0 modules then you're screwed. |
| 20:40.37 | ManxPower | Un1x: There you go |
| 20:40.44 | ManxPower | fix your sip.conf |
| 20:40.50 | Deeewayne | Un1x: I gotta go. good luck |
| 20:40.50 | *** part/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 20:40.56 | [TK]D-Fender | ManxPower: But he got his sip.conf from BKW, it *must* be good! |
| 20:41.03 | Un1x | its not loading because of an errorr in my sip.conf but the problem is its how bkw wanted the sip.conf |
| 20:41.21 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 20:41.23 | Un1x | [TK]D-Fender i didn't assume it must be good its just he said it was fine the way it was, since i got service from asterlink... :( |
| 20:41.40 | [TK]D-Fender | Un1x: And the reason we're not seeing that file now is? |
| 20:41.43 | *** join/#asterisk awk_r (n=rawk@nat/digium/x-98c63817151b7a10) |
| 20:42.22 | [TK]D-Fender | wonders if there is a prize for hitting 20 pastebins |
| 20:42.31 | Akiyuki | I have a device listening on port 5060, is it possible to have asterisk use 5070 for incoming? IE, my trunk provider would reply back on 5070? |
| 20:42.46 | Un1x | [TK]D-Fender http://pastebin.ca/1255520 sip.conf |
| 20:42.53 | [TK]D-Fender | Akiyuki: What device is talking over 5060? |
| 20:43.12 | [TK]D-Fender | Un1x: that's better not be everything... |
| 20:43.26 | Akiyuki | Another router on our network that is a SIP proxy |
| 20:43.26 | ManxPower | Un1x: There are 57.4 billion sample sip.confs out there. Stary by looking at sip.conf.sample |
| 20:43.35 | Akiyuki | We are trying to setup a test-devel server in the same network |
| 20:43.53 | [TK]D-Fender | Akiyuki: then yes you can have * bind to another port |
| 20:43.56 | Un1x | [TK]D-Fender its not |
| 20:44.02 | Un1x | sorry pasted half accidenly |
| 20:44.08 | [TK]D-Fender | Un1x: Please pastebin more, masking only passwords |
| 20:44.54 | Akiyuki | [TK]D-Fender: How will our trunk provider know that we are bound to a different port? |
| 20:44.54 | c4t3l | jeez! today is taking forever! |
| 20:45.05 | oej | ManxPower! Hello! |
| 20:45.09 | [TK]D-Fender | Akiyuki: the port * tells them |
| 20:45.19 | Un1x | [TK]D-Fender, http://pastebin.ca/1255522 |
| 20:45.35 | [TK]D-Fender | c4t3l: 1s/s like usual. |
| 20:45.46 | *** join/#asterisk bcrochet (n=bcrochet@cpe-069-132-204-022.carolina.res.rr.com) |
| 20:46.11 | Akiyuki | [TK]D-Fender: Sweet , thanks dawg |
| 20:46.34 | [TK]D-Fender | Un1x>No category context for line 2 of /etc/asterisk/sip.conf |
| 20:46.36 | [TK]D-Fender | hrm |
| 20:47.49 | Un1x | [TK]D-Fender i fixed it it was a blank space :( |
| 20:47.53 | Un1x | just one problem this [Nov 13 16:51:26] WARNING[20997]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 20:48.37 | Akiyuki | chan_sip.c:1787 __sip_xmit: sip_xmit of 0x996f7c0 (len 423) to 194.221.62.198:5060 returned -2: Bad file descriptor |
| 20:48.46 | Akiyuki | That is the error that we got after changing our side to use 5061 |
| 20:49.05 | [TK]D-Fender | Akiyuki: And you changed your forwarding to match? |
| 20:49.10 | Akiyuki | Yeah |
| 20:49.23 | Akiyuki | tcp;WAN_IP/WAN_SUBNET-5061>192.168.1.109-5061 |
| 20:49.24 | Akiyuki | udp;WAN_IP/WAN_SUBNET-5061>192.168.1.109-5061 |
| 20:49.27 | Un1x | [TK]D-Fender can you help me with the echo cancelation thing? |
| 20:50.14 | [TK]D-Fender | Akiyuki: pastebin SIP debug of the actual communication |
| 20:50.49 | krondorl | [8none1]: Got your message, (Sorry had to step out for a sec to talk to boss) I looked at that code, but we do not want to start playing with code if possible.. I don't see any answer so am unsure if it works. |
