| 00:04.37 | NormanASD | Quick quesion. I'm using externip in my sip.conf, and it doesn't seem to be using this IP in the SDP packets. I'm getting "c=IN IP4 <internal_ip>" |
| 00:04.55 | mog | hey has anyone bought that chan_skype thing? |
| 00:04.56 | *** join/#asterisk LoBo_JeDi (n=LoBo_JeD@208.51.215.226) |
| 00:05.00 | *** part/#asterisk LoBo_JeDi (n=LoBo_JeD@208.51.215.226) |
| 00:06.02 | *** join/#asterisk necudeco (n=chatzill@200.121.75.218) |
| 00:06.40 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:08.05 | *** part/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net) |
| 00:10.17 | fifer | I'm having a VM issue |
| 00:10.33 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:10.39 | fifer | WHen a VM is being recorded it just keeps generating these errors: |
| 00:10.43 | justinu|laptop | NormanASD: is localnet= set? |
| 00:10.48 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 00:10.49 | fifer | format_wav_gsm.c243 update_header Unable to find our position |
| 00:11.05 | fifer | And just creates both an empty txt and wav files |
| 00:11.40 | NormanASD | justinu|laptop: yes. I may have found the problem. externip goes under general, correct? |
| 00:11.54 | fifer | I have the format set to wav49 |
| 00:12.06 | justinu|laptop | NormanASD: i think so |
| 00:12.10 | fifer | This setup has been working fine for hte last 8 months, just stoped working this week |
| 00:12.11 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.223) |
| 00:12.36 | NormanASD | justinu|laptop: i've been fighting this all night. i think i'm getting close. the SDP looks good now. |
| 00:12.58 | *** join/#asterisk DaneM (n=DaneM@70.135.56.182) |
| 00:13.25 | justinu|laptop | at least you know what SDP is :) |
| 00:13.55 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:14.09 | DaneM | Hello all. I have 2 phone lines, split (using a 2-line splitter), with each individual line going into a X100P PCI card. Both lines handle incoming calls, however, only line 1 will pick up. Any ideas? |
| 00:14.11 | NormanASD | I read (skimmed, it's long) the SIP RFC. |
| 00:14.39 | docelmo | Ya dont use the X100P card |
| 00:14.45 | docelmo | get a TDM400P |
| 00:14.51 | NormanASD | Sadly, I'm doing Asterisk -> NAT -> Internet -> NAT -> SIP ATA |
| 00:14.59 | DaneM | hmmm...why's that? |
| 00:15.16 | NormanASD | However, I think it's now working after modifying the RTP ports in the other firewall. |
| 00:15.23 | fifer | Any ideas on the Voicemail issue? |
| 00:15.45 | docelmo | DaneM uhh they suck |
| 00:15.46 | justinu|laptop | yeah, in a double nat situation you'll have to use port forwarding on at least one side |
| 00:16.09 | NormanASD | Yes, I did get that, thankfully I have a static IP on one end. |
| 00:16.11 | docelmo | And putting 2 in the same machine.. Your just asking for head aches |
| 00:16.38 | DaneM | OK...anyway, I'd like to get it working with the hardware I'm using if that's possible. Card 1 works perfectly with all my extensions that use it, but card 2 won't answer the phone. |
| 00:16.53 | docelmo | Can you see it in asterisk? |
| 00:17.47 | DaneM | that's weird...I saw it a while ago; now I get this error when I type, "ztcfg -vv": |
| 00:18.15 | docelmo | There ya go.. |
| 00:18.21 | DaneM | Channel 1 is already configured as 'FXS Kewlstart' at line 11 |
| 00:18.34 | DaneM | yeah...I suppose that's a problem :-p |
| 00:18.39 | DaneM | brb...looking |
| 00:18.52 | docelmo | You should have something like channel 1 and 2 |
| 00:19.01 | docelmo | 0 should be your pseudo |
| 00:19.45 | DaneM | nodnod. It was saying that before. I'm checking for the cause now. |
| 00:20.36 | DaneM | ok...fixed it. I'll try calling now... |
| 00:20.49 | DaneM | (as soon as the other line is free) |
| 00:23.11 | Skyelar | woohoo, that was easy: just wrote a small zaptel app that takes a line off hook and then sleeps forever |
| 00:23.33 | DaneM | OK...it's seeing both channels now, but it's still not picking up. |
| 00:27.40 | *** join/#asterisk droops_mobile (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 00:30.36 | DaneM | hmmmm...I think I found the problem: the jack is messed up. No dial tone from that line when I plug a phone in. |
| 00:30.57 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 00:31.11 | docelmo | um, dude.. FXO is CO not phone |
| 00:31.31 | fifer | anyone know what would cause format_wav_gsm.so to suddenly have issues? |
| 00:31.49 | fifer | When trying to record a VM it is giving these errors: |
| 00:31.57 | fifer | format_wav_gsm.c243 update_header Unable to find our position |
| 00:31.58 | Skyelar | justinu|laptop: thanks for your answers before - have now solved my problem: wrote a little program that uses ZT_HOOK to take an FXO port off-hook and then waits forever; in combination with "zap destroy channel X", I'm all sorted :) |
| 00:32.23 | justinu|laptop | kewl |
| 00:38.35 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 00:42.15 | *** join/#asterisk robin_sz (n=you@adsl.redpoint.org.uk) |
| 00:47.39 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:58.39 | *** join/#asterisk sloth_ (n=sloth@cpe-69-203-212-182.nyc.res.rr.com) |
| 01:00.39 | *** join/#asterisk jusse (n=jusse@190.40.212.137) |
| 01:00.39 | jusse | hola |
| 01:03.41 | jusse | im use amportal |
| 01:04.07 | jusse | i need help |
| 01:07.15 | Qwell | jusse: type /topic |
| 01:11.01 | jeremy_g | hi |
| 01:11.04 | jeremy_g | anyone home |
| 01:11.28 | jeremy_g | when i type show sip channels |
| 01:11.34 | jeremy_g | i see |
| 01:11.50 | jeremy_g | IP - Username/ANI - ... |
| 01:12.54 | jeremy_g | is there a way, a person calls in and i can make changes to asterisk configuration so that Username shown in the channel can be different each time it calls |
| 01:13.04 | jeremy_g | bye |
| 01:27.40 | *** join/#asterisk mv00 (i=HiTMaN@gateway/tor/x-aadcfcd173ab953d) |
| 01:28.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.154) |
| 01:32.33 | *** join/#asterisk Adam12 (n=adam@d150-182-137.home.cgocable.net) |
| 01:33.21 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 01:37.51 | riddlebox | is faxing possibe over voip? |
| 01:39.39 | *** join/#asterisk cvaldess (n=hello@75.Red-88-18-160.staticIP.rima-tde.net) |
| 01:39.43 | cvaldess | Hi |
| 01:39.56 | Adam12 | hi |
| 01:39.58 | cvaldess | need a litle help with SIP registration |
| 01:40.09 | cvaldess | register => user[:secret[:authuser]]@host[:port][/extension] |
| 01:41.11 | cvaldess | my user is 307314134213@telefonica.net secret: blabla and authuser: authuser |
| 01:42.13 | cvaldess | register command read the first @ and think there is host after it |
| 01:43.31 | cvaldess | canīt do register => 307314134213@telefonica.net:blabla:authuser@192.168.1.1:5070 |
| 01:44.40 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 01:46.04 | arcanine | add extension on your register cmd |
| 01:46.28 | cvaldess | the problem is with the two @ |
| 01:46.41 | cvaldess | exten go to s |
| 01:50.01 | *** join/#asterisk vlt (n=daniel@dslb-088-073-239-158.pools.arcor-ip.net) |
| 01:53.08 | *** join/#asterisk JSabines (i=JSabines@189.158.185.137) |
| 01:53.18 | arcanine | cvaldess : this is on the dialplan? |
| 01:53.29 | cvaldess | sip.conf |
| 01:53.51 | kronic | anyone know much about skills-based routing? |
| 01:56.15 | inv_Arp | cvaldess: telephonica is that in FL? |
| 02:05.38 | cvaldess | telefonica.es |
| 02:09.26 | *** join/#asterisk vlt (n=daniel@dslb-088-073-249-088.pools.arcor-ip.net) |
| 02:14.35 | arcanine | have anyone tested a dialogic jct-2e1 on 1.4 beta |
| 02:19.39 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 02:20.00 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-153-250-45.red.bezeqint.net) |
| 02:24.33 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 02:25.06 | CunningPike | ~seen aptura |
| 02:25.23 | jbot | aptura is currently on #asterisk (5h 46m 29s). Has said a total of 1 messages. Is idling for 5h 33m 59s, last said: 'do you have it saves on pastebin'. |
| 02:27.02 | aptura | hi |
| 02:27.11 | *** join/#asterisk kimo_sabe (i=nick@zappa.azrackspace.net) |
| 02:38.16 | *** join/#asterisk deny_ (n=andy@adsl-69-210-249-96.dsl.chcgil.ameritech.net) |
| 02:40.29 | deny_ | how would i modprobe my fxo card |
| 02:42.13 | deny_ | i got a digum (i think)100 |
| 02:43.05 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 02:43.21 | *** join/#asterisk Essobi (i=kstone@216.26.137.75) |
| 02:43.30 | *** part/#asterisk Essobi (i=kstone@216.26.137.75) |
| 02:46.34 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 02:46.42 | cekc | so, what exactly is a wink? |
| 02:48.36 | De_Mon | when someone dials an outside line, I set the callerID before initiating the dial. As a result the src of the call is always the callerid name/number instead of the staff member who made the call |
| 02:48.43 | *** join/#asterisk kannan (i=1001@61.8.147.164) |
| 02:49.21 | De_Mon | how do I setup my dialplan to identify the actual caller? |
| 02:54.00 | *** join/#asterisk jahani (n=ja@adsl196-213-242-217-196.adsl196-16.iam.net.ma) |
| 02:54.03 | jahani | hi |
| 02:54.28 | jahani | can askterisk so termanation on h323 gateways? |
| 02:55.03 | kimo_sabe | jahani: not out of the box I don't think, but yeah |
| 02:55.27 | jahani | what you mean not out of the box? |
| 02:56.16 | kimo_sabe | jahani: I think the h323 channel driver isn't included in asterisk-proper.. Or maybe it is now but you still need some extra stuff to compile it |
| 02:57.37 | jahani | ok |
| 02:57.39 | jahani | thankyou |
| 02:59.54 | *** join/#asterisk techie (n=gus@voipops.net) |
| 03:00.06 | De_Mon | jahani check voip-info.org |
| 03:00.12 | De_Mon | ??h323 |
| 03:00.42 | kimo_sabe | jahani: hmm, that is a good question. Why h323? |
| 03:04.04 | *** join/#asterisk kamileon (n=kamileon@c-71-207-204-15.hsd1.al.comcast.net) |
| 03:10.39 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 03:17.52 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 03:20.05 | cekc | my asterisk box is dialing outgoing numbers too fast for the phone company to hear, how do I slow that down? |
| 03:20.23 | kimo_sabe | cekc: using a digium card? |
| 03:20.31 | cekc | yeah, on a T1 |
| 03:21.11 | X-Rob_ | I'm seriously confused about what a T1 is then. |
| 03:21.21 | X-Rob_ | You've got your T1 plugged into a TE11x card? |
| 03:21.54 | *** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se) |
| 03:22.13 | cekc | TE110P |
| 03:22.24 | JT | do you also have a channel bank cekc ? |
| 03:22.35 | JT | or are you using PRI ISDN signalling? |
| 03:22.42 | cekc | em_w signalling |
| 03:23.08 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:24.16 | cekc | I setup an extension so that when I dial 9 it calls Zap/g1 (I have group 1 setup) and then I get dialtone. If I manually dial a number fast I noticed that I still hear dialtone until about the 3rd digit and the call doesn't go through. If I dial slowly it works |
| 03:24.34 | X-Rob_ | cekc, ahha. |
| 03:24.45 | X-Rob_ | zapata.conf, 'toneduration' |
| 03:25.11 | cekc | know what the default is by any chance? |
| 03:25.28 | X-Rob_ | says 100 in /usr/src/asterisk/configs/zapata.conf.sample |
| 03:25.50 | cekc | how come I don't have a sample file! |
| 03:26.13 | X-Rob_ | coz you didn't install from source? |
| 03:26.16 | JT | because you don't have /usr/srv/asterisk ? |
| 03:26.20 | JT | src even |
| 03:26.33 | cekc | I installed from subversion |
| 03:30.53 | cekc | ah, I read that as /etc instead of /usr/src |
| 03:31.10 | kronic | anyone know about implementing skills-based routing? |
| 03:32.32 | cekc | now that sounds like a pretty neat idea |
| 03:32.49 | kimo_sabe | kronic: skills-based? Like, keep the n00bs out of the l33t h4x?!??! |
| 03:33.48 | kimo_sabe | kronic: seriously though, what is that? |
| 03:33.56 | kronic | http://www.voip-info.org/tiki-index.php?page=PBX%20Skill%20Based%20Routing |
| 03:34.20 | cekc | I think he means say you have a tech support line and someone needs OSX support, only try to ring extensions for people who know how to support it |
| 03:34.31 | cekc | now I will of course click the link to see if I was right |
| 03:35.01 | kimo_sabe | kronic: ah, direct the n00b to their l33test h4x0r available. |
| 03:35.09 | kimo_sabe | I'll lay off the l33t sp33k now |
| 03:36.26 | cekc | so... how do I tell if my caller id isn't setup right, or if the phone company just isn't sending it? |
| 03:37.27 | kronic | prior to the Dial Cmd, perhaps do a NoOp(${CALLERID(num)}) |
| 03:37.54 | cekc | I mean for incoming calls |
| 03:39.08 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 03:39.22 | cekc | I don't know what it's called, but my box is providing dial tone and the DID is dialed on incoming calls. Can I even receive caller id with this method? |
| 03:39.22 | kimo_sabe | cekc: analog line? Are you waiting for the second ring to pick up? |
| 03:39.28 | cekc | T1 |
| 03:41.55 | cekc | I do say that this channel was right in telling me that getting a T1 would eliminate my echo program. Voice quality with a T1 and SIP phones is excellent. |
| 03:42.32 | cekc | somehow I am the first customer of my phone company to want a digital signal and not want the T1 converted to copper |
| 03:43.48 | kimo_sabe | cekc: wow, where abouts the globe are you at? |
| 03:43.55 | cekc | California |
| 03:51.25 | kimo_sabe | cekc: hmm, callerid=asreceived in your zapata.conf? |
| 03:53.46 | *** join/#asterisk bmg505 (n=leon@c1-54-12.rndf.isadsl.co.za) |
| 04:00.44 | kronic | found what I was looking for, ICD |
| 04:05.54 | JT | cekc: suburbia? |
| 04:06.15 | orlock | suburbia! |
| 04:06.36 | JT | cekc: sucks that they didn't give you a PRI though :( |
| 04:07.21 | [TK]D-Fender | CAS T!... *shudder* |
| 04:07.38 | [TK]D-Fender | cekc : Kick their asses and make them switch to PRI.. |
| 04:08.24 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 04:17.55 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 04:18.13 | CrazyTux | Hello, does anyone here use editplus as an editor? I'm looking to see if anyone here had a .STX file for editplus for AEL2? |
| 04:19.01 | *** join/#asterisk WAudette (n=WAudette@71.237.146.239) |
| 04:19.43 | [TK]D-Fender | CrazyTux : Welcome to #AELlovers, population : YOU. |
| 04:19.45 | [TK]D-Fender | :D |
| 04:19.54 | CrazyTux | [TK]D-Fender, thank you. |
| 04:20.09 | [TK]D-Fender | CrazyTux : Seriously I don't think anyone's cared enough to do it..... |
| 04:20.16 | file | [TK]D-Fender: yay turkey |
| 04:20.32 | [TK]D-Fender | file : g0bbl3 g0bbl3 |
| 04:20.43 | CrazyTux | [TK]D-Fender, Hmm, whys that. |
| 04:21.44 | [TK]D-Fender | CrazyTux : Because it offers nothing taht can't be done in standard extension logic, and was still "experimental". the problem with "free-form" coding is it lets people get "creatively sloppy". |
| 04:22.27 | CrazyTux | [TK]D-Fender, Yea, but its all compiled into one standard form, so I don't see a difference? |
| 04:23.50 | [TK]D-Fender | CrazyTux : Yeah another syntax to the same end. Why bother? Better to have spent the time redesigning chan_sip or something |
| 04:24.16 | CrazyTux | [TK]D-Fender, Whys that? |
| 04:24.23 | file | or perfecting the best recipe for muffins |
| 04:24.30 | CrazyTux | lol. |
| 04:24.37 | CrazyTux | So you guys are telling me AEL2 is worthless? |
| 04:25.25 | [TK]D-Fender | Thats it! AEL2 is just a modeling engine for making better muffins! |
| 04:25.59 | CrazyTux | ... |
| 04:26.27 | file | [TK]D-Fender: EXACTLY! |
| 04:27.21 | file | [TK]D-Fender: how's the weather up your way? |
| 04:27.27 | file | up.. over... |
| 04:27.39 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
| 04:27.46 | CrazyTux | I don't get it... why is AEL2 so shitty? |
| 04:27.53 | CrazyTux | Or are you guys just biased? |
| 04:27.58 | CrazyTux | Or whats so bad about AEL2 |
| 04:28.12 | file | CrazyTux: nobody said it was, and I didn't make a comment at all... just depends |
| 04:28.12 | murf | CrazyTux: we are biased, we admit it! |
| 04:28.37 | file | personally I grew up with the standard extensions.conf, so it's what I use |
| 04:29.42 | CrazyTux | murf, steven murphy? |
| 04:29.50 | file | that's murf! |
| 04:29.52 | murf | That be me. |
| 04:31.32 | murf | CrazyTux: there's a whole generation of users that got dragged into extensions.conf kicking and screaming, and they'll die before they go thru that again. And that's OK. For the guys irritated by extensions.conf, we have AEL. |
| 04:32.02 | [TK]D-Fender | CrazyTux : So very few people use it taht when those who figure "yeah I wanna feel cool and like a programmer" and fail come in for help and realize the rest of us knew better and just stuck with "old-faithful" it makes you wonder why you bother in the first place. |
| 04:32.33 | CrazyTux | murf, mind if I private message you? |
| 04:32.35 | file | we'll always be together! |
| 04:32.41 | dalbaech | I can completely understand that argument. I believe in the "old-faithful" as you put it. It does everything I need... |
| 04:32.43 | file | together in the PBX core. |
| 04:32.57 | murf | CrazyTux: private away..! |
| 04:36.21 | De_Mon | http://pastebin.ca/195913 |
| 04:36.33 | De_Mon | why does this always use the trunk teliax |
| 04:37.36 | De_Mon | it's like the function is IF(cond=val?fval) |
| 04:37.56 | *** part/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai) |
| 04:39.23 | kimo_sabe | murf: horray for AEL |
| 04:41.16 | murf | hey, guys, don't mistake my aquiescence for agreement... I have several reasons why using AEL is better, mostly QA and cost related... but I don't want to preach to audiences that aren't listening, either. |
| 04:41.51 | file | murf: don't tell them it can take over the world |
| 04:41.53 | file | oh, whoops |
| 04:44.09 | murf | file: you just ***had*** to let it slip! |
| 04:44.43 | file | >_< |
| 04:47.06 | De_Mon | ael is probably what n korea used to come up with their world domination plans |
| 04:48.47 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:51.31 | jahani | how to fix this error : |
| 04:51.32 | jahani | You do not appear to have the sources for the 2.6.8-022stab078.10 kernel installed. |
| 04:51.32 | jahani | make[1]: *** [linux26] Error 1 |
| 04:51.33 | jahani | ? |
| 04:52.29 | cekc | jahani: which linux do you have |
| 04:52.37 | De_Mon | not debian |
| 04:52.41 | jahani | centos 4 |
| 04:52.54 | cekc | type: yum install kernel-source |
| 04:53.01 | jahani | ok thx |
| 04:53.46 | cekc | and let me know if that actually works |
| 04:57.21 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:57.32 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 05:05.55 | sx-wks | what a joke http://www.heise-security.co.uk/news/79212 |
| 05:09.38 | JT | what's the joke? |
| 05:10.02 | jahani | cekc not work yum install kernel-source |
| 05:10.20 | sx-wks | JT: they expect using some software that only works on windows to do their dids... |
| 05:10.29 | jahani | No Match for argument: kernel-source |
| 05:10.30 | jahani | Nothing to do |
| 05:11.31 | jaxzn | sx-wks, yeah this sentence is probably the lamest "The ISPs of the persons under investigation will then slip the program onto their computers. " |
| 05:11.56 | JT | sx-wks: looks like they're targeting skype in particular |
| 05:12.15 | sx-wks | yeah. that's lame |
| 05:12.44 | sx-wks | and totally useless, considering real criminals already use vpns and stuff |
| 05:13.02 | JT | it doesn't do it at the network layer |
| 05:13.03 | jaxzn | they should then slip something to get my bank numbers, good luck with that |
| 05:13.14 | JT | looks like it intercepts it closer to an audio level at the pc |
| 05:13.34 | sx-wks | JT: as long as it can be installed... |
| 05:13.39 | jaxzn | JT, in order for it to work it has to be installed/deployed |
| 05:13.45 | JT | sure |
| 05:14.15 | jaxzn | that means if they can install wiretap whats preventing them from installing keyboard sniffer, lets say |
| 05:14.23 | jaxzn | in turn they become the criminals |
| 05:14.39 | jaxzn | the whole thing is a big joke as sx-wks pointed |
| 05:14.50 | sx-wks | jaxzn: running openbsd or something will make the thing moot :D |
| 05:15.32 | jaxzn | sx-wks, criminals use much more sophisticated means of communication then skype |
| 05:15.38 | jaxzn | hehe |
| 05:15.41 | sx-wks | jaxzn: right |
| 05:16.04 | jaxzn | skype is for l33t hax0r kiddies |
| 05:16.10 | sx-wks | haha |
| 05:16.48 | JT | how will law enforcement using keyboard sniffers turn them into criminals? |
| 05:17.13 | jaxzn | JT its invasion of privacy, who knows what data they are collecting |
| 05:17.20 | JT | wtf |
| 05:17.34 | JT | if they've got a court order, your pirvacy is irrelevant |
| 05:17.45 | jaxzn | JT, no it isn't |
| 05:17.54 | jaxzn | JT, court order is for specific thing |
| 05:17.55 | JT | of course it is |
| 05:17.55 | sx-wks | the real question is, given they target windows... you should be safe as long as you use something else :D |
| 05:18.13 | jaxzn | JT, obviously you dont understand the law |
| 05:18.30 | JT | if they have teams covertly following you around everywhere, i think the courts have already decided "for the greater good of the community, this person(s)' privacy will suffer" |
| 05:18.36 | JT | jaxzn: bullshit |
| 05:18.56 | *** join/#asterisk dalbaech (i=dalbaech@serverchimps.org) |
| 05:19.39 | jaxzn | JT, law is not simple in real life |
| 05:19.50 | JT | sure |
| 05:19.50 | jaxzn | JT, read a book or two |
| 05:20.15 | JT | but being a newbie shouting "privacy!!!111one" does not summarise the law well either |
| 05:20.18 | JT | of fuck off |
| 05:20.19 | JT | seriously |
| 05:20.23 | JT | you stupid git |
| 05:20.35 | JT | i was happy to discuss this |
| 05:20.40 | JT | but you've taken to insults |
| 05:20.43 | JT | so fuck off |
| 05:20.56 | jaxzn | ok l33t kiddie |
| 05:21.01 | JT | l33t? |
| 05:21.10 | JT | you seriously have no idea what you're talking about |
| 05:21.18 | JT | so quick to pass judgement |
| 05:21.20 | JT | tisk tisk |
| 05:23.12 | *** join/#asterisk freebsd_fan (n=unsure@sw69x3.foto.gu.se) |
| 05:23.29 | jahani | i got this error when i did ./configure inasterisk folder : configure: error: *** termcap support not found |
| 05:23.32 | jahani | how can i fix it? |
| 05:43.05 | *** join/#asterisk bun-bun (n=bun@unaffiliated/bun-bun) |
| 05:49.05 | *** part/#asterisk jahani (n=ja@adsl196-213-242-217-196.adsl196-16.iam.net.ma) |
| 05:51.11 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 05:51.19 | *** join/#asterisk LakeSolon (n=blake@64-83-227-227.dhcp.stcd.mn.charter.com) |
| 05:51.24 | PakiPenguin | hello :) |
| 05:53.25 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 05:53.28 | drcode | hi all |
| 05:53.37 | drcode | for sip users what .conf file I need to config? |
| 05:53.41 | drcode | pc to pc |
| 05:53.49 | drcode | hi all |
| 05:56.29 | kaldemar | sip.conf and extensions.conf |
| 05:57.46 | drcode | thanx |
| 05:59.06 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:01.06 | shellshark | PakiPenguin: long time no see ;-) |
| 06:01.15 | shellshark | PakiPenguin: how have you been brother? |
| 06:04.14 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:07.21 | PakiPenguin | hey shellshark |
| 06:07.24 | PakiPenguin | been busy |
| 06:07.26 | PakiPenguin | how about you? |
| 06:08.11 | shellshark | same ;) |
| 06:08.24 | stephane_ | jour |
| 06:08.39 | shellshark | heya stephane_ |
| 06:09.56 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 06:17.47 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 06:18.01 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:22.48 | *** join/#asterisk VijayG (i=vijay@202.131.145.234) |
| 06:22.52 | VijayG | hello |
| 06:23.10 | VijayG | i am facing some issues dialing from sip trunks |
| 06:23.26 | VijayG | it seems request is not going out properly from sip trunks |
| 06:23.33 | VijayG | any idea what could be the issue |
| 06:24.25 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:25.38 | VijayG | HEllo Anyone there?????? |
| 06:28.46 | VijayG | ??????????????????? |
| 06:28.48 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 06:29.06 | VijayG | hello |
| 06:29.13 | VijayG | Anyone there?????? |
| 06:29.17 | VijayG | i am facing some issues dialing from sip trunks |
| 06:29.22 | VijayG | <PROTECTED> |
| 06:29.26 | VijayG | any idea what could be the issue |
| 06:29.28 | Inverted | once an agi script is running, how do you get user input from the keypad? |
| 06:29.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:30.02 | VijayG | through a VOIP gateway |
| 06:30.04 | VijayG | analog phones |
| 06:31.50 | VijayG | ???????????????? |
| 06:32.45 | shellshark | Inverted: Read() |
| 06:33.00 | shellshark | VijayG: what does the CLI say? |
| 06:33.17 | VijayG | Time out |
| 06:33.21 | Inverted | shellshark: so you just execute Read() from within the agi script? |
| 06:33.35 | VijayG | moreover i am not able to even register at other end |
| 06:33.39 | shellshark | VijayG: firewall / proxy / nat issue? |
| 06:33.44 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:33.49 | VijayG | everything is clear |
| 06:33.58 | shellshark | Inverted: should do it |
| 06:34.01 | VijayG | on IAX trunks its just going fine |
| 06:34.17 | VijayG | on IAX i am able to make calls |
| 06:34.22 | VijayG | register and do anything i want |
| 06:34.29 | VijayG | but when i try the same through SIP |
| 06:34.33 | shellshark | sounds like your ISP is blocking SIP |
| 06:34.34 | VijayG | nothing is happening |
| 06:34.44 | shellshark | i've seen that happen before |
| 06:34.55 | VijayG | you mean 5060 is blocked |
| 06:34.57 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
| 06:35.01 | skywriter | does any voip comapny support asterisk if they support sip?? |
| 06:35.21 | shellshark | skywriter: in theory |
| 06:35.33 | shellshark | VijayG: could be |
| 06:35.42 | VijayG | ok |
| 06:35.47 | shellshark | VijayG: not saying that for sure, but it sounds highly likely |
| 06:35.58 | VijayG | what else could be the possibility |
| 06:36.07 | VijayG | because .conf files are not an issue |
| 06:36.13 | VijayG | i have checked that 100's of times |
| 06:37.31 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:38.15 | VijayG | is there anything else that could be guessed |
| 06:38.28 | VijayG | as what the problem could be |
| 06:38.33 | VijayG | other than ISP |
| 06:38.39 | VijayG | i will check on that part |
| 06:38.53 | *** join/#asterisk compu73rg33k (n=compu73r@d53-64-54-188.nap.wideopenwest.com) |
| 06:40.27 | shellshark | VijayG: have you talked with your voip provider to see if they are even seeing any packets coming from your side? |
| 06:41.04 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 06:41.29 | VijayG | nothing is reaching them |
| 06:41.49 | shellshark | something along the way is dropping the packets then |
| 06:42.31 | shellshark | if you have no packet filter running on your end, and your voip provider is not filtering packets (i.e. only letting in certain IP's), then it has to be your ISP |
| 06:42.40 | shellshark | why can't you just run IAX? |
| 06:42.42 | VijayG | moreover the problem is that it is not even allowing to register |
| 06:43.00 | shellshark | right, as a register goes over port 5060 ;) |
| 06:43.10 | VijayG | one of the vendor is giving a very good price for UK dialing |
| 06:43.19 | VijayG | and he supports only sip |
| 06:43.33 | VijayG | thats why i am planning to switch over to sip |
| 06:43.33 | shellshark | what's his price? |
| 06:44.00 | VijayG | for UK he is offering me 1.2 cents |
| 06:44.01 | VijayG | at 6/6 |
| 06:44.37 | shellshark | you can get that from voxee.com over IAX |
| 06:44.53 | VijayG | i think they are offering at 30/6 |
| 06:45.03 | VijayG | we have good amount of calls going to voicemail |
| 06:45.23 | VijayG | so, do not want to pay for 30 seconds when it is getting detected in 3-4 seconds |
| 06:45.54 | *** join/#asterisk ToTo (n=ToTo@host203-49-dynamic.0-87-r.retail.telecomitalia.it) |
| 06:46.21 | VijayG | Also one more thing, i need to know |
| 06:46.24 | shellshark | well those are pretty much your choices ;) |
| 06:46.30 | VijayG | in 1.2.10 version |
| 06:46.48 | VijayG | i was using the g729 educatiional license |
| 06:46.50 | shellshark | go with voxee and work with IAX, or fight your ISP to unblock SIP |
| 06:46.58 | VijayG | ok, i will do that |
| 06:47.04 | shellshark | ah, the Intel IPP codec |
| 06:47.05 | VijayG | <PROTECTED> |
| 06:47.09 | VijayG | ya |
| 06:47.29 | VijayG | by just putting the .so file in right place |
| 06:47.36 | VijayG | now in 1.2.12.1 |
| 06:47.49 | VijayG | it seems it needs some .h or .c files as well |
| 06:48.13 | VijayG | do you think putting the .so file in right place will allow g729 transcoding on my server |
| 06:48.25 | VijayG | in 1.2.12.1 |
| 06:48.36 | VijayG | in 1.2.10 it was working fine |
| 06:49.01 | VijayG | but there was no concept of .c files in 1.2.10 like wise in 1.2.12.1 |
| 06:50.14 | shellshark | you may have to re-compile against the new source tree |
| 06:51.08 | VijayG | ok |
| 06:52.57 | Inverted | from within my perl script can I just say EXEC Playback filenametoplay |
| 06:54.05 | shellshark | Inverted: use Asterisk::AGI |
| 06:54.12 | Inverted | oic you have to use print |
| 06:54.29 | shellshark | Inverted: google for Asterisk::AGI |
| 06:54.38 | shellshark | Inverted: it will make your life so much simpler ;) |
| 06:54.52 | Inverted | shellshark: ive seen it, just trying to figure out the inner workings a bit, then I'll start using it :) |
| 06:55.29 | shellshark | as long as you've got the time to waste, i guess ;) |
| 06:55.51 | Inverted | you can tell from my perl code that I come from a C background |
| 06:56.00 | shellshark | hehe |
| 06:56.06 | Inverted | perl seems really sloppy to me |
| 06:56.08 | shellshark | I come from a perl backgroudn ;) |
| 06:56.19 | shellshark | well perl can be really structured, or really sloppy |
| 06:56.34 | shellshark | there are about 10 different ways to write the same thing |
| 06:56.45 | Inverted | i can see where it is great for manipulating strings. my background comes from programming microcontrollers, where it would be worthless |
| 06:56.47 | shellshark | so it all depends really on the coder |
| 06:56.56 | shellshark | right |
| 06:57.07 | shellshark | my area is AGI and CGI |
| 06:57.19 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 06:57.19 | shellshark | perl fills that niche quite well ;) |
| 06:57.44 | Inverted | ive dabbled a bit with php, for web stuff its nice |
| 06:57.57 | Inverted | but my friend swears by ruby |
| 06:58.21 | shellshark | php has no real data structures |
| 06:58.43 | shellshark | and i'm a big fan of code and content seperation, which is near impossible with php |
| 06:59.00 | shellshark | layout, content, and application should be completely seperate |
| 07:00.15 | shellshark | i've not had much exposure to ruby though |
| 07:00.48 | Inverted | i personally dont like the lack of structure of how perl code looks. one guys code can look drastically different from another's, and hence take more time to read. like how you can put the conditionals after the data |
| 07:01.13 | *** join/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 07:01.57 | shellshark | Inverted: right |
| 07:02.17 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 07:03.13 | shellshark | for a shop filled with perl guys though, it shouldnt be too much of a problem |
| 07:06.47 | *** join/#asterisk HaMYaI_ (n=HaMYaI@ppp-58.8.5.193.revip2.asianet.co.th) |
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| 07:14.18 | *** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 07:16.52 | *** part/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 07:20.09 | mosty | is there a way for an agent to join a queue other than via their phone? (eg from a web app or similar?) |
| 07:24.45 | *** join/#asterisk lorinc (n=ang@caracas-4802.adsl.interware.hu) |
| 07:35.33 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-58-55.dsl.tul2ok.sbcglobal.net) |
| 07:45.43 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 07:45.46 | SaTLaN32 | hi |
| 07:46.01 | SaTLaN32 | need help with mysql that don't work from asterisk |
| 07:46.11 | SaTLaN32 | i have the log here: http://pastebin.ca/196029 |
| 07:46.38 | SaTLaN32 | i preform few queries from the same db and the same table, all exept one are ok. |
| 07:48.05 | mosty | which line is the error? |
| 07:51.18 | SaTLaN32 | it start with the statement in line 33 |
| 07:51.21 | *** join/#asterisk packetman (n=trevor@d57-225-189.home.cgocable.net) |
| 07:51.24 | SaTLaN32 | select group from... |
| 07:51.32 | *** join/#asterisk oej (n=oej@193.213.21.232) |
| 07:54.03 | mosty | looks like you have multiple threads using the same connection, and mysql is not quite thread safe |
| 07:54.14 | *** join/#asterisk VijayG (i=vijay@202.131.145.234) |
| 07:54.18 | VijayG | hi |
| 07:55.14 | SaTLaN32 | so how do i fix this? to open a new connection for every query? |
| 07:55.18 | mosty | satlan32: see this page http://dev.mysql.com/doc/refman/5.0/en/threaded-clients.html |
| 07:56.57 | stephane_ | re |
| 07:59.04 | mosty | satlan32: i could be wrong, though |
| 07:59.09 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:59.26 | SaTLaN32 | i'll read it and try |
| 07:59.27 | SaTLaN32 | thanks |
| 08:00.20 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:01.45 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
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| 08:14.35 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
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| 08:18.46 | ksh | hi |
| 08:18.58 | VijayG | hi |
| 08:19.17 | ksh | my asterisk is behaving strange after updateing to 1.2.12.1 |
| 08:19.30 | ksh | its seems like a socket problem |
| 08:19.31 | VijayG | whats happenign |
| 08:20.16 | ksh | once the ast. runs, its fine. but after some time or a adsl reconnect it wont register with my sip peers and doesnt seem to accept packets anymore |
| 08:20.26 | ksh | im on BSD (OpenBSD 3.9) here. |
| 08:23.04 | ksh | i even noticed the asterisk tries to register by using the old Internet IP when sending REGISTER UDP packets |
| 08:23.04 | oliviert | hi guys |
| 08:23.33 | ksh | as i woke up today i found the usual state. which seems strange |
| 08:23.33 | oliviert | i do have on hold music working between my SIP phones extensions |
| 08:24.06 | ksh | Name/username Host Dyn Nat ACL Port Status |
| 08:24.06 | ksh | sipgate-out/xxxxxxx 217.10.79.9 N 5060 OK (160 ms) |
| 08:24.06 | ksh | 1001 (Unspecified) D N 0 UNKNOWN |
| 08:24.06 | ksh | 1000/1000 10.1.3.250 D N 5060 UNKNOWN |
| 08:24.06 | ksh | XXX*CLI> sip show registry |
| 08:24.08 | ksh | Host Username Refresh State |
| 08:24.11 | ksh | sipgate.de:5060 xxxxxxxx 120 Unregistered |
| 08:24.13 | ksh | XXX*CLI> iax2 show registry |
| 08:24.16 | ksh | Host Username Perceived Refresh State |
| 08:24.18 | ksh | 192.246.69.186:4569 xxxxxxxx 88.64.57.9:4569 60 Registered |
| 08:24.21 | ksh | 10:16:10.194954 88.x.x.x.sip-ksh > 217.10.79.9.sip: udp 595 |
| 08:24.23 | ksh | 10:16:10.280836 217.10.79.9.sip > 88.x.x.x.sip-ksh: udp 390 (DF) [tos 0x10] |
| 08:24.24 | oliviert | but, for an incoming call, coming from a zaptel analog line, if the sip phone put on Hold, the music is not played on the corresspondant analog phone, any idea ? |
| 08:24.52 | ksh | any ideas ? |
| 08:24.54 | oliviert | ksh: use a pastebot to insert your code: pastebin.ca is one |
| 08:25.03 | ksh | okay. |
| 08:31.28 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 08:32.01 | qdk | ksh: you sure that it isnt the SIP phone, which is stupid? |
| 08:32.19 | ksh | absolutly |
| 08:32.28 | ksh | the packets comming in |
| 08:32.43 | shellshark | ksh: you could make a phantom extension to go right to music on hold |
| 08:32.48 | ksh | but asterisk wont even notice them on "sip debug ip" or verbose and debug level 4 |
| 08:33.03 | qdk | shellshark: tell oliviert that? |
| 08:33.07 | ksh | shellshark: already did that. |
| 08:33.24 | shellshark | ksh: and calling it from the SIP phone you get music? |
| 08:33.24 | ksh | its just like asterisk wont accept packets anymore. |
| 08:33.36 | shellshark | what version of asterisk also |
| 08:34.15 | ksh | shellshark: no no, the sipphone is dead. packets are coming in, but asterisk doesnt respond to them. just like aster. wont receive them. im running 1.2.12.1 |
| 08:34.27 | ksh | that was the thing i posted above. |
| 08:34.43 | ksh | sip registry state is "unregistered" while the peer says its "OK" |
| 08:34.58 | qdk | ksh: btw. now that i think of it.. i had the exact same problem on my FreeBSD 6.1 with * 1.2.12... i see "pings" from the ITSP on my SIP account (using tcpdump) entered in my asterisk, but no reply from *... |
| 08:35.21 | ksh | same to my expention 1000 (10.1.3.250), this is the phone which is unregistered. |
| 08:35.44 | ksh | once i stop asterisk, not "restart now" and wait a couple of minutes and start it again |
| 08:35.46 | shellshark | sounds like your phone is braindead |
| 08:36.00 | ksh | it works. so the problem is just after a while |
| 08:36.05 | shellshark | have you tried rebooting your phone when this happens? |
| 08:36.08 | ksh | shellshark: no, belive me, its the asterisk |
| 08:36.22 | ksh | i even got my softphones here. all timeout |
| 08:36.38 | shellshark | network packet throttling? |
| 08:36.40 | qdk | ksh: I had a HARD TIME just placing a call out.. softphone -> asterisk -> ITSP (using SIP) |
| 08:36.55 | shellshark | QoS problem? |
| 08:37.34 | ksh | sec.. phone |
| 08:38.11 | qdk | shellshark: strange suggestions... we are both telling you that there is NO traffic at all going out. |
| 08:39.04 | shellshark | qdk: no, he's saying there is traffic going, but after a while it stops |
| 08:39.10 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 08:40.13 | shellshark | so i'm lead to believe that perhaps there is some QoS / packet throttling going on, and perhaps the switch is shutting down traffic on that given port or port range to the destination switch port |
| 08:40.18 | shellshark | i've seen it done before |
| 08:40.28 | shellshark | not that it's highly likely, but it's possible |
| 08:45.36 | qdk | shellshark: tcpdump looks at all the traffic touching the interface on the machine, so whatever the port in the switch does isnt relevant until some packets are seen in tcpdump. |
| 08:46.15 | shellshark | qdk: if the packets are dropped on the switch you wouldnt see them with tcpdump silly ;) |
| 08:46.32 | shellshark | switch being network switch in my context, not talking about "phone" switch |
| 08:46.38 | shellshark | maybe you got confused |
| 08:47.29 | ksh | okay, sorry. |
| 08:47.35 | ksh | there is indeed QoS going on |
| 08:47.53 | ksh | but it doesnt matter if i disable it, the same problem relies. |
| 08:48.15 | shellshark | odd |
| 08:48.16 | ksh | as the QoS is just on the external interface and the clients are experiencing the same issues on the local network |
| 08:48.26 | shellshark | gotcha |
| 08:48.31 | qdk | shellshark: ksh shellshark: no no, the sipphone is dead. packets are coming in, but asterisk doesnt respond to them. just like aster. wont receive them. im running 1.2.12.1 |
| 08:49.11 | ksh | qdk: could you solve your problems ? |
| 08:49.33 | shellshark | ksh: like i said, it's not highly likly, but it was worth looking into |
| 08:49.40 | shellshark | likely* |
| 08:50.45 | VijayG | ShellShark, you remember, we discussed the same problem around 2 hours back |
| 08:51.05 | VijayG | and you were telling the problem could be that ISP is blocking 5060 port |
| 08:51.09 | VijayG | or SIP |
| 08:51.23 | ksh | :) |
| 08:51.38 | shellshark | VijayG: the problem I discussed with you was regarding the g729 codec after you upgraded asterisk, and your ISP blocking _all_ SIP traffic |
| 08:51.42 | shellshark | VijayG: this is different |
| 08:51.57 | VijayG | G729 codec is different thing |
| 08:52.07 | shellshark | both are different than his issue |
| 08:52.10 | VijayG | my problem was the calls are going out through IAX |
| 08:52.19 | shellshark | ksh: are your phones local to the asterisk box? |
| 08:52.19 | VijayG | not through SIP |
| 08:52.27 | qdk | ksh: im not sure our problem is a the same place, but im almost sure that its related... my problem is dialing in/out through an external ITSP where i have a SIP account... if i use monitoring on my asterisk, asterisk sees the connections as down, though i can dial OUT if i dont monitor that account... never got it to dial in, coz i didnt wanna debug an potentiel limit at my ITSP, so im working on a new setup where i have 100% control and the test some more... |
| 08:52.27 | qdk | <PROTECTED> |
| 08:52.28 | VijayG | not even getting registered |
| 08:53.14 | shellshark | VijayG: right, he's getting some packets through, but after some time, it stops responding |
| 08:53.22 | shellshark | VijayG: so it's completely different issue |
| 08:53.28 | VijayG | ok |
| 08:55.12 | ksh | okay, lets see. altq is removed. |
| 08:55.16 | ksh | shellshark: phones are local |
| 08:55.54 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 08:55.54 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 08:56.25 | shellshark | ksh: so Internet uncertainties are really out of the equation ;) |
| 08:57.12 | ksh | qdk: you cant receive calls ? but can dial out ? |
| 08:57.25 | qdk | ksh: if you JUST upgraded from a version where it worked, why are you changing everything else? |
| 08:57.29 | qdk | ksh: yes |
| 08:57.54 | ksh | shellshark: think so |
| 08:58.13 | ksh | qdk: i c. sip is working, or is RTP the problem ? |
| 08:58.14 | qdk | ksh: well the call i placed giong out died shotrly after the call was active in both ends. |
| 08:58.29 | ksh | ahhh. that seems familiar to me |
| 08:58.37 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 08:59.09 | qdk | ksh: and the tcpdump went crazy with something like no route to host or something like that (cant recall precise). |
| 08:59.10 | ksh | qdk: im changing altq, because its known to make trouble for me. |
| 08:59.34 | qdk | ksh: ok, i had no OS QoS or fw enabled. |
| 09:00.19 | ksh | qdk: okay. you need to know exactly where thi no route to host comes from. this basically cant be |
| 09:00.21 | qdk | it was a fresh installed freebsd 6.1 with recompiled kernel and world... all inside a VMware server. |
| 09:00.30 | ksh | or you are sending internal packets over the internet |
| 09:00.39 | shellshark | ksh: you are using openbsd? |
| 09:00.42 | ksh | and the gateway refuses by no route to host |
| 09:00.44 | qdk | ksh: well some ICMP errors. |
| 09:00.50 | ksh | shellshark: yes |
| 09:01.08 | shellshark | well you guys know that asterisk is best suited for linux right? |
| 09:01.14 | ksh | sure |
| 09:01.23 | ksh | but it worked before. |
| 09:01.36 | shellshark | so some problems are to be expected on bsd's, because they are not as well tested |
| 09:01.49 | shellshark | right, it worked before, but you upgraded, correct? :P |
| 09:01.56 | ksh | yep |
| 09:02.20 | ksh | it just makes no sense, that it works and then suddenly stops |
| 09:02.43 | ksh | qdk: i would guess a NAT problem |
| 09:03.04 | ksh | the vmware is bridged, right ? did you use "nat=yes" and set the externalip=.... |
| 09:03.17 | shellshark | ksh: i think your issue may be a bug with the version you upgraded to, probably specific to bsd |
| 09:03.32 | *** join/#asterisk pph (n=pph@81.255.164.157) |
| 09:03.43 | shellshark | ksh: nat=always seems to work better than nat=yes |
| 09:04.25 | qdk | ksh: NAT is a problem going in... this is out.. AND on the machine with the service and no NAT. |
| 09:04.52 | qdk | ksh: no i use vmware with "physical" interfaces. |
| 09:05.36 | ksh | always, cant recall this option |
| 09:05.50 | qdk | ksh: anyway, im doing my setup in a non NAT setup on a physical machine using * 1.4, so i see how it works there. |
| 09:06.04 | qdk | ksh: 2 sec.. ill fire up my vmware. |
| 09:06.13 | ksh | qdk: i would also suggest this. use a bridged network |
| 09:06.43 | ksh | qdk: tcpdump on the host and guest side for sip packets and their source and destination adress |
| 09:06.57 | ksh | ensure there is absoluty no problem there |
| 09:06.59 | qdk | ksh: sorry, it IS bridged which is what i also meant by "physical". |
| 09:07.07 | ksh | ah, okay. |
| 09:07.21 | qdk | no NAT or some crap like that. |
| 09:08.07 | qdk | and i will NEVER look towards a problem being NAT-related, when i get packets IN and non out. |
| 09:08.24 | ksh | wonderful. anyways, make sure "canreinvite=no" |
| 09:08.54 | ksh | dont say that |
| 09:09.34 | ksh | im suggesting this, because asterisk can send raw packets by specifing a externalip=... |
| 09:09.36 | qdk | no matter what options you have or do not have, there should always be some kind of reply to the request, unless a firewall drops the packets. |
| 09:10.17 | ksh | indeed. but the firewall would drop the packet if its comming from internal |
| 09:10.38 | ksh | just try to figure out which ip address is used when sending them to you ITSP |
| 09:12.16 | qdk | ksh: well im not really looking to sovle any problems here. If BSD keep giving me a hard time in a controlled setup, ill just switch back to Linux again. |
| 09:13.09 | shellshark | qdk: linux is best when working with asterisk |
| 09:13.23 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 09:13.32 | qdk | shellshark: define 'best'? |
| 09:13.53 | VijayG | :) |
| 09:14.33 | shellshark | qdk: asterisk was designed to work with linux best, per Mark Spencer |
| 09:14.41 | shellshark | qdk: ask him why ;) |
| 09:14.58 | shellshark | (you can talk to him on instant messanger easy enough) |
| 09:15.58 | shellshark | i need sleep... |
| 09:16.00 | shellshark | night all |
| 09:16.13 | ksh | nite |
| 09:16.17 | *** join/#asterisk don_martino (n=martin@e181030031.adsl.alicedsl.de) |
| 09:16.37 | ksh | im also out. later |
| 09:16.38 | *** join/#asterisk pph (n=pph@81.255.164.157) |
| 09:16.55 | *** join/#asterisk jgoo (n=e4b80e21@87.203.103.190) |
| 09:16.56 | VijayG | :B |
| 09:17.02 | don_martino | hello world! i have a more general question about voip. how much bandwidth should i reserver for voip? |
| 09:17.03 | jgoo | hey guys, anyone have expeirence with pstn simulators? |
| 09:17.11 | r0d3nt|m | and 80% of all vm's will produce crappy asterisk audio streams.. although I haven't tested the latest versions of asterisk.... |
| 09:18.09 | shellshark | don_martino: there is a nice VoIP bandwidth calculator out there that cisco made... it depends on what codec you're using |
| 09:18.59 | shellshark | don_martino: cisco's calculator figures in IP, UDP, and RTP overhead for you also, which is pretty handy in getting a true idea of real-world bandwidth |
| 09:19.08 | don_martino | shellshark: thanks. i'll ask google now :-) |
| 09:19.18 | shellshark | don_martino: it's free to use, and it's the most comprehensive calculator i've seen to date |
| 09:19.23 | shellshark | web-based too |
| 09:19.29 | shellshark | :) |
| 09:19.31 | *** join/#asterisk pph (n=pph@81.255.164.157) |
| 09:21.19 | *** join/#asterisk tillo (i=tillo@85-218-17-14.dclient.lsne.ch) |
| 09:22.39 | jgoo | I have a virtual console, I am guessing the setup, which is already on there, is config'd for US, so I should just change my default zone to US in the configs... |
| 09:22.55 | don_martino | hhmmmm i wonder if womens voices require more bandwidth ... |
| 09:23.30 | shellshark | don_martino: depends on if you're doing silence suppression or not i would imagine |
| 09:24.00 | don_martino | lol |
| 09:24.51 | jgoo | don_martino: because they talk more? |
| 09:24.55 | jgoo | XD |
| 09:25.25 | don_martino | jgoo: because their voice is more colorful (i mean frequencies, not choice of words) |
| 09:26.15 | jgoo | oh, they *are* more colourful though ;-) |
| 09:26.20 | don_martino | so in theory mens voices can be compressed more efficently |
| 09:27.06 | don_martino | (leaving out the fact that men already leave out a lot of redundancy already) |
| 09:27.15 | jgoo | yeah, I guess if they have more entropy, females will be tougher to compress. So you should hire Barry White types to work at your call center |
| 09:27.37 | jgoo | relax and arouse your clientelle whilst lowering your system requirements |
| 09:27.40 | jgoo | eeeuch, that sounded sick |
| 09:27.44 | don_martino | and saving money on traffic! |
| 09:27.49 | MrChimpy | jgoo: whateve floats your boat sister |
| 09:28.06 | jgoo | that is why 900 numbers are so expensive... someone has to pay the bandwidth! |
| 09:28.26 | don_martino | which prooves my theory :-) |
| 09:31.20 | jgoo | so, barry white on hold music is cost effective, and lulls your angry customers who have been waiting. kerching, write that in the book! ;-) |
| 09:31.46 | jgoo | whereas celine dion kills your network, and makes your customers want to kill you |
| 09:32.25 | MrChimpy | "celine dion kills" is a good cover all. |
| 09:32.33 | don_martino | and true |
| 09:34.31 | don_martino | HA! i found a calculator and i knew the recommended 256kbit for 2 simultaneous connections is way too much |
| 09:35.53 | r0d3nt|m | well for ULAW/ALAW not entirely too much.. |
| 09:36.08 | don_martino | 'scuse me? |
| 09:36.12 | *** join/#asterisk ZX81 (n=ZX81@62-101-126-239.ip.fastwebnet.it) |
| 09:36.29 | ZX81 | grrrrrrr Install ncurses to use the menu interface! |
| 09:36.43 | ZX81 | I have installed every single package that relates in any way to ncurses |
| 09:36.57 | ZX81 | dev, dbg, bin, base, etc etc |
| 09:37.00 | ZX81 | 4,5 |
| 09:37.03 | ZX81 | everything |
| 09:37.26 | ZX81 | stupid debian |
| 09:38.26 | qdk | ZX81: will you get to the part about your problem? |
| 09:38.47 | ZX81 | :) |
| 09:38.55 | ZX81 | can't make menuconfig on zap 1.4 |
| 09:39.06 | ZX81 | webpage here says: http://www.howtoforge.com/linux_vserver_debian |
| 09:39.11 | ZX81 | ncurses-base, and libncurses5-dev: Needed for "make menuconfig" when compiling kernel. |
| 09:39.21 | ZX81 | but both are installed and latest version |
| 09:39.27 | qdk | don_martino: there is usually no jump between 128kb and 256kb, and 128kb is not enough for 2 calls using G711. |
| 09:39.31 | don_martino | r0d3nt|m: i don't know a whole lot about voip. i just know there are 2 people coming on thursday who will mount 2 ugly boxes and try to establish a connection between those two boxes over the internet. IPSEC will fail badly. PPTP is too insecure for my taste, so I set up an SSL-Tunnel for them through which they may do as pleases them most. for voip do work through this tunnel i must reserve some bandwidth for it and 256 of 768 is more t |