| 20:50.53 | bcrochet | My TDM400 has stopped functioning. Can anyone help me troubleshoot? I've done everything I know. |
| 20:51.45 | Akiyuki | http://pastebin.ca/1255528 |
| 20:52.36 | Un1x | hey has the set caller id command/fuynction changed? |
| 20:52.40 | [TK]D-Fender | Akiyuki: Your attempt to narrow the IP is clearly not helping. |
| 20:53.11 | [TK]D-Fender | Un1x: "upgrade.txt" "readme.txt" , "channelvariables.txt" |
| 20:53.19 | Un1x | ok |
| 20:54.08 | Akiyuki | [TK]D-Fender: What do you mean? I dont understand. |
| 20:55.04 | [TK]D-Fender | Akiyuki: Means you tried to debug a single IP and we aren't getting the full messages because they aren't coming in on that IP |
| 20:55.17 | Akiyuki | oh, i just did it by hostname |
| 20:56.07 | harry_v | why in the world does vm say I have one new message and then i press the number option to listen to it that it does not play it and the vm continues on with its options? I thought this bug was removed long ago. |
| 20:56.12 | Akiyuki | What can I do to do that? [TK]D-Fender |
| 20:56.25 | [TK]D-Fender | Akiyuki: open sip debug for EVERYTHING |
| 20:57.05 | Akiyuki | I did "set sip debug" |
| 20:57.09 | *** join/#asterisk drzed (n=drzed@synflood.homelinux.org) |
| 20:57.14 | *** join/#asterisk cvnet (n=dahitler@74.210.103.241) |
| 20:57.15 | drzed | hi theree |
| 20:57.25 | Akiyuki | All I get are these errors now, [Nov 13 15:57:03] WARNING[27593]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0x996f7c0 (len 423) to 194.221.62.198:5060 returned -2: Bad file descriptor |
| 20:57.30 | Akiyuki | Really destroying SIP dialog '2eaf87195498baf770aebb0d5692c124@127.0.0.1' Method: REGISTER |
| 20:57.50 | drzed | how do i tell (in extensions.conf) asterisk to add a prefix to a dialed number |
| 20:58.04 | cvnet | hello all, is there a way to find out pattern of payphones? my toll free DID charges me 75 cents if paypal |
| 20:58.10 | [TK]D-Fender | drzed: just put it in your dial |
| 20:58.23 | Akiyuki | [TK]D-Fender: Ah, found it http://pastebin.ca/1255534 |
| 20:59.03 | [TK]D-Fender | Akiyuki: Contact: <sip:SIP@192.168.1.109:5061> <-- you have clearly not set your system up to work properly behind NAT |
| 20:59.06 | [TK]D-Fender | Akiyuki: read up : |
| 20:59.08 | [TK]D-Fender | ~sipnat |
| 20:59.08 | 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 |
| 20:59.21 | Akiyuki | I have nat=yes |
| 20:59.26 | [TK]D-Fender | Akiyuki: iNOT ENOUGH |
| 20:59.30 | [TK]D-Fender | Akiyuki: Go read the guide |
| 20:59.38 | harry_v | dont forget rtp |
| 20:59.40 | [TK]D-Fender | Akiyuki: -- Registration for 'jimisanchez@sip.sipdiscount.com' timed out, trying again (Attempt #29) <-- its why all your reg attempts time out |
| 21:00.17 | drzed | [TK]D-Fender: "exten => _9.,1,Dial(Zap/g1/${EXTEN:1}0)" <-- this does not seem to add the prefix 0 i need |
| 21:00.35 | [TK]D-Fender | drzed: Indeed... becaue you put it AFTER |
| 21:01.05 | Akiyuki | [TK]D-Fender: I dont see what applies in this situation though. |
| 21:01.12 | [8none1] | krondorl: You could always implement your own parking lot in a dialplan |
| 21:01.15 | [TK]D-Fender | Akiyuki: All of it. |
| 21:01.29 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 21:02.06 | krondorl | [8none1]: I'll have to think about that, but as we are completely re-writing our dialplans it might be how to go. |
| 21:02.20 | *** join/#asterisk km2 (n=x@32.178.49.79) |
| 21:02.28 | drzed | [TK]D-Fender: you mean i'd have to put it before the }? |