| 09:39.31 | don_martino | want to share |
| 09:39.39 | qdk | ZX81: make menuselect? |
| 09:39.41 | ZX81 | as well as every other package with ncurses somewhere in the title |
| 09:39.42 | ZX81 | yeah |
| 09:39.45 | ZX81 | zaptel 1.4 |
| 09:39.55 | ZX81 | heh |
| 09:40.04 | ZX81 | soz menuselect |
| 09:40.07 | qdk | ZX81: you wrote menuconfig... its not your kernel. |
| 09:40.08 | ZX81 | keep typing menuconfig |
| 09:40.09 | ZX81 | :) |
| 09:40.12 | ZX81 | yeah I know |
| 09:40.15 | don_martino | qdk: my problem is that i don't know which compression these 2 boxe use. so i have to think worst case |
| 09:40.15 | ZX81 | I get them mixed up |
| 09:40.30 | ZX81 | but make menuconfig always says no target for menuconfig |
| 09:40.30 | ZX81 | :) |
| 09:40.40 | ZX81 | so then I remember to do menuselect |
| 09:40.40 | qdk | don_martino: and so does you calc i guess. |
| 09:40.41 | ZX81 | :) |
| 09:41.18 | ZX81 | http://pastebin.ca/196100 |
| 09:41.19 | ZX81 | :) |
| 09:41.21 | r0d3nt|m | i've ran voip over many vpn's, including IPSEC quite effectively.. with cisco's and netgear's and software products.... |
| 09:41.23 | r0d3nt|m | don_martino, |
| 09:41.27 | qdk | ZX81: another time remember to poste what you wrote and the actuall error. |
| 09:41.33 | ZX81 | :) |
| 09:41.37 | ZX81 | read the pastebin |
| 09:41.42 | ZX81 | for enhanced accuracy |
| 09:41.43 | ZX81 | :) |
| 09:41.50 | don_martino | qdk: the calculator i found offers me G.711 PC; 64kbps uncompressed with 10ms packet duration as worst case and this comes to 192kbit for 2 voice paths |
| 09:41.57 | ZX81 | r0d3nt: OpenVPN, UDP works well too |
| 09:42.37 | ZX81 | qdk: I've even done make clean, ./configure, make menuselect |
| 09:42.43 | ZX81 | but same schtuff |
| 09:42.43 | qdk | ZX81: ok, look at your paths. I had to make a fix for BSD installation due to missing PATHs. |
| 09:42.52 | ZX81 | ok |
| 09:43.02 | ZX81 | i.e. read the makefile |
| 09:43.02 | don_martino | r0d3nt|m: oh the problem is not IPSEC itseld. it's the capabilities of these 2 ugly boxes and and the existing infrastructure |
| 09:43.07 | r0d3nt|m | ZX81, i'm glad you know, it, as i mentioned, some software products... openvpn is one of them, and tinc... |
| 09:43.20 | r0d3nt|m | don_martino, i understand |
| 09:44.03 | r0d3nt|m | also done some voip over ssh tunnels |
| 09:45.02 | ZX81 | hmmm config.log under menuselect has conftest.c:61:20: error: curses.h: No such file or directory |
| 09:45.21 | r0d3nt|m | gentoo ftw ! |
| 09:45.23 | don_martino | r0d3nt|m: i also thought about that but how would i route gre through an ssh tunnel |
| 09:45.56 | tillo | I've a B410P card. mISDN says "<3>mISDN: INTERNAL ERROR in /cross/tmp/misdn/drivers/isdn/hardware/mISDN/stack.c:1187" and "<4>mISDNd: addr(f0000) prim(f1980) failed err(-92)". does anyone have an idea of what's going on? card leds are blinking ininterruptly. http://www.rafb.net/paste/results/5kRMlr43.html is the complete kernel log. |
| 09:46.01 | don_martino | r0d3nt|m: don'T forget, i can't route voip protocls, i have to route ipsec or pptp |
| 09:46.03 | r0d3nt|m | i never did that... gre over ssh tunnel |
| 09:46.14 | r0d3nt|m | why ? |
| 09:46.16 | qdk | ZX81: does it work without using menuselect? |
| 09:46.44 | don_martino | r0d3nt|m: i because these 2 ugly boxes must talk to each other. and i don't have control over them (which is what makes them so ugly) |
| 09:46.55 | r0d3nt|m | aye sir |
| 09:47.05 | ZX81 | qdk: yeah, but no hardware so I wanted to turn off some of the modules |
| 09:47.09 | ZX81 | strange |
| 09:47.09 | r0d3nt|m | well you'll be building your network around those boxes than |
| 09:47.10 | qdk | ZX81: that sound much like a $PATH issue. |
| 09:47.10 | ZX81 | http://pastebin.ca/196101 |
| 09:47.23 | ZX81 | it finds initscr in lcurses |
| 09:47.34 | ZX81 | but then loses lcurses on the next test |
| 09:47.46 | qdk | ZX81: turn off or not compile them? there is a big difference. |
| 09:47.50 | *** join/#asterisk korty (n=k@196.211.12.66) |
| 09:48.05 | ZX81 | not compile |
| 09:48.05 | ZX81 | :) |
| 09:48.45 | ZX81 | maybe the second test should have -lcurses as well no? |
| 09:48.47 | qdk | ZX81: sorry, but why waste time? its a lot faster just to compile it all, and get on with your business. |
| 09:48.55 | r0d3nt|m | ya |
| 09:49.12 | ZX81 | or, fix the bug and submit a bugtrack entry so others can use it too |
| 09:49.13 | ZX81 | :) |
| 09:49.21 | don_martino | r0d3nt|m: but i found a rough scematic of these boxes. i will have to check my contract and possible warranty seals. the way this looks these 2 boxes are nothing but an embedded linux with asterisk and there is a CF-Card in the upper right corner of the board. i hope i can get my hands on it :-) |
| 09:49.40 | qdk | ZX81: there is _no_ performance impace on have all the modules compiled. Just dont load those you dont use. |
| 09:49.42 | r0d3nt|m | don_martino, neat |
| 09:50.02 | ZX81 | yeah I know |
| 09:50.13 | korty | hi all... can anybody help me? i have a gentoo installed machine with Asterisk 1.2.12.1 and Zaptel 1.2.9.1, running kernel 2.6.17-gentoo-r8... I have a couple of ISDN's that run perfectly in their, but trying to start the tdm04b, 4 port fxo card, as soon as i do a modprobe wctdm, i get wcfxs FATAL, module not found... can anybody help?? |
| 09:50.18 | ZX81 | but want a small binary for asterisk |
| 09:50.26 | r0d3nt|m | ZX81, and why would think it's a bug ? |
| 09:50.36 | qdk | ZX81: ok, then go through Makefile and configure and figure out why i cant find the header file. |
| 09:50.43 | ZX81 | yeah all good |
| 09:50.45 | ZX81 | doing it now |
| 09:51.00 | ZX81 | heh |
| 09:51.12 | don_martino | korty: did you do a depmod -a after compiling the module |
| 09:51.12 | ZX81 | make clean doesn't clean /menuselect |
| 09:51.17 | qdk | ZX81: im pretty sure that you either dont have it, or its in the "wrong" place. |
| 09:51.20 | ZX81 | make distcl;ean does |
| 09:51.25 | ZX81 | *distclean |
| 09:51.32 | ZX81 | all good |
| 09:51.33 | ZX81 | fixed now |
| 09:51.40 | qdk | ZX81: good. |
| 09:51.42 | korty | don_martino, cant say i did... |
| 09:51.49 | ZX81 | qdk: yes |
| 09:51.50 | ZX81 | :) |
| 09:52.06 | don_martino | what's the path of your tdm04b module? |
| 09:52.10 | ZX81 | works fine now |
| 09:52.10 | ZX81 | :D |
| 09:52.17 | ZX81 | ~adn |
| 09:52.24 | jbot | i guess adn is is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 09:52.29 | korty | don_martino, it is gentoo, so it inserts everything for you in the correct place, i cant even find the module wcfxs in /lib/modules... |
| 09:52.40 | korty | don_martino, one sec... |
| 09:52.45 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:53.22 | korty | don_martino, /lib/modules/2.6.17-gentoo-r8/misc |
| 09:53.35 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 09:55.15 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:55.32 | don_martino | korty: interesting: go to that directoy and try "insmod <filename>" then |
| 09:55.50 | korty | don_martino, have a look at my ls... |
| 09:55.57 | korty | cwain.ko torisa.ko wct4xxp.ko wcusb.ko ztd-eth.ko |
| 09:55.58 | korty | pciradio.ko wcfxo.ko wctdm.ko xpp ztd-loc.ko |
| 09:55.58 | korty | qozap.ko wct1xxp.ko wctdm24xxp.ko zaphfc.ko ztdummy.ko |
| 09:55.58 | korty | tor2.ko wct4xxp wcte11xp.ko zaptel.ko ztdynamic.ko |
| 09:56.06 | ZX81 | I posted a quick note so that it appears in google the next time someone searches |
| 09:56.08 | ZX81 | D |
| 09:57.02 | korty | asterisk misc # modprobe wctdm |
| 09:57.02 | korty | FATAL: Module wcfxs not found. |
| 09:57.23 | *** part/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:57.25 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:57.36 | backblue | korty: modprobe -l| grep wctdm |
| 09:57.36 | korty | and i have had this card working on zaptel 1.2.6 before they changed the wcfxs to wctdm |
| 09:57.54 | backblue | dont do ls, probably you are not on the correct path |
| 09:57.56 | korty | asterisk misc # modprobe -l | grep wctdm |
| 09:57.56 | korty | /lib/modules/2.6.17-gentoo-r8/misc/wctdm24xxp.ko |
| 09:57.56 | korty | /lib/modules/2.6.17-gentoo-r8/misc/wctdm.ko |
| 09:58.01 | don_martino | korty: it has to be in /lib/modules/`uname -r`. did you emerge your driver or did you type "make" yourself? |
| 09:58.07 | backblue | korty: depmod -ea |
| 09:58.15 | backblue | korty: modprobe -vvv wctdm |
| 09:58.22 | *** join/#asterisk Benj0 (n=BenjO@lev92-1-81-57-180-205.fbx.proxad.net) |
| 09:58.44 | ZX81 | backblue or lsmod |grep wctdm |
| 09:58.45 | ZX81 | :) |
| 09:58.46 | korty | don_martino, emerged zaptel with my new kernel of course, so it compiled it for the correct kernel... dont worry, i aint a dumbass.. |
| 09:58.46 | Benj0 | hi |
| 09:58.58 | backblue | whatever |
| 09:59.08 | korty | Linux asterisk 2.6.17-gentoo-r8 #1 Thu Oct 5 19:30:01 SAST 2006 i686 Pentium III (Coppermine) GNU/Linux |
| 09:59.13 | backblue | lsmod it's only for loaded modules. |
| 09:59.14 | jgoo | oooh, I am in CLII mode, and I get lots of == Connect attempt from '127.0.0.1' unable to authenticate |
| 09:59.33 | korty | asterisk misc # modprobe -vvv wctdm |
| 09:59.33 | korty | FATAL: Module wcfxs not found. |
| 09:59.44 | backblue | wcfxs |
| 09:59.54 | backblue | modprobe -l wcfxs |
| 10:00.00 | backblue | modprobe -l|grep wcfxs |
| 10:00.14 | don_martino | korty: not using portage does not make you dumb. but it seems odd that portage didn't build all your modules |
| 10:00.18 | korty | asterisk misc # modprobe -l wcfxs |
| 10:00.18 | korty | asterisk misc # modprobe -l | grep wcfxs |
| 10:00.18 | korty | asterisk misc # |
| 10:00.29 | backblue | so, you found your problem. |
| 10:00.32 | korty | but if i download zaptel from the site.... |
| 10:00.42 | korty | there is NO wcfxs module.. |
| 10:00.45 | korty | no such thing.. |
| 10:00.56 | backblue | it has not compiled that module |
| 10:00.57 | don_martino | korty: can't help you then i thought it's a problem with how you did your installation but it seems it's a problem of what you install |
| 10:00.58 | korty | only in older versions as they have replaced it with wctdm |
| 10:01.24 | backblue | dont know then, dont use that |
| 10:01.27 | korty | backblue, i know, but i look in source, and that module does not exsist... |
| 10:01.36 | korty | i have aliased it modules.conf as well... |
| 10:01.37 | korty | :( |
| 10:01.49 | tillo | maybe an alias or something in modules.d? |
| 10:01.56 | korty | tried |
| 10:02.12 | backblue | remove the alias, and try again |
| 10:02.24 | korty | one sec.. |
| 10:02.50 | korty | #alias wcfxs wctdm |
| 10:02.51 | korty | alias wct2xxp wct4xxp |
| 10:02.59 | korty | asterisk etc # modprobe wctdm |
| 10:03.00 | korty | FATAL: Module wcfxs not found. |
| 10:03.23 | korty | it is like when they converted to wctdm, they forgot to rename is in the actual module... |
| 10:03.37 | backblue | check the source |
| 10:03.48 | backblue | it has to be wcfxs somewhere |
| 10:03.58 | tillo | grep for wcfxs in the source and in the modules.conf/modules.d |
| 10:04.01 | tillo | exactly |
| 10:04.14 | korty | like i said, i have search and what i can find is that they replaced wcfxs with wctdm... |
| 10:04.50 | backblue | check again, i think it has something there |
| 10:04.52 | korty | download the latest zaptel, and i guarentee you wont find that module in it... |
| 10:04.52 | korty | nuffin... |
| 10:05.00 | korty | well.. nuffin yet... |
| 10:05.02 | korty | hehe... |
| 10:05.04 | korty | let me check.. |
| 10:05.05 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 10:05.13 | don_martino | it is odd that modules.symbols would list a not installed module... you COULD (i'm not saying it's the best way to solve your problem) search modules.symbols for all symbols exported by your unpresent module, then go to the source and figure out which module is supposed to export those symbols |
| 10:05.15 | ZX81 | wcfxs should be an alias for wctdm |
| 10:05.44 | backblue | korty: do your System.map updated and all of that? |
| 10:06.14 | backblue | some of the modules wctdm it's trying to load, needs wcfxs i think |
| 10:06.46 | *** join/#asterisk adorah (n=admin@84.94.127.28.cable.012.net.il) |
| 10:07.58 | korty | YES, i found it....!!!!!! |
| 10:08.08 | korty | did a grep in etc... |
| 10:09.07 | don_martino | and what did that grep show you? |
| 10:09.12 | korty | took the alias out of, modprobe.conf, modules.conf and modules.d/zaptel, and i do a modprobe wctdm, and it finds it... |
| 10:09.22 | don_martino | cool |
| 10:09.38 | don_martino | i can stop emerging it now then? :-) |
| 10:09.40 | korty | #alias wctdm wcfxs |
| 10:09.46 | korty | just took that out everywhere and it works... |
| 10:09.57 | korty | interesting... |
| 10:09.59 | Benj0 | Somone knows if RTP must pass throught asterisk ? |
| 10:10.00 | korty | yes, thanx don_martino ... |
| 10:10.08 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 10:10.10 | korty | perhaps i should post this up as a bug or something... |
| 10:10.16 | don_martino | yes |
| 10:10.21 | korty | or should i leave it in your guys hands... |
| 10:10.27 | don_martino | or you write a patch which corrects the installation |
| 10:10.51 | backblue | korty: that was what i have sayed you to do. |
| 10:11.27 | korty | lol |
| 10:11.29 | korty | hehe... |
| 10:11.35 | korty | backblue, :) thanx |
| 10:11.49 | korty | yeah... just not to include that alias... |
| 10:11.49 | tillo | same thing here |
| 10:12.01 | korty | an alias for wctdm and not for wcfxs... |
| 10:12.13 | korty | ok, so like where can i post this error up... or something? |
| 10:12.14 | tillo | in change, now you will resolve my problem |
| 10:12.25 | tillo | http://www.rafb.net/paste/results/5kRMlr43.html |
| 10:13.32 | don_martino | tillo: what is that? your isdn driver going haywire? |
| 10:15.27 | don_martino | oh great! somebody just pulled a plug somethere in the other office ... |
| 10:15.45 | tillo | well the card is new and I don't know how to work with it at all, and I have to install it before other technicians can configure asterisk... but the card is blinking |
| 10:15.53 | tillo | and I don't think blinking is a good sign |
| 10:16.01 | don_martino | wow my phone still isn't ringing ... |
| 10:16.19 | tillo | don_martino: prepare the excuse list |
| 10:16.35 | tillo | "We ran out of dial tone and we're and waiting for the phone company to deliver another bottle." |
| 10:16.51 | jgoo | okidoki folks, I have a pstn sim, I am dialling into it, and it calls out on a line, I HEAR it ringing (which must be from asterisk) but asterisk -vvvvvvvvvr shows nothing |
| 10:16.53 | korty | don_martino, dude, posting it up on asterisk/digium forum =P |
| 10:16.53 | jgoo | is that right? |
| 10:16.59 | don_martino | tillo: so you don't even know if anything is broken? btw: here goes my phone gtg |
| 10:17.02 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:17.30 | *** part/#asterisk don_martino (n=martin@e181030031.adsl.alicedsl.de) |
| 10:25.36 | *** join/#asterisk Cyt (n=danielcy@85.75.176.202) |
| 10:28.56 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 10:29.48 | Cyt | Please, what this message means? -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from 66.35.63.247 |
| 10:31.22 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 10:33.27 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 10:33.59 | *** join/#asterisk batitrak (i=fallen@helios.edu.uoc.gr) |
| 10:34.24 | batitrak | hey there...anyone can help me with pf? |
| 10:36.00 | *** part/#asterisk bun-bun (n=bun@unaffiliated/bun-bun) |
| 10:37.15 | qdk | batitrak: not really the place for it, but fine by me, now that i know pf. ;-) |
| 10:39.50 | *** join/#asterisk dovid (n=dovi5988@barak.cellcom.co.il) |
| 10:44.30 | *** join/#asterisk Ebola (n=Ebola@host81-132-187-57.range81-132.btcentralplus.com) |
| 10:53.15 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 10:57.53 | stephane_ | re |
| 10:58.20 | dovid | re ? |
| 10:59.51 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-248-212.telkomadsl.co.za) |
| 11:02.34 | *** join/#asterisk thibault_ (n=thibault@pai34-3-82-231-33-117.fbx.proxad.net) |
| 11:02.54 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 11:06.43 | *** join/#asterisk Dovid (n=dovi5988@barak.cellcom.co.il) |
| 11:12.05 | bXi | anybody around who knows visdn? |
| 11:12.20 | backblue | visdn should not be used yet. |
| 11:12.53 | bXi | why not if i may ask? |
| 11:13.03 | Dovid | can anyone explian what it is |
| 11:13.09 | Dovid | VISDN that is |
| 11:13.15 | backblue | because its not finish |
| 11:13.22 | backblue | it does not suport much things yet |
| 11:13.26 | backblue | use mISDN |
| 11:13.43 | backblue | you have 3 options, misdn bristuff visdn |
| 11:13.43 | bXi | isnt misdn just for connecting a phone to the isdn card? |
| 11:13.46 | backblue | i use misdn for all |
| 11:13.49 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 11:13.54 | backblue | misdn it's for everything |
| 11:14.03 | backblue | E1 cards |
| 11:14.06 | backblue | octo-bri cards |
| 11:14.06 | bXi | hmmm ok |
| 11:14.08 | bXi | BUT |
| 11:14.14 | bXi | visdn works good so far |
| 11:14.25 | bXi | cept for when i call the number assigned to my isdn connection |
| 11:14.28 | bXi | then i get this |
| 11:14.29 | backblue | well, i dont use it, because it dont gives me what misdn gives |
| 11:14.30 | bXi | Oct 10 13:13:31 WARNING[10274]: file.c:587 ast_readaudio_callback: Failed to write frame |
| 11:15.34 | bXi | on the phone i'm calling from i get a number does not respond |
| 11:15.41 | bXi | but asterisk does see its getting a call |
| 11:37.00 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:40.38 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:41.03 | *** join/#asterisk VijayG (i=vijay@202.131.145.234) |
| 11:42.35 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 11:45.19 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:45.29 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 11:46.10 | blitzrage | morning all. Anyone use the ChanIsAvail() app in 1.2? For some reason it doesn't fall through to the next priority when the end point is not registered. I have autofallthrough=yes as well. |
| 11:46.55 | Dovid | i am using it in 1.2.9 and works fine |
| 11:47.07 | Dovid | r u uisng n+101 ? |
| 11:47.14 | hwt | how do i rewrite the SIP from? |
| 11:47.38 | hwt | in a new call leg from the dialplan. |
| 11:48.01 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:48.08 | Dovid | hwt: dont understand the question |
| 11:48.50 | Dovid | u wana rewrite the CID ? |
| 11:49.00 | hwt | when a number has call forwarding on, i set up a new call leg in the dialplan. |
| 11:49.23 | hwt | Dovid: well, i'd prefer to just send out a new invite with another A-number in the SIP from. |
| 11:50.03 | hwt | (a new call leg to the number that the subscriber has forwarded it to) |
| 11:50.36 | Dovid | im lost. to techincal for me. try to explain a lil more spimpler |
| 11:51.02 | Dovid | what r u tryin to get end result. if what is set then to do what ? |
| 11:51.04 | blitzrage | Dovid: no.......... I'm not using priority jumping since that is waaaaaaay old |
| 11:51.43 | hwt | ok, let me ask the question in another way: how do you guys do call forwarding in the dialplan? |
| 11:51.45 | Dovid | blitzrage: cant teach an old dog new tricks. so what are you trying to do ? i am allways up for learning |
| 11:52.01 | blitzrage | Dovid: sorry, don't really have time to teach right now :) |
| 11:52.05 | Dovid | hwt: meaning that if they arent avail to send it else where |
| 11:52.31 | Dovid | blitzrage: wasnt looking for a lesson just what function r u tryin to use. i like to learn new ones and google em |
| 11:52.32 | hwt | Dovid: yeah, either that or if the subscriber has redirected all call to e.g. his cell phone. |
| 11:52.38 | blitzrage | hwt: set var in DB, check that before placing call to end point, if active, forward call to forwarding number instead of end point |
| 11:52.46 | blitzrage | Dovid: ChanIsAvail() |
| 11:53.00 | hwt | blitzrage: yeah, but then the A-number will still be the number that dialed in. |
| 11:53.05 | Dovid | hwt: i believe that if it is done on the phone that it is set in the astdb |
| 11:53.35 | hwt | blitzrage: and when billing the subscriber, we will try billing a number outside our network. |
| 11:53.38 | Dovid | blitzrage: all i know for chanisavail is n+101 i guess i have some reading to do |
| 11:53.58 | Dovid | hwt: u may wana write ur own AGI |
| 11:54.25 | hwt | Dovid: uh, i think not. |
| 11:54.42 | hwt | Dovid: running AGI-scripts adds a lot of overhead. |
| 11:54.47 | blitzrage | Dovid: you don't want to use n+101... thats call priority jumping, and it totally limits how you write a dialplan. Use the 'n' priority and priority labels. |
| 11:54.48 | hwt | Dovid: it should be possible with asterisk "internals". |
| 11:54.50 | Dovid | hwt: okies. |
| 11:55.05 | blitzrage | hwt: the A-number? |
| 11:55.15 | hwt | blitzrage: the calling number. |
| 11:55.33 | blitzrage | why would that be billed to? |
| 11:55.45 | blitzrage | you are not capturing the right data then |
| 11:56.05 | hwt | A invites B, B invites the number which B has been forwarded to (e.g. on another provider's network). |
| 11:56.54 | hwt | in the billing we will se an invte from: <calling party on another provider network>@domain.com to: <some cell phone>@gateway.com |
| 11:57.14 | hwt | and we want to bill <number which as been forwarded>@domain.com |
| 12:01.09 | *** join/#asterisk txotin (n=egomez@56.Red-80-38-204.staticIP.rima-tde.net) |
| 12:01.12 | txotin | hi all |
| 12:02.13 | Dovid | hello txotin |
| 12:02.48 | txotin | I have a problem with asterisk, incoming calls work fine, but on outgoing calls, when the receiver asnwer, asterisk automatically close the connection... |
| 12:03.14 | Dovid | hmm |
| 12:03.17 | txotin | I'm in Spain and have Wildcard X100p clone.. |
| 12:03.32 | Dovid | what do u mean by asterisk closes the connection ? |
| 12:03.42 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:03.50 | Dovid | can u post ur realative dial plan in pb |
| 12:03.52 | Dovid | ~pb |
| 12:03.59 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 12:03.59 | txotin | this.. |
| 12:03.59 | Dovid | morning TK |
| 12:03.59 | txotin | Called 1/696857645 |
| 12:03.59 | txotin | <PROTECTED> |
| 12:03.59 | txotin | <PROTECTED> |
| 12:03.59 | txotin | <PROTECTED> |
| 12:04.07 | txotin | sorry for the flush.. |
| 12:04.15 | Dovid | txotin please use pb |
| 12:04.23 | txotin | ups.. sorry |
| 12:06.12 | [TK]D-Fender | *yawn* |
| 12:06.25 | Dovid | lol |
| 12:06.31 | Dovid | TK: Sleepless night ? |
| 12:07.25 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 12:08.34 | blitzrage | [TK]D-Fender: yo |
| 12:08.57 | txotin | Dovid: here can see my config and asterisk log: http://pastebin.ca/196211 |
| 12:08.58 | [TK]D-Fender | blitzrage: I don't want to be at work, I just want...... |
| 12:09.25 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:09.30 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 12:09.45 | blitzrage | ! ! ! |
| 12:11.18 | blitzrage | [TK]D-Fender: hehehe... you ever use ChanIsAvail() ? I'm finding if the end point is not registered, it just drops the connection instead of going to the next priority and reporting something in ${AVAILSTATUS}. I'm not using priority numbers or jumping. |
| 12:12.03 | blitzrage | [TK]D-Fender: well, found the problem... its still priority jumping |
| 12:12.18 | blitzrage | DOH! I don't have priorityjumping=no at the top |
| 12:12.19 | blitzrage | wtf! |
| 12:12.24 | [TK]D-Fender | blitzrage: It won'te tell you if its registered, it'll only say if its on a call. |
| 12:13.01 | blitzrage | [TK]D-Fender: it will based on the AVAILSTATUS... if its not avail (reg'd or not), it should return a non-zero value |
| 12:13.24 | tzanger | morning |
| 12:13.47 | tzanger | blitzrage: I have a patch in the works for Dial() with priorities |
| 12:14.18 | blitzrage | tzanger: wicked! |
| 12:14.19 | tzanger | where you set a call priority (or it receives a default) and if a Dial() for that channel comes thorugh and its of higher priority it will drop the call in progress (optinally warning them) and then placing the call |
| 12:14.30 | Dovid | TK: will chanisavail work for a SIP connection if u arent registerd |
| 12:14.31 | Dovid | ? |
| 12:14.32 | blitzrage | tzanger: I think I saw that in the bug tracker. Jim pointed it at me. |
| 12:14.32 | tzanger | avoids the issues of channel race with ChanIsAvail |
| 12:14.40 | blitzrage | tzanger: aye |
| 12:15.15 | [TK]D-Fender | Dovid: It'll return unavail if you aren't, but it nevers gives details as to why you aren't available. |
| 12:15.49 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:15.59 | blitzrage | ahhhhhh ha! Dialplan on this box is using a combo of priority numbering with jumping, and some of it is updated... |
| 12:16.05 | Dovid | ah ok. i am concerd for 911 routing. i have pri. to go out thru pots but if its down i have a few SIP connections |
| 12:16.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:16.10 | blitzrage | So I need priorityjumping on for now I guess |
| 12:16.13 | blitzrage | how stupid.... |
| 12:16.29 | txotin | There are parts in the asterisk configuration dependant on the country where asterisk is connected rather than zaptel.conf? |
| 12:16.32 | blitzrage | [TK]D-Fender: correct -- I just want to know if its avail or not... I don't care of the details :) |
| 12:16.40 | [TK]D-Fender | blitzrage: Shouldn't need to, and you can enable for this option only with the "j" parm. |
| 12:17.00 | blitzrage | [TK]D-Fender: other way around though... this part of the dialplan is not priority jumping, the rest of it is |
| 12:17.26 | [TK]D-Fender | blitzrage: Even more reason to keep cleaning up.. |
| 12:18.00 | blitzrage | [TK]D-Fender: fuck that, I should still be in bed |
| 12:18.01 | Dovid | TK: I once tried to write a system off some one else's config's. BIG MISTAKE !!! |
| 12:18.13 | [TK]D-Fender | blitzrage: You're flogging a dead equine ;) |
| 12:18.31 | blitzrage | [TK]D-Fender: oh well, issue resolved, time to go back to bed |
| 12:20.01 | Dovid | . |
| 12:25.43 | txotin | Can anyone see my post on http://pastebin.ca/196211 and check if my dialplan is correct for outgoing calls, please.. |
| 12:26.17 | *** join/#asterisk pph (n=pph@81.255.164.157) |
| 12:30.16 | [TK]D-Fender | txotin: Your 2XX series lines will never be called, and unless you have priority jumping enabled (which you shouldn't) your 1XX series ones shouldn't either. But for now does it do what you want? |
| 12:32.54 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 12:32.55 | *** join/#asterisk L-info (n=linfo@62.69.102.99) |
| 12:34.10 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 12:35.10 | txotin | my problem is, when I make a outgoing call, when my peer answer, Asterisk hangup the call... |
| 12:35.33 | blitzrage | I was all by myself... |
| 12:35.39 | blitzrage | Noooo one was loooooooking.... |
| 12:35.42 | blitzrage | I was all by myself |
| 12:36.09 | Qwell | rofl |
| 12:36.15 | blitzrage | Qwell: !!! |
| 12:36.15 | Qwell | blitzrage++ |
| 12:36.21 | blitzrage | hahaha |
| 12:36.35 | Qwell | god, I haven't heard that song in years, heh |
| 12:36.37 | blitzrage | Qwell: I leave for Astricon in 7 days |
| 12:36.41 | Qwell | w00t |
| 12:36.47 | blitzrage | next tuesday |
| 12:36.56 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 12:37.13 | tamp4x | Oct 9 15:39:39 WARNING[1217588144]: rtp.c:1358 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. |
| 12:37.19 | tamp4x | what does that error usually mean |
| 12:37.40 | blitzrage | tamp4x: can't transcode from g729 to ulaw |
| 12:37.53 | tamp4x | i have g729 licenses though |
| 12:38.11 | blitzrage | show g729 |
| 12:38.39 | blitzrage | and does 'show translation' show a path? |
| 12:39.15 | tamp4x | 1/1 encoders/decoders of 20 licensed channels are currently in use |
| 12:40.10 | blitzrage | oh... its trying to native bridge... ? yah... 256 and 4 are not native |
| 12:40.32 | tamp4x | what does that mean , not native |
| 12:40.39 | blitzrage | native means the same |
| 12:40.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:40.48 | blitzrage | ulaw-ulaw, g729-g729 |
| 12:40.55 | Qwell | blitzrage: You gonna send me that song? :p |
| 12:41.17 | tamp4x | ahh yes i remember its my stupid carrier |
| 12:41.19 | blitzrage | Qwell: don't have a copy of it... I was just singing it out loud :) |
| 12:41.22 | Qwell | lame |
| 12:41.32 | blitzrage | my room mate probably has a copy of it somewhere though |
| 12:43.52 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 12:44.34 | tzanger | fooking skype |
| 12:44.56 | blitzrage | lame |
| 12:45.00 | blitzrage | who the heck uses skype? |
| 12:45.05 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-76629.otenet.gr) |
| 12:45.12 | tzanger | my counterpoint in NC |
| 12:45.19 | tzanger | not for phone though, just IM |
| 12:45.24 | denon | blitzrage: every end-luser who thinks they've "discovered voip" |
| 12:45.25 | tzanger | it's not like I'm on enough IM networks as is |
| 12:45.27 | denon | its almost as bad as vonage |
| 12:45.29 | tzanger | denon: hahaha AMEN |
| 12:45.44 | blitzrage | quite |
| 12:45.50 | denon | then we get to hear the, "Yeah, voip sounds really bad .. poor quality" |
| 12:45.57 | jgoo | anyone here use POLYCOM hardphones? I moved to a new country, and I need a heads on on what the voltage/amp requirements are on the device, methinks 12v .5A |
| 12:46.05 | denon | so I retort, "voip sounds jsut fine, ulaw, on my 7960s on a gig-e switch" |
| 12:46.07 | jgoo | the power-over-lan flavour |
| 12:46.25 | denon | and they dont have a clue what I'm talking about, nor do they understand the quality control issues of the Internet |
| 12:46.39 | blitzrage | yay lusers |
| 12:46.49 | blitzrage | fuck.... I got woke up way too early... and now I have no motivation to work |
| 12:46.50 | tzanger | jgoo: they don't have much int he way of power use |
| 12:47.08 | jgoo | blitzrage: what happens if you wake up way too late? |
| 12:47.19 | blitzrage | jgoo: lol, I'll tell you when it happens :) |
| 12:47.34 | jgoo | :-D, now, back to fastagi |
| 12:47.44 | blitzrage | hrmmm... anyone happen to have a good link on using IPtables to share a pair of internet connections? |
| 12:47.48 | denon | blitzrage: I've got no sympathy for you - I ended up working through the night from home, and now I'm back at the office |
| 12:48.06 | blitzrage | denon: I've done that a bunch too... and I work at home :) |
| 12:48.16 | denon | if you got any sleep, you're leaps and bounds ahead :) |
| 12:48.24 | denon | yeah, it sucks - I dont do it as much anymore |
| 12:48.25 | blitzrage | :D |
| 12:48.35 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:48.40 | txotin | bye all, thanks for your attention.. |
| 12:49.20 | jgoo | ok, so I hooked up a pstn sim to my tdm04b |
| 12:49.23 | blitzrage | I guess I'll go try to find the parts to attach my computer to the stereo so I can play some music |
| 12:49.29 | jgoo | of which I have 2 |
| 12:49.46 | jgoo | and, well, the lights flash on the pstn sim, BUT, on the back of my *box absolutely nothing flashes *tear* |
| 12:49.47 | jgoo | is that normal? |
| 12:52.24 | jgoo | perhaps the voltages from the sim are not right? these settings were tested on dialogic cards... |
| 12:55.49 | jgoo | Executing Goto("SIP/269-0854xxx0", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp("SIP/269-0854xxx0", "Dial failed due to CHANUNAVAIL") in new stack |
| 12:56.05 | jgoo | hrm, I get that from trying to make an out call, methinks the trunk is not config'd right |
| 12:56.47 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 12:57.32 | jgoo | what does a flash button on a phone do? |
| 12:57.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.154) |
| 12:58.20 | blitzrage | flashes the line (like when you hold the button down for a second to answer Call Waiting) |
| 12:59.01 | *** join/#asterisk purserj (n=purserj@29.124.233.220.exetel.com.au) |
| 12:59.05 | purserj | evening all |
| 12:59.10 | jgoo | aah thanks |
| 13:01.29 | oliviert | hey guys, still have a problem with musiconhold and zaptel channels |
| 13:01.56 | jgoo | blitzrage: I have an all circuits busy message, any ideas? I have been checking all posts about it online |
| 13:02.04 | jgoo | my in and out -bounds seem ok |
| 13:02.07 | oliviert | on incoming calls, i ring a group, then a SIP softphone take the line, put on hold, but the music don't play back |
| 13:02.13 | jgoo | the trunk is what worriesme, it seems... too empty the config |
| 13:02.29 | oliviert | while on internal calls, between sip phone, it works |
| 13:02.43 | purserj | Is there any known issues between asterisk installs and the ekiga softphone? |
| 13:03.17 | [TK]D-Fender | jgoo: You only showed us a Goto which says nothing of WHY. pastebin the WHOLE thing ploease... |
| 13:03.18 | *** join/#asterisk cian (n=cian@cian.ws) |
| 13:03.28 | oliviert | i think that the problem is in zaptel.conf, but even if i configure the musiconhold, it does not change anything |
| 13:04.22 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 13:04.28 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-241-102.red.bezeqint.net) |
| 13:05.30 | jgoo | [TK]D-Fender: I will pastebin now, thanks |
| 13:05.56 | jgoo | I thought the CHANUNAVAIL was a key thing :p |
| 13:08.39 | Cyt | Please: What this error means? -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" |
| 13:09.25 | jgoo | [TK]D-Fender: http://pastebin.ca/196286 |
| 13:10.03 | *** join/#asterisk necudeco (n=chatzill@200.121.75.218) |
| 13:10.07 | oliviert | [TK]D-Fender: any idea about my musiconhold problem with zaptel channels ? |
| 13:10.48 | jgoo | ssssh don't distract him ;-) |
| 13:11.11 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:11.28 | jgoo | I can see the lines, you know, where it fails, but I have been faffing around with the prefixes and rules, because I fixed it once on an old setup, but right now I fear any more tinkering would do more hard than good, so I reset everything |
| 13:11.35 | oliviert | jgoo: sorry dude |
| 13:11.50 | [TK]D-Fender | jgoo: line #47 - -- Executing Dial("SIP/269-084ce1a0", "ZAP/g0/6|120|r") in new stack |
| 13:12.09 | [TK]D-Fender | jgoo: Do you think that a SINGLE DIGIT phone # is VALID?! |
| 13:12.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.02 | [TK]D-Fender | jgoo: And AMP/FreePBX is NOT supported here. Please read the channel topic. |
| 13:13.39 | jgoo | [TK]D-Fender: how do you know that a single phone number is not valid? |
| 13:14.00 | [TK]D-Fender | oliviert: I'd be nice to SEE the problem. |
| 13:14.25 | [TK]D-Fender | jgoo: I didn't say s single phone number, I said single DIGIT. |
| 13:14.28 | jgoo | the fact is, any number fails, I'd expect to hear my telcos 'number not found, you moron, lol' message, not asterisk saying something, but thanks anywho |
| 13:14.39 | jgoo | I only ask herebecause last time in a normal *install I had the same problem |
| 13:14.50 | [TK]D-Fender | jgoo: You have a PRI by any chance? |
| 13:14.51 | oliviert | [TK]D-Fender: well everything looks like easy to me, but it does not work, then i should miss something |
| 13:15.18 | jgoo | [TK]D-Fender: no this is hooked up to a PSTN sim (was a normal line, but for now a PSTN sim) |
| 13:15.24 | jgoo | I will have PRI, eventually |
| 13:15.39 | jgoo | I have this running on my own home setup on BRI, no problems, on suse |
| 13:16.01 | [TK]D-Fender | jgoo: Ok, lets look at the oterh side. Show me what interfaces are in "g0". |
| 13:16.19 | jgoo | ... interfaces? |
| 13:16.28 | oliviert | [TK]D-Fender: well my config is simple: 4 analog lines for incoming calls, and VOIP for outgoing calls, outgoing is working, if i place a call, the musiconhold feature is working, |
| 13:17.03 | oliviert | [TK]D-Fender: but if i answer an incoming call, the musiconhold don't play when i hold the line |
| 13:17.08 | [TK]D-Fender | jgoo: Zaptel. Show me whats in "g0" |
| 13:17.15 | jgoo | aah gotcha |
| 13:17.47 | jgoo | you mean zaptel.conf? |
| 13:18.15 | [TK]D-Fender | oliviert: I heard you the first time and for the last time I'm going to tell you to **SHOW ME** THE DAMNED PROBLEM or stop asking how to fix it. |
| 13:18.39 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.154) |
| 13:18.53 | jgoo | [TK]D-Fender: loadzone= us \ defaultzone= us << my entire zaptel.conf... |
| 13:18.54 | [TK]D-Fender | jgoo: Zapata.conf |
| 13:19.04 | jgoo | thought so, ok, there is no g0 context there |
| 13:19.51 | jgoo | and [trunkgroups] is empty |
| 13:19.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.154) |
| 13:19.51 | jgoo | shall I pastebin the zapata? |
| 13:20.05 | purserj | if I pastebin an error message could someone tell me what it means? |
| 13:20.17 | jgoo | http://pastebin.ca/196293 |
| 13:20.29 | jgoo | [TK]D-Fender: that is the zapata.conf |
| 13:21.12 | jgoo | (I was playing with the context, from-pstn and from-zaptel....) didn't understand too much, only that I need it for some DID features |
| 13:21.31 | [TK]D-Fender | jgoo: And like a true AMP/FreePBX user you don't think to give me the INCLUDED files.... |
| 13:22.05 | jgoo | [TK]D-Fender: I did |
| 13:22.10 | jgoo | they are both emmpty |
| 13:22.27 | jgoo | I checked..., I may be a lame trixbox user, but that is as lame as I go! |
| 13:22.32 | jgoo | I feel so segregated... ;-) |
| 13:22.48 | [TK]D-Fender | jgoo: EMPTY? Congratulations you have no frigging channels defined! |
| 13:23.00 | jgoo | exactly my worry... so that is weird :-) |
| 13:23.09 | purserj | http://pastebin.ca/196294 <-- My problem, sorry for bothering :) |
| 13:23.33 | jgoo | well, my first out of the box trixbox worked... so I am just very stupid this time, thanks |
| 13:23.49 | jgoo | a chanel is a trunk? |
| 13:24.20 | [TK]D-Fender | purserj: Looks like the caller hung up while listening to VM prompts. Seems perfectly normal. Any reason you believe otherwise? |
| 13:24.34 | purserj | yar, they keep reporting "Cannot connect to host" |
| 13:24.45 | [TK]D-Fender | jgoo: Not the place to be asking that..... They have a book and a channel..... |
| 13:24.46 | purserj | two different people on ekiga softphones |
| 13:24.58 | jgoo | [TK]D-Fender: np, thanks for the sanity check |
| 13:25.24 | purserj | However one person on a hardware voip manages to connect fine, I'm new to asterisk so I was just checking to see if I hadn't stuffed up |
| 13:25.29 | [TK]D-Fender | jgoo: You would be far better served to ditch Trixbox while the gettings good... |
| 13:25.32 | oliviert | [TK]D-Fender: ok, when you got 2 mins, tell me what conf files do you want me to show you, thanks |
| 13:26.56 | [TK]D-Fender | oliviert: If you can't think of what to show us there is something seriously wrong. CLI out put of the failed calls. Dialplan where applicable. Config files for the failed bits. "ls" output of the folders where files are expected to be. Etc. |
| 13:27.23 | *** join/#asterisk eset (n=adam@ip545186e3.direct-adsl.nl) |
| 13:27.30 | *** join/#asterisk zoid_99 (n=cch123@br0.asteriasgi.com) |
| 13:27.42 | jgoo | thanks man |
| 13:28.47 | *** join/#asterisk kgpsathish (n=sathish@59.92.143.145) |
| 13:28.47 | eset | i have a newbie question - does SIP INFO actually carry DTMF tones, or is it just numerical data? |
| 13:28.53 | [TK]D-Fender | eset: Just data |
| 13:28.57 | eset | ok |
| 13:29.20 | eset | and if you use RTP for DTMF, si this audio? |
| 13:29.22 | [TK]D-Fender | eset: Its a popular "out of band" format. RFC2833 is the more common choice. |
| 13:29.38 | eset | ok |
| 13:29.43 | [TK]D-Fender | eset: I believe so. Not sure on the specific phrasing |
| 13:29.56 | eset | ok, thanks a lot |
| 13:30.05 | [TK]D-Fender | eset: np |
| 13:30.34 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 13:33.02 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 13:33.22 | oliviert | [TK]D-Fender: here they are: http://pastebin.ca/196299 |
| 13:33.40 | pingwin | from the console, how do I kill a sip connection where the sip phone has been disconnected already? (basically it's stuck in a phone menu) |
| 13:34.02 | [TK]D-Fender | pingwin: "soft hangup [channel]" |
| 13:34.16 | pingwin | awesome, thank you! |
| 13:34.26 | oliviert | [TK]D-Fender: i did not ls the directory where files have to be, as musiconhold work in other context than ZAP, |
| 13:35.51 | [TK]D-Fender | oliviert: Show me your musiconhold.conf and the output of a call that DOES get the MoH, and one that DOESN'T. |
| 13:36.01 | [TK]D-Fender | pingwin: np |
| 13:36.48 | *** join/#asterisk ltd (n=z@202-161-29-10.dyn.iinet.net.au) |
| 13:37.34 | kgpsathish | I am using SIP with Aiudiocodes gateway,I am getting an error essage |
| 13:37.34 | kgpsathish | as " Got SIP response 486 "Busy Here" back from 192.168.100.195". |
| 13:37.59 | *** join/#asterisk dir (n=dir@124.106.223.190) |
| 13:40.41 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
| 13:41.09 | oliviert | [TK]D-Fender: you already help me, investigating what you ask me for, my default class is working, nut the one i defined before wasn't, i think i'll be able to find the problem then, thanks for all, and sorry for disturbing |
| 13:41.18 | jgoo | anyone here using fastagi? 'tis a great thing |
| 13:41.48 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:41.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:42.05 | skywriter | i succefully registered to an voip provider |
| 13:42.42 | skywriter | but when i try to call out i get Auto fallthrough, channel 'SIP/joe-0822ed48' status is 'CHANUNAVAIL' |
| 13:43.27 | jgoo | [TK]D-Fender: I just saw your advice about ditching trix, the only reason I wanted to use it was... in case of issues, to have a procedure to make a new setup from scratch, you know... since I am using fastagi anyway for development, I wanted the PBX to be almost hotswappable... |
| 13:43.37 | skywriter | <PROTECTED> |
| 13:43.42 | jgoo | but I appreciate the advice, perhaps I need to install a system and then disk image it... |
| 13:43.53 | jgoo | skywriter: are you using trixbox? |
| 13:44.03 | *** part/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 13:44.05 | jgoo | run genzaptelconf |
| 13:44.07 | skywriter | no i m using asterisk |
| 13:44.08 | [TK]D-Fender | jgoo: Thats what tar-ing /etc/asterisk does even FASTER. 2 second backups. |
| 13:44.48 | jgoo | [TK]D-Fender: aaah. yeah. that sounds good too... *whistle* |
| 13:45.15 | blitzrage | anyone know where JunK-Y's photo album is? Astricon 2005 pics is what I'm after |
| 13:45.31 | jgoo | I have a box setup at home, but, just one sip extension, one voip provider, and I can make calls from my normal phone over broadvoice / some local voip provider / normal phone network, that is all I use it for |
| 13:45.33 | Juggie | the website was down blitzrage last i looked |
| 13:46.07 | blitzrage | Juggie: grrrr... I'm trying to find some pictures of it :) |
| 13:47.12 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-204-200.telkomadsl.co.za) |
| 13:48.56 | [TK]D-Fender | blitzrage I'll get you mine when I get back home. |
| 13:49.14 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 13:49.16 | blitzrage | [TK]D-Fender: coo, thx! |
| 13:49.17 | [TK]D-Fender | blitzrage: Oops, actually that wasn't Astricon.... |
| 13:49.31 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 13:49.32 | [TK]D-Fender | blitzrage: Was thinking of the "Meet Asterisk" one... |
| 13:49.33 | blitzrage | oops :) |
| 13:49.36 | blitzrage | ahhhh yes |
| 13:49.42 | blitzrage | I'm looking for a specific picture actually |
| 13:49.49 | blitzrage | but Junky has it :) |
| 13:52.28 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 13:53.44 | *** join/#asterisk prh (n=paul@X80.mjr.org) |
| 13:54.17 | prh | anyone here know if the chan_gsm_bt or chan_bluetooth work with asterisk 1.2.12? |
| 13:54.27 | prh | (even where to get chan_gsm_bt) |
| 13:55.05 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 13:58.24 | *** join/#asterisk jsmith (n=jsmith@000-242-432.area3.spcsdns.net) |
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| 13:59.18 | jeremy_g | Mercestes!!! u there |
| 13:59.29 | Mercestes | yes....yes I am. |
| 14:00.56 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 14:02.08 | *** join/#asterisk chrislh (n=christo@courgette.jml.net) |
| 14:03.11 | *** join/#asterisk intralanman (n=lanman@pool-72-82-69-74.nrflva.east.verizon.net) |
| 14:03.24 | chrislh | lspci shows that I have uhci controllers. lsmod shows usbcore and uhci, but I don't have usb-uhci loaded - and modprobe fails. How can I get this module installed on a slackware system? |
| 14:04.16 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 14:04.30 | SplasPood | Has anyone ported app_cepstral to 1.4 yet? |
| 14:04.41 | chrislh | this is for ztdummy timing (no zaptel hardware installed) |
| 14:04.53 | [TK]D-Fender | chrislh: Thats something to ask in #slackware . Theya re very helpful on this stuff there. |
| 14:05.26 | jsmith | SplasPood: Asterisk 1.4 has a speech module already, so all you'll need is the license and module from Cepstral, as far as I know. |
| 14:05.31 | chrislh | [TK]D-Fender: they are, but I got no reply today.. will try again. thanks |
| 14:05.39 | SplasPood | jsmith: I thought that was for recognition, not output |
| 14:05.43 | [TK]D-Fender | chrislh: np, wish I had an answer for you. |
| 14:05.48 | chrislh | :) |
| 14:06.01 | jsmith | SplasPood: I know there's one for speech rec, but I thought there was one for Cepstral as well. |
| 14:06.26 | jsmith | SplasPood: I know Business Edition has it, and I thought they were going to put it in 1.4 as well, but I could be mistaken. |
| 14:06.50 | SplasPood | well the speech api is def in 1.4, but I'm not certain if it goes both ways |
| 14:07.37 | jgoo | hurrah! and the fastagi script is alive :-) cheers folks |
| 14:08.37 | SplasPood | not a lot of 1.4 docs on voip-info yet |
| 14:09.29 | jsmith | Nope... |
| 14:09.41 | SplasPood | The generic speech recognition engine is implemented in the res_speech.so module |
| 14:09.41 | SplasPood | . |
| 14:09.41 | SplasPood | This module connects through the API to speech recognition software, that is |
| 14:09.41 | SplasPood | not included in the module. |
| 14:09.46 | SplasPood | seems to be only for recog |
| 14:10.53 | jsmith | Yeah, looks like I was wrong. |
| 14:13.05 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:13.37 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:17.11 | jmls | in the UK (BT IDSN) if someone withholds their number when calling in, is there any way of knowing what the number is ? Someone claims to be able to retrieve the number on an inbound call even if the number is withheld. His argument is "the police and fire brigade do it" - I say that's a special dispensation from BT to those services. I could be wrong - I hope I'm wrong .... |
| 14:17.11 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 14:20.57 | SplasPood | jsmith: When I first saw the API I thought the same thing.. then was disappointed |
| 14:21.51 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 14:21.59 | jsmith | SplasPood: I know the code is out there, because it's part of ABE -- I just don't know if Digium and/or Cepstral are going to release it. I do know, however, that it works better than app_cepstral. |
| 14:24.19 | SplasPood | app_cepstral always worked well for me... I've already gone in and replaced most of my use of it with static files.. but in some places it really needs to be dynamically generated |
| 14:27.13 | jsmith | file: Hey, you might know... any chance of the Cepstral code making it to 1.4? Obviously we'd still need a license... |
| 14:28.06 | file | from a software standpoint I don't see why not, from a business/licensing standpoint I don't know |
| 14:34.04 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 14:34.23 | vooduhal | Does DeadAgi call the agi script at that line or wait until the call is completed? |
| 14:34.34 | jsmith | vooduhal: It calls it at that line |
| 14:34.41 | vooduhal | K. Thank you. |
| 14:36.22 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:37.13 | vooduhal | If I needed to move a file after the call is completed would I put the DeadAgi call after the Dial, or in the h extension? |
| 14:39.40 | *** join/#asterisk JoshuaFuller (n=jdf@jdf.arccgi.com) |
| 14:39.59 | JoshuaFuller | Good morning folks. |
| 14:40.13 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 14:40.15 | JoshuaFuller | I'm trying to get a little help with an IAXY. |
| 14:40.26 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:40.40 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 14:40.51 | JoshuaFuller | Anyone home ? |
| 14:42.29 | FlatFoot | no i'm at work ( Ha ha ) |
| 14:42.36 | JoshuaFuller | I'm having problems with my Cisco Catalyst 2950 and my IAXy. |
| 14:42.41 | JoshuaFuller | :) |
| 14:42.44 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:42.44 | JoshuaFuller | Me too. |
| 14:42.56 | FlatFoot | cisco and me don't mix |
| 14:43.03 | *** join/#asterisk eliXier (i=GTI16V@gti.twice-irc.de) |
| 14:43.04 | Dr-Linux | hey all young guys! |
| 14:43.13 | FlatFoot | not in australia are you ? |
| 14:43.18 | JoshuaFuller | I wish I had another switch right now. |
| 14:43.24 | JoshuaFuller | I have 2 2950's. |
| 14:43.51 | JoshuaFuller | The IAXy module we have has some issue with the ethernet packets getting corrupted through the 2950. |
| 14:43.59 | JoshuaFuller | I don't know why yet. |
| 14:44.06 | *** join/#asterisk xnon (i=xnon@200.82.222.85) |
| 14:44.24 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:45.07 | JoshuaFuller | Anyone in here have any luck with 2950 and an IAXy ? |
| 14:45.24 | JoshuaFuller | I think I'm up the creek on this one :) |
| 14:45.50 | brad_mssw | 2950? you mean the cisco catalyst switch, or does 2950 refer to something else? |
| 14:46.00 | JoshuaFuller | yeah my Catalsyt. |
| 14:46.01 | brad_mssw | btw, i've had nothing but issues with IAXy's |
| 14:46.23 | JoshuaFuller | I plug my IAXy in the back of a 2950 and it works fine. |
| 14:46.27 | JoshuaFuller | Its strange. |
| 14:47.02 | brad_mssw | the IAXy's I have just go offline for no reason after a few hours ... you have to unplug them and plug them back in |
| 14:47.25 | brad_mssw | we've canned them ... I would say it's a problem with the IAXy, not your switch |