| 21:02.34 | *** join/#asterisk BAJoseph (n=BAJoseph@adsl-66-143-100-157.dsl.hstntx.swbell.net) |
| 21:02.49 | [TK]D-Fender | drzed: Considerably |
| 21:02.58 | krondorl | good night all, gotta run.. |
| 21:03.21 | drzed | but woulldn't that strip the first 10 digits of my dialed number? |
| 21:03.45 | [TK]D-Fender | drzed: You don't seem to be paying close attention to my answers.... |
| 21:04.16 | [TK]D-Fender | drzed: it is not part of your VARIABLE REFERENCE. You want it to occur BEFORE it. |
| 21:04.34 | drzed | *do'h* |
| 21:04.40 | Un1x | [TK]D-Fender can you just help me solve this ,,,.... chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 21:04.56 | Un1x | the card i have is a TDM22B |
| 21:05.19 | drzed | [TK]D-Fender: thx alot |
| 21:05.33 | Akiyuki | [TK]D-Fender: I am still getting the error even after adding in the extempip and localnet |
| 21:06.34 | [TK]D-Fender | drzed: You're welcome. This is a good lesson to learn. |
| 21:07.07 | Akiyuki | Still getting SIP@... etc |
| 21:10.22 | Un1x | [inbound-asterlink] |
| 21:10.22 | Un1x | ; in 1.4 you can use SIP_HEADER function. You have X-DNIS, X-ANI2 |
| 21:10.22 | Un1x | exten => 18007551851,1,Dial(Zap/1,30) |
| 21:10.34 | Un1x | is that not a valid dial plan?, like someones trying to dial in on my did and they getbusy |
| 21:11.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:12.55 | [TK]D-Fender | Un1x: Go look in CLI with SIP debug enabled and watch the call |
| 21:14.19 | *** join/#asterisk esaym-acer (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:14.34 | Akiyuki | I added externip and localnet to sip.conf and after restarting, still seeing the Contact <sip:SIP@...> |
| 21:16.09 | [TK]D-Fender | Akiyuki: Clearly done things wrong. |
| 21:16.40 | Akiyuki | What could be wrong? |
| 21:16.42 | Akiyuki | nat=yes |
| 21:16.42 | Akiyuki | extempip=70.63.90.226 |
| 21:16.42 | Akiyuki | localnet=192.168.1.0/24 |
| 21:16.42 | Akiyuki | quality=yes |
| 21:16.47 | [TK]D-Fender | Akiyuki: Everything. |
| 21:17.35 | [TK]D-Fender | Akiyuki: Starting with the fac you spelled so many parameters wrong to begin with |
| 21:18.26 | Akiyuki | I copied and pasted from the wiki you showed.. |
| 21:19.08 | [TK]D-Fender | Akiyuki: Which link? |
| 21:19.32 | Akiyuki | whoops |
| 21:19.39 | Akiyuki | I had my cursor over it when i was on IM |
| 21:19.40 | Akiyuki | :D |
| 21:20.15 | Akiyuki | changed it to be externip now |
| 21:21.08 | Akiyuki | and qualify |
| 21:22.10 | Akiyuki | Still getting the SIP@192 |
| 21:23.41 | Akiyuki | But actually now it is saying Contact: <sip:SIP@70.63.90.226:5061> |
| 21:24.24 | [TK]D-Fender | Akiyuki: waiting for you to show me something complete & useful... |
| 21:25.08 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 21:25.25 | Akiyuki | http://pastebin.ca/1255557 |
| 21:25.28 | FruitBasket | I have one of my sip id's configured to only be able to receive one call (busy-level=1; call-limit=1; limitonpeer=yes), and sometimes they can neither make nor receive a call.. I've looked into it, and it's a call getting "stuck" in Asterisk. Today we had one that I had to soft hangup, and the CDR records the call as 9732 seconds. The phone provider, however, recorded it as 5.2 minutes. wtf? |
| 21:28.03 | Akiyuki | Also getting this error now --- |
| 21:28.04 | Akiyuki | [Nov 13 16:26:57] WARNING[28727]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0x941e9b0 (len 580) to 192.168.1.158:31578 returned -2: Bad file descriptor |