| 14:47.28 | JoshuaFuller | Tech support site mentioned this "The S101i IAXy is currently known to have problems when connected to Cisco 2950 switches. The ethernet frames from the IAXy get corrupted during traversal to the switch. This issue is not resolvable through a firmware update." |
| 14:47.58 | brad_mssw | you're better off with an unlocked Linksys PAP2, it's cheaper and more reliable |
| 14:48.10 | brad_mssw | (plus it offers 2 lines instead of 1) |
| 14:48.10 | JoshuaFuller | yeah no matter what I do it just won't talk to my cisco equipment except the 7960 phones. |
| 14:48.28 | JoshuaFuller | Cool thanks for the recommendation. |
| 14:48.36 | JoshuaFuller | I'm going to check it out now. |
| 14:48.53 | brad_mssw | the line quality of the IAXy is aweful, we tried to use it to hook up a fax machine (it just goes to a TDM400P to get an external line), and nothing would work |
| 14:48.58 | JoshuaFuller | Those are the ones that get sent with Vonage and other VIOP providers right ? |
| 14:48.58 | brad_mssw | a PAP2 works flawlessly though |
| 14:49.11 | JoshuaFuller | cool |
| 14:49.16 | *** join/#asterisk Soda_AFK (i=heyo@ivr94-1-81-57-178-223.fbx.proxad.net) |
| 14:49.20 | brad_mssw | JoshuaFuller: yeah, but just buy them from voipsupply or something, it's the PAP2-NA, which is not locked to any provider |
| 14:49.44 | JoshuaFuller | Cool. I bought all our voip stuff from those guys. |
| 14:50.00 | JoshuaFuller | I think this Iaxy too actually. |
| 14:50.07 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 14:50.18 | [TK]D-Fender | screw the PAP2. SPA-2002 avoids all those troubles and has more functionality. |
| 14:50.20 | JoshuaFuller | I have a PRI line coming in today. |
| 14:50.55 | brad_mssw | [TK]D-Fender: I'd go with the 2100 if you're going for a higher-end ATA ... don't know what advantages 2002 has over PAP2 |
| 14:51.03 | JoshuaFuller | I'm trying to get ready for it. I think the fax is just going to have to wait until I get some other way of tying it into the new asterisk server we are setting up. |
| 14:51.18 | *** join/#asterisk mtaht4 (n=m@h-72-244-145-197.phlapafg.covad.net) |
| 14:51.24 | [TK]D-Fender | 2100 = bas idea. it has an integrated router and last I checked you could only run SIP on the WAN interface. |
| 14:51.30 | JoshuaFuller | I have a TDM400 already for our POTS lines that are going away today. |
| 14:51.45 | JoshuaFuller | Can I just buy a new module for that too as an alternative ? |
| 14:52.11 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:52.19 | JoshuaFuller | I'd rather have a CAT5 line though hitting right next to our fax since that means I won't have to run any phone wire around the office. |
| 14:52.30 | brad_mssw | [TK]D-Fender: 2100 supports T.38 though, which is nice, the 2002 doesn't, and no one said you need to use the router portion, just plug in the WAN interface to your internal network and go |
| 14:52.40 | brad_mssw | don't plug anything into the LAN side |
| 14:52.47 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 14:52.49 | supjigatr | Hello. |
| 14:53.10 | [TK]D-Fender | JoshuaFuller: You can plug an RJ11 into an RJ45 jack you know. just run CAT5EE and skip cat3. You just need to patch it right from your termination point. |
| 14:53.25 | JoshuaFuller | I forget about that. |
| 14:53.53 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 14:53.54 | JoshuaFuller | How well does a fax work over G.711u ? |
| 14:53.54 | supjigatr | Is there a way to setup a polycom in asterisk to have a line button. For example have ext 1025 be on line2 on the polys on 2 or three phones including one analog on a zap interface? |
| 14:54.16 | brad_mssw | JoshuaFuller: Yeah, but you can use a PAP2, or SPA-2002/2100 for that fax machine as well ... 2100 if you require T.38 and have T.38 abilities |
| 14:54.20 | RoyK | [TK]D-Fender: ee? |
| 14:55.00 | brad_mssw | JoshuaFuller: otherwise a high-quality network and g711u works fine for me |
| 14:55.23 | JoshuaFuller | brad_mssw:Whats the 2100 ? |
| 14:55.27 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 14:56.06 | JoshuaFuller | A Cisco 2100 router ? |
| 14:56.25 | brad_mssw | JoshuaFuller: no, the sipura SPA-2100 ATA |
| 14:56.30 | JoshuaFuller | ohhh... |
| 14:56.34 | JoshuaFuller | OK |
| 14:57.32 | [TK]D-Fender | supjigatr: "lines" don't exist in SIP. Just registrations and call channels |
| 14:58.06 | JoshuaFuller | I think I'm going to get a 2100. |
| 14:58.07 | [TK]D-Fender | supjigatr: Your meaning is unclear. |
| 14:58.23 | JoshuaFuller | Anyone have a 2100 and how well does it work ? |
| 14:58.36 | supjigatr | TK: what I am trying to do is have 3 polys register to the same account. |
| 14:59.31 | JoshuaFuller | I just read about that. Multiple SIP registrations where not supporting without a patch I believe. |
| 14:59.32 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:59.44 | mut | what in windows plays gsm files? |
| 15:00.14 | [TK]D-Fender | supjigatr: Unless you're successfully running SLA in * 1.4 you can't |
| 15:00.31 | [TK]D-Fender | supjigatr: Just make 3 reg's and make them ring at the same time. |
| 15:00.51 | [TK]D-Fender | mut: Winamp |
| 15:00.58 | mut | sorry |
| 15:01.01 | [TK]D-Fender | (IIRC) |
| 15:01.02 | mut | convert wav to gsm |
| 15:01.03 | mut | not play |
| 15:01.09 | JoshuaFuller | Hey Thanks for all your help guys, I'm going to try to get my hands on a 2100. |
| 15:01.12 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.228) |
| 15:01.13 | [TK]D-Fender | mut: : Audacity |
| 15:01.22 | supjigatr | TK: Thanks. |
| 15:01.44 | JoshuaFuller | That looks like it will work great. I'll use it for the fax machine and the conference room phone :) |
| 15:02.32 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-54447d944a40973f) |
| 15:03.08 | DarKnesS_WolF | anyone have IAXy here ? S100I .. i want to know the default IP to conf. it |
| 15:03.19 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-3d7526b4b6548ec6) |
| 15:07.13 | JoshuaFuller | DarKnesS_WolF: I'm configuring mine right now. |
| 15:07.22 | *** join/#asterisk p1p (i=pip@64.200.16.100) |
| 15:07.27 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:07.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:07.38 | JoshuaFuller | It setup to do DCHP from the facatory. |
| 15:08.11 | DarKnesS_WolF | JoshuaFuller: oh :-s |
| 15:08.18 | DarKnesS_WolF | hehe so i need DHCP server damn it |
| 15:08.47 | JoshuaFuller | It comes with most routers. |
| 15:09.00 | JoshuaFuller | Windows servers and linux server have it included too. |
| 15:09.32 | DarKnesS_WolF | oh so i need to install DHCPD :-s not good :-s |
| 15:09.44 | JoshuaFuller | Its not to hard. |
| 15:09.55 | DarKnesS_WolF | i know but why :P |
| 15:10.01 | qdk | its actually boringly easy. |
| 15:10.05 | JoshuaFuller | The first time I did it , it took 30min - 1 hour I think. |
| 15:10.19 | JoshuaFuller | With VLANs :) |
| 15:10.26 | DarKnesS_WolF | it took 10 mi nto get DHCP up and running ;-) in mandriva 1 min using drakwizard :) |
| 15:10.54 | rob0 | For small sites, dnsmasq is a pretty slick little DHCP daemon. |
| 15:10.54 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 15:10.55 | JoshuaFuller | We use Webmin on some of our servers. |
| 15:11.07 | DarKnesS_WolF | so then i can add a static IP ? |
| 15:11.08 | JoshuaFuller | To configure the DHCP and firewall settings. |
| 15:11.13 | qdk | JoshuaFuller: :-( |
| 15:11.15 | DarKnesS_WolF | or it should always be a DHCP ? |
| 15:11.32 | [TK]D-Fender | TAKES ME 1 MINUTE TO COPY &PASTE MY DHCPD.CONF FROM MY MAIN SERVER TO WHEREVER i NEED IT :) |
| 15:11.37 | JoshuaFuller | It will request a DHCP address by default. |
| 15:12.22 | JoshuaFuller | Yeah webmin is not really all that spiffy sometimes... On the main servers we do most administration through SSH only. |
| 15:12.43 | DarKnesS_WolF | JoshuaFuller: oh so i can't give it a static IP ? |
| 15:12.58 | [TK]D-Fender | All hail the caps-lock key! |
| 15:12.59 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 15:13.36 | [TK]D-Fender | Actually Webmin does DHCPD.CONF pretty well IMO. Many thinkgs you wouldn't want it for but the rest, hey, why not. |
| 15:13.43 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 15:13.47 | jmls | [TK]D-Fender: thought you were just angry .. |
| 15:13.59 | JoshuaFuller | You can't force it as far as I know until it actually gets an IP from the DHCP server. Then you can provision it to stay a static IP. |
| 15:14.11 | ellisdee | in the event that i reload the musiconhold files...does asterisk require a reboot? |
| 15:14.27 | ellisdee | or just a config file rehash/reload? |
| 15:14.34 | [TK]D-Fender | ellisdee: You need to do : "reload res_musiconhold.so" |
| 15:15.24 | DarKnesS_WolF | done :P |
| 15:15.27 | ellisdee | nice |
| 15:15.28 | ellisdee | thanks man |
| 15:17.08 | [TK]D-Fender | ellisdee: np |
| 15:17.12 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 15:18.01 | *** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 15:18.05 | tmccrary | hey |
| 15:18.34 | DarKnesS_WolF | i don't get it why i need iaxprov.conf JoshuaFuller ? |
| 15:18.35 | tmccrary | With the email settings for voicemail, is it possible to bcc another email for archival purposes? |
| 15:20.09 | jsmith | tmccrary: Sure... set up an alias on your email server ;-) |
| 15:22.23 | tmccrary | that really sucks :( |
| 15:22.48 | tmccrary | in that case, unless I filter it, i'd catch ALL the email for each box |
| 15:23.09 | tmccrary | open source to the rescue I suppose :) |
| 15:26.00 | *** join/#asterisk andrebarbosa_ (n=andrebar@62.48.215.223) |
| 15:26.40 | RoyK | tmccrary: SMOP |
| 15:28.34 | rob0 | Postfix has sender_bcc_maps, so you could only archive the voicemail easily. |
| 15:29.39 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 15:31.25 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 15:31.49 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 15:33.58 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:34.48 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 15:35.22 | *** join/#asterisk Splas (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:39.55 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:41.59 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-76629.otenet.gr) |
| 15:42.08 | jgoo | is it ok to ask agi Q's here? |
| 15:42.31 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 15:42.44 | *** join/#asterisk Ard0gx (n=adiaz0@ns.sinergis.com.py) |
| 15:43.37 | *** join/#asterisk E-Rage (n=erage@nat-gw.seattle.identitext.com) |
| 15:44.58 | E-Rage | Anyone using Mitel SIP phones (5224) and have gotten the MWI light working? |
| 15:46.06 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 15:47.56 | NormanASD | DarKnesS_WolF: doesn't the iaxy use dhcp to get an IP? |
| 15:48.17 | mog | it can |
| 15:48.25 | NormanASD | by default, i mean. |
| 15:48.40 | mog | yes |
| 15:48.43 | NormanASD | that's how i set mine up. |
| 15:49.24 | NormanASD | thus to get it's default ip, DarKnesS_WolF, check your dhcp server. |
| 15:51.13 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:56.08 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 15:58.22 | jeremy_g | is it possible to change sip_uri =sip:3001@192.168.0.84:5060, |
| 15:58.36 | key2 | jeremy_g: from where |
| 15:59.32 | jeremy_g | i want to change this 3001@192. to 505@111 |
| 15:59.45 | jeremy_g | how do i change that: |
| 15:59.53 | jeremy_g | is it the CALLERID |
| 16:00.12 | jeremy_g | is this CALLERID=505@111 stupid? |
| 16:01.13 | [TK]D-Fender | jeremy_g: the 3001 & 505 is the callerID, and yes you can change it, |
| 16:01.34 | jeremy_g | but i want to change the latter part also that comes after @ |
| 16:01.50 | VibroMax | I want to try out asterisk using the asterisk @home iso and also want to try installing from tarball can I do this on one system ? |
| 16:01.51 | jeremy_g | how do i do that |
| 16:02.11 | jeremy_g | VibroMax:yes you can but use different ports |
| 16:03.18 | jeremy_g | [TK]D-Fender:i want to change the host part in that like sip:505@111.12.12.12 |
| 16:03.43 | jeremy_g | is there something like HOSTID=111.12.12.12 in * or some way to do it |
| 16:03.59 | [TK]D-Fender | jeremy_g: Why? |
| 16:04.05 | VibroMax | jeremy_g: I was thinking along the line of installing the tarball to a fedora core partition. I dont really need the two to be running simultaneously |
| 16:04.50 | [TK]D-Fender | VibroMax: Not without partitioning on your HD or something else like that. |
| 16:06.04 | jeremy_g | [TK]D-Fender:to let the gateway know which ip to switch to |
| 16:06.11 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 16:06.41 | [TK]D-Fender | jeremy_g: the IP you see is *'s IP. * is NOT a proxy, it is a B2BUA. Learn this and quick. |
| 16:07.04 | jeremy_g | [TK]D-Fender:the * box is talking to another sip box which does routing between medias. it looks at the sip uri and then routes me to my home network. i need a way to tell it that this is my home network id |
| 16:07.13 | jeremy_g | damn, i am so tired by now |
| 16:07.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:07.32 | jeremy_g | DNID, EXTENSION.... been playing all day |
| 16:07.50 | jeremy_g | i wudnt have to do this if there was a way to change the DNID |
| 16:08.09 | jeremy_g | [TK]D-Fender:if you know anything that can help, plz tell now. |
| 16:10.33 | *** join/#asterisk jgoo_ (n=e4b80e21@foodtecsolutions.com) |
| 16:11.49 | fenlander | Hi guys. I'm chasing an rtp bug where rfc2833 stops flowing after a period of time - 5 to 20 minutes from the start of the call (in 1.4beta) - It looks like the seqno test is failing - any ideas? is this a known problem? |
| 16:13.22 | jeremy_g | fenlander:which asterisk release u using |
| 16:15.18 | *** join/#asterisk Qwell[] (i=qwell@nat/digium/x-7f84e9c04d74f488) |
| 16:15.18 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:18.10 | Dr-Linux | question, can i get callerID in below monitor line? : |
| 16:18.11 | Dr-Linux | exten => s,2,SetVar(CALLFILENAME=/home/pasport${EXTEN:1}-${TIMESTAMP}) |
| 16:18.11 | VibroMax | jeremy_g: from what I have read so far the asterisk@home iso wipes all data from the drive before installation I wanted to shrink the partition after the install and put fedora on the second partition |
| 16:18.28 | Dr-Linux | exten => s,2,SetVar(CALLFILENAME=/home/pasport${EXTEN}-${TIMESTAMP}) |
| 16:19.27 | Corydon-w | TIMESTAMP is deprecated, use STRFTIME() instead |
| 16:20.32 | Dr-Linux | Corydon-w: thanks, but can i get caller ID or called id, actually when i sort a bunch of calls, i wanna recognize them. what you suggest? |
| 16:21.16 | jeremy_g | VibroMax:asterisk at home bootable cd;) |
| 16:21.21 | Corydon-w | CALLERID(num) |
| 16:21.41 | [TK]D-Fender | Dr-Linux: There is a function for that you know. This is Asterisk 101 stuff I reall can't believe you're asking this.... and SetVar is deprecated too.... |
| 16:21.42 | jeremy_g | VibroMax:reads the configs from hd and rest is like a live distro |
| 16:22.22 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:22.36 | jeremy_g | fenlander:are u even there? i know dtmf |
| 16:22.40 | Dr-Linux | :S |
| 16:22.58 | jeremy_g | Dr-Linux:change ur friggin nick, u are as pathetic as me |
| 16:23.28 | jeremy_g | fenlander:there are issues in 1.10 relesaes and prolly in .9 as well |
| 16:24.23 | pingwin | should anyone running 1.2 look seriously at moving to 1.4? |
| 16:24.29 | Dr-Linux | jeremy_g: i'm dull :P but fine with my all * servers :P |
| 16:24.30 | pingwin | is there a real benefit? |
| 16:24.33 | jsmith | pingwin: Yes |
| 16:24.33 | FlatFoot | mut: i think audacity |
| 16:24.55 | mut | yea.. |
| 16:25.18 | mut | you lagged like an hour and a half |
| 16:25.20 | mut | :P |
| 16:25.28 | FlatFoot | mut: sorry was a bit behind , had the screen rolled up |
| 16:25.34 | mut | ah |
| 16:25.37 | mut | heh i hate when i do that |
| 16:25.50 | FlatFoot | just me in PLANK mode |
| 16:26.01 | pingwin | jsmith any particular reasons? I'm using SIP, queues and talking on a zaptala / pri outbound line |
| 16:26.25 | jsmith | pingwin: Many improvements to the SIP channel and queues |
| 16:26.34 | [TK]D-Fender | pingwin: You running fairly stable? |
| 16:26.51 | pingwin | yeah, just launched the system about a week ago |
| 16:26.58 | pingwin | still doing tweeks of course |
| 16:30.31 | [TK]D-Fender | pingwin: Then I'd wait for a while past the 1.4 official release before thinking of moving. Stabilize first, transition later. |
| 16:31.03 | jsmith | pingwin: Test the 1.4 betas, but don't run it in production until it's released and had a chance to settle down. |
| 16:31.20 | pingwin | werd, just curious. have read about a few of the new 1.4 features and some sound really cool |
| 16:31.55 | pingwin | jsmith unfortunately I don't have the spare pc's and phones to do any sort of beta testing. it's all or nothing sort of environment |
| 16:32.10 | pingwin | sure that's ideal, but for most the ideal situation doesn't exist |
| 16:32.18 | jsmith | pingwin: Then pick your poison :-) |
| 16:32.28 | pingwin | hehe I'm used to it :P |
| 16:32.41 | [TK]D-Fender | pingwin: Well wait for the official release before touching it. I'd clearly not spend time with the beta in your position. look at it later I guess |
| 16:32.58 | jmls | pingwin: if you have no spare pc around, then wtf are you going to do if/when the pc hosting asterisk crashes ? Wait for 4 weeks while you order another one ? |
| 16:33.04 | pingwin | [TK]D-Fender i will, just curious about stability of 1.4 and if it's a good option to migrate |
| 16:33.34 | pingwin | jmls i'll fix the crashed machine. |
| 16:33.42 | [TK]D-Fender | pingwin: Well the itea is always good in one way, but you always risk the stability you have in so doing. Telling us you don't have a test box obviously makes this harder. |
| 16:33.48 | pingwin | jmls usually i can piece together a base machine from anything crashed |
| 16:34.12 | jmls | yikes. My users would *kill* me if we are down for anything more than 5 minutes |
| 16:34.15 | pingwin | [TK]D-Fender yep. i'll wait for a stable 1.4 and probably a few post releases after |
| 16:34.38 | pingwin | jmls same here. I run a web hosting company like this also. minimum resources |
| 16:35.01 | pingwin | jmls just configure really well and I know my shit so I usually don't run into too many problems |
| 16:35.25 | pingwin | jmls one of my old servers at my last job had over 410 days uptime before the dumbass after me did a reboot |
| 16:36.01 | pingwin | [TK]D-Fender yeah, I'd love to have a test environment. but obviously none such exists for me |
| 16:36.51 | *** part/#asterisk Ard0gx (n=adiaz0@ns.sinergis.com.py) |
| 16:37.06 | jmls | pingwin: you can never account for hardware failures. Last one I had was the SCSI raid controller (Adaptec) which trashed the data on *both* of the mirrors. Had to reinstall from scratch once we'd sourced more disks. That server was down for 4 hours. Never again. |
| 16:37.24 | jmls | (oh, and sourced another RAID controller) |
| 16:37.51 | jmls | Much cheaper in the long run to have a spare server as a hot spare |
| 16:38.38 | pingwin | jmls had that happen before, but I ran a backup disk. so I recovered from that in just a few hours. using a very small spare ide hard drive until I could get the new parts |
| 16:38.56 | pingwin | well going to lunch guys. |
| 16:39.01 | jmls | yeah - that's the point: "a few hours" |
| 16:39.13 | jmls | not "5 minutes" |
| 16:39.22 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-71-74.dsl.irvnca.pacbell.net) |
| 16:39.26 | DasTech | morning |
| 16:39.35 | jmls | evening ;) |
| 16:39.48 | DasTech | anyone by chance know where I can get the app_flite.c fie |
| 16:39.56 | DasTech | to make flite work in asterisk |
| 16:40.32 | DasTech | flite much better then Festival |
| 16:40.39 | DasTech | much cleaner and faster |
| 16:40.54 | DasTech | also anyone get sphinx to work with asterisk ? |
| 16:41.41 | jmls | yikes! slow down ! :) |
| 16:41.42 | Flauto | dastech, i was using flite |
| 16:42.02 | Flauto | but it stopped working with 1.2 |
| 16:42.08 | Flauto | but not 1.4 |
| 16:42.09 | DasTech | why saw was ? |
| 16:42.15 | DasTech | ok |
| 16:42.31 | DasTech | well I know they use it in trixbox but thats linux |
| 16:42.36 | DasTech | I am building on fbsd |
| 16:43.38 | Flauto | okay |
| 16:43.41 | Flauto | i never used that |
| 16:43.49 | Flauto | so, i can not tell you anything about it |
| 16:44.01 | DasTech | how did you get it to work with linux |
| 16:44.08 | DasTech | I can modify it for bsd |
| 16:44.24 | DasTech | did you rewrite the festicl.c file ? |
| 16:44.33 | DasTech | app_festival ? |
| 16:44.37 | jgoo_ | oh, I wanted to ask, on the status of Flite and Festival |
| 16:44.59 | DasTech | FLite is better |
| 16:45.07 | DasTech | brb |
| 16:45.44 | Flauto | if you search asterisk weather flite |
| 16:45.47 | Flauto | you will get it |
| 16:45.52 | Flauto | but it is for trixbox |
| 16:45.53 | fenlander | jeremey_g: oops sorry - popped out. It's a known problem and has been fixed in svn. Thanks for replying! |
| 16:46.16 | jgoo_ | I am using trixbox... does it come with it? (I am in #freepbx if you want to answer in there) |
| 16:46.23 | Flauto | i am still using it but i could not get app_flite.c to work with 1.4 |
| 16:46.41 | Flauto | jgoo, i think so |
| 16:46.46 | DasTech | I need to get the app_flite.c file |
| 16:46.52 | DasTech | I can fix it |
| 16:47.08 | jgoo_ | Flauto: can I pick your brains in #freepbx for a bit |
| 16:47.08 | Flauto | destech, do you use any messengers, i can send you one |
| 16:48.18 | jgoo_ | do I need that you think? (latest trix) I think I can call channel.execute("Flite...."); to run Flite from fastAGI |
| 16:48.21 | DasTech | yahoo |
| 16:48.38 | DasTech | tierco is the nic |
| 16:48.39 | Flauto | give me your yahoo id |
| 16:48.54 | jgoo_ | Flauto: you use jabber/gizmo/gmail? |
| 16:48.55 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 16:49.17 | Flauto | jgoo, i have been trying to use gtalk on asterisk |
| 16:49.22 | Flauto | but not working |
| 16:49.32 | jgoo_ | gizmo does it :-) |
| 16:49.33 | Flauto | i am still trying now on another computer |
| 16:49.39 | Flauto | hope it works |
| 16:49.48 | Flauto | gizmo does gtalk> |
| 16:49.53 | jgoo_ | yep |
| 16:49.53 | Flauto | really |
| 16:50.00 | Flauto | how it works |
| 16:50.09 | Flauto | maybe i can just use them instead |
| 16:50.13 | jgoo_ | they all use xml, jabber |
| 16:50.23 | jgoo_ | and SIP compatible |
| 16:50.26 | jgoo_ | for talking |
| 16:50.44 | jgoo_ | plus you get free calls to landlines with gizmo, if you are a regular user, crazy good |
| 16:51.03 | DasTech | tier_co or unixbehr |
| 16:51.06 | jgoo_ | I am just itching to uninstall skype for ages now |
| 16:51.07 | DasTech | sorry |
| 16:51.13 | DasTech | my brain hurts |
| 16:51.23 | jgoo_ | slowly migrating my contact list to gizmo when they jump ship |
| 16:51.41 | *** join/#asterisk lukketto (n=lukketto@host97-106-dynamic.59-82-r.retail.telecomitalia.it) |
| 16:52.31 | *** join/#asterisk Bpedersen (n=bp@82-192-171-74.sk.dsl.struer.net) |
| 16:53.10 | Inverted | why would this call from an agi not be working? print "EXEC Read userID|audiofile|10|skip|2|15"; |
| 16:54.18 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 16:54.19 | ellisdee | how is everyone defeating the reset of passwords for users after ntpd has adjusted the time on the systems? |
| 16:54.21 | diclophis-work | hello all |
| 16:56.11 | *** join/#asterisk kannan (i=1001@61.8.147.164) |
| 16:57.21 | Flauto | jgoo, use uplink to work with asterisk |
| 16:57.26 | DasTech | anyone working on 1.4 to make sure it works on fbsd |
| 16:59.54 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-219-146.red.bezeqint.net) |
| 17:00.13 | jgoo_ | Flauto?? |
| 17:00.13 | Dr-Linux | [TK]D-Fender: i want defualt system time in this function, what's the syntax? |
| 17:00.17 | Dr-Linux | i.e. STRFTIME([<epoch>][,[<timezone>][,<format>]]) |
| 17:00.17 | jgoo_ | uplink? |
| 17:00.52 | jgoo_ | Flauto: for some reason: char option = getOption("test-main-intro", "123"); |
| 17:00.54 | jgoo_ | plays nothing |
| 17:01.07 | jgoo_ | and channel.exec("Flite('Hello asterisk user, how are you today?')"); / channel.exec("Festival('Hello asterisk user, how are you today?')"); |
| 17:01.12 | jgoo_ | both also play nothing... :s |
| 17:01.27 | ellisdee | what os are you using? |
| 17:01.30 | ellisdee | gentoo? |
| 17:01.37 | Dr-Linux | Corydon-w: any clue? |
| 17:02.08 | jgoo_ | "AGI Script Executing Application: (Flite('Hello asterisk user, how are you today?')) Options: () |
| 17:02.39 | ellisdee | Flite('Hello........') maybe? |
| 17:02.47 | Corydon-w | Dr-Linux: STRFTIME(||%c) should be enough, I think |
| 17:02.49 | jgoo_ | ?? that is what I wrote |
| 17:03.12 | jgoo_ | ellisdee: you see a problem with the code? it doesn't throw an error, I just... don't hear anything. |
| 17:03.15 | Inverted | jgoo_: why would this call from an agi not be working? print "EXEC Read userID|audiofile|10|skip|2|15"; |
| 17:03.23 | jgoo_ | nor my recordings, which I sox'd |
| 17:03.23 | ellisdee | oh |
| 17:03.30 | Corydon-w | Dr-Linux: or just STRFTIME() |
| 17:03.31 | ellisdee | i had a similiar problem |
| 17:03.39 | ellisdee | jgoo, i had a gentoo box that had similiar problems |
| 17:03.46 | jgoo_ | Inverted: I looked at that, and didn't see why not. Inverted: all I say is, use fastAGI, nicer coding interface |