| 21:28.07 | Akiyuki | [Nov 13 16:26:59] NOTICE[28727]: chan_sip.c:7517 sip_reg_timeout: -- Registration for 'jimisanchez@sip.sipdiscount.com' timed out, trying again (Attempt #12) |
| 21:29.59 | *** join/#asterisk jyfletcher (n=justin@2105ds4-ar.0.fullrate.dk) |
| 21:30.07 | *** join/#asterisk goat_srv (n=goat_srv@unaffiliated/musikgoat) |
| 21:30.10 | *** part/#asterisk goat_srv (n=goat_srv@unaffiliated/musikgoat) |
| 21:30.42 | Akiyuki | [TK]D-Fender: What else can I pate for you? |
| 21:31.27 | *** join/#asterisk goat_srv (n=goat_srv@unaffiliated/musikgoat) |
| 21:31.39 | [TK]D-Fender | Akiyuki: you didn't include your configs, |
| 21:31.45 | [TK]D-Fender | Akiyuki: But I'm off heading home. |
| 21:31.48 | [TK]D-Fender | Back soon. |
| 21:35.00 | *** part/#asterisk goat_srv (n=goat_srv@unaffiliated/musikgoat) |
| 21:35.30 | *** join/#asterisk viator (n=viat0r@pool-71-123-76-43.wma.east.verizon.net) |
| 21:35.40 | viator | we have an avaya r7 system but now we want to be able to record all calls can we use an asterisk system in conjunction with our current system |
| 21:38.03 | viator | can i just do trunks >avaya>asterisk>phones or trunks>asterisk>avaya phones |
| 21:38.48 | viator | technically we only have to record 3 lines |
| 21:39.04 | thedonvaughn | viator: we have an avaya definitey system we do the same thing with asterisk. asterisk is our gateway between the CO and the avaya pbx so it records all are calls for us. We just have PRI trunks from avaya to asterisk and from asterisk to CO |
| 21:40.15 | viator | hmmm |
| 21:40.41 | viator | just lloking for a cost effective way to do this |
| 21:42.40 | viator | i think i might* lol be in a little over my head ahem |
| 21:42.54 | viator | maybe ill just call digium or somthing |
| 21:47.32 | jjshoe | Qwell |
| 21:47.45 | Qwell | jjshoe |
| 21:47.53 | jjshoe | Qwell just accepted. |
| 21:48.34 | Qwell | woot |
| 21:48.43 | Qwell | where's my finders fee? |
| 21:48.50 | jjshoe | what do you want? |
| 21:48.55 | Qwell | nothing |
| 21:49.04 | jjshoe | you know I'll sleep with you any time |
| 21:49.05 | Qwell | You already give me enough, if you know what I mean. *wink* |
| 21:49.53 | [8none1] | thinks he's in the wrong channel |
| 21:50.04 | Qwell | [8none1]: what? |
| 21:50.21 | harry_v | my uncle always asked for a 20% finders fee for finding the financial institution to fund businesses |
| 21:50.39 | [8none1] | Qwell: Couldn't tell for a sec if this was a jobs channel or sex channel. ;) jk |
| 21:51.00 | Qwell | [8none1]: he gives me free candy. |
| 21:51.04 | *** join/#asterisk hi365_m (n=hi365@213.151.61.149) |
| 21:52.21 | Qwell | and by sleep he means... |
| 21:52.25 | Qwell | well, I don't know what he means. |
| 21:52.35 | Qwell | jjshoe: sicko |
| 21:53.20 | Qwell | jjshoe: go to Appleton, buy my friend a beer, and we'll call it even - I was never able to :p |
| 21:54.13 | Qwell | (though I did convince the cab driver to go by his house so I could take pictures and freak him the hell out) |
| 21:54.25 | Qwell | ((that was so full of win)) |
| 21:56.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:00.51 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 22:05.46 | *** join/#asterisk Tuxofred (n=Fred@ip-80-236-223-137.dsl.scarlet.be) |
| 22:05.48 | Tuxofred | Hello |
| 22:06.08 | Tuxofred | I'm looking a public asterisk server to test my sip phone.. |
| 22:07.33 | hardwire | voxter |
| 22:07.55 | *** join/#asterisk telecos (n=sergio@84.166.219.87.dynamic.jazztel.es) |
| 22:09.17 | *** join/#asterisk johann8384 (n=jonathan@75-120-52-110.dyn.centurytel.net) |