| 17:03.54 | ellisdee | jgoo, resorted to a centos solution that had the same syntax.. worked flawless. |
| 17:04.00 | Dr-Linux | Corydon-w: i tried that already, but |
| 17:04.01 | jgoo_ | ellisdee: aaaah. this is ... the naughty T word... trixbox |
| 17:04.02 | ellisdee | jgoo, i am not certain on the exact differences.. |
| 17:04.20 | jgoo_ | ellisdee: it is * syntax though |
| 17:04.23 | jgoo_ | os agnostic |
| 17:04.29 | ellisdee | erm |
| 17:04.36 | jgoo_ | Flite() is * syntax ain't it? |
| 17:04.49 | Inverted | jgoo_: I am actually using eagi |
| 17:05.11 | jgoo_ | eagi? hrm what is that? I am using a fastAGI server, I love it... great stuff... if it works :p |
| 17:05.54 | Dr-Linux | hhm... |
| 17:06.02 | Dr-Linux | Corydon-w: this format comes : 46-Tue Oct 10 17:24:11 2006.wav |
| 17:06.28 | Corydon-w | Dr-Linux: yeah, that's the default system format |
| 17:06.46 | Corydon-w | Dr-Linux: if you want a different format, you need to specify that format |
| 17:06.53 | Inverted | jgoo_: eagi gives you access to the audio stream |
| 17:07.36 | Flauto | jgoo, sorry, i am slow |
| 17:07.38 | Flauto | i am working |
| 17:07.43 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 17:07.51 | Dr-Linux | Corydon-w: i want something like this MM-DD-YYYY |
| 17:07.57 | Optic | well, the office is in full swing, and no crashes so far! |
| 17:08.23 | jgoo_ | Flauto: think nothing of it |
| 17:08.50 | jgoo_ | I am at office too, at least until midnight :/ |
| 17:08.58 | Corydon-w | Dr-Linux: so something like %m-%d-%Y |
| 17:09.16 | *** join/#asterisk lero (n=rootz@200.192.160.100) |
| 17:09.34 | lero | anyone here use munin with asterisk scripts? |
| 17:10.01 | Corydon-w | Dr-Linux: see the manpage for strftime(3) |
| 17:10.37 | Dr-Linux | Corydon-w: yeah i'm seeing that page |
| 17:12.12 | Dr-Linux | hhm.. |
| 17:12.18 | *** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 17:12.36 | Dr-Linux | Corydon-w: STRFTIME(,,%F) is just fine |
| 17:12.37 | Dr-Linux | 46-2006-10-10.wav |
| 17:12.46 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:15.02 | krondorl | Anyone ever have an issue with the phones getting all locked up causing the inability to call in (busy signal) or call out (called failed message). The CLI shows that there are constant calls being processed yet there is nothing happening. |
| 17:15.21 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 17:15.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.100) |
| 17:17.25 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 17:27.02 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 17:29.04 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 17:29.14 | backblue | hi |
| 17:29.22 | backblue | anyone have allready collected 1 digit with disa? |
| 17:31.30 | syzygyBSD | krondorl: yes |
| 17:37.05 | p1p | anyone ever setup a bogen UTI1 with a linksys ata? |
| 17:37.11 | *** join/#asterisk mtaht4 (n=m@h-72-244-145-197.phlapafg.covad.net) |
| 17:37.34 | *** join/#asterisk cian (n=cian@cian.ws) |
| 17:38.03 | krondorl | syzygyBSD: What was the cause? And how does one stop it? |
| 17:39.07 | syzygyBSD | well, mine was caused by an agent's phone not correctly tranfering a call, tying up a zap channel |
| 17:40.07 | krondorl | Did that cancel the call but left everything still open? |
| 17:40.26 | Flauto | [Oct 10 12:37:12] WARNING[16967]: loader.c:376 load_dynamic_module: Module 'app_flite.so' did not register itself during load |
| 17:40.26 | Flauto | [Oct 10 12:37:12] WARNING[16967]: loader.c:608 load_resource: Module 'app_flite.so' could not be loaded. |
| 17:40.37 | krondorl | Ie: No one on the phone but system still though they were? |
| 17:40.38 | Flauto | this is what i got when i restarted my asterisk |
| 17:40.47 | Flauto | so, flite is not loaded |
| 17:41.15 | syzygyBSD | krondorl: it made it hangup the call, but asterisk still thought the zap channel was in use, and wouldn't accept any new calls on it |
| 17:41.48 | *** join/#asterisk lrizzo (n=luigi@88-149-142-59.f5.ngi.it) |
| 17:43.27 | krondorl | any known fix for that problem? |
| 17:46.42 | Inverted | is asterisk-perl not in cpan? |
| 17:47.57 | Inverted | guess not yet |
| 17:48.16 | ghenry | nope Inverted |
| 17:48.28 | ghenry | I think it's going to be very soon |
| 17:48.28 | ghenry | I offered to help |
| 17:49.50 | ghenry | I sometimes wish it was written by someone else with more time ;-) |
| 17:50.33 | beyond | hi all! can AEL be used in production sites or is it still considered as experimental? |
| 17:50.47 | *** join/#asterisk txotin (n=egomez@56.Red-80-38-204.staticIP.rima-tde.net) |
| 17:50.50 | Qwell[] | beyond: as of 1.4, I believe the "experimental" was removed from ael |
| 17:50.53 | txotin | hi all |
| 17:51.20 | beyond | Qwell, very cool |
| 17:51.57 | [TK]D-Fender | Qwell : But how many will truely care anout it? |
| 17:52.03 | [TK]D-Fender | about* |
| 17:52.08 | *** join/#asterisk angom (n=angom@red-corp-200.79.141.244.telnor.net) |
| 17:52.12 | Inverted | ghenry: someone is doing a remake of it |
| 17:54.14 | kannan | anyone has asterisk settings mdoel for Broadvox, be very thankful for it can't get calls out yet |
| 17:56.30 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 17:57.19 | Qwell[] | [TK]D-Fender: exactly 6 people |
| 17:57.28 | *** part/#asterisk lrizzo (n=luigi@88-149-142-59.f5.ngi.it) |
| 17:57.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:01.42 | syzygyBSD | krondorl: sorry for the delay, I had to revert back a couple versions |
| 18:01.52 | syzygyBSD | went from 1.2.12 to 1.2.9 |
| 18:03.09 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 18:03.30 | Assid | err.. anyone remember the sipdiscount/voipbuster network site which gives free luthiana |
| 18:04.00 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:08.20 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:12.03 | *** join/#asterisk MIXX941 (n=Mark@unaffiliated/mixx941) |
| 18:12.13 | ghenry | Inverted: Someone is, cool. I never liked the code at all. All a bit oldstyle OO Perl. You know who? |
| 18:14.56 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 18:15.33 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-71-74.dsl.irvnca.pacbell.net) |
| 18:19.03 | Inverted | ghenry: talk to Corydon, I believe he helps with it |
| 18:23.15 | krondorl | syzygyBSD: Do feel that is the only way to fix the problem? |
| 18:24.00 | syzygyBSD | there if your problem is the same as mine, there is also a patch available, I just prefered to revert and will upgrade once the patch in in the latest stable |
| 18:24.10 | *** join/#asterisk fugitivo (n=ajf@190.48.182.225) |
| 18:24.17 | fugitivo | hello |
| 18:24.23 | syzygyBSD | high |
| 18:25.46 | *** join/#asterisk sloth (n=sloth@pool-162-83-220-153.ny5030.east.verizon.net) |
| 18:26.07 | ghenry | Corydon-w: You helping with a new Asterisk::AGI ? |
| 18:26.15 | fugitivo | in my asterisk server sometimes the sip module freezes, if I do sip reload it doesn't do anything, sip debug doesn't show anything neither, only way to fix that is restarting asterisk, any idea? |
| 18:26.21 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:26.49 | fugitivo | i |
| 18:26.53 | fugitivo | i'm using asterisk 1.2.12.1 |
| 18:26.58 | sloth | Hello, I have a quick one: When I pull a value from astdb into a variable, the variable hold that value however it is quoted. I need to remove these quotes. Any idea? |
| 18:28.00 | stephane_ | re |
| 18:29.56 | *** join/#asterisk flujan_ (n=flujan@internet.nube.com.br) |
| 18:31.00 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
| 18:31.19 | japerry | anyone know how to get Asterisk Debug numbers to show in the logs |
| 18:31.22 | *** join/#asterisk samsn (n=kein@84-73-90-64.dclient.hispeed.ch) |
| 18:31.32 | japerry | I'm trying to figure out what numbers incoming callers are pressing |
| 18:32.25 | japerry | we're getting a very strange issue where most callers can get through the IVR menu, but some phone systems don't work at all. They get to our main number, and any digit they dial will not get any response, basically putting IVR through a loop until it hangs up |
| 18:32.56 | *** join/#asterisk RoyK (n=roy@ti211310a080-6054.bb.online.no) |
| 18:33.43 | MIXX941 | does anyone know any external ethernet gateways with two FXO ports? or if one had a single FXO port but could take from two phone lines over the same port and split them to two SIP/IAX channels to allow the two lines to get into * |
| 18:35.04 | *** join/#asterisk soylentgreen (n=fgast@dslb-082-083-086-081.pools.arcor-ip.net) |
| 18:35.06 | xming | sipura2k? |
| 18:38.57 | krondorl | is there a time limit for the voicemail system before it stops recording? |
| 18:39.26 | fugitivo | krondorl: maxmessage in /etc/asterisk/voicemail.conf |
| 18:39.49 | fugitivo | anyone having problems with sip being freezed? |
| 18:40.03 | wmandra | freezed?? |
| 18:40.16 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 18:41.13 | fugitivo | yes |
| 18:41.24 | fugitivo | sip reload doesn't do anything, sip debug neither |
| 18:41.27 | fugitivo | restarting asterisk is the solution |
| 18:41.43 | MIXX941 | xming: SPA-2000? I mean two phone lines, not two phones. |
| 18:42.08 | flujan_ | hi guys... I am having problems with the jitter in my asterisk box. So my company bought a switch that supports TOS. |
| 18:42.15 | japerry | has *anyone* else seen an issue with asterisk not accepting digits being dialed from incoming callers? |
| 18:42.28 | flujan_ | I change the tos=lowdelay in the iax.conf but the packages are not been marked as lowdelay in the tos. |
| 18:42.38 | flujan_ | I dump some of them using the ethereal tool |
| 18:43.07 | Splas | Anyone here using the ChanSkype thing? |
| 18:43.14 | flujan_ | Did you see this problem before? |
| 18:44.36 | flujan_ | I dunno why this is happening. |
| 18:45.34 | krondorl | fugitivo: Is there a system default if there is no entry?? |
| 18:49.23 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-17eafb115242c964) |
| 18:49.55 | fugitivo | krondorl: i don't think so, if it's, it's too long |
| 18:50.37 | krondorl | my problem is people are getting cut off after about 18 seconds and there is no entry in the voicemail.conf.. |
| 18:51.06 | fugitivo | do you see any message on the cli? |
| 18:51.30 | krondorl | we think it might be the fact that the person's voice is not recording and the system thinks it's dead air. |
| 18:52.00 | fugitivo | get into the cli and check if you see any message |
| 18:52.27 | fugitivo | you should see something like "recording too long, cutting... " or something like that |
| 18:54.58 | krondorl | nope nothing like that is showing... |
| 18:56.00 | fugitivo | asterisk -rvvvvvvvvvv |
| 18:59.09 | *** join/#asterisk TexasJay (n=me@ns1.accu-com.com) |
| 18:59.54 | *** join/#asterisk oQPa (n=Spain@237.Red-83-44-33.dynamicIP.rima-tde.net) |
| 19:00.34 | TexasJay | Hey, what's wrong with this syntax: |
| 19:00.34 | TexasJay | exten => *1,1,Wait(2) |
| 19:00.35 | TexasJay | exten => *1,2,AgentCallbackLogin(||${CALLERIDNUM}) |
| 19:00.38 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 19:00.54 | TexasJay | it still prompts me for an extension :/ |
| 19:01.15 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:02.28 | *** join/#asterisk ys0987 (i=ys0987@61.247.229.200) |
| 19:03.09 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 19:03.12 | drcode | hi all |
| 19:03.31 | drcode | well , I read about tree users that I can set in sip.conf |
| 19:03.45 | drcode | If I want to use pc to pc , what users I need to set? |
| 19:03.48 | drcode | peer,frendly ? |
| 19:04.05 | ys0987 | Hi everyone!! Need some help on IVR to capture 4 digit id of caller. Tried using pattern matching with background. Did not work. Can anyone help? |
| 19:04.08 | fugitivo | TexasJay: try with ${CALLERIDNUM}@context |
| 19:04.50 | TexasJay | What if I don't know the context? |
| 19:04.52 | drcode | any eaxmple what I need to set ? |
| 19:04.59 | TexasJay | I mean, I have several contexts. |
| 19:05.27 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:06.10 | ys0987 | Hi all!! Need some help on IVR |
| 19:06.15 | flujan_ | guys someone having problems with asterisk not tagging packages in the TOS field? |
| 19:06.27 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 19:07.36 | ys0987 | Hi everyone!! Need some help on IVR to capture 4 digit id of caller. Tried using pattern matching with background. Did not work. Can anyone help? |
| 19:09.24 | syzygyBSD | ys0987: paste the part of the dialplan you are using to pastebin |
| 19:09.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:09.34 | *** join/#asterisk cian (n=cian@cian.ws) |
| 19:10.57 | *** join/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net) |
| 19:11.00 | *** join/#asterisk myiagy (n=myiagy@201.64.81.120) |
| 19:11.33 | Crad | Hey is there a way to log SIP packets in asterisk? I want to see the actual SIP invitation being sent. |
| 19:11.43 | Crad | (or received) |
| 19:12.05 | wunderkin | sip debug |
| 19:12.10 | Crad | thanks |
| 19:12.16 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 19:12.19 | drcode | hi all |
| 19:12.31 | drcode | I have setup [xliet1] in astrkis |
| 19:12.34 | sloth | ys0987: you can use the Read() cmd |
| 19:12.41 | drcode | now I can connect from x-lite |
| 19:12.58 | drcode | I want to do pc to pc , so both will connect into asterisk? |
| 19:13.21 | sloth | ys0987: nevermind I onle half read your request |
| 19:14.29 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 19:14.37 | jm|home | hullo :) |
| 19:14.53 | jm|home | can I make my IP phone use the FXO to make the analogue phones ring? |
| 19:15.03 | jm|home | kinda like an intercom |
| 19:16.33 | *** part/#asterisk Seyr|afk (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 19:17.33 | ys0987 | exten => 2005,1,Answer() |
| 19:17.33 | ys0987 | exten => 2005,2,Noop(${EXTEN}) |
| 19:17.33 | ys0987 | ;exten => 2005,3,GoTo(_XXX,1) |
| 19:17.33 | ys0987 | exten => _XXX,1,Noop(${EXTEN}) |
| 19:17.33 | ys0987 | exten => _XXX,2,Background(why-no-answer-mystery) |
| 19:17.33 | ys0987 | ;exten => _XXX,3,SayDigits(${EXTEN}) |
| 19:17.35 | ys0987 | exten => _XXX,3,SetVar(testvar=${EXTEN}) |
| 19:17.37 | ys0987 | exten => _XXX,4,SayDigits(${testvar}) |
| 19:17.39 | ys0987 | exten => _XXX,5,Hangup |
| 19:17.41 | wunderkin | you need a fxs module to connect a phone, fxs = station = phone, fxo = co = phone line |
| 19:17.54 | jm|home | wunderkin: I get it |
| 19:18.16 | jm|home | wunderkin: I can't 'control' the signals on the FXO 'line' |
| 19:18.50 | ys0987 | syzygyBSD . after dialing 2005 from sip phone i heard sound file |
| 19:19.26 | ys0987 | but as soon as i press any digit it again replay whole sound and then say the digit |
| 19:20.21 | ys0987 | any help plz |
| 19:20.59 | intralanman | ys0987: you really should be using pastebin for stuff like that |
| 19:21.02 | intralanman | ~pb |
| 19:21.09 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:21.46 | wmandra | hey all..... anyone have any suggestions on getting outbound callerid working with voicepulse connect via sip????? |
| 19:22.21 | jm|home | hmm |
| 19:22.58 | jm|home | I have parallel analogue phones on my Zap extension |
| 19:23.12 | pingwin | i ran a macro for ourgoing calls to set the caller id info |
| 19:23.12 | japerry | so... anyone have an issue where DTMF digits are not being transmitted on an incoming call? |
| 19:23.17 | jm|home | how do I make Asterisk realise that someone else has picked up? |
| 19:23.51 | japerry | jm|home: get rid of the analog phones and get simple digital ones |
| 19:23.59 | jm|home | :) |
| 19:24.04 | jm|home | japerry: I wish I could! |
| 19:24.11 | jm|home | $$ :( |
| 19:24.17 | syzygyBSD | ys0987: well, the extension is only 3 numbers long |
| 19:24.33 | syzygyBSD | and pastbin that not paste into the channel |
| 19:24.46 | drcode | if I want to use pc to pc . what I need to set in exention.conf? |
| 19:24.48 | japerry | ys0987: hey you there? |
| 19:24.49 | jm|home | it should re-check the status of the line before it does vm-nobodyavail |
| 19:24.51 | ys0987 | ya thats fine |
| 19:25.11 | japerry | ys0987: are you trying to get asterisk to repeat back the digits you're dialing in? |
| 19:25.27 | ys0987 | ya |
| 19:25.45 | japerry | ys0987: curious, are you doing it for shits and giggles or is there a problem with people calling in? |
| 19:25.51 | ys0987 | but it again replay the whole sound file and the say that digits |
| 19:26.13 | ys0987 | i dont want to hear the whole sound file |
| 19:26.13 | syzygyBSD | ys0987: add a set(digittimeout=5) |
| 19:26.22 | wmandra | pingwin: i have Set(CALLERID(num)=XXXXXXXXXX) before the dial but the callerid doesn't show up on |
| 19:26.54 | drcode | any help |
| 19:27.10 | wmandra | does voicepulse need sendrpid=true in sip.conf |
| 19:27.31 | ys0987 | what will happen then |
| 19:27.53 | jm|home | hmmm |
| 19:28.42 | jm|home | so there's a laptop in the kitchen; where one of the analogue phones are ... it uses wireless to get the wLAN but has an wired ethernet on it, too. I could have it bridge the two and plug an IP phone into it |
| 19:28.50 | jm|home | that'd be one phone down |
| 19:29.02 | jm|home | the one in the lounge is DECT analogue .... what are those adaptors called? |
| 19:30.00 | jm|home | grr @ skype everything |
| 19:30.33 | ys0987 | i wanna capture those digit and then take action accordingly "japerry" |
| 19:34.34 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:36.09 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 19:37.54 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 19:38.51 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 19:39.03 | *** join/#asterisk saftsack (n=oliver@p54A7D488.dip.t-dialin.net) |
| 19:40.06 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 19:41.02 | *** join/#asterisk bmg505 (n=leon@c1-106-11.rndf.isadsl.co.za) |
| 19:44.52 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 19:45.53 | *** join/#asterisk sysreq (n=sysreq@142-217-213-125.telebecinternet.net) |
| 19:45.55 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 19:47.03 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 19:47.07 | drcode | k |
| 19:47.09 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 19:47.24 | drcode | I setup an ext. [tutorial] |
| 19:47.26 | drcode | exten => 1222,1,Dial(SIP/test1) |
| 19:47.34 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:47.47 | drcode | I try to dail 1222 , no answer |
| 19:48.21 | Dr-Linux|work | drcode: try debug |
| 19:48.28 | drcode | where? |
| 19:48.39 | Dr-Linux|work | drcode: on asterisk consol |
| 19:48.58 | Dr-Linux|work | drcode: when you dial 1222, what you see on consol? |
| 19:49.12 | jm|home | can I not make [from-zap] check the status of the line to ensure that there IS still an event2 before it does Answer -> Voicemail ? |
| 19:50.53 | TexasJay | AgentCallbackLogin() doesn't logout an old agent if a new agent picks the same extension does it? |
| 19:51.29 | Corydon-w | agentcallbacklogin is deprecated |
| 19:52.34 | ernie_ | why does in iax.conf the identifier of a provider has to be the same as the username? otherwise i get: Rejected connect attempt from 80.92.86.39, who was trying to reach '38837901005094@ |
| 19:52.39 | anthonyl | you should really just use agentlogin |
| 19:52.50 | TexasJay | agentlogin eh? Hmm... |
| 19:53.01 | *** join/#asterisk flujan_ (n=flujan@internet.nube.com.br) |
| 19:53.34 | flujan_ | does IAX2 really supports the TOS tag? |
| 19:53.37 | TexasJay | Is there a corresponding AgentLogout ? |
| 19:54.05 | flujan_ | I add the tos=lowdelay to my iax.conf but the packages are note been tagged with it. |
| 19:54.17 | flujan_ | someone already used the tos tag? |
| 19:54.39 | *** join/#asterisk svenna_ (n=svenna@p548D2872.dip0.t-ipconnect.de) |
| 19:54.52 | anthonyl | TexasJay, no there is not.. all of that is handled by chan agent once the agent is logged in |
| 19:55.18 | *** join/#asterisk arisjr (n=arisjr@galois.wahtec.com.br) |
| 19:55.28 | TexasJay | anthonyl: do you mean that if a logged in agent tries to log in again, it logs them out? |
| 19:57.53 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 19:59.02 | TexasJay | it appears that AgentLogin only keeps the user logged in until they hang up |
| 19:59.18 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-71-74.dsl.irvnca.pacbell.net) |
| 19:59.59 | flujan_ | hum... the iax2 debug show the tos of packages? |
| 20:00.04 | arisjr | hi there. where can I find some internet docs for diagnosing asterisk performance problems? |
| 20:00.10 | jsmith | flujan_: Which version of Asterisk are you using? |
| 20:00.31 | flujan_ | jsmith, 1.2.9.1 |
| 20:00.42 | jsmith | flujan_: Yes, that should work. |
| 20:00.43 | DasTech | ok who here has asterisk on fbsd |
| 20:00.51 | DasTech | I need a zaptel.conf file |
| 20:00.58 | jsmith | flujan_: Did you reload the IAX2 driver or restart Asterisk after making the change? |
| 20:01.02 | DasTech | its still not in the src like it shold be |
| 20:01.28 | flujan_ | I just set asterisk reload in the CLI. |
| 20:01.48 | flujan_ | jsmith, how can I debug it? |
| 20:02.32 | jsmith | flujan_: Try restarting, just to be sure... There are several things in the IAX driver that a "reload" won't catch. |
| 20:02.50 | jsmith | flujan_: Next, pull up the packets in a packet sniffer and see if they've got the ToS bits set |
| 20:04.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:05.37 | flujan_ | jsmith, ok i will try it with ethereal |
| 20:06.10 | fugitivo | again sip in my asterisk is frozen |
| 20:06.24 | fugitivo | i have to restart it |
| 20:06.25 | fugitivo | i |
| 20:06.33 | fugitivo | im the only one with that problem? |
| 20:06.57 | DasTech | ok no one here running asterisk on fbsd |
| 20:07.08 | DasTech | I need the lines for loader.conf |
| 20:07.10 | DocHolliday | maybe.. i'm running an older version though :) |
| 20:07.19 | DasTech | for zaptel |
| 20:07.22 | jsmith | fugitivo: That's not normal. Is there anything in the logs that would indicate why it's happening? |
| 20:07.28 | DasTech | to load the drivers |
| 20:07.46 | jsmith | DasTech: I didn't know there were working Zaptel drivers for FreeBSD |
| 20:08.04 | DasTech | yes they are in ports and there is a svn tree |
| 20:08.13 | DasTech | for over 3 years now |
| 20:08.18 | fugitivo | jsmith: sip debug doesn't work at all |
| 20:08.20 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 20:08.32 | fugitivo | jsmith: it seems that sip is blocked |
| 20:08.33 | DasTech | /usr/ports/misc/zaptel |
| 20:08.57 | DasTech | libpri and asterisk 1.2.12.1 are also in ports |
| 20:09.05 | fugitivo | jsmith: it started to happen with the last version (1.2.12.1) |
| 20:09.12 | *** join/#asterisk ambriento (n=melcon@201-27-80-238.dsl.telesp.net.br) |
| 20:09.47 | jsmith | fugitivo: That's not normal... I run dozens of Asterisk systems, and I've never seen anything like that. |
| 20:10.08 | fugitivo | jsmith: me neither, until now |
| 20:10.43 | jsmith | fugitivo: Did you check the logs? |
| 20:10.52 | jsmith | fugitivo: Did you turn the verbose and debug levels up? |
| 20:11.40 | DasTech | I am also porting freepbx to freebsd |
| 20:12.46 | fugitivo | jsmith: yes, i don't see anything |
| 20:14.11 | *** join/#asterisk _bk_ (i=bk@cleartext.overthewire.org) |
| 20:15.12 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 20:18.50 | *** join/#asterisk RoyK (n=roy@ti211310a080-6054.bb.online.no) |
| 20:19.14 | TexasJay | does anyone have a link to a decent agent login/logout script? i'm having problems creating one of my own. |
| 20:21.31 | *** join/#asterisk S^P (n=Masood@203.148.73.236) |
| 20:22.02 | jm|home | DasTech: cool! |
| 20:22.22 | S^P | CDR simple logging enabled. << I added cdr_mysql but its not logging |
| 20:22.24 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 20:22.37 | jm|home | zaptel-1.0_1 A FreeBSD Driver for FXO, FXS, BRI and PRI Telephony Cards |
| 20:23.38 | DasTech | got it loaded |
| 20:24.56 | japerry | so geh, another issue, anyone have problems with the digium e&m cards not passing caller id? callerid=asrecieved |
| 20:25.16 | jm|home | stupid mpg123 |
| 20:25.26 | qdk | jm|home: me2.. had any problems with * on FBSD? |
| 20:25.33 | *** join/#asterisk saftsack (n=oliver@p54A7D488.dip.t-dialin.net) |
| 20:25.37 | jm|home | qdk: I'm not wholly there, yet |
| 20:25.37 | qdk | jm|home: use native mp3? |
| 20:25.43 | jm|home | yeah, I might |
| 20:25.50 | jm|home | I get my IP deskphone tomorrow |
| 20:25.55 | jm|home | I've been arsing around with X-Lite thus far |
| 20:26.02 | qdk | jm|home: ok. |
| 20:26.10 | jm|home | I have been able to work calls from work on my home phone line |
| 20:26.19 | qdk | jm|home: ok, i also only test with softphones atm. |
| 20:26.38 | jm|home | qdk: I have no sip or aix accounts either :S |
| 20:26.45 | qdk | jm|home: ok, i had some issues with a SIP-account from a unknown setup. |
| 20:27.01 | qdk | jm|home: oh, ok... im like SIP/IAX only. |
| 20:27.20 | jm|home | yeah, full IP is probably 'easier' |
| 20:27.23 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 20:27.30 | jm|home | don't want to change my home number |
| 20:27.36 | jm|home | for non-regional or whatever |
| 20:27.37 | DasTech | now I have to build a box and get 1.4 to work |
| 20:27.40 | jm|home | not yet, anyway |
| 20:27.46 | DasTech | then I have 2 diff ver to work from |
| 20:27.48 | qdk | jm|home: well i have * with SS7 running on some linuxboxes. |
| 20:27.48 | jm|home | DasTech: vmware? |
| 20:28.14 | DasTech | I have to make the ports first |
| 20:28.30 | qdk | DasTech: thats easy.. well i compiles fine here on my FreeBSD 6.2 PRERELEASE with SMP/AMD64 |
| 20:28.35 | qdk | i = it |
| 20:28.37 | jm|home | (: |
| 20:28.49 | DasTech | 1.4 does |
| 20:29.08 | DasTech | but we need it in the ports tree also |
| 20:29.15 | qdk | and 1.2.12 works too.. but that was on 6.1 |
| 20:29.21 | qdk | DasTech: why? |
| 20:29.32 | jm|home | I was 'advised' that OpenPBX is the way to go ... |
| 20:29.34 | DasTech | I have 6.2 with asterisk 1.2.12.1 |
| 20:29.37 | qdk | DasTech: it takes forever to get a port release. |
| 20:29.46 | DasTech | not for me |
| 20:29.56 | DasTech | I had it to max andhe submits |
| 20:30.20 | DasTech | I know MaxSo |
| 20:30.22 | qdk | jm|home: OpenPBX is a word for a lot more than just * |
| 20:30.25 | jm|home | DasTech: port VMware Server, please ;) |
| 20:30.26 | DasTech | he is a ports commiter |
| 20:30.40 | DasTech | I thought it was in port |
| 20:31.01 | jm|home | don't think so |
| 20:31.01 | qdk | jm|home: VMware needs to many modules, so its linux only atm. |
| 20:31.06 | jm|home | vmware3 was the last one iirc |
| 20:31.14 | qdk | old crap. |
| 20:31.20 | jm|home | qdk: sure but vmnet etc. were ported |
| 20:31.37 | DasTech | <PROTECTED> |
| 20:31.47 | qdk | jm|home: ok. cool. |
| 20:32.11 | DasTech | ok |
| 20:32.15 | jm|home | Skype pisses me off |
| 20:32.35 | jm|home | I don't WANT usb |
| 20:32.49 | *** join/#asterisk sloth_ (n=sloth@pool-162-83-220-153.ny5030.east.verizon.net) |
| 20:33.16 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 20:34.01 | DasTech | ok now onto fixing freepbx port |
| 20:34.08 | DasTech | so it installs right |
| 20:35.38 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 20:37.05 | *** join/#asterisk oQPa (i=Ftv@237.Red-83-44-33.dynamicIP.rima-tde.net) |
| 20:41.11 | *** join/#asterisk SomethingISODD (n=Dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 20:41.29 | SomethingISODD | hello does anyone know where i can get a software call generator that will call random numbers?? |
| 20:41.48 | SomethingISODD | I need to place bulk calls |
| 20:42.17 | saftsack | has asterisk 1.4 a webinterface for configuring the dialplan? |
| 20:42.19 | qdk | SomethingISODD: callfiles? |
| 20:42.33 | qdk | SomethingISODD: though it sounds liek you wanna spam people. |
| 20:42.38 | jsmith | SomethingISODD: Build one with Asterisk, or buy somethink like Hammer |
| 20:42.40 | SomethingISODD | qdk, nota ctually |
| 20:42.53 | SomethingISODD | qdk, i need to test a softswitch |
| 20:42.57 | qdk | saftsack: you can probably find some 3rd part. |
| 20:43.00 | SomethingISODD | its going to go and come back to asterisk |
| 20:43.13 | qdk | SomethingISODD: SIP-accounts? |
| 20:43.21 | SomethingISODD | ya |
| 20:43.29 | qdk | SomethingISODD: google sipp |
| 20:43.32 | SomethingISODD | but it needs to be random numbers |
| 20:43.38 | SomethingISODD | can sipp handle random umbers |
| 20:43.42 | qdk | dunno |
| 20:43.46 | qdk | havent tried it yet |
| 20:43.48 | *** part/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 20:43.53 | saftsack | qdk, know it but will there be a from house solution? |
| 20:44.21 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 20:44.24 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 20:44.33 | SomethingISODD | qdk, ok thanks |
| 20:44.41 | qdk | saftsack: no idea, i hope not... it will take valuable time from some actually work on asterisk. |
| 20:44.47 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:45.29 | saftsack | hrhr |
| 20:45.45 | mattchis | Can anyone help me with an issue involving voicemail alerts to email? |
| 20:45.52 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:46.02 | hoobastooba | mattchis: what is your question? |
| 20:46.19 | *** join/#asterisk flujan_ (n=flujan@internet.nube.com.br) |
| 20:46.27 | flujan_ | jsmith, hi... I am back... :) |
| 20:46.42 | jsmith | flujan_: What did you find? |
| 20:47.11 | flujan_ | jsmith, please, take a look at: http://pastebin.ca/196717 it is not marking the TOS right? |
| 20:47.39 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 20:47.39 | *** join/#asterisk Dovid (n=dovi5988@barak.cellcom.co.il) |
| 20:48.05 | jsmith | flujan_: I don't know what to tell you -- it's always worked for me. |
| 20:48.29 | Dovid | flujan_: whats the issue ? |
| 20:49.32 | flujan_ | hi Dovid , and thanks jsmith |
| 20:49.45 | flujan_ | i want to use the TOS to iax.conf |
| 20:49.56 | flujan_ | but asterisk is not marking the packages... as you can see. |
| 20:49.58 | Dovid | TOS bits ? |
| 20:49.58 | flujan_ | :( |
| 20:50.02 | flujan_ | yeap! |
| 20:50.18 | Dovid | never used that. |
| 20:50.20 | Dovid | u want it for QOS ? |
| 20:50.31 | flujan_ | yes |
| 20:50.33 | Dovid | i know it works on SIP. never used IAX much |
| 20:50.44 | Dovid | what problems are ru running in to ? |
| 20:51.02 | flujan_ | big jitter... I am having gaps during the calls... |
| 20:51.04 | flujan_ | :( |
| 20:51.16 | Dovid | and without tos bits works good ? |
| 20:51.40 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 20:51.42 | ambriento | flujan_, that pastebin looks like no marks at all |
| 20:51.49 | *** part/#asterisk _bk_ (i=bk@cleartext.overthewire.org) |
| 20:51.52 | wmandra | flujan: if your switches/routers are capable conigure them to handle the QoS and don't bother with the TOS bit from * |
| 20:52.15 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 20:52.17 | Dovid | flujan_: it can very well be a router issue. I was banging my head against the wall for 2 days with a client |
| 20:52.28 | Dovid | $300 router and it was the problem |
| 20:52.37 | Dovid | switched to $50.00 one and worked like a charm |
| 20:53.16 | flujan_ | ambriento, yes this is the problem... no marks at alll.. |
| 20:53.39 | flujan_ | Dovid, whe are using a linksys $600 SO BIG PROBLEM... LOL!!!! |
| 20:54.10 | ambriento | flujan_, who is the SRC and DST? |
| 20:54.17 | Dovid | Flujan_: thats what my client had. i had him use a junky one from circuit city and it solved all issues |
| 20:54.28 | Dovid | any way u can drop a seperate DSL line for them ? |
| 20:54.29 | ambriento | I mean, which one is the server? |
| 20:54.54 | *** join/#asterisk sloth_ (n=sloth@pool-162-83-220-153.ny5030.east.verizon.net) |
| 20:55.22 | flujan_ | http://pastebin.ca/196728 my iax.conf |
| 20:55.35 | flujan_ | ambriento, 192.168.1.6 is the server 192.168.1.103 is my station. |
| 20:56.43 | ambriento | flujan_, then the pastebin shows that your station isn't marking the TOS field |
| 20:56.53 | ambriento | what about the other way? |
| 20:57.30 | flujan_ | ambriento, so I need to allow my softphone to mark it? |
| 20:57.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:58.43 | ambriento | flujan_, what do you have between your station and your server? a switch? a hub? |
| 20:58.51 | Dovid | ur phone needs to support it as well |
| 20:59.43 | flujan_ | ambriento, http://pastebin.ca/196732 neither asterisk is marking it |
| 20:59.58 | *** join/#asterisk reticent (n=reticent@207-232-98-114.ip.van.radiant.net) |
| 21:00.13 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:00.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:00.31 | reticent | is anyone else having a problem activating digium G729 licenses? I keep getting sporadic connection resets while talking to their SSL server |
| 21:00.50 | flujan_ | ambriento, I have two srw22... linksys switchs. |
| 21:01.17 | flujan_ | Dovid, ok... I will check the phone too. |
| 21:01.19 | reticent | which causes their register problem to generate an errouneous error, "libcurl-gnutls is not available. please install." and quit |
| 21:01.41 | flujan_ | Dovid, but it is strange that asterisk is not doing it |
| 21:01.44 | flujan_ | Dovid, marking the packages. |
| 21:01.53 | *** part/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 21:03.09 | flujan_ | ambriento, any idea? |
| 21:03.58 | flujan_ | ambriento, a linksys srw224g4 |
| 21:04.12 | ambriento | flujan_, that was not a IAX2 packet |
| 21:04.35 | ambriento | or an* IAX2 packet :) |
| 21:05.25 | *** join/#asterisk fifo_ (n=myr@dsl-145-238-13.telkomadsl.co.za) |
| 21:06.20 | fifo_ | the free gpl asterisk, how many concurrent "users" can it handle when acting as a PBX? |
| 21:06.51 | Nugget | fifo_: How fast is a car? |
| 21:06.55 | fifo_ | with handle i mean, business quality stability |
| 21:07.12 | fifo_ | depends on the car, in this case open source asterisk |
| 21:07.14 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 21:07.28 | Nugget | give it a minute, the answer will jump out at you. |
| 21:07.33 | fifo_ | i am sorry, but i admit i am new to voice comm |
| 21:07.38 | [TK]D-Fender | fifo_ : Just as many as the nopn-GPL one. |
| 21:07.52 | flujan_ | ambriento, hum... my mistake |
| 21:07.59 | flujan_ | asterisk is marking the packages... not my phone |
| 21:08.00 | [TK]D-Fender | fifo_ : Hundreds? Thousands? depends on hardware and what you're asking it to do. |
| 21:08.03 | fifo_ | so i might ask a couple of questions you guys feel to be ovious |
| 21:08.19 | [TK]D-Fender | fifo_ : No worries. What do you want to use * for? |
| 21:08.42 | fifo_ | [TK]D-Fender: i want to set it up for a business with about 40 phones. what setup would be good? iow, what hardware. |
| 21:08.45 | flujan_ | ambriento, I can enable this feature from asterisk configuration? |
| 21:08.46 | TexasJay | does anyone have a link to a decent agent login/logout script? i'm having problems creating one of my own. |
| 21:09.06 | [TK]D-Fender | fifo_ : Easy task. What kind of lines were you thinking of? What kind of phones? |
| 21:09.09 | fifo_ | especially for allowing inbound/outbound calls (from/to PSTN) and internally through the computer/software (ie asterisk) |
| 21:09.30 | *** join/#asterisk earthsound (n=one_down@138.26.35.115) |
| 21:09.47 | earthsound | ping kram |
| 21:09.49 | fifo_ | [TK]D-Fender: lines? normal analog phone lines. no isdn if that is what you mean. |
| 21:09.50 | aptura | TK you have setup more then one line on a ip500 going to say a sip/did and zap/did before? |
| 21:10.22 | [TK]D-Fender | fifo_ : how many lines? What kind of phones for desks? |
| 21:10.39 | [TK]D-Fender | aptura : Sure |
| 21:11.06 | [TK]D-Fender | aptura : I've done jsut about everything there is to do with Polycom's |
| 21:11.20 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 21:11.23 | fifo_ | [TK]D-Fender: was thinking of purchasing new phones for the front offices, and the back office managers, then perhaps apapters for the rest (to lower costs) |
| 21:11.45 | sevard | Is the order I'm supposed to install.. libpri, zaptel, asterisk, or zaptel, libpri, asterisk? |
| 21:11.51 | Lawbringer | hey guys, quick question do all the system voice prompts have to be in GSM? |
| 21:12.05 | fifo_ | lines. not sure. what would be recommended? i don't know what they currently have. |
| 21:12.08 | Lawbringer | or can i use wave or 711a |
| 21:12.27 | Lawbringer | really i'm just trying to save anybit of CPU i can |
| 21:12.49 | Lawbringer | Veryold machine.. might even break records :D |
| 21:12.59 | fifo_ | let's say i want to accept two calls inbound, and make two calls outbound. can that be done with a single analog line? |
| 21:13.19 | [TK]D-Fender | fifo_ If you can its suggested you use ISDN-PRI (not BRI) for incoming lines. For phones weI suggest either Polycom or Aastr SIP phones. Model dependant on needs and local pricing/budget |
| 21:13.22 | Lawbringer | fifo_ not over a single PSTN live |
| 21:13.51 | Lawbringer | fifo_, you had dsl or could use voip then yes |
| 21:13.52 | ambriento | Lawbringer, you can have your voice prompts in any format for asterisk >1.2 |
| 21:14.01 | [TK]D-Fender | Lawbringer : Encode in the the format you will use the most whatever that happens to be. |
| 21:14.01 | aptura | [TK]D-Fender I have setup a sip.conf extention for line 2. Setup the phone for that extention wich is my last for of my did number. What I want to know was should I use the username/password of the DID provider for that extention context and point that context to the right one in extensions.conf ? I tried this a few days ago and my phone locked up. |
| 21:14.28 | aptura | I will wait untill you are done. |
| 21:14.30 | fifo_ | Lawbringer: with voip do i simply connect asterisk with a "gateway" at a voip service provider? |
| 21:15.18 | Lawbringer | ambriento, how will voicemail() no that vm-intro.gsm has changed to vm-intro.wav? |
| 21:15.19 | [TK]D-Fender | aptura : typically I suugest you just connect * to your DID provider and just forward calls to a SINGLE registration on the phone. Rarely is it worth it to maintain sperate registrations on the phone. |
| 21:15.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:15.53 | Lawbringer | fifo_, yep they do the breaking out to pstn instead of you, can save quite a chunk of cash like this |
| 21:15.54 | [TK]D-Fender | fifo_ : You don't need a VoIP provider in your scenario from what you're describing. You jsut want to use * as a normal PBX and use normal lines. |
| 21:16.06 | aptura | TK okay thats what i did. |
| 21:16.23 | Lawbringer | [TK]D-Fender, he can't make four calls on one pstn line. |
| 21:16.24 | fifo_ | with outbound calls, is it possible to say have calls handled/routed differently when you entered a special "code", like when i enter 0, and dial i dial normal number, but when i enter 4 and dial, i dial through a special gateway? |
| 21:16.29 | [TK]D-Fender | aptura : But you want a dedicated key on the IP 501 just for that DID that no-one else will access? |
| 21:16.48 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.128.38) |
| 21:16.53 | [TK]D-Fender | fifo_ : yes, you can take any kind of number you want and do pretty mcuh whatever you want with it. |
| 21:17.02 | Lawbringer | fifo_, yes that's possible read asterisk the future of tele |
| 21:17.05 | fifo_ | iow, when i entered 4 and dial 02222 i dial another phone on another asterisk network? |
| 21:17.12 | Lawbringer | it's a free download somewhere |
| 21:17.20 | ambriento | Lawbringer, if you have more than one format for a given voice prompt, * will take the least expensive one |
| 21:17.22 | aptura | The asterisk box will be a home phone and bussiness phone system. Press line one and the called party will see my CID or line one and thay will see my home CID. That is what I am trying to acomplsih. |
| 21:17.34 | [TK]D-Fender | fifo_ User dials "1234" on Tuesday night and its raining? Ok, the system will use line 4 to dial 987654321. Just because you feel like it. |
| 21:17.48 | Dr-Linux|work | anyone tried 1.4beta? |
| 21:18.02 | aptura | TK, yes no one else will access. not that is will be a issue. |
| 21:18.56 | [TK]D-Fender | aptura : Ah. Ok, you want to make it easy to choose your identity for outbound calls. Sure, make another reg for it. I'd suggest you then use 1 line key per reg and say 5 calls per key. |
| 21:18.56 | Lawbringer | ambriento, expensive, how does it rate? via cpu or how much it cost me? and how would it know :) |
| 21:18.56 | fifo_ | is it possible to route it to another phone directly, using an asterisk gateway, which in turn uses another gateway, which in turn.. etc..? |
| 21:18.56 | fifo_ | basically just want to know how flexible asterisk is. |
| 21:18.56 | [TK]D-Fender | fifo_ You can do pretty much anything. |
| 21:19.00 | fifo_ | [TK]D-Fender: cool :> |
| 21:19.13 | [TK]D-Fender | fifo_ : Mine used to make COFFE. I'm sure you're needs won't be too much :P |
| 21:19.18 | ambriento | Lawbringer, try CLI> show translations to see the cost of transcoding |
| 21:19.18 | [TK]D-Fender | COFFEE* |
| 21:19.23 | fifo_ | one more thing. from you guys' experience. how stable is asterisk in a high demanding environment? |
| 21:19.40 | Lawbringer | ambriento, ohh thanks.. look interesting |
| 21:19.42 | ambriento | Lawbringer, for the CPU :) |
| 21:19.45 | [TK]D-Fender | fifo_ : Works great for all the companies I consult to. Also use ti at my day job |
| 21:19.58 | [TK]D-Fender | fifo_ : And I' run it at home as well naturally. |
| 21:20.00 | ambriento | Lawbringer, and check CLI> show file formats as well |
| 21:20.10 | jmls | Dr-Linux|work: we use 1.4 pre-alpha ;) in a production enviroment with 4000+ calls per day. Rock solid. Current uptime is 3 weeks. (We took it down for some hw maintenance) |
| 21:20.27 | aptura | my asterisk box is registered inbound with my did provider. Outbound on line 2 should be setup with my did providers user/pass and the outbound context? |
| 21:20.30 | jmls | using queues / dynamic queue members and jabber |
| 21:20.41 | Lawbringer | ambriento, i don't want any transcoding. i'm running on a 5x86 133mhz |
| 21:20.44 | ambriento | Lawbringer, if you have a channel using G711, and you have G711 prompts, there will be no translation |
| 21:20.57 | Lawbringer | ahhh |
| 21:21.02 | Dr-Linux|work | jmls: can i /msg you? |
| 21:21.05 | Nugget | asterisk is quite solid if you're careful about what hardware you buy and how you set it up. It's hard to just say that asterisk is stable as a blanket statement. |
| 21:21.09 | [TK]D-Fender | aptura : You want to use * on that line 2 for VM, etc? You *could* have the phone reg to your DID provider directly if you wanted. |
| 21:21.10 | ambriento | ~native prompt |
| 21:21.29 | [TK]D-Fender | Lawbringer : Dear God you're asking for trouble... ULAW or bust. |
| 21:21.30 | jmls | fifo_: see my comments to Dr-Linux|work. We currently have 120+ sip extensions / phones |
| 21:21.34 | jmls | Dr-Linux|work: yup |
| 21:21.53 | *** join/#asterisk JonR800 (n=jon@p1mp.org) |
| 21:21.53 | Lawbringer | ambriento, so since i want everything to be g711 to save overall cpu cycles |
| 21:21.57 | [TK]D-Fender | fifo_ : Indeed I've seen many 100-200 ext systems around. |
| 21:21.57 | Nugget | if you throw a bunch of clone x100p cards into a spare dell optiplex you had laying around, then deploy grandstream phones to all your desks you're on the fast track to being fired, committing suicide, or being committed. |
| 21:22.06 | *** join/#asterisk sloth_ (n=sloth@pool-162-83-220-153.ny5030.east.verizon.net) |
| 21:22.12 | aptura | [TK]D-Fender yes vm. Sure I could have it registered directly. Would need to make changes for the firewall to port to the phone. |
| 21:22.13 | fifo_ | (if not using a voip provider) what hardware would i need to have the effect of two lines with PBX (call switching/waiting, called id, switchboard, voice mail, etc,) when using analog lines. |
| 21:22.23 | [TK]D-Fender | Nugget : I hear you can get all 3 in a bundled package now ;) |
| 21:22.27 | Nugget | haha |
| 21:22.31 | fifo_ | s/called/caller |
| 21:22.40 | Lawbringer | [TK]D-Fender, well it's all setup and running well just wondered if i can squeze a third line :) |
| 21:22.51 | aptura | [TK]D-Fender but I would perfer to register against the asterisk box and with the did user/pass |
| 21:22.52 | [TK]D-Fender | aptura : Yeah... thats not such a great idea. Run it through *... less trouble. |
| 21:23.18 | fiber0pti | I'm looking for some feedback on a new operator panel. Anyone want to give it a whirl? |
| 21:23.29 | aptura | fiber0pti sure |
| 21:23.40 | [TK]D-Fender | fiber0pti : show |
| 21:23.42 | DasTech | Fiber is it opensource |
| 21:23.44 | DasTech | show it |
| 21:23.46 | intralanman | fiber0pti: what's it written in? |
| 21:23.48 | DasTech | lets see |
| 21:24.06 | intralanman | flash againn? or ajax or something? |
| 21:24.12 | fiber0pti | It's written in java. It's not open source at this time, however, we're thinking about making the free version open source once we get it to a useable version |
| 21:24.16 | fifo_ | and does asterisk have a console where you can look at the calls active and control them (such as canceling, switching, snooping, etc)? any web interfaces to make this available to any1 authenticated? |
| 21:24.26 | fiber0pti | It can be downloaded at www.i9technologies.com/isymphony |
| 21:24.40 | ambriento | Lawbringer, check http://www.voip-info.org/wiki/view/Asterisk+Native+Sounds |
| 21:24.40 | [TK]D-Fender | fifo_ : there is FOP (FlashOperatorPanel) thats good for that, so yes. |
| 21:24.40 | Nugget | fifo_: several, but none that are as polished or stable as you're probably envisioning. |
| 21:24.49 | fiber0pti | There are a few more features we're currently adding and we're also changing the layout drastically. |
| 21:25.00 | aptura | I wonder how many asterisk applications have been written to interface with windows applications. |
| 21:25.02 | fiber0pti | Any feedback is welcome |
| 21:25.19 | Lawbringer | ambriento, is the show translation relative to the machine it's running on or just a general guide? |
| 21:25.32 | aptura | fiber0pti do you have a url? |
| 21:25.33 | fiber0pti | fifo_: iSymphony is a real-time operator panel. Check it out at www.i9technologies.com/isymphony |
| 21:25.40 | Qwell[] | Lawbringer: it's calculated when asterisk starts, I believe. It's definitely for that machine though |
| 21:25.52 | fiber0pti | fifo_: you can even barge and record calls with iSymphony |
| 21:26.00 | arisjr | hi folks, is the message "channel.c: Avoiding deadlock for '<channel num>'" normal? |
| 21:26.14 | justinu|laptop | heh |
| 21:26.24 | Lawbringer | and since i'm in the rest of the world i should be using alaw right? |
| 21:26.37 | ambriento | where are you? |
| 21:27.14 | Lawbringer | ambriento, UK |
| 21:27.14 | [TK]D-Fender | Qwell[] : Go fix/add somthing so you can release 1.2.13 or 1.4! |
| 21:27.18 | ambriento | Lawbringer, are you planning to talk to PSTN? |
| 21:27.22 | Qwell[] | [TK]D-Fender: I am :p |
| 21:27.30 | fiber0pti | aptura: Yeah, it's at www.i9technologies.com/isymphony |
| 21:27.32 | [TK]D-Fender | Qwell : yay |
| 21:27.36 | [TK]D-Fender | \o/ |
| 21:27.48 | fiber0pti | aptura: You can even see the screenshots there. |
| 21:27.49 | Lawbringer | ambriento, yes it's mostly going to be to pstn but over a gateway so locally i'm pure voip |
| 21:27.55 | DasTech | you need to make a java client for it also |
| 21:27.56 | arisjr | folowed by ""WARNING: channel.c: Avoided deadlock for '0x81b2a48', 10 retries! |
| 21:28.07 | DasTech | so that unix users can run it |
| 21:28.21 | DasTech | i see the windows client and the unix server |
| 21:28.26 | DasTech | but no unix client |
| 21:28.37 | aptura | fiber yea see it. downloading jvm |
| 21:28.40 | ambriento | Lawbringer, I see, well, if you have bandwidth and UK uses a-Law, go forward :) |
| 21:28.53 | ambriento | we use a-Law in here too |
| 21:28.55 | fiber0pti | aptura: Great. Don't hesitate to ask for help you need it. |
| 21:29.37 | aptura | What I want to see is a CID pop up on the windows client if the phone is in a dark enviroment. ip500s are not known for good displays. |
| 21:29.53 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 21:29.59 | [TK]D-Fender | Qwell[] : In as much as we all know 1.4 will come "when its done" or whatever, is progress for 1.4 going near the pace you've been hoping for to keep with the expected release date? |
| 21:30.09 | jmls | aptura: use 1.4 and jabber to send a message to an IM running on the client |
| 21:30.40 | Lawbringer | ambriento, weell for the amount of call i'll be making and getting i'm more worried about cpu than bandwidth.. the machine is and old terminal server totally slient.. :) nice little setup but under powered |
| 21:30.43 | arisjr | "Oct 4 17:10:11 WARNING[20336] channel.c: Avoided deadlock for '0x81b2a48', 10 retries!" |
| 21:30.44 | aptura | jmls if company policy states you can ron a IM |
| 21:30.44 | arisjr | Is this message normal for you guys? |
| 21:31.04 | jbalcomb | [TK]D-Fender: another multi-queues questsion.. when my reps put people on hold it plays the same hold music for all queues |
| 21:31.05 | Lawbringer | ambriento, and i thought that american used uLaw |
| 21:31.18 | [TK]D-Fender | jbalcomb : thats configurable per-queue |
| 21:31.23 | jmls | aptura: embed an IM into your apps using IPworks XMPP control if you're using windows |
| 21:32.15 | jbalcomb | [TK]D-Fender: yeah, it plays right hold music, the reps answers, and then if the rep hits hold again it plays /our/ hold music |
| 21:32.22 | fifo_ | fiber0pti: can isymphony serve it's purpose from machines other than the asterisk host? |
| 21:32.32 | jbalcomb | [TK]D-Fender: after a year, someone finally noticed and thinks its a problem. |
| 21:32.32 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 21:32.36 | [TK]D-Fender | jbalcomb : Ah. the agent's MoH asserts itself.... dang... |
| 21:32.52 | justinu|laptop | that's still not fixed? damn |
| 21:32.56 | [TK]D-Fender | jbalcomb : not sure on that... try setting the MoH in the dialplan and see if it takes over. |
| 21:33.06 | jbalcomb | [TK]D-Fender: I'm about to tell them I need a seperate phone server for each client |
| 21:33.23 | ambriento | Lawbringer, in north-america - yes :) I'm in south-america |
| 21:33.24 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:33.34 | [TK]D-Fender | jbalcomb : Oh FFS... thats obcene |
| 21:33.36 | fiber0pti | fifo_: I don't understand your question. Can it be used with something other than asterisk? |
| 21:34.03 | fifo_ | fiber0pti: no. i have asterisk "PBX" on host A. can i run isymphony on host B? |
| 21:34.08 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 21:34.18 | jbalcomb | [TK]D-Fender: true statement |
| 21:34.18 | fifo_ | and then control asterisk on host A? |
| 21:34.38 | [TK]D-Fender | jbalcomb : I know who to attribute these statements to don't I? |
| 21:34.40 | fiber0pti | fifo_: Yes, you can run it from anywhere as long as you can reach the asterisk host |
| 21:35.11 | fifo_ | fiber0pti: i see. costs? |
| 21:35.53 | jbalcomb | [TK]D-Fender: If you guessed the DoOer, you guessed right!! |
| 21:35.58 | fiber0pti | fifo_: Free at the moment. There will be a price associated with advanced functionality like permissions, queue handling, etc.. |
| 21:36.43 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 21:36.52 | fifo_ | erm. what setup do you need if you want to have call waiting for multiple callers? like those you get at customer care centres? |
| 21:37.37 | *** join/#asterisk paulhuynh (n=paul@68.82.4.138) |
| 21:38.13 | paulhuynh | is anyone here using atsreisk with Cavalier |
| 21:38.19 | [TK]D-Fender | fifo_ : No special needs. |
| 21:38.31 | jbalcomb | filo_: more than one phone line or a PRI |
| 21:38.40 | awannabe | has anyone got the snom 360 login/logout to work properly? |
| 21:38.51 | [TK]D-Fender | fifo_ : With * you get call queues, IVR optiosn, MOH, tons of stuff. Basically anything you might expect from a system that costs tons more. |
| 21:38.59 | jbalcomb | fifo_: and card that support more than one phone line or a PRI |
| 21:39.08 | fifo_ | i understand |
| 21:39.31 | Dovid | ~bristuff |
| 21:39.33 | jbot | from memory, bristuff is a patch collection with a hfc-pci driver supporting NT and TE mode ISDN. See http://junghanns.net/ for more info |
| 21:39.36 | TexasJay | Is there a command to show what variables Asterisk has in memory? |
| 21:39.56 | [TK]D-Fender | fifo_ : how your phone users will juggle calls around depends on the phone's you give them, but decent SIP phones can handle handful or dozens of calls depending. |
| 21:39.58 | jbalcomb | [TK]D-Fender: yeah, too damn bad its got retarded support for reporting, multi-queues, and config management |
| 21:39.58 | fifo_ | back to my previous question (main and last probably). for a basic setup like: the effect of two lines (outbound + inbound) with PBX (*). analog lines used. what hardward do i need? |
| 21:40.30 | Dovid | fifo_: what codecs, connfrences, call recording ? |
| 21:40.48 | fifo_ | doesn't matter. conferences maybe. call recording maybe. |
| 21:40.49 | jbalcomb | [TK]D-Fender: *sigh* atleast I didn't pay a vendor 50K for the same helpless feeling. |
| 21:40.52 | [TK]D-Fender | jbalcomb : You mean too bad for your "super-duper-call-center-with-1000000-clients-who-each-would-be-happy-if-not-for-that-bitch" |
| 21:40.55 | Dr-Linux|work | hhm.. thinking about 1.4beta |
| 21:41.12 | Dovid | Dr: wouldnt use it for work enviroment |
| 21:41.14 | Dr-Linux|work | any other dude tried 1.4beta? |
| 21:41.17 | aptura | [TK]D-Fender now the address in line two is the contect in sip.conf? I put in the DIDs user/pass and pointed the server1 address to my asterisk box. |
| 21:41.18 | [TK]D-Fender | jbalcomb : '"TM" |
| 21:41.27 | fifo_ | i want to basically connect asterisk to the phones, then use asterisk for the features. then i have two fones connected to asterisk. and then have asterisk connect via analog lines to the PSTN. what hardware would i need. |
| 21:41.34 | aptura | its not registering. |
| 21:41.35 | Dovid | fifo_: call confrencing eats a lot, g729 does as well |
| 21:41.50 | Dovid | for call recording depending on the volume u may need lots of disk space |
| 21:41.54 | paulhuynh | hello |
| 21:41.57 | [TK]D-Fender | aptura : No. You mane a 2nd "normal" looking reg that does not resemble your external account at all and reg your phone to that. |
| 21:42.07 | jbalcomb | [TK]D-Fender: haha.. yeah, i didn't really consider our setup to be that special when I first got here. Now I realize that we really do need someone who is something of an Asterisk specialist. |
| 21:42.08 | TexasJay | Is there a command to show what variables Asterisk has in memory? |
| 21:42.09 | paulhuynh | how complicated is it to get h323 trucking to work with asterisk |
| 21:42.16 | [TK]D-Fender | aptura : You aren't "bridging" or "prozying" the calls through *. |
| 21:42.20 | fifo_ | Dovid: so for those i need a strong/big machine. besides that, what hardware do i need phone/line wise. |
| 21:42.23 | paulhuynh | my provider can support h323 only |
| 21:42.27 | aptura | [TK]D-Fender yes thats what I did originally |
| 21:42.39 | jbalcomb | [TK]D-Fender: Preferably a developer who can do good work in the dial plan and maybe patch the C code for our needs. |
| 21:42.47 | Dovid | paulhuynhL u want h323 trunking between asterisk boxes ? |
| 21:42.52 | Dovid | ah ok |
| 21:43.10 | Dovid | fifo_: what kind of phone lines ? |
| 21:43.17 | Dovid | POTS ? ISDN ? VOIP ? |
| 21:43.17 | aptura | [TK]D-Fender so orginally i made a seperate extention for line 2 |
| 21:43.20 | fifo_ | Dovid: analog |
| 21:43.23 | fifo_ | POTS |
| 21:43.36 | Dovid | i would get the sangoma A200 |
| 21:43.50 | Dovid | i would get echo can. its a worthy invesment |
| 21:43.54 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 21:44.02 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 21:44.11 | Dovid | i personaly like sangoma over digium |
| 21:44.15 | ambriento | TexasJay, try evaluate'em with NoOp() |
| 21:44.16 | paulhuynh | H.323 version 2 Fast start, MGCP, NCS. If you can get your PBX to support that, than we may be able to deliver the VOIP via T1 |
| 21:44.28 | Dovid | thier support was real good when i needed them, and i have not heard the best about digium |
| 21:44.49 | paulhuynh | telco company is using adit600 box to get the dial tone to my asterisk and i'm trying to just go direct |
| 21:44.58 | [TK]D-Fender | jbalcomb : We'll I've got part 1 in hand. Part 2 is yours..... |
| 21:45.18 | fifo_ | Dovid: (for arguments sake for my understanding) to switch that to isdn i just need a different card to connect the line to asterisk right? and for VOIP nothing, as i would connect asterisk to VOIP provider? |
| 21:45.24 | [TK]D-Fender | aptura : then you should go back to doing it that way |
| 21:45.33 | ambriento | TexasJay, exten => 1234,1,NoOp( ${VARNAME}) ; and you'll have the value of that VARIABLE |
| 21:45.37 | ambriento | ~variables |
| 21:45.40 | jbot | variables are at /usr/src/asterisk/doc/README.variables |
| 21:45.40 | Dovid | fifo_: correct |
| 21:45.44 | jbalcomb | [TK]D-Fender: yeah, that sound like the theory behind sending mostly 19 year old country bumpkins to fight our wars. |
| 21:46.25 | TexasJay | ambriento, jbot: No, I mean a list of current variables that Asterisk can reference. |
| 21:46.26 | [TK]D-Fender | jbalcomb : To coin Katty "that does not parse" |
| 21:46.33 | fifo_ | Dovid: so the basic idea is, the line to asterisk connecter is a choice between ISDN/POTS. then alternatively for "certain" calls say internatonal you can have asterisk use VOIP provider? |
| 21:46.45 | fifo_ | Dovid: then you just connect the phones with asterisk? |
| 21:47.10 | [TK]D-Fender | fifo_ : Yeah sure. Route specific kinds of calls alternately to maximize your cost effectivness |
| 21:47.11 | TexasJay | Not global variables that I can reference. Ones that Asterisk created (like ${AGENTBYCALLERID_<#>} |
| 21:47.16 | Lawbringer | ok so if i say allow=alaw only in sip.conf and everything else is in alaw format is this the lowest cpu i can use? |
| 21:47.24 | Dovid | fifo_: u can use VOIP for all of ur calls (both in and outbound). |
| 21:47.36 | aptura | [TK]D-Fender okay im setting up to register against that extention I setup in sip.conf. As far as its context is concerned the sip.conf context should point to my DIDs context in extentions.conf right? |
| 21:47.54 | Dovid | personaly I only use VOIP and i use a belnd of providers for lowest cost |
| 21:48.08 | Dovid | one provider for origination and several diffrent ones for outbound |
| 21:48.16 | fifo_ | Dovid: belnd? |
| 21:48.17 | [TK]D-Fender | fifo_ : So for instance you have a bunch of local lines which you'd use for local calls, and 3 VoIP providers each providing your the cheapest rates in their respective territories. * could then choose the best resource for any give call based on what you want. |
| 21:48.40 | ambriento | TexasJay, in /usr/src/asterisk/doc/README.variables you'll have all the variables |
| 21:48.54 | [TK]D-Fender | aptura : taht reg should have its own cotext witha more direct access to dialing out so as to feel 'natural" |
| 21:49.33 | Dovid | ambriento: i think he wants a list of current variables in asterisk memories i.e. if he does Set(foo=${EXTEN}) |
| 21:50.00 | fifo_ | in asterisk configuration, what is the concept (in the way it works) behind differentiating what type of outbound call you have. in other words, how do you "teach" asterisk to differentiate? |
| 21:50.03 | ambriento | Lawbringer, I guess so Lawbringer. try the disallow=all first then allow=alaw |
| 21:50.06 | jsmith | TexasJay: Check out the DumpChan() application |
| 21:50.07 | ambriento | oops |
| 21:50.21 | ambriento | Dovid, hmm.. I see |
| 21:50.23 | *** join/#asterisk saftsack (n=saftsack@p54A7D471.dip.t-dialin.net) |
| 21:51.07 | [TK]D-Fender | fifo_you seperate by pattern matching. You can tell it to sen everything like "9XXXXXXXX" out one provider, everything "8XXXXXXXX" out another, etc. super specifica, super general, its all up to you |
| 21:51.17 | Dovid | fifo_: we were all newbies at some point or another |
| 21:51.55 | fifo_ | Dovid: :> |
| 21:52.32 | [TK]D-Fender | fifo_ : You can tell it that if you dial 123 it'll send 465 out your normal phone lines. You can have ti so it'll send all calls to one area code out normal lines, and others through a VoIP provider. You can twist and warp the whole system just about any which way you want. |
| 21:52.43 | TexasJay | jsmith: So there's nothing built in, eh? Drat :( |
| 21:52.50 | jbalcomb | [TK]D-Fender: ok, it works. i took it out when i made my beta queue macro. doh & sshhh... |
| 21:53.08 | fiber0pti | aptura: Get it installed? |
| 21:53.22 | [TK]D-Fender | jbalcomb : So Dialplan initiated overrides take prority over the Agent? |
| 21:53.25 | jsmith | TexasJay: I'm not sure I understand what you're saying... the DumpChan() application is built into Asterisk. |
| 21:53.28 | fifo_ | [TK]D-Fender: and this is done in an "easy" to learn fashion, not like configuring hardware parameters like for certain dial-in cards? the ^AT/42../1! kind? |
| 21:53.43 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 21:53.43 | [TK]D-Fender | fifo_ Considerably easier. |
| 21:53.56 | Dovid | fifo_: A LOT EASIER |
| 21:54.03 | Dovid | if u know linux it wont take u long |
| 21:54.04 | [TK]D-Fender | fifo_ : Go download "THE BOOK" for reference. |
| 21:54.06 | [TK]D-Fender | ~book |
| 21:54.08 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:54.25 | jbalcomb | [TK]D-Fender: yes |
| 21:54.27 | Dovid | i didnt know linux at all and it took me 2-3 months to learn the basics |
| 21:54.32 | aptura | [TK]D-Fender I have line two as Address 0607 auth userID 0607 pass xxx label 0607. numlinekey=1 callsper=1 server adress is same. In my sip.conf the context 0607 with the user/pass is the same. My question is if the context in the [0607] points to the DID contect in extentions.conf how is it going to authenticate? |
| 21:54.39 | [TK]D-Fender | jbalcomb : \o/ |
| 21:54.56 | [TK]D-Fender | aptura : I suggest 5 calls/key.... |
| 21:55.17 | [TK]D-Fender | aptura : Make another entry for that reg. |
| 21:55.31 | aptura | okay |
| 21:55.40 | aptura | in sip.conf |
| 21:55.40 | fifo_ | [TK]D-Fender: oh, i have that one :> |
| 21:55.57 | aptura | you mean make a reg for the did provider |
| 21:56.07 | [TK]D-Fender | fifo_ : then read over the sections about dialplan creation and pattern matching. it'll explain much. |
| 21:56.08 | aptura | with the dids user/pass |
| 21:56.21 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-219-146.red.bezeqint.net) |
| 21:56.35 | fifo_ | [TK]D-Fender, Dovid: thx alot |
| 21:56.47 | [TK]D-Fender | aptura : You should know how to have * reg to a provider..... then you set up a context for your PHONE'S reg that will exclusively dial out that provider and set our callerid for you, etc.... |
| 21:57.20 | fifo_ | erm, what would two basic phones (enough to switch to each other and dial out and receive, with caller id and call waiting, etc) and a card cost me about. just an estimate? i just want to start playing around to get a feel for this. |
| 21:57.58 | aptura | I can dial out on the did thats not a issues. Its just bridging two context my phones and my DID's is what im trying to do. |
| 21:59.30 | [TK]D-Fender | aptura : that should be easy.... youmake an inbound context for your DID provider, take the call and then jsut hard-dial your 2nd reg on your phone. end of story. |
| 21:59.48 | [hC] | is it possible to push someone to the top of a queue based on their caller id? |
| 21:59.52 | [hC] | say, for vip people? |
| 22:00.18 | [hC] | oh. |
| 22:00.19 | [hC] | found it. |
| 22:00.27 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 22:01.41 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:01.59 | Dovid | fifo_: u dont need a card. u can go striclty voip and use soft phones which u can get for free |
| 22:02.57 | fifo_ | Dovid: but if you dont want to go voip. |
| 22:03.07 | aptura | [TK]D-Fender Line 2 is now not giving a fast busy tone and I can dial out but dont see anything in cli and line one, which I am ringing is not rining. |
| 22:03.16 | Dovid | fifo_: for learning u dont need |
| 22:03.32 | Dovid | have a look at voip supply |
| 22:04.03 | Dovid | for a TDM400P with 2 fxo (FXO is for incoming lines and FXS if u wana connect a pots phone to asterisk) |
| 22:04.06 | fifo_ | Dovid: but the PSTN part is what i want to learn about actually. the rest is quite simple. even before i came here |
| 22:04.16 | fifo_ | it is the PSTN part the i was the most unsure about |
| 22:05.02 | [TK]D-Fender | aptura : You should be reg-ing to * and not your DID provider. |
| 22:05.35 | Dovid | fifo_: the dial planing is pretty much the same. the card is slightly diffrent. |
| 22:06.04 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 22:06.08 | fifo_ | Dovid: name of * supported cards? PRI and analog. |
| 22:06.52 | aptura | TK, line two is registered aginst its own user/pass and context in sip.conf. Dont see it registered in sip show peers however. |
| 22:07.03 | fifo_ | is asterisk kernel 2.6 compatible? |
| 22:07.21 | [TK]D-Fender | aptura : then you've done something wrong. bad server IP or somthing? |
| 22:07.41 | aptura | same ip as line 1 |
| 22:07.44 | [TK]D-Fender | fifo_ 2.4/2.6 yes |
| 22:07.59 | [TK]D-Fender | aptura : Not sure what to tell you... its simply not registered... |
| 22:08.07 | aptura | :) |
| 22:08.08 | fifo_ | anyway, thx alot |
| 22:08.10 | fifo_ | gonna go sleep now |
| 22:08.14 | fifo_ | getting l8t |
| 22:09.13 | Dovid | fifo_: if ur going with pots then look at the TDM400P |
| 22:09.22 | Dovid | fifo_: Night |
| 22:09.29 | fifo_ | Dovid: thx alot :> |
| 22:10.01 | *** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 22:10.12 | E-Rage | Anybody have experience with Mitel SIP phones? |
| 22:10.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:12.59 | ambriento | omg I need to eat something quickly |
| 22:13.11 | Dovid | lol |
| 22:13.14 | Dovid | likewise |
| 22:13.31 | Dovid | but work here comes first - hence all the cigarete buts in the ash tray |
| 22:14.25 | ambriento | :) |
| 22:15.47 | ambriento | supermarket: Here I go! |
| 22:16.05 | Dovid | and how i wish i had my friend jack here now |
| 22:16.18 | Dovid | jack daniels that is |
| 22:16.23 | ambriento | hahahhaa |
| 22:16.40 | ambriento | I have my friend Johnny in here :) |
| 22:17.27 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:18.44 | *** join/#asterisk warmbowski (n=warmbows@dsl231-055-233.sea1.dsl.speakeasy.net) |
| 22:19.18 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 22:20.12 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 22:22.20 | Dovid | so do i |
| 22:22.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:22.30 | Dovid | he solves all solutions |
| 22:22.47 | ambriento | hahahaha |
| 22:23.37 | ambriento | you know, my nick comes from a mispelled spanish word for hungry |
| 22:25.57 | Dovid | lol |
| 22:26.12 | aptura | anyone here have a ip500 and have two lines setup? |
| 22:26.45 | Dovid | i had 601 |
| 22:26.46 | Inverted | does anyone know how to use asterisk-perl to append audio to a previously recorded file? |
| 22:26.48 | Dovid | whats ur question |
| 22:36.50 | *** join/#asterisk adorah (n=admin@84.94.127.28.cable.012.net.il) |
| 22:38.40 | *** join/#asterisk Adam12 (n=adam@d150-182-137.home.cgocable.net) |
| 22:40.02 | Adam12 | Anybody around that has Cisco ATA 186 experience? |
| 22:48.49 | Dovid | Adam12: i dont but whats ur question |
| 22:50.48 | Adam12 | Just received a used ATA from a friend of mine, and I've been trying to reset it to get web access but it keeps prompting me for a password. He doesnt' remember it. I was just curious if anybody knew a way of hard resetting the firmware without having to go through Cisco. |
| 22:51.23 | Dovid | hmm. i dont know them but i am sure some where on the device there is a reset button |
| 22:51.57 | Adam12 | There isn't |
| 22:52.21 | Dovid | argh: dont know anyother way. try google. u sure he changed it from the defaut ? |
| 22:52.22 | Adam12 | There is a way to do it using the phone connected to the ata, but it prompts for a password I do not have :| |
| 22:52.44 | Adam12 | I don't know the default either. I tried all the combinations I could think of. Google has yielded no results. |
| 22:52.49 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 22:53.09 | rene1 | ~seen oej |
| 22:53.36 | jbot | oej <n=oej@62.92.148.159> was last seen on IRC in channel #asterisk, 1d 4h 16m 5s ago, saying: 'YES!'. |
| 22:54.22 | Dovid | Adam12: googling now |
| 22:54.59 | Dovid | my connection is real slow. look here |
| 22:55.03 | Dovid | http://corp.deltathree.com/productsandservices/manuals/2port.pdf#search=%22cisco%20186%20ata%20quick%20start%20guide%22 |
| 22:56.32 | justinu|laptop | Press the function button on the CiscoATA. |
| 22:56.32 | justinu|laptop | Press the digits 322873738 (FACTRESET) then press # on your telephone keypad. |
| 22:56.32 | justinu|laptop | Press 3 on your telephone keypad to confirm that you want to reset the CiscoATA, then hang up the phone. |
| 22:56.36 | justinu|laptop | your google search skills suck |
| 22:58.11 | Adam12 | It prompts me for a password after I enter the FACTRESET code |
| 22:58.22 | aptura | I need somone that can clue in to my line2 issue. Want to use line 2 ONLY for my did assigned number to dialout/dialin. |
| 22:58.23 | Adam12 | Which is the password I am trying to reset, ironically. |
| 22:58.40 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 22:59.57 | aptura | justinu|laptop are you a polycom master? :) |
| 23:00.12 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 23:00.21 | justinu|laptop | i've seen a few of them |
| 23:00.40 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 23:00.54 | *** join/#asterisk purserj (n=purserj@29.124.233.220.exetel.com.au) |
| 23:02.18 | aptura | Gad, man whats worse. Explaining instructions to somone who has bad english skills or bad technical skills! |
| 23:02.36 | justinu|laptop | Adam12: another choice paste |
| 23:02.42 | justinu|laptop | 2) Make sure the unit is not a "locked" unit. Starting with v2.16, Cisco introduced a setting that would permit the device to be locked such that it is impossible to recover or reset the password under any circumstances. The rumor is that a certain large VoIP long distance company demanded that this be added so that their customers would stop selling the ATA-186 units that came with their inexpensive introductory offer. This "locking" setting is |
| 23:03.02 | justinu|laptop | sounds like you're SOL |
| 23:03.17 | aptura | justinu|laptop who? |
| 23:03.29 | justinu|laptop | the cisco ata dude |
| 23:03.43 | aptura | okay |
| 23:05.28 | justinu|laptop | what's worse is wasting your time trying to explain something who neither knows english and has no technical skills |
| 23:06.57 | Adam12 | Yeah, it looks that way, other than desoldering the EEPROM and dumping a hex output of it.... which means I should probably just look for a decent SIP phone and be done with it. |
| 23:08.00 | justinu|laptop | or just get an unlocked ata... they're about 60USD |
| 23:09.08 | *** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 23:09.46 | Adam12 | Yeah, that's another option. |
| 23:13.15 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:15.52 | *** part/#asterisk warmbowski (n=warmbows@dsl231-055-233.sea1.dsl.speakeasy.net) |
| 23:28.18 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:30.35 | *** join/#asterisk Dovid[Laptop] (n=dovi5988@barak.cellcom.co.il) |
| 23:30.52 | Dovid[Laptop] | ~seen aptura |
| 23:30.54 | jbot | aptura is currently on #asterisk (3h 56m 20s). Has said a total of 36 messages. Is idling for 27m 12s, last said: 'okay'. |
| 23:31.59 | *** join/#asterisk sloth (n=sloth@cpe-69-203-212-182.nyc.res.rr.com) |
| 23:35.24 | purserj | has anyone experienced problems with people using ekiga to connect to an asterisk install:? |
| 23:35.51 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:37.35 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 23:39.32 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 23:39.50 | cekc | I just asked my phone company to give me caller id and they say the only way they can do it is after they dial the DID number they would dial * and then the callerid info. Do I have to make a special dialplan to tell asterisk that the extra digits are caller ID or does it already know how to handle that? |
| 23:40.04 | *** join/#asterisk zane_ (n=zane@dsl092-019-067.sfo1.dsl.speakeasy.net) |
| 23:40.05 | THX2000 | Anyone know why I'd be getting "CallerID feed failed: success" on a zap card that previously handled cid fine? |
| 23:42.19 | sloth | THX: dunno, but sometimes i get that, but most other times it passes |
| 23:42.23 | sloth | cid just fine |
| 23:42.26 | zane_ | hi. I realize this sounds like a "just google it" question, but i've been searching around for quite some time and been unable to find anything. My question is: Is anyone aware of any soft or hard phones out there that actually support RSA authentication in IAX? or is it just supported in the protocol, but no one has taken advantage of it yet? |
| 23:42.47 | THX2000 | its consistent...i haven't received a cid on this card for days now |
| 23:43.38 | sloth | really, weird. Have you plugged in a cid capable phone on that line to make sure its not the telco? |
| 23:46.32 | *** join/#asterisk Cyt (n=danielcy@athedsl-162176.otenet.gr) |
| 23:46.35 | THX2000 | That was gunna be my next step...im workin remotely. So I just bounced the guy an email to see if he can check that for me |
| 23:47.04 | sloth | I assume that this is a digium tdm400p? |
| 23:47.13 | THX2000 | yea :/ |
| 23:48.05 | sloth | For a new project i decided to purchase the Sangoma card.....so far so good. But I have not really played with it too much yet |
| 23:49.06 | THX2000 | This was my first project w/ it and so far i've been less than pleased. Just sent out an rma for one of the modules on the card...I'm hoping some of my problems subside after I get the new one in |
| 23:50.59 | sloth | My first Asterisk project with a tdm400p was so bad I thought I might have to refund their money, but after much tweaking everything is "ok" |
| 23:51.30 | file | <PROTECTED> |
| 23:52.30 | THX2000 | I've been having to test this thing daily to make sure calls are getting terminated...This one module will work great for a weak then just die, which makes troubleshooting a bitch cause i'd have to wait a week to see if the changes i did made any difference |
| 23:52.41 | THX2000 | *week |
| 23:54.12 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 23:54.41 | sloth | That annoying!! |
| 23:55.52 | THX2000 | The sangoma card looks pretty sweet, looks like the price is right on the rhino's too |
| 23:56.38 | sloth | Yeah and their support is good, I have no problem with Digiums support though. |
| 23:57.07 | sloth | how many lines do you have with your tdm400p? |
| 23:57.19 | THX2000 | Yeah, the guy i talked to seemed pretty cool. I'm just not looking to become buddies w/ a support staff :P |
| 23:57.22 | THX2000 | just 2 |
| 23:57.49 | sloth | i hear that.\ |
| 23:58.20 | sloth | Do you have echo on these lines? |
| 23:59.42 | THX2000 | I got it pretty well worked out...cept the guy was complaining that when calling in on his cell he was getting echo. I haven't heard any when calling in, nor do i know why he would be |