| 22:12.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:13.25 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 22:14.08 | johann8384 | Anyone on here in St. Louis, MO. and want a job? |
| 22:14.08 | [TK]D-Fender | Tuxofred: Why would it have to be an * server? |
| 22:14.23 | [TK]D-Fender | Katty: MEW |
| 22:14.51 | [TK]D-Fender | johann8384: Contact Katty here... may be interested |
| 22:16.16 | harry_v | johann8384 you mean asterisk support job? |
| 22:16.30 | *** part/#asterisk Tuxofred (n=Fred@ip-80-236-223-137.dsl.scarlet.be) |
| 22:16.40 | johann8384 | Yeah, full-time VOIP Engineer, I know of two openings. |
| 22:17.15 | FruitBasket | johann: elaborate? |
| 22:19.50 | Akiyuki | [TK]D-Fender: http://pastebin.ca/1255594 can you take a look at my sip.conf and exten.conf now? |
| 22:20.35 | *** join/#asterisk zmitya (n=mitya@dsl5400B777.pool.t-online.hu) |
| 22:20.43 | zmitya | hi all |
| 22:20.47 | [TK]D-Fender | Akiyuki: "register" must occur AFTER everything else under [general] |
| 22:21.05 | zmitya | I'm trying to build asterisk-addons-1.2 |
| 22:21.29 | zmitya | asterisk 1.4.22 was compiled with --prefix=/usr/local/asterisk |
| 22:21.47 | zmitya | when I do a make it says: http://paste.debian.net/21412/ |
| 22:21.58 | Qwell | zmitya: Why are you trying to use -addons 1.2 with Asterisk 1.4? |
| 22:22.08 | *** part/#asterisk ys76 (n=ys76@ubuntu/member/ys76) |
| 22:22.18 | zmitya | Qwell: is it wrong for that version ? |
| 22:22.20 | Qwell | yes |
| 22:22.24 | zmitya | hmm :( |
| 22:22.31 | zmitya | do I need asterisk 1.6 ? |
| 22:22.39 | Qwell | asterisk and asterisk-addons 1.4 |
| 22:23.14 | zmitya | oh, I just found that link with google, did not know that it has the same vesion number |
| 22:23.34 | zmitya | sorry, I'm a bit newbie to asterisk (used it a long time ago...) |
| 22:23.36 | zmitya | thanks |
| 22:23.45 | Akiyuki | [TK]D-Fender: Ok, I changed and restarted. Not seeing any SIP errors or messages about trying to register now. Is there a way to see if it registered? like show peers? |
| 22:24.46 | [TK]D-Fender | Akiyuki: "sip show registry" |
| 22:25.05 | Akiyuki | Nothing listed. |
| 22:25.13 | *** join/#asterisk propellerhead (n=yogurt2u@200.80.189.153) |
| 22:25.58 | Akiyuki | I am also no longer seeing the time out errors in the log file either. |
| 22:28.38 | zmitya | Qwell: I did: svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.4 asterisk-addons-1.4 |
| 22:28.53 | zmitya | how can I define that asterisk is in /usr/local/asterisk ? |
| 22:29.00 | Qwell | --with-asterisk |
| 22:29.42 | Akiyuki | [TK]D-Fender: Ah there we go. Now I am getting s@70.. for the Contact |
| 22:30.04 | zmitya | Qwell: great, thanks !! |
| 22:30.30 | Akiyuki | [TK]D-Fender: Now, sip show registry says sip1.sipdiscount.com:5060 jimisanchez 120 Request Sent |
| 22:34.57 | Akiyuki | Still getting this error though: chan_sip.c:7517 sip_reg_timeout: -- Registration for 'jimisanchez@sip1.sipdiscount.com' timed out, trying again (Attempt #17) |
| 22:37.04 | bcrochet | I keep getting this message: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 22:37.04 | bcrochet | My TDM400 was working before. |
| 22:37.14 | bcrochet | The card is connected to power. |
| 22:37.25 | bcrochet | Any ideas? |
| 22:38.20 | bcrochet | Plus... when I load wctdm I get this: Freshmaker failed register test |
| 22:38.53 | bcrochet | Should I go back to an earlier version of zaptel? |
| 22:40.35 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:41.11 | *** join/#asterisk grantm (n=grant@66.29.180.194.static.utahbroadband.com) |
| 22:42.31 | Akiyuki | [TK]D-Fender: i tried using a bogus username/password but didnt get an error message saying invalid, just a time out. What else can I change? |
| 22:43.55 | [TK]D-Fender | Akiyuki: You could realize that you keep make changes, testing them, and then telling me they don't work. All of this without SHOWING ME. |
| 22:44.10 | murdock_ut | I've got a question on the Dial application. |
| 22:44.14 | murdock_ut | exten => 2190,n,Dial(IAX2/ICSSLC->SLC/1199&IAX2/ICSSLC->SLC/1172,30,${TRUNK_OPTIONS_IN}) |
| 22:44.17 | [TK]D-Fender | murdock_ut: go for it |
| 22:44.41 | murdock_ut | This seems to call both phone but the second one rings once then hangs up. |
| 22:45.03 | [TK]D-Fender | murdock_ut: I'm not quite sure "->" islegal |
| 22:45.06 | Akiyuki | [TK]D-Fender: I showed you my sip.conf and my extensions.conf I will repaste again |
| 22:45.22 | [TK]D-Fender | aiksa[LV]: You showed it to me when it was wrong and I told you to change things. |
| 22:45.28 | [TK]D-Fender | Akiyuki: ^^ |
| 22:45.31 | murdock_ut | If I just have it call one extension it works fine. |
| 22:45.38 | [TK]D-Fender | Akiyuki: You didn't show me the "after" |
| 22:45.51 | [TK]D-Fender | Akiyuki: and please permanetly remove all the commented lines |
| 22:46.09 | [TK]D-Fender | murdock_ut: pastebin the failed call's CLI output at verbose 10 |
| 22:46.13 | [TK]D-Fender | ~pb |
| 22:46.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:46.20 | Akiyuki | oh |
| 22:46.20 | Akiyuki | :D |
| 22:46.28 | Akiyuki | [TK]D-Fender: http://pastebin.ca/1255621 |
| 22:47.03 | [TK]D-Fender | Akiyuki: Clean up the junk and repaste |
| 22:50.32 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 22:50.54 | Akiyuki | [TK]D-Fender: http://pastebin.ca/1255628 |
| 22:52.25 | [TK]D-Fender | Akiyuki: And you haven't shown the current failure, or described what you have forwarded to * |
| 22:52.57 | murdock_ut | [TK]D-Fender: http://pastebin.com/dc1aebd4 |
| 22:53.12 | Akiyuki | [TK]D-Fender: http://pastebin.ca/1255632 |
| 22:54.04 | murdock_ut | Lets see how may pastbin's we can send to [TK]D-Fender |
| 22:54.31 | [TK]D-Fender | murdock_ut: -- Executing Answer("IAX2/209.181.82.1:2317-8", "") in new stack <- the first one ANSWERS the call explicitly |
| 22:54.32 | Akiyuki | :D |
| 22:54.39 | [TK]D-Fender | murdock_ut: first come, first served |
| 22:55.00 | murdock_ut | Shouldn't both phones ring until one is answered? |
| 22:55.19 | [TK]D-Fender | murdock_ut: Both phones? You are calling another * box. the DIALPLAN is answering the call! |
| 22:55.27 | murdock_ut | Ah, |
| 22:55.32 | murdock_ut | wait |
| 22:55.41 | [TK]D-Fender | murdock_ut: You are not passing progrewss back by dialing an inside phone, your dialplan has issued the explicit answer |
| 22:56.00 | [TK]D-Fender | murdock_ut: So you've shot yourself in the foot |
| 22:56.35 | murdock_ut | So I shouldn't have this on the receiving * box dialplan: exten => _11XX,1,Answer() |
| 22:56.37 | [TK]D-Fender | murdock_ut: The receiving exten should not be answering or issuing audio, only doing what it has to to reach the inside Dial command |
| 22:56.44 | [TK]D-Fender | murdock_ut: Correct |
| 22:57.24 | murdock_ut | [TK]D-Fender: That makes sense. I thought you always started a dialplan with Answer(). I guess this is one of the rare exceptions. |
| 23:01.51 | Akiyuki | [TK]D-Fender: What else am I missing for yuo? |
| 23:03.21 | FruitBasket | does Asterisk support SIP over TCP/IP and RTP over UDP? |
| 23:04.33 | Akiyuki | bbiab |
| 23:06.21 | harry_v | What is the most reliable asterisk interface being used right now. |
| 23:06.29 | *** join/#asterisk farkus____ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 23:06.52 | [TK]D-Fender | harry_v: CLI. Hasn't failed anyone yet... |
| 23:06.57 | harry_v | hehe |
| 23:07.09 | [TK]D-Fender | murdock_ut: You're welcome |
| 23:07.17 | harry_v | yea, but when you start getting busy... |
| 23:07.25 | harry_v | ive always used sli |
| 23:07.27 | harry_v | cli |
| 23:10.33 | [TK]D-Fender | ok, off for a few hours. BBL |
| 23:13.13 | murdock_ut | [TK]D-Fender: Thanks! So normally you do start your dialplan with answer() correct? |
| 23:16.24 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 23:17.27 | murdock_ut | [TK]D-Fender: IN this: exten => _11XX,n,SIPAddHeader("Alert-Info:<http://www.notused.com>\;info=alert-external\;x-line-id=0") |
| 23:17.27 | murdock_ut | exten => _11XX,n,Dial(SIP/${EXTEN},20,TtWw) Is there a way to specify a different context? |
| 23:17.40 | murdock_ut | oops |
| 23:18.14 | murdock_ut | In the dial application is there a way to specify a different context? |
| 23:20.05 | Assimilate | isn't it SIP/${EXTEN}@Context? |
| 23:21.10 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:21.12 | murdock_ut | Assimilate: Let me try. |
| 23:24.03 | murdock_ut | No, it didn't like that. |
| 23:24.07 | murdock_ut | :q |
| 23:27.05 | *** join/#asterisk kornelak (n=karl@199.33.79.4) |
| 23:37.34 | Assimilate | murdock_ut, I use exten => s,2,Dial(SIP/${ARG1}@broadvoice,180,th) to dial the number through my broadvoice context |
| 23:37.50 | kornelak | Hello, again, everyone! I've found some more weirdness. |
| 23:38.14 | kornelak | Moving from 1.4.21.2 to 1.4.22, suddenly MeetMe no longer prompts for a pin # on conferences. Has anyone seen this? |
| 23:39.08 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 23:41.24 | rene- | hello |
| 23:41.40 | rene- | when setting lspci latency_timer |
| 23:41.41 | kornelak | I run the MeetMe application as "MeetMe(|DIMopsx|)" |
| 23:42.03 | rene- | a bigger timer means lower precedence? |
| 23:50.33 | *** join/#asterisk sprbck (n=sprbck@81.92.203.249) |
| 23:50.37 | sprbck | hello |
| 23:51.38 | sprbck | I need some help with caller-id via Verizon POTS lines and Asterisk.. I can't seem to get it, but on a directly connected phone it shows. Anyone has got it working? |
| 23:52.27 | *** part/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
| 23:53.40 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 23:54.07 | kornelak | I always thought that calls from an analog line would send Caller*ID info between the 1st and 2nd ring. |
| 23:54.07 | drmessano | Isnt a verizon problem, it's a callerid problems.. Try the wiki |
| 23:54.21 | kornelak | Is * not waiting long enough? |
| 23:54.33 | Katty | Qwell: 4 bars into 71 |
| 23:55.50 | sprbck | I get callerid on a variety of lines, analog or isdn, on multiple countries. It's the first time I've came across this. And it does show only after 1-2 seconds, on the phone. |
| 23:56.28 | drmessano | sprbck: If you're getting the CallerID on a PHONE, then the CallerID is working.. |
| 23:56.48 | sprbck | yes, It tells me that the provider is putting it there |
| 23:57.04 | drmessano | So its an asterisk config issue |
| 23:57.16 | drmessano | Nothing unique to verizon |
| 23:57.18 | sprbck | yes, so it seems, and I've tried to search about this a *lot* |
| 23:57.34 | sprbck | trued differente cidsignalling settings, but no luck |
| 23:57.38 | sprbck | tried |
| 23:57.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |