| 00:00.00 | Starscreamer | I've used it for some pretty light stuff before |
| 00:00.02 | Starscreamer | web server |
| 00:00.03 | joeyjones | i've persoanlly used centos and debian before and found them both very stable |
| 00:00.07 | Starscreamer | F.O.G server etc |
| 00:00.12 | Starscreamer | and it's pretty cool |
| 00:00.40 | joeyjones | centos is pretty solid though |
| 00:00.42 | Starscreamer | I'm no linux guru and I tend to use Ubuntu because it's the one that gets rammed down my neck every way I turn ha |
| 00:01.07 | Starscreamer | I heard the community is great for centos |
| 00:01.13 | joeyjones | Starscreamer: i'd suggest centos, it's get s a lot of support and is very stable |
| 00:01.15 | Starscreamer | but I've never really used it |
| 00:01.34 | joeyjones | where ubuntu is mainly desktop, so server stuff might not be quite as good |
| 00:01.41 | Starscreamer | I noticed that it's used for most dedicated linux server providers |
| 00:01.44 | joeyjones | it's all the same thing really |
| 00:01.50 | Starscreamer | I hear ya |
| 00:01.53 | joeyjones | just it uses yum instead of apt |
| 00:02.19 | Starscreamer | I've never used yum on a server before |
| 00:02.21 | Starscreamer | only desktop |
| 00:02.24 | Starscreamer | like SUSE |
| 00:02.47 | WIMPy | The german Windows? :-) |
| 00:02.54 | joeyjones | it is likely to be best to compile from source, but when i tried to compile asterisk i had tons of issues and gave up |
| 00:03.17 | Starscreamer | on CentOS? |
| 00:05.53 | joeyjones | that was on debian |
| 00:06.12 | joeyjones | i almost always have troubles when i compile from source though |
| 00:07.29 | citywok | any recommendations for good cordless phones? preferably dect over wifi |
| 00:07.52 | WIMPy | citywok: That is a contradiction in itself. |
| 00:08.02 | citywok | dect instead of wifi* |
| 00:08.09 | WIMPy | But you surely want dect, not wifi. |
| 00:08.12 | *** join/#asterisk Nombrandue (~Satan@ip174-71-68-157.om.om.cox.net) |
| 00:08.27 | citywok | yea, the wording could have been better. i didn't mean i wanted dect using wifi, i meant dect instead of wifi |
| 00:08.48 | WIMPy | You usually should be able to use any handset. The right base is the interesting part. |
| 00:08.50 | citywok | we use aastra 6757i CT's, and i am kind of tired of the handset, it's not that great. |
| 00:09.02 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:09.12 | WIMPy | Just replace the handsets? |
| 00:09.35 | citywok | it uses some weird pairing mechanism |
| 00:09.53 | WIMPy | Not GAP compliant? |
| 00:09.53 | citywok | i'm not sure it can be done, though i haven't looked in to it that deeply. maybe. |
| 00:10.19 | citywok | "Employing secure WDCT cordless technology," |
| 00:10.38 | WIMPy | Whatever that means. |
| 00:10.49 | citywok | yea, it's not under their dect family header either |
| 00:11.12 | WIMPy | You need the US version? |
| 00:11.34 | citywok | yea, US |
| 00:11.53 | WIMPy | That somewhat limits your choice. |
| 00:12.43 | shmaltz | is there an android port for asterisk? |
| 00:13.13 | citywok | to run asterisk on android, or to run a softphone? |
| 00:13.29 | shmaltz | citywok, on android |
| 00:13.30 | WIMPy | shmaltz: I think, there actually is. Or at least it has been done. No access to telephony features, however. |
| 00:13.47 | shmaltz | not looking for telephony part, just the ip part |
| 00:13.50 | citywok | yea i imagine it would be purely ip |
| 00:14.08 | citywok | but i'm not sure why i would ever need to do that |
| 00:14.49 | shmaltz | citywok, no need other than walk around with a pbx for presentations or impress someone elses girlfriend |
| 00:15.17 | citywok | haha, get a tiny ass atom based desktop :P -- and i don't know of many girls you could impress with that :P |
| 00:15.30 | shmaltz | lol |
| 00:15.52 | citywok | and i live in nerd-dom, most girls i know are it/programming/computer nerds |
| 00:16.02 | shmaltz | well if all of a sudden I press a button on my tab and their phone rings with their grandmas cid and it connect her to Obama |
| 00:16.13 | shmaltz | citywok, where is that? |
| 00:16.29 | citywok | Redmond, WA |
| 00:16.48 | citywok | little company based here, you may have heard of them before. They make the windowzzzz and the tree sixty |
| 00:17.42 | shmaltz | never heard of them :o |
| 00:17.53 | shmaltz | they trade on any of the markets? |
| 00:18.37 | citywok | nahh |
| 00:18.48 | citywok | little shop in some guys garage |
| 00:18.54 | citywok | i hear he dropped out of the U.W. |
| 00:19.13 | shmaltz | a drop out is your boss? |
| 00:19.31 | citywok | I'm in the #asterisk channel, do you really think i work for bill/steve? :P |
| 00:19.53 | shmaltz | I have no clue, anything flies |
| 00:20.30 | citywok | haha, although i do have a microsoft alias, i am not a msft employee, we are a msft vendor |
| 00:22.31 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 00:28.20 | *** join/#asterisk greezmunkey (~greezmunk@173.241.161.62) |
| 00:40.49 | *** join/#asterisk Ta^3 (~tacvbo@gponr9-203-11-242.iusacell.net) |
| 00:57.08 | *** join/#asterisk `paul (~kutimoy@119.93.15.82) |
| 01:00.21 | `paul | Im testing vicidial with sangoma. and getting a "Got SIP response 603 "Decline (licensing capacity exceeded)" when I emailed them about this they say that Its a bug cause the sangoma service is treating the "options" sip message as one connection. (Trial version of sangoma is just one connection hence the error). Is there a way for asterisk to not send the "options" message? |
| 01:05.44 | leifmadsen | `paul: sure, just hack the code ;) |
| 01:05.50 | leifmadsen | beyond that, I don't know |
| 01:05.56 | `paul | >< |
| 01:06.50 | voxter | I believe if you disable qualify= it wont send OPTIONS packets |
| 01:06.55 | voxter | although I could absolutely be wrong. |
| 01:07.06 | *** join/#asterisk diemos (ad0d8a31@gateway/web/freenode/ip.173.13.138.49) |
| 01:07.19 | `paul | hmmm |
| 01:07.20 | diemos | lame you can see my IP |
| 01:08.14 | *** part/#asterisk Starscreamer (~Starscrea@5e030669.bb.sky.com) |
| 01:09.08 | joeyjones | lol |
| 01:09.29 | joeyjones | my ip is a lie~ |
| 01:09.32 | leifmadsen | voxter: hmmm perhaps, for some reason I was thinking that was NOTIFY heh |
| 01:09.39 | leifmadsen | wtf do I know though? |
| 01:09.47 | leifmadsen | it's late and I just ate chocolate |
| 01:10.30 | voxter | leifmadsen: hm.. yeah, youre probably right. up until yesterday, i believed asterisk would actually attempt to find the best codec negotiation it could based on the codec already being used (ie if you were on a ulaw call and making another, it would pref ulaw for that too, even if it wasnt first in the list) - i was wrong. |
| 01:10.57 | voxter | leifmadsen: apparently it will pick the first codec in the list, and transcode if necessary, rather than evaluating and deciding! |
| 01:11.16 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 01:13.31 | leifmadsen | voxter: yep, it depends on how you configure the outbound channel -- they are not related in terms of codec negotiation. That is something that is going to get worked on for 1.10 |
| 01:13.47 | leifmadsen | the list is the only thing that matters at this point, with first listed has highest weighted |
| 01:13.52 | leifmadsen | and now I'm going to hang with the wife, lates! |
| 01:14.44 | WIMPy | Oh, the book isn't THAT bad... |
| 01:14.47 | WIMPy | ;-) |
| 01:15.49 | voxter | leifmadsen: enjoy your evening :) |
| 01:32.08 | *** join/#asterisk wesphillips (~wphillips@adsl-75-53-131-55.dsl.hstntx.sbcglobal.net) |
| 01:41.31 | *** join/#asterisk rcsheets (~rcsheets@madeleine.picosecond.org) |
| 01:53.34 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 01:57.52 | *** join/#asterisk Inssomniak (~dave@199.19.173.132) |
| 01:58.08 | Inssomniak | anyone running asterisk in a vmware esxi? |
| 02:05.08 | shmaltz | Inssomniak, it shouldn't be a problem, I haven't tried it but I don't see why it shouldnt work |
| 02:07.53 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 02:09.04 | atan | For my outbound context I have DIAL(SIP/voipms/${SIPPEER(${SIPID}:prefix)}${EXTEN}, 60) but it's not pulling prefix from sip.conf. Can anyone suggest what I might have forgotten? |
| 02:10.02 | WIMPy | I guess you only want to use ${prefix}. |
| 02:12.34 | atan | I forgot Set(SIPID=${SIPCHANINFO(peername)}) but it's still not running for me =| must of forgotten something |
| 02:17.24 | atan | WIMPy, thanks for the suggestion but ${prefix} inside my dialplan is not pulling prefix=1905 from sip.conf peer |
| 02:19.36 | atan | Maybe you can't just invent your own sip.conf vars |
| 02:20.43 | WIMPy | I never tried it, but my understanding was that it sets a simple variable in any channel created by that peer. |
| 02:27.23 | `paul | RejectReason: SIP INVITE without custom header <--- what does this mean? |
| 02:29.23 | Inssomniak | Im reading mixed results on esxi/asterisk |
| 02:29.38 | Inssomniak | I have it going so far no real issues but not load tested yet |
| 02:34.50 | atan | What's wrong with exten =>_NXXXXXX,n,Goto(${DIALSTATUS},1) exten =>_NXXXXXX,n(CHANUNAVAIL),Playback(cannot-complete-as-dialed)? |
| 02:34.58 | atan | Says channel does not exist |
| 02:35.03 | shmaltz | something wrong with the list? |
| 02:35.43 | WIMPy | You goto an extension but define a priority. |
| 02:37.09 | atan | We're good now ^_^ |
| 02:50.10 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 03:07.48 | *** join/#asterisk n0cturnal (~n0c@2403:dc00:ffff:fffc:226:18ff:fe3a:3bad) |
| 03:09.16 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 03:19.01 | *** join/#asterisk fofware (~Fabian@host210.190-225-14.telecom.net.ar) |
| 03:22.59 | atan | Still at a loss as to how I can invent a variable for sip.conf peers. Is it even possible? |
| 03:23.35 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:23.52 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 03:25.14 | atan | Lord. Someone already said it. ^_^ my bad! setvar = variable=value |
| 03:52.26 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 03:52.36 | *** part/#asterisk k-man (~jason@unaffiliated/k-man) |
| 03:58.53 | WIMPy | Err. I thought that was what you were doing the whole time. |
| 04:03.10 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 04:03.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:06.25 | p3nguin | heh |
| 04:06.44 | WIMPy | ho |
| 04:07.34 | *** part/#asterisk wesphillips (~wphillips@adsl-75-53-131-55.dsl.hstntx.sbcglobal.net) |
| 04:07.39 | dan__t | Hi. |
| 04:18.24 | *** join/#asterisk copland (~jawa78@c-174-58-3-18.hsd1.fl.comcast.net) |
| 04:21.43 | copland | I am having issues with trying to get sip clients to connect to the asterisk server. The user/pass are right clients just time out there connections and nothing is displayed in the logs any ideas |
| 04:27.11 | p3nguin | firewall? |
| 04:27.12 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 04:32.02 | copland | no it on the same network I might just redo the install |
| 04:32.24 | copland | since it not on production environment just a sandbox |
| 04:32.38 | p3nguin | Just because it is on the same network does not mean there is no firewall. |
| 04:33.22 | p3nguin | Get on the box and run "iptables -L" to see what the firewall is doing. |
| 04:35.17 | copland | It not doing anything out of the ordinary |
| 04:36.04 | copland | nothing showing as blocked or dropped |
| 04:36.38 | copland | I probably screwed something up when running the at Google voice script for the 2nd time |
| 04:38.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 04:38.34 | *** join/#asterisk tyman (~tyler@adsl-065-015-255-051.sip.hsv.bellsouth.net) |
| 04:42.00 | *** join/#asterisk mjobin (~mjobin@64.229.249.200) |
| 04:42.35 | tyman | If i have a mailbox=6001@default statement in my sip.conf, shouldn't a call to VoiceMailMain() from exten => 6001 know what mailbox i need and ONLY ask for password? |
| 04:42.48 | mjobin | hi there ! I'm looking for the guy from quebec taking care of the servers. I met him once but forgot his name. |
| 04:43.01 | mjobin | wrong channel, sorry |
| 04:43.02 | tyman | it asks me for mailbox then passwd each time |
| 04:43.56 | p3nguin | I hate that. |
| 04:44.18 | tyman | works as expected with VoiceMailMain(${CALLERID(num)}@default) (eg only asks for pwd) |
| 04:44.55 | tyman | p3nguin: responding to me or mjobin? |
| 04:45.20 | p3nguin | I was referring to the VoiceMailMain() thing. |
| 04:46.37 | tyman | p3nguin: yeah, don't see the point in the sip directive unless it passes you thru...no big deal unless you're setting callerid in sip.conf as DID |
| 04:47.05 | p3nguin | The mailbox setting is for MWI. |
| 04:47.25 | p3nguin | mailbox subscriptions |
| 04:47.27 | tyman | then you need the vmail directive (unless you want to write a lookup in the astdb or something |
| 04:48.07 | tyman | p3nguin: so do you just use the CALLERID(num) deref? |
| 04:48.21 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-tdvwvopquxpxppti) |
| 04:48.23 | p3nguin | I use VoiceMailMain(${CALLERID(num)}@default) |
| 04:48.50 | p3nguin | My problem is similar to your initial complaint. If I call my own voicemail and press * to check my messages, I have to again enter in the mailbox then the password. |
| 04:49.10 | tyman | p3nguin: ok...that's what i've been doing as a temp solution...if they had DID in callerid then i could rewrite in the Dial on PSTN calls |
| 04:51.08 | p3nguin | I feel like if I already reached the mailbox of a specific person (in this case, me), if I press * for the purpose it was intended, I should only need to enter in the password rather than first the mailbox number and then the password. |
| 04:51.59 | tyman | p3nguin: y, prob could figure a way to do this with simple macro...I mean, to ultimately give VoiceMailMain what it wants (your EXTEN) |
| 04:52.25 | p3nguin | Well, it wants my mailbox number -- it doesn't care about extension numbers. |
| 04:52.57 | tyman | right....yes...that was incorrect...i'm just using matching exten/mbox numbers |
| 04:55.05 | tyman | you could store a EXTEN to MBOX lookup in the ASTDB under globals, and lookup that within VoiceMailMain |
| 05:15.08 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 05:18.16 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 05:26.44 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 05:26.50 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 05:27.30 | *** part/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 05:27.42 | gladier | hey guys - i'm trying to look at getting tcp support in 1.4 |
| 05:28.07 | gladier | i know that someone was writing a patch for 1.2 but can't see any specifics for 1.4 |
| 05:37.42 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 05:44.46 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 05:48.48 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:51.12 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 06:32.55 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 06:40.48 | *** join/#asterisk fofware (~Fabian@host210.190-225-14.telecom.net.ar) |
| 06:42.37 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:44.51 | atan | I don't suppose there's a way to forcefully set CDR(duration)? ^_^ |
| 06:46.30 | atan | tyman, if you're still messing with mailboxes here's what I use: http://pastie.org/1381784 |
| 06:47.20 | tyman | atan: nice...that's perf |
| 06:47.45 | atan | I used *99 because that's what people are used to where I live. The local telco uses *99 for their crummy service. |
| 06:48.06 | atan | They actually charge like $5/month for a mailbox with 10 messages. $10 for 25 messages, and like $20 for 100 messages. |
| 06:48.12 | atan | None of which save more than 10 days. |
| 06:48.21 | atan | It's like... grr. If I had a permit to kill. |
| 06:48.47 | atan | tyman, I don't suppose you know anything much about CDR do ya? ^_^ I'm on a wild hunt to hack mine |
| 06:49.54 | tyman | atan: I know what you mean about that...(caugh) cisco telephony |
| 06:50.36 | tyman | atan: not really much on cdr yet...just getting into asterisk really...learning fast though |
| 06:50.59 | atan | ^_^ it's well worth learning |
| 06:52.53 | tyman | atan: i will for sure, just haven't gotten there...and will be doing that via postgres. Getting ready to get my configs in ARA and get that dialed in...then CDR |
| 06:53.23 | tyman | atan: CDR is last on the list because i'm not in production with anything much yet |
| 06:54.00 | tyman | atan: heard mixed info (mostly dated) on ARAs stability. Heard anything on ARA stability in 1.8.1? |
| 06:54.30 | atan | What the heck is ARA? |
| 06:54.50 | tyman | most of the info i read on the subject appears to be relvent to 1.[46] |
| 06:55.08 | atan | I can't comment on ARA since I haven't a sweet clue what it is :P |
| 06:55.33 | atan | I've been using 1.8 for a month or two. So far things are smashing. |
| 06:55.51 | atan | The video phone users have never been happier. And I swear it sounds better.... but I'm just nuts. |
| 06:57.26 | tyman | http://www.asterisk.org/doxygen/asterisk1.4/AstARA.html |
| 06:58.38 | tyman | atan: probably sounds better because you get that subtle cues with lipsync and expressions that make those consonants more clear intuitively |
| 06:59.05 | tyman | atan: which video phones are you using? VTX? |
| 06:59.47 | tyman | I love * so far...my brain is just exploding with ideas |
| 07:00.21 | ChannelZ | gets the mop |
| 07:01.12 | kaldemar | tyman: where did you hear mixed info on realtime stability? |
| 07:01.20 | *** join/#asterisk titter (~Justin@c-98-208-153-116.hsd1.fl.comcast.net) |
| 07:02.32 | tyman | kaldemar: i'll try to find the link that I read, and was corroborated by the instructor at my Asterisk Advanced class that i'm in this week at digium |
| 07:02.55 | kaldemar | atan: you could use custom CDR and have your own variable for a modified duration. |
| 07:05.08 | tyman | kaldemar: my plans for * have asterisk ARA as a big part of them...are you using ARA in production? |
| 07:05.26 | kaldemar | tyman: no, but many do. |
| 07:07.18 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 07:07.24 | tyman | kaldemar: i'll be checking it out next week. Just planning on getting my dCAP this week and then checking it out next week. |
| 07:07.39 | atan | tyman, Nortels =\ |
| 07:08.29 | tyman | atan: meridians all the way |
| 07:09.11 | atan | tyman, cost was my issue otherwise I'd be all over something nicer |
| 07:10.30 | tyman | atan: we deal with cisco ucm all the time...don't see the value |
| 07:19.51 | *** part/#asterisk gladier (~gladier@pedobear.molested.me) |
| 07:25.48 | atan | Are there any cheap 411 services out there? |
| 07:25.56 | atan | Paying $1 per lookup seems a bit much. Hmm. |
| 07:26.16 | atan | I could play a recording "Use Google to find a number, it's far cheaper!!" but that seems taboo |
| 07:29.02 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:33.35 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 07:33.36 | cmnky | $1 per lookup is retarded |
| 07:35.05 | cmnky | especially when its a computer doing the query |
| 07:37.14 | luckman212 | hi folks. can someone tell me the right command to use to svn checkout asterisk 1.8.1.1 from digium's source? i thought it might be "svn co http://svn.asterisk.org/svn/asterisk/branches/1.8 /usr/src/asterisk" but not sure that's right |
| 07:38.52 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:38.54 | schmidts | good morning |
| 07:38.56 | *** join/#asterisk Beltechs (~Beltechs@166.190.193.180) |
| 07:44.31 | Beltechs | could a Misc Destination be a URL ? |
| 07:45.00 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:45.31 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:48.41 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 07:50.29 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 08:01.01 | kaldemar | Beltechs: a misc destination? |
| 08:02.03 | kaldemar | Beltechs: oh, freepbx stuff. ask in #freepbx. |
| 08:03.44 | *** part/#asterisk `paul (~kutimoy@119.93.15.82) |
| 08:04.42 | *** join/#asterisk joobie (~joobie@CPE-124-176-177-112.vic.bigpond.net.au) |
| 08:04.56 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 08:06.31 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:11.54 | EmleyMoor | Here's an odd one. When I make a call using ekiga that Asterisk routes over DAHDI, I get no sound. I can make calls that go by SIP or IAX OK, and make DAHDI calls from DAHDI handsets |
| 08:16.40 | EmleyMoor | Likewise with another SIP phoe |
| 08:16.42 | EmleyMoor | phone |
| 08:18.43 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 08:21.19 | EmleyMoor | Using 1.6.2.9 if it helps |
| 08:24.34 | aimka | EmleyMoor: Does your softphone receives any RTP packet ? |
| 08:25.14 | EmleyMoor | aimka: What's the best way to tell? |
| 08:25.28 | aimka | hummmm, whireshark maybe :-/ |
| 08:25.36 | aimka | wireshark |
| 08:26.18 | EmleyMoor | But why the difference if the outgoing medium is DAHDI? |
| 08:27.04 | kaldemar | rtp debug in CLI is probably an easier way to see if asterisk is sending any RTP packets. |
| 08:27.43 | kaldemar | EmleyMoor: pastebin a CLI output of a call with sip debug and verbosity enabled. |
| 08:28.15 | EmleyMoor | How verbose? |
| 08:29.09 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 08:29.19 | EmleyMoor | Will 3 do? |
| 08:29.33 | kaldemar | 3 is fine. |
| 08:34.06 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:35.39 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:37.33 | EmleyMoor | I can't make any sense of this - all the SIP debug during the call is not for anything to do with the call |
| 08:40.56 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 08:41.10 | EmleyMoor | Ah, maybe there's something now - pastebin URL will follow |
| 08:41.20 | kaldemar | maybe activity from other phones or OPTIONS messages because of qualify? enable debug for the particular peer only (with sip set debug peer <peername>) if there's other activity. |
| 08:41.37 | EmleyMoor | Even better, will try that |
| 08:42.31 | *** join/#asterisk mAniAk-_- (~arrr@77.240.212.222) |
| 08:43.04 | mAniAk-_- | does the zaptel modules come with the zaptel package in debian? |
| 08:46.41 | kaldemar | mAniAk-_-: it only seems to have the utilities. |
| 08:48.19 | EmleyMoor | http://pastebin.com/3zeJ9QNV |
| 08:52.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:52.53 | mAniAk-_- | kaldemar: had to compile then with the zaptel-source package |
| 08:53.34 | kaldemar | EmleyMoor: looks like you're not even getting a call through. are you dialing a valid number? |
| 08:54.25 | EmleyMoor | kaldemar: Yes. Calls do go - can even get it to ring my mobile, but no sound |
| 08:54.58 | kaldemar | mAniAk-_-: zaptel is really old, it was renamed to DAHDI long ago. you can find DAHDI in the backports repository. |
| 08:55.22 | kaldemar | EmleyMoor: you're pastebin doesn't look like a call that goes through. |
| 08:55.52 | mAniAk-_- | kaldemar: an old install of asterisk had a disk crash last nigth, backups only on configs and we lack real in house knowledge of asterisk |
| 08:56.00 | *** join/#asterisk haakonnes (~haakonnes@137.174.34.95.customer.cdi.no) |
| 08:56.21 | mAniAk-_- | kaldemar: so just trying to get it up and running agian |
| 08:56.30 | kaldemar | mAniAk-_-: what version of asterisk are you setting up? |
| 08:56.31 | EmleyMoor | It does go through, though |
| 08:57.09 | kaldemar | EmleyMoor: answer the call then. |
| 08:57.22 | kaldemar | and pastebin again. |
| 08:57.51 | EmleyMoor | It is answered. Also, when I call something I can answer, and do, likewise happens - will demonstrate |
| 08:57.54 | mAniAk-_- | kaldemar: 1.4, debian stable |
| 08:58.16 | mAniAk-_- | kaldemar: think its the same that was running on the one that crashed |
| 08:58.16 | kaldemar | mAniAk-_-: the 1.4 version in lenny? |
| 08:58.19 | mAniAk-_- | yes |
| 08:59.35 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 09:00.06 | kaldemar | go for zaptel then. at least in vanilla asterisk, DAHDI support came with 1.4.22. lenny has a 1.4.21.2-based version. |
| 09:00.18 | *** join/#asterisk m_tadeu (~quassel@89.181.100.210) |
| 09:02.45 | mAniAk-_- | ty |
| 09:02.48 | mAniAk-_- | any ideas? http://pastie.org/1381951 |
| 09:03.06 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 09:04.43 | EmleyMoor | http://pastebin.com/kbHhgWNR |
| 09:05.02 | EmleyMoor | (I called a phone I have control of, and answered, on that one |
| 09:17.56 | kaldemar | mAniAk-_-: plenty. some config file seems to try to include a file that doesn't exist now. look for #include directives in your config files to know which one it is. |
| 09:18.19 | kaldemar | mAniAk-_-: the musiconhold warning is not dangerous. |
| 09:19.55 | kaldemar | mAniAk-_-: was there a database to store some configs? extconfig.conf will give you hints on what it tries to do with databases. |
| 09:21.35 | mAniAk-_- | hmm yeah there were some there |
| 09:21.44 | mAniAk-_- | couldnt dump the database tho |
| 09:21.55 | mAniAk-_- | could only select * and copy paste the text |
| 09:22.49 | m_tadeu | I have asterisk+2pc's with softphone in my lan. One of them is getting registered with the lan ip, but the other is being registered with the external ip. why is this? |
| 09:24.11 | kaldemar | EmleyMoor: what is DAHDI/4? an analog line? don't see much of anything that indicates issues. what is lacking is and indication of answer though. |
| 09:25.44 | kaldemar | m_tadeu: what address do the soft phones register to? both to the LAN address? |
| 09:26.24 | hrhrhr | is there a way to have a single speaker/dialer for a call group |
| 09:26.30 | hrhrhr | i.e. sales |
| 09:26.39 | m_tadeu | kaldemar: indeed. both the server and the pc's are in the same subnet |
| 09:26.40 | hrhrhr | im sure 20 phones don't need to be ringing all at once |
| 09:31.10 | kaldemar | hrhrhr: make the phones ring without sound. more of a phone feature. |
| 09:31.23 | *** join/#asterisk Tim_Toady (~moi@188.4.146.42.dsl.dyn.forthnet.gr) |
| 09:31.45 | hrhrhr | kaldemar: be nice to have some sort of aural indication tho? |
| 09:32.02 | hrhrhr | maybe i keep one phone on :P |
| 09:34.32 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 09:34.58 | z4nD4R | hi somebody help with snom 360 and SRTP? |
| 09:37.41 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-110.dsl.net.pk) |
| 09:38.00 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:40.44 | joobie | guys in the states, how much is ISDN for 10 lines? |
| 09:40.46 | joobie | like an average monthly cost |
| 09:40.47 | *** join/#asterisk Starscreamer (~starscrea@webmail.mackellar.co.uk) |
| 09:45.03 | *** join/#asterisk edgars_ (edgars@ns.dtg.lv) |
| 09:45.05 | edgars_ | yo |
| 09:45.50 | edgars_ | how much resources i need in virtual environment for 30-50 users? maximum concurrent cals 20 |
| 09:58.31 | *** join/#asterisk infobot (ibot@rikers.org) |
| 09:58.31 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1.1 (2010/12/15), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 10:02.04 | edgars_ | joobie: how much broad? :> |
| 10:06.25 | mechbangirc | hello i downloaded phpagi-2.20 how do i install it? |
| 10:07.35 | z4nD4R | hi somebody help with snom 360 and SRTP? |
| 10:16.36 | kaldemar | mechbangirc: not much to install. copy it somewhere and include it in your own php. |
| 10:17.15 | mechbangirc | kaldemar: thanks |
| 10:18.17 | EmleyMoor | kaldemar: It is an analog line. It appears I may be missing the answer indication, as you say - if I could fix that, it may help |
| 10:19.41 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 10:22.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:23.51 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 10:23.51 | EmleyMoor | Just to summarise: when my Asterisk box calls out over the analog line on DAHDI/4, it gets no indication that the call has been answered. Probably in consequence, a SIP phone used to make the call remains silent. What do I need to do to get the answer indication? |
| 10:27.38 | *** join/#asterisk Starscreamer (~starscrea@mailgate.mackellar.co.uk) |
| 10:29.09 | mechbangirc | EmleyMoor: indications.conf use your country code |
| 10:29.14 | *** part/#asterisk mechbangirc (~mechbangi@mbl-82-58-110.dsl.net.pk) |
| 10:33.41 | EmleyMoor | That's not helped, btw |
| 10:34.23 | ectospasm | EmleyMoor: gets answered from which side? |
| 10:34.59 | EmleyMoor | ectospasm: Far, obviously |
| 10:36.06 | ectospasm | so the SIP phone is attached to the far end? |
| 10:36.46 | ectospasm | ... |
| 10:36.51 | EmleyMoor | No - the SIP phone is at the near end. What is at the far end is not important. It's the fact that the call is sent over DAHDI that is |
| 10:37.09 | ectospasm | yes, but the SIP phone and DAHDI are independent |
| 10:37.18 | EmleyMoor | Not entirelf |
| 10:37.21 | EmleyMoor | entirely |
| 10:37.54 | EmleyMoor | The SIP phone is irrelevant other than the way the problem manifests |
| 10:38.02 | ectospasm | if DAHDI doesn't receive ringing from the telco between it and the far end, it won't signal to the SIP phone that it's trying, and ringing the far end (180 ringing) |
| 10:38.10 | kleszcz | found a place with backdoor on openbsd http://img696.imageshack.us/img696/9687/openbsdbackdoorlocated.jpg ];> |
| 10:38.43 | EmleyMoor | If DAHDI *does* receive ringing and fails to pass it on, what's wrong? |
| 10:39.18 | EmleyMoor | (this is "blanket", not specific to any particular number called) |
| 10:39.44 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 10:42.21 | ectospasm | EmleyMoor: I'd need to see chan_dahdi.conf (et al.), and Asterisk CLI of such a failure |
| 10:42.38 | EmleyMoor | http://pastebin.com/kbHhgWNR |
| 10:43.27 | EmleyMoor | I will pastebin chan_dahdi.conf - but what exactly woult "et al." constitute |
| 10:43.49 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:45.15 | EmleyMoor | http://pastebin.com/CHGXtjML - chan_dahdi.conf |
| 10:45.20 | ectospasm | EmleyMoor: any files #include'd in chan_dahdi.conf |
| 10:45.52 | EmleyMoor | Ah - there is one - I will obfuscate private stuff therein and pastebin |
| 10:46.26 | ectospasm | EmleyMoor: can you please remove the comments, those are utterly useless: egrep -v '^[[:space:]]*;|^$' chan_dahdi.conf |
| 10:48.08 | EmleyMoor | Right - will redo and do likewise with the other |
| 10:48.49 | EmleyMoor | http://pastebin.com/Z9Qt8LVs |
| 10:49.10 | ectospasm | turn off callprogress |
| 10:49.16 | EmleyMoor | http://pastebin.com/RCzKhuuX |
| 10:49.27 | EmleyMoor | OK, will try |
| 10:49.49 | ectospasm | I've only seen that parameter cause problems on analog trunks |
| 10:51.02 | EmleyMoor | ectospasm: That was it :-) |
| 10:51.28 | ectospasm | EmleyMoor: no problem! Thanks for choosing Asterisk! |
| 10:52.00 | EmleyMoor | wonders if his faxes might work now |
| 10:52.34 | ectospasm | wishes faxing wasn't so necessary in the world |
| 10:52.39 | ectospasm | there are better ways now |
| 10:52.44 | *** join/#asterisk joeyjones (~joeyjones@S01060016365a2351.gv.shawcable.net) |
| 10:53.04 | EmleyMoor | I totally agree - but there are others who don't |
| 10:53.10 | ectospasm | EmleyMoor: yep |
| 10:53.28 | ectospasm | Mainly because technological signatures aren't trusted |
| 10:53.48 | ectospasm | even though they're harder to forge than classical signatures. |
| 10:54.30 | ectospasm | hell, at Walmart, whenever I choose credit as the method of payment, I always leave a simple line stroke as my signature |
| 10:59.52 | *** join/#asterisk bmg505 (~leon@196-209-44-79.dynamic.isadsl.co.za) |
| 10:59.57 | *** join/#asterisk Starscreamer (~starscrea@mailgate.mackellar.co.uk) |
| 11:00.53 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 11:01.10 | Amnesia | hi, where do you define the mail server for the voicemail serveremail option? |
| 11:01.37 | *** join/#asterisk fish9370 (~Miranda@mbox.fpg.ru) |
| 11:02.28 | ectospasm | Amnesia: in your MTA for the system |
| 11:02.49 | ectospasm | Amnesia: MTA like sendmail, postfix, exim, etc. |
| 11:03.04 | EmleyMoor | T.30 T1 timeouts now (!) |
| 11:03.14 | Amnesia | but what if it does not run on the machine asterisk is running on? |
| 11:03.19 | fish9370 | who know what I must to do if I take some function/file from Asterisk project to my own project? |
| 11:04.01 | ectospasm | Amnesia: Asterisk uses sendmail by default, so if your Asterisk system MTA isn't configured to send to your actual local mail server, you'll need to change that (outside of Asterisk) |
| 11:04.40 | Amnesia | well the mta runs on another server |
| 11:05.09 | ectospasm | Amnesia: your Asterisk system needs to know where to send stuff, you'll need an MTA that simply forwards to your actual mail server. |
| 11:05.44 | Amnesia | ah I see |
| 11:05.47 | ectospasm | fish9370: any of the ops should be able to answer that question, but they may not be available for three or more hours since it's just after 0500 US CST (GMT-6) |
| 11:05.55 | Amnesia | is there any tutorial for this? |
| 11:06.07 | ectospasm | Amnesia: there are tons for your given MTA |
| 11:06.09 | EmleyMoor | If it were possible to interface something like HylaFAX to the inbuilt Fax apps, I'd try that way. |
| 11:06.25 | Amnesia | hm I'm installing sendmail now |
| 11:06.50 | fish9370 | ectospasm: ok.. |
| 11:06.51 | ectospasm | Amnesia: sendmail is the big bad grandaddy of MTAs, it may be easier to install something simpler |
| 11:07.01 | Amnesia | hm |
| 11:07.07 | Amnesia | what about postfix:P? |
| 11:07.13 | Amnesia | thats even more advanced^^ |
| 11:07.14 | ectospasm | Amnesia: I'm partial to postfix |
| 11:07.14 | WIMPy | Simpler that sendmail? |
| 11:07.25 | ectospasm | WIMPy: I meant in terms of configuration |
| 11:07.35 | ectospasm | WIMPy: sendmail has always been rather arcane |
| 11:07.38 | WIMPy | Still |
| 11:07.43 | ectospasm | postfix is straightforward IMO |
| 11:07.50 | Amnesia | but ehm, asterisk looks on a mta on the local machine, then that mta should forward it to the main mta and that mta should handle the job? |
| 11:07.51 | ectospasm | ...or, *more* straightforward |
| 11:08.01 | ectospasm | Amnesia: yes |
| 11:08.20 | ectospasm | Amnesia: if you can find a versio of it for your system, I always like sSMTP |
| 11:08.26 | WIMPy | I prefer postfix for receiving mail, but for simple things like a smart host I find senmail easier. |
| 11:08.31 | Amnesia | so I should google something like: forward mail request to mta sendmail/postfix ? |
| 11:08.58 | ectospasm | Amnesia: yep |
| 11:09.05 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 11:12.02 | *** join/#asterisk _schmidts (~schmidts@lmlo.sil.at) |
| 11:12.51 | Amnesia | ectospasm, email relay host? |
| 11:14.02 | WIMPy | Or smarthost |
| 11:14.11 | ectospasm | Amnesia: yeah, that's what I know it as |
| 11:15.58 | Amnesia | hm can't find any good tutorial on it:/ |
| 11:15.59 | Amnesia | it' |
| 11:16.05 | Amnesia | s all slightly different |
| 11:18.18 | Amnesia | http://www.postfix.org/SASL_README.html#client_sasl |
| 11:18.24 | WIMPy | For sendmail just put the other host in to DS. |
| 11:18.26 | Amnesia | is that the same? |
| 11:19.41 | EmleyMoor | Hmmm... I now appearto be missing caller IDs on my DAHDI analog line |
| 11:20.39 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 11:22.10 | EmleyMoor | Any clues on caller ID detection on DAHDI in the UK? |
| 11:24.05 | *** join/#asterisk bmg505 (~leon@196-209-99-143.dynamic.isadsl.co.za) |
| 11:25.08 | *** join/#asterisk tyman_ (~tyler@adsl-065-015-255-051.sip.hsv.bellsouth.net) |
| 11:27.25 | *** join/#asterisk joeyjones (~joeyjones@93.186.171.52) |
| 11:34.08 | Amnesia | I've installed postfix now and have entered the main mta |
| 11:34.13 | Amnesia | is there nothing else I have to do |
| 11:34.22 | Amnesia | making a user called voicemail or something? |
| 11:42.54 | *** join/#asterisk jkroon (~jkroon@dsl-241-252-245.telkomadsl.co.za) |
| 11:45.00 | ectospasm | Amnesia: first things first, does "sendmail -t ..." work from the Linux shell, you receive the test e-mail and everything? |
| 11:45.30 | Amnesia | I'm trying postfix |
| 11:47.45 | Amnesia | gonna try sendmail now, sec |
| 11:58.37 | EmleyMoor | I'm using an AEX410P card and am not getting caller ID on my FXO |
| 12:04.25 | z4nD4R | hi somebody help with snom 360 and SRTP? |
| 12:08.24 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 12:09.29 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 12:09.37 | kjs | is there a way of purging voicemail from the command line? |
| 12:10.29 | z4nD4R | WIMPy: any idee with my poroblem ... srtp and snom 360? |
| 12:11.36 | z4nD4R | *ideas offcourse |
| 12:15.04 | EmleyMoor | Any patches to fix UK caller ID on 1.6.2.9? |
| 12:19.01 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-169.telkomadsl.co.za) |
| 12:19.51 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 12:22.25 | ectospasm | EmleyMoor: why not just upgrade to 1.6.2.15 and see if it's already been released? |
| 12:24.34 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 12:25.19 | dimm | asterisk not dial from some peer in LAN to some outgoing numbers |
| 12:25.55 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 12:28.07 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 12:28.46 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:29.38 | *** join/#asterisk linuxplatform (~centoslin@1x-193-157-207-182.uio.no) |
| 12:32.33 | EmleyMoor | Quite a big "just" but being considered |
| 12:33.43 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:35.59 | z4nD4R | somebody help with snom 360 and SRTP? - my log file http://pastebin.com/1yhvh19c |
| 12:42.37 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:45.48 | *** join/#asterisk linuxplatform (~centoslin@1x-193-157-207-182.uio.no) |
| 12:51.26 | angryuser | Good day |
| 12:52.19 | angryuser | When provider dia an INVITE to asterisk with sip username, and the field TO has the DID of the destination, asterisk should parse that DID and call it right ? |
| 12:52.44 | angryuser | *When provider do |
| 12:53.18 | *** part/#asterisk mAniAk-_- (~arrr@77.240.212.222) |
| 12:53.33 | angryuser | I have something like this: |
| 12:53.49 | angryuser | INVITE sip:990000000002@83.167.156.247 SIP/2.0 |
| 12:54.10 | angryuser | field TO is: To: <sip:33253440000@83.167.31.20> |
| 12:54.24 | angryuser | you see that DID is 3325xxxxxx |
| 12:54.31 | *** join/#asterisk [netman] (~netman@86.Red-88-19-41.staticIP.rima-tde.net) |
| 12:54.43 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 12:54.44 | angryuser | but right now my asterisk considers that 990000000xxx DID is called |
| 12:54.52 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:54.59 | angryuser | Is there any way to for asterisk to look at TO field ? |
| 12:55.12 | angryuser | force* |
| 12:56.48 | haakonnes | z4nD4R: I've never used SRTP, but this seems like something you should look at: |
| 12:56.51 | haakonnes | [Dec 16 13:29:43] WARNING[1589]: chan_sip.c:8380 process_sdp: Matched device setup to use SRTP, but request was not! |
| 12:57.18 | haakonnes | z4nD4R: seems like your phone is not trying to use srtp, but asterisk is? |
| 13:00.48 | z4nD4R | haakonnes: asterisk is set to use srtp, but the phones too ( snom 360 snom 370 ) |
| 13:00.59 | z4nD4R | haakonnes: i enable RTP encryption |
| 13:01.21 | haakonnes | I'm sorry, but I've never set my foot in a snom |
| 13:02.50 | z4nD4R | haakonnes: when i open WS and sniff this, WS show 488 not accerable here |
| 13:03.36 | z4nD4R | haakonnes: this show phones to |
| 13:15.23 | z4nD4R | haakonnes: sry this ist the right log http://pastebin.com/wURkcQUK |
| 13:15.37 | z4nD4R | somebody help with snom 360 and SRTP? - my log file http://pastebin.com/wURkcQUK |
| 13:16.15 | haakonnes | You can see asterisk sends a 488 Not acceptable here because the snom is not sending any srtp handles in the sdp |
| 13:16.25 | *** join/#asterisk jpmcallister (~jpmcallis@200.242.28.231) |
| 13:17.22 | jpmcallister | When I make calls using an IAX client I'm not hearing the calling tone. |
| 13:18.45 | z4nD4R | haakonnes: but in first invite asterisk send a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6qbkp3l+mK8rOrbZLzmRoS1AVmsohcKYDFrJs7Gj |
| 13:20.20 | jpmcallister | I don't hear a tone indicatino that * is dialing when I make a call using a IAX client. What should I configure? |
| 13:27.07 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:28.26 | haakonnes | z4nD4R: I think asterisk is waiting for a RTP/SAVP instead of RTP/AVP |
| 13:28.32 | haakonnes | but I am no expert on srtp |
| 13:29.37 | z4nD4R | haakonnes: u thing set RTP/SAVP as mandatory? |
| 13:30.09 | haakonnes | where? |
| 13:30.54 | z4nD4R | in phones |
| 13:31.12 | haakonnes | can you define that? |
| 13:31.43 | z4nD4R | yes |
| 13:31.50 | z4nD4R | i defined this |
| 13:31.50 | haakonnes | then i would try that yes |
| 13:32.08 | z4nD4R | but new message |
| 13:32.10 | z4nD4R | SRTP unprotect: authentication failure |
| 13:34.22 | *** join/#asterisk [netman] (~netman@247.Red-81-47-145.staticIP.rima-tde.net) |
| 13:37.56 | z4nD4R | i becomme message.. "[Dec 16 14:36:45] WARNING[23950]: res_srtp.c:346 ast_srtp_unprotect: SRTP unprotect: authentication failure |
| 13:37.56 | z4nD4R | " but phones show as encrypted call... Question is... is this encrypted or not? |
| 13:38.49 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 13:41.15 | haakonnes | http://forum.snom.com/index.php?showtopic=5013 |
| 13:48.15 | z4nD4R | haakonnes: i have the same result.... work with 80 auth. tag, but nor with 32 tag. ... i becomme message.. "[Dec 16 14:36:45] WARNING[23950]: res_srtp.c:346 ast_srtp_unprotect: SRTP unprotect: authentication failure |
| 13:48.32 | z4nD4R | i muss install higher asterisk version? |
| 13:49.22 | haakonnes | seems like asterisk is not dynamic on the bit tag yet |
| 13:49.51 | haakonnes | so you need to choose between either 80 or 32 |
| 13:50.07 | haakonnes | if you want 32, you need to change asterisk source as the article suggests |
| 13:50.41 | z4nD4R | is someting difrent in SRTP when i use 80 bit tag? |
| 13:51.06 | z4nD4R | if i use 80bit tag in current version off asterisk ( 1 |
| 13:51.11 | z4nD4R | 1.8.0 |
| 13:51.33 | z4nD4R | are the calls encrypted? |
| 13:51.34 | haakonnes | do they go through? |
| 13:51.56 | z4nD4R | what? |
| 13:52.08 | haakonnes | can you call another phone and answer? |
| 13:52.29 | z4nD4R | yes |
| 13:52.50 | haakonnes | then it's probably encrypted yes. But you could check with wireshark to be sure |
| 13:52.57 | z4nD4R | snom show lock, that indicate encrypted call |
| 13:53.13 | z4nD4R | asterisk show this res_srtp.c:346 ast_srtp_unprotect: SRTP unprotect: authentication failure |
| 13:53.26 | z4nD4R | ws show only UDP packets |
| 13:53.55 | haakonnes | yes, if you select one of the rtp packets, and in wireshark go to the telephony menu |
| 13:54.07 | z4nD4R | no rtp packet are here |
| 13:54.12 | z4nD4R | only UDP packet |
| 13:54.30 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 13:54.41 | haakonnes | ok? are you limiting the search then? |
| 13:54.45 | haakonnes | in wireshark? |
| 13:55.16 | z4nD4R | i have displeyed are packet from konversation |
| 13:55.37 | z4nD4R | upd... try i decode as RTP? |
| 13:55.56 | haakonnes | wireshark should recognize the udp voice packets as rtp |
| 13:57.13 | z4nD4R | haakonnes: this is the sniff http://www.sendspace.com/file/ew6ect |
| 13:57.43 | z4nD4R | server: 192.168.0.250 |
| 13:57.43 | z4nD4R | snom 360: .12 |
| 13:57.43 | z4nD4R | snom 370: .13 |
| 14:01.58 | z4nD4R | haakonnes: i try to analyze, but i here only nois... |
| 14:04.19 | haakonnes | Yes, only noise.. So it's probably encrypted |
| 14:05.38 | tuxx- | hi guys, we have an analog phone connected on a asterisk box with a TDM8 card. We get the following error when the analog phone is calling, the call is silenced for a second, and then continues. Anyone have a clue what this is about? Google gives me nothing. If you need a dahdi debug log, we can supply you with that (have to make it though ;-) http://pastie.org/1382555 |
| 14:05.54 | z4nD4R | haakonnes: i thig too |
| 14:07.05 | *** join/#asterisk [netman] (~netman@110.Red-83-55-245.dynamicIP.rima-tde.net) |
| 14:10.46 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 14:11.07 | *** join/#asterisk [SySteM] (~antoine@188.240.47.244) |
| 14:11.13 | [SySteM] | Hi |
| 14:11.58 | Amnesia | I've just installed postfix, but how do I configure it to relay to the main mta in order to send emails to the user when there's a voicemail message for that person? |
| 14:12.30 | [SySteM] | On wich method i can give a vocal message (from var) to a agent on queue depend var from a caller (he choose before entering in queue). Please ? |
| 14:12.35 | z4nD4R | haakonnes: http://www.sendspace.com/file/gyyrzx and this, this wold by unecryted |
| 14:13.57 | stix | Anyone else than me having problems with blind transfers after upgrading to asterisk 1.6.2.15? |
| 14:15.25 | haakonnes | z4nD4R: yes. I can hear you hang up |
| 14:16.44 | z4nD4R | haakonnes: the same like me... big thx :) |
| 14:20.28 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:22.37 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 14:23.47 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 14:24.07 | stix | Okay after I upgraded from 1.6.2.12 to 1.6.2.15, asterisk drops all calls I try to transfer. This doesn't happen if I make an attended transfer. I have tried with several clients. Anyone? |
| 14:24.59 | *** part/#asterisk ks3 (~ks3@74.203.195.1) |
| 14:25.32 | phix | stix: featuremap => blindxfer? |
| 14:25.49 | stix | phix, what does that mean? |
| 14:26.09 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 14:26.22 | [SySteM] | Please.. i want to customize queue-reporthold by var for each user ? |
| 14:26.47 | phix | stix: ummmm have you enabled blind transfer support in your conf file? |
| 14:27.23 | stix | I am using freepbx, so I don't really know |
| 14:27.31 | stix | it worked before and now it doesn't |
| 14:27.32 | leifmadsen | stix: have you looked at the issue tracker to see if someone has already reported that? It could very well be a bug. |
| 14:27.44 | stix | leifmadsen, I have seen this: http://forums.digium.com/viewtopic.php?f=1&t=76412&p=151488&hilit=transfer&sid=e14d3699c4fcd07d4fb55eb5fde0352a#p151488 |
| 14:27.51 | leifmadsen | that's not the issue tracker ;) |
| 14:27.55 | stix | looks like it may be the same |
| 14:28.13 | stix | no sorry :) |
| 14:28.15 | *** part/#asterisk ks3 (~ks3@74.203.195.1) |
| 14:28.17 | leifmadsen | that looks like different functionality to me |
| 14:28.27 | stix | cheking issues now |
| 14:29.12 | *** join/#asterisk CheBuzz_Home (~CheBuzz_H@68-29-20-159.pools.spcsdns.net) |
| 14:30.22 | CheBuzz_Home | I have installed asterisk on an ubuntu 10.04 box, but the base install is pretty open, right? Is there a good guide to securing the default install? |
| 14:31.34 | leifmadsen | CheBuzz_Home: follow the instructions at http://ofps.oreilly.com |
| 14:31.55 | leifmadsen | CheBuzz_Home: feel free to offer feedback on the page as well. The new instructions run Asterisk as a non-root installation. |
| 14:32.10 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:32.18 | leifmadsen | CheBuzz_Home: there is some information in the Security chapter, and there should be information about iptables somewhere in the book. |
| 14:32.54 | russellb | ~newbook |
| 14:32.54 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/ |
| 14:33.37 | russellb | ~fail2ban |
| 14:33.37 | infobot | fail2ban is, like, a program to ban people using iptables based on information in logs: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk. For information on setting up fail2ban with Asterisk, see the Security chapter of Asterisk: The Definitive Guide - http://ofps.oreilly.com/titles/9780596517342/ch25.html |
| 14:33.46 | leifmadsen | infobot: no, newbook is Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ |
| 14:33.46 | infobot | okay, leifmadsen |
| 14:33.49 | leifmadsen | ~newbook |
| 14:33.49 | infobot | i guess newbook is Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ |
| 14:34.00 | leifmadsen | grrrr, I forgot to remove the crap at the beginning |
| 14:34.06 | leifmadsen | looks at russellb to fix his mistake :) |
| 14:34.09 | russellb | you fix it |
| 14:34.14 | leifmadsen | I forget how :) |
| 14:34.24 | russellb | leifmadsen: newbook is <reply> .... |
| 14:34.44 | leifmadsen | isn't that what I just did? :) |
| 14:34.54 | russellb | no |
| 14:34.58 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:34.58 | russellb | literally, <reply> whatever |
| 14:35.03 | leifmadsen | ah |
| 14:35.16 | leifmadsen | infobot: no, newbook is <reply> Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ |
| 14:35.16 | infobot | okay, leifmadsen |
| 14:35.18 | russellb | ~thwack leifmadsen |
| 14:35.18 | infobot | ACTION hits leifmadsen on the eye with a iMac |
| 14:35.22 | leifmadsen | ~newbook |
| 14:35.23 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ |
| 14:35.26 | leifmadsen | sick |
| 14:35.39 | leifmadsen | ON the eye?! |
| 14:35.44 | leifmadsen | the humanity! |
| 14:36.14 | stix | leifmadsen, I can't seem to find any bug submitted on issues.asterisk.org - can you assist? It may be something simple missing in a conf. |
| 14:36.41 | leifmadsen | stix: if you provide some information perhaps. I'm working on triaging issues right now though. |
| 14:37.10 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:37.37 | stix | leifmadsen, okay. Well calls are disconnected when a user tries to transfer. This does not happen on attended transfers. |
| 14:38.11 | leifmadsen | stix: good place to start is console output with debug level logging showing what happens when you do the transfer, and if it is SIP channels, a sip trace as well |
| 14:38.29 | stix | leifmadsen, I have already done that |
| 14:38.45 | stix | but can't really see why it is happening anyway |
| 14:38.46 | leifmadsen | ok, then the next step is to show me |
| 14:38.52 | stix | oki |
| 14:38.53 | leifmadsen | I can't read minds |
| 14:38.58 | leifmadsen | (yet) |
| 14:38.58 | stix | oh :) |
| 14:39.28 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 14:40.52 | stix | leifmadsen, http://pastebin.com/VwAUbB9R. SIP 3167 calls 3164 - 3164 tries to blind transfer to 3165. SIP debug is from 3164. |
| 14:41.23 | leifmadsen | stix: where is the console output with debug level logging? |
| 14:42.15 | stix | leifmadsen, it is here: http://pastebin.com/X3YpwJZX |
| 14:43.56 | leifmadsen | stix: there is no debug level logging |
| 14:44.03 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 14:44.17 | stix | I thought so |
| 14:44.53 | stix | core set debug 9? |
| 14:45.23 | leifmadsen | after you've enabled it in logger.conf |
| 14:45.25 | leifmadsen | on the console |
| 14:46.47 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 14:46.48 | stix | debug is set for the full-log |
| 14:47.20 | [SySteM] | nothing to help me ? |
| 14:47.40 | joeyjones | ~encryption |
| 14:47.45 | joeyjones | ~security |
| 14:47.45 | infobot | Security is way over rated, Heck I imagine you still run windows on your regular PC don't lecture about security |
| 14:47.47 | [SySteM] | i search how give a personnalised message to the agent on queue wich receive the caller |
| 14:49.52 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 14:52.15 | stix | leifmadsen, there it is: http://pastebin.com/4TbKX4Yw |
| 14:52.17 | stix | a lot |
| 14:53.21 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 14:56.05 | *** join/#asterisk garymc (~chatzilla@host81-148-106-225.in-addr.btopenworld.com) |
| 14:59.02 | *** join/#asterisk yonahw (~yonahw@www.mcatrack.com) |
| 14:59.10 | yonahw | ~itsp |
| 14:59.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:59.14 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-lllbbbrrkwptbrkl) |
| 14:59.24 | yonahw | ~itsplist-us |
| 14:59.24 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 14:59.32 | leifmadsen | stix: this is too confusing with all this freepbx BS -- you should reproduce it with a simple dialplan |
| 15:00.30 | stix | leifmadsen, I am not sure I can figure that out |
| 15:01.12 | leifmadsen | stix: then you'll just have to wait for pabelanger to write the transfer tests and see if he runs into it too |
| 15:01.37 | stix | okay |
| 15:02.04 | pabelanger | back scrolls |
| 15:04.31 | stix | pabelanger, why are you writing transfer tests? |
| 15:05.46 | pabelanger | for the testsuite |
| 15:06.48 | [SySteM] | .. there is no possibility for my ask ? |
| 15:07.23 | stix | So what can I do in the meantime, I have an entire company who cannot perform blind transfers now |
| 15:07.37 | pabelanger | rollback? |
| 15:09.08 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:09.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:10.12 | stix | pabelanger, a dependency nightmare on centos 55 |
| 15:12.27 | stix | I might have figured it out |
| 15:14.39 | leifmadsen | that's why I don't use RPMs :) |
| 15:14.42 | leifmadsen | compiling ftw |
| 15:16.47 | stix | leifmadsen, yes but it's usually problemfree :) |
| 15:17.37 | leifmadsen | sounds like a local test plan issue to me |
| 15:19.34 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:20.19 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:20.31 | tuxx- | hi guys, we have an analog phone connected on a asterisk box with a TDM8 card. We get the following error when the analog phone is calling, the call is silenced for a second, and then continues. Anyone have a clue what this is about? Google gives me nothing. If you need a dahdi debug log, we can supply you with that (have to make it though ;-) http://pastie.org/1382555 |
| 15:20.34 | moos3 | rpm do suck, compile |
| 15:26.29 | Amnesia | moos3, compiling only rocks when using stow:P |
| 15:28.11 | CheBuzz_Home | What .so provides the symbol ast_aji_get_client ? |
| 15:28.27 | stix | leifmadsen and pabelanger, I have managed to downgrade to 1.6.2.12 and now I have no problems at all |
| 15:28.47 | stix | sorry to 1.6.2.14 |
| 15:30.06 | pabelanger | stix: what time of transfer are you doing? |
| 15:30.12 | pabelanger | s/time/type/ |
| 15:30.42 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 15:31.28 | stix | SIP REFER, if that is what you mean |
| 15:31.33 | stix | blind transfer |
| 15:31.56 | pabelanger | Are you using features.conf? |
| 15:32.24 | stix | I don't think so. Sorry I am using freepbx |
| 15:33.00 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 15:33.04 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 15:33.58 | stix | I see "blindxfer=##" in there |
| 15:34.02 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:35.05 | leifmadsen | stix: are you transferring using DTMF or the Transfer key on the phone? |
| 15:35.05 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:35.18 | stix | transfer key on the phone and softphones |
| 15:35.23 | stix | so not ## |
| 15:36.05 | pabelanger | I'm working on DTMF transfers ATM. |
| 15:36.06 | stix | that was what I mean by the SIP REFER - but not sure it was the same :) |
| 15:36.11 | stix | okay |
| 15:36.20 | pabelanger | which I just ran against branches/1.6.2 and passed |
| 15:36.35 | stix | pabelanger, 1.6.2.15 also? |
| 15:37.28 | pabelanger | Un-tested. We usually only run against the branch, not tags |
| 15:37.42 | leifmadsen | 1.6.2.16-rc1 exists now too |
| 15:37.43 | stix | anyway, I don't know if it's a bug. But after I upgraded from 1.6.2.14 to 1.6.2.15, it didn't work. When I downgraded again, it worked again |
| 15:38.00 | stix | okay |
| 15:38.10 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 15:38.28 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
| 15:39.18 | WindBack | My SIP provider is sending me periodically an OPTIONS message. My Asterisk is answering it with 404 NOT Found. How I can configure it to answer this message with 200 OK |
| 15:40.12 | leifmadsen | WindBack: enable the extension it is looking for in your default context |
| 15:46.11 | WindBack | leifmadsen: and what applications should I run at this extension? |
| 15:46.27 | WindBack | noop for example? |
| 15:46.52 | leifmadsen | WindBack: sure |
| 15:46.56 | moos3 | anyone know anyone that offers unlimited 1800 inbound calling ? |
| 15:47.58 | SuPrSluG | moos3:no. that's the point of an 800 number. you pay for the call |
| 15:48.02 | WindBack | leifmadsen: To create the extension, I'm using the TO field of received OPTIONS message, is it ok? |
| 15:48.18 | leifmadsen | WindBack: try it :) |
| 15:49.33 | *** join/#asterisk zigzapgabe (~gabe@c-71-57-35-169.hsd1.il.comcast.net) |
| 15:49.51 | moos3 | SuPrSluG: its not about paying, but when you pay 2.9 cent at a high volume of inbound calls at like 1200 hours a month, 2.9 gets expensive |
| 15:51.42 | SuPrSluG | moos3: you may be able to find a cheaper rate, but i doubt anyone will offer fixed rate |
| 15:52.52 | WindBack | leifmadsen: it is still not working :( |
| 15:53.48 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:53.50 | WindBack | leifmadsen: perhaps I should add it to the peer context |
| 15:54.00 | WindBack | instead of default context |
| 15:56.52 | devmod | trying to collect info on a deadlock but 'core show locks' isnt available, any ideas? |
| 15:57.50 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 15:57.57 | leifmadsen | devmod: enable DEBUG_THREADS in menuselect |
| 15:59.03 | devmod | leifmadsen, k thanks |
| 15:59.23 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:59.36 | WindBack | leifmadsen: When * receive the OPTIONS i see the following message: "Looking for in default (domain 192.169.0.2:5060)". Seems that it is looking for an empty extension in default context |
| 15:59.44 | devmod | also, on the docs says safe_asterisk would add the -g option to asterisk but It doesnt seem to do it |
| 15:59.53 | *** join/#asterisk quenenni (~atchoum@85.234.197.189) |
| 16:01.38 | _schmidts | windback there was an issue on the tracker for this problem and AFAIK its allready fixed |
| 16:03.54 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e7.bcn.adamo.es) |
| 16:05.13 | *** join/#asterisk quenenni (~atchoum@unaffiliated/quenenni) |
| 16:08.57 | *** part/#asterisk quenenni (~atchoum@unaffiliated/quenenni) |
| 16:08.57 | *** join/#asterisk quenenni (~atchoum@unaffiliated/quenenni) |
| 16:12.33 | *** join/#asterisk fofware (~Fabian@host210.190-225-14.telecom.net.ar) |
| 16:14.10 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:30.03 | *** join/#asterisk tyman (~tyler@nat/digium/x-higggwnnldssvfac) |
| 16:30.05 | *** part/#asterisk tyman (~tyler@nat/digium/x-higggwnnldssvfac) |
| 16:33.36 | *** join/#asterisk grayhame (~grayhame@74-94-250-169-Nashville.hfc.comcastbusiness.net) |
| 16:35.07 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 16:35.56 | *** join/#asterisk n1gak (~lawrence@189.162.186.156) |
| 16:37.27 | *** join/#asterisk Starscreamer (~starscrea@mailgate.mackellar.co.uk) |
| 16:38.38 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 16:39.35 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:39.42 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 16:40.51 | moos3 | anyone have a recommendation for a sip cordless handset ? |
| 16:41.26 | WIMPy | moos3: Yes. Don't do it. Go dect. |
| 16:41.55 | moos3 | WIMPy: i need something that i can take with me to my barn |
| 16:43.21 | WIMPy | You can get sip dect bases, but quality seems to come at a price there. What country? |
| 16:44.19 | moos3 | WIMPy: us |
| 16:45.02 | WIMPy | That only leaves a pots cordless phone on an ATA as 2nd choice. |
| 16:45.12 | moos3 | k |
| 16:46.24 | n1gak | Are you saying WIMPy that there are no DECT 6.0 SIP bases? (I'm looking into this exact question for a friend in the US) |
| 16:46.45 | WIMPy | There are, but they seem to be rather cotly. |
| 16:46.52 | WIMPy | cost |
| 16:47.01 | *** join/#asterisk Tim_Toady (~moi@188.4.146.42.dsl.dyn.forthnet.gr) |
| 16:47.03 | n1gak | ahhh |
| 16:47.45 | WIMPy | If it wasn't US, I'd suggest using an isdn dect base. They are cheap and well tested, but I'd be surprised if an US version existed. |
| 16:48.48 | leifmadsen | Aastra makes one I believe |
| 16:48.57 | leifmadsen | Aastra 93i if I remember correctly |
| 16:49.43 | WIMPy | Not known to google. |
| 16:51.04 | n1gak | <PROTECTED> |
| 16:52.05 | n1gak | KX TGP550 is the prod code on that |
| 16:52.12 | WIMPy | Actually it seems the Siemens things are actually usable *IF* you only have one handset. |
| 16:52.22 | *** part/#asterisk CheBuzz_Home (~CheBuzz_H@68-29-20-159.pools.spcsdns.net) |
| 16:54.02 | WIMPy | I have been saying they are complete crap, but after I set up a 2nd base so there is only one mobile per base, it seems to work. |
| 16:54.18 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:54.49 | n1gak | so, I wanted to come back to another point you made: You seemed to Pooh-pooh the WiFi SIP-phone out of hand in favor of DECT. Justification? |
| 16:55.31 | WIMPy | Wifi and realtime don't mix. It gives you a lot of jitter and good chances for packet loss. |
| 16:55.40 | n1gak | OK |
| 16:56.09 | WIMPy | dect gives better quality on longer distances and the battery live is incomparable. |
| 16:56.44 | *** join/#asterisk johani (~johani@h-201-129.A176.priv.bahnhof.se) |
| 16:57.02 | WIMPy | Wifi phones tend to only give you 24h of standby. Dect phones easily give yuo two weeks. |
| 16:58.04 | n1gak | Point taken |
| 16:59.03 | WIMPy | I see only two points pro Wifi: 1. It's geeky :-) and 2. you can add as many APs as you like. |
| 16:59.23 | n1gak | and you can take your handset to places where there are free WiFi APs far far away from your house |
| 17:00.02 | WIMPy | And give away your SIP account while doing so. |
| 17:00.24 | WIMPy | But, yes. It is possible. |
| 17:00.52 | WIMPy | I guess that's the geeky part. |
| 17:01.24 | WIMPy | I'm not even sure if any of them support WPA2. |
| 17:06.43 | n1gak | OK ... I'm sold :) |
| 17:07.01 | _Corey_ | The Panasonic SIP units are very good |
| 17:07.07 | _Corey_ | We've sold a bunch of them |
| 17:07.26 | joeyjones | personally i'd like a wifi SIP phone |
| 17:12.02 | joeyjones | but i'd need atleast WPA support |
| 17:13.31 | joeyjones | but damn, SIP really need encryption if only during authentication |
| 17:14.17 | WIMPy | WPA is available, just not sure about WPA2. |
| 17:15.04 | moos3 | does anyone by chance have a query for cdr to only find the inbound calls |
| 17:15.05 | joeyjones | WIMPy: WPA should be secure enough anyway |
| 17:15.48 | WIMPy | joeyjones: Well they are nice gadgets, but I really don't think you want to use them on a daily basis. |
| 17:16.35 | joeyjones | WIMPy: it would be more for when travelling and needing to make longdistance calls |
| 17:16.45 | joeyjones | i plan to use my cellphone as it is |
| 17:18.07 | WIMPy | I tend to dial to a MOH extension before making a real call to check if I'd be able to understand ppl. |
| 17:18.36 | WIMPy | But if you're sitting at a friends and using their AP it can be quite useful. |
| 17:19.38 | joeyjones | WIMPy: call quality is fine on HSPA+ anyway |
| 17:19.53 | joeyjones | and in canada they carriers are not allowed to throttle or block traffic |
| 17:20.01 | joeyjones | atleast as far as i've been able to research |
| 17:21.26 | WIMPy | Yes, HSPA will probably work a lot better than wifi. |
| 17:24.40 | WindBack | My SIP provider sends periodically OPTIONS message to qualify the trunk. I dont know why, my * is sending 404 instead of 200 OK. Any idea? http://www.pastebin.ca/2021350 |
| 17:25.36 | WIMPy | WindBack: Does the response matter in any way? |
| 17:27.04 | WindBack | WIMPy: probably no, but it will be better if it will answer sending 200 OK |
| 17:27.18 | *** part/#asterisk Holos (~chad@ip-70-38-12-204.static.privatedns.com) |
| 17:27.30 | WindBack | WIMPy: seems that * is looking for an empty extension in default context |
| 17:27.55 | WindBack | WIMPy: Looking for in default (domain 192.169.0.2:5060) |
| 17:38.44 | joeyjones | WIMPy: have you played with encryption for * much? |
| 17:40.29 | WIMPy | I've got TLS active. SRTP was working as well, if only for a certain amount of time. |
| 17:43.15 | joeyjones | WIMPy: what * version? |
| 17:43.46 | WIMPy | 1.8.0 to TRUNK |
| 17:44.12 | WIMPy | TLS has been working with 1.6 as well. |
| 17:44.30 | joeyjones | iirc i'm running 1.6 |
| 17:44.55 | joeyjones | yeah, 1.6.2.9-1~bpo50+3 |
| 17:46.27 | WIMPy | Should be enough for TLS. |
| 17:49.01 | joeyjones | WIMPy: i'm guessing it's a sompile option, not a config option though... |
| 17:49.07 | joeyjones | *compile |
| 17:50.00 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:50.08 | WIMPy | I didn't have to select something manually, but there will surely be dependencies. |
| 17:50.26 | joeyjones | WIMPy: how did you enable it? |
| 17:50.49 | WIMPy | Generate a certificate and set it up in sip.conf. |
| 17:53.08 | lordvadr | Is there a way to get cdr_odbc to store to two different databases? |
| 17:54.15 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 17:55.15 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 17:59.05 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:05.09 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 18:05.17 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 18:05.37 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 18:11.56 | *** join/#asterisk b_d (~brian@mail.bsg.bm) |
| 18:12.08 | b_d | hi all |
| 18:12.50 | b_d | can anyone point me in the direction of some examples of how I can present an ivr to the callee? ie. active call acceptance |
| 18:13.16 | p3nguin | Use Playback() and/or BackGround(). |
| 18:13.25 | Qwell | p3nguin: callee.. |
| 18:13.26 | WIMPy | What about google? |
| 18:13.41 | WIMPy | There should be tons of examples. |
| 18:13.45 | WIMPy | oh |
| 18:13.47 | b_d | tried, maybe I'm searching for the wrong thing? |
| 18:14.05 | WIMPy | Called how? |
| 18:16.22 | b_d | well it is like a courtesy phone, but when it goes off-hook, it is automatically dialing an IVR script (phpagi), then the call is place from there, once the callee picks up, I want to do "Do you wish to accept, Press 1 yes, Press 2 no" |
| 18:20.38 | *** part/#asterisk jpmcallister (~jpmcallis@200.242.28.231) |
| 18:20.43 | b_d | currently I am using "originate" from within phpagi and then bridging |
| 18:27.21 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 18:32.30 | b_d | <PROTECTED> |
| 18:35.13 | *** join/#asterisk oelewapperke (wapper@85-158-215-1.powered-by.benesol.be) |
| 18:35.21 | *** join/#asterisk astuser169 (~astuser16@mail.falconsteel.com) |
| 18:35.21 | oelewapperke | how can one see transitions in the dialplan in asterisk 1.6 ? |
| 18:35.34 | p3nguin | core set verbose 4 |
| 18:35.45 | p3nguin | Then makes calls and watch the CLI. |
| 18:35.45 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 18:36.05 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 18:36.29 | b_d | no ideas on the active call acceptance? |
| 18:36.38 | astuser169 | can anyone tell me why the freepbx gui timesout on a fresh asteriskNOW install? i do nothing other than install take out disc and goto gui and it keeps timing out when u click on different tabs |
| 18:36.53 | p3nguin | b_d: You could use the FollowMe() code. |
| 18:37.11 | b_d | FollowMe(), ok, I will check it out |
| 18:37.22 | oelewapperke | p3nguin: any idea how I could debug this: phone sends SIP REFER to do a "blind transfer" and asterisk just hangs up |
| 18:37.46 | oelewapperke | if "core set verbose 4" works it's not even hitting the dialplan |
| 18:37.51 | p3nguin | b_d: The part about the phone dialing by itself is dependent upon one of two possible choices: either the phone can be configured for auto-dialing, or the channel driver has to support hotline. |
| 18:38.18 | jsidhu | i've got an extension that says the user in on the phone. Seems like it thinks the extension is still on a call, looking at the output of 'core show channel SIP/100-08b17750' its sayting that its a ParkedCall on 71, Blocking in: ast_waitfor_nandfds. How do i get this to hangup? how can i just kill this call/channel? |
| 18:38.27 | jsidhu | in/is* |
| 18:39.20 | p3nguin | jsidhu: soft hangup in 1.4, or something about request channel hangup on later branches that I don't use. |
| 18:39.36 | astuser169 | anyone know why freepbx gui keeps timing out on a fresh AsteriskNOW install? |
| 18:40.00 | p3nguin | #asterisknow might be more appropriate. |
| 18:40.05 | b_d | p3nguin: that part is already sorted, I'm using an ata to autodial, and it works, just now I need to get the callee acceptance part sorted |
| 18:40.17 | astuser169 | thanks p3nguin i didn't know there was one i will go there now |
| 18:40.34 | *** part/#asterisk astuser169 (~astuser16@mail.falconsteel.com) |
| 18:41.56 | p3nguin | b_d: FollowMe does exactly what you asked about. Traditionally, a caller calls some extension, which doesn't answer, which runs FollowMe... FollowMe tells the caller "Please wait while I try to reach the person you are calling," and rings the phone on the other end. When the callee picks up, it says, "You have a call from ... Press 1 to accept ..." |
| 18:42.35 | p3nguin | Sounds like half your work is already done for you; you'll just have to modify it to do exactly what you need. |
| 18:43.03 | p3nguin | Of course, this is just one approach. |
| 18:43.52 | jsidhu | thanks p3nguin: that did it. |
| 18:45.00 | p3nguin | jsidhu: If one call is blocking a device from taking more calls, you might consider looking at your call-limit settings on the devices. |
| 18:45.31 | b_d | p3nguin: ok thanks |
| 18:54.02 | *** join/#asterisk msetim (~setim@187.59.14.84) |
| 18:54.39 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:57.37 | *** join/#asterisk jpmcallister (~jpmcallis@200.242.28.231) |
| 19:00.08 | jpmcallister | Anyone having that problem with IAX2 and asterisk 1.8.1: NOTICE[14592]: chan_iax2.c:8618 reg_source_db: IAX/Registry astdb host:port invalid - '10.1.21.127:4569' |
| 19:00.48 | Katty | ugggh |
| 19:01.18 | Katty | Qwell: ER? ;) |
| 19:03.46 | Qwell | Katty: ohai |
| 19:03.49 | *** join/#asterisk The_Boy_Wonder (~sloth@asterisk/batman-developer/dvossel) |
| 19:05.50 | Katty | Qwell: hai |
| 19:05.54 | Katty | Qwell: i need vodka :? |
| 19:05.57 | Katty | Qwell: :/ |
| 19:06.30 | Qwell | I'm fresh out :( |
| 19:07.27 | Katty | bummer :< |
| 19:09.16 | p3nguin | I only have a small amount left of Stoli and of Grey Goose. I pretty much quit drinking when I spent a week in the hospital with kidney problems. |
| 19:09.33 | Katty | i'll bring maple syrup |
| 19:15.47 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 19:16.20 | *** join/#asterisk thegoat (~thegoat@c-71-224-170-221.hsd1.pa.comcast.net) |
| 19:16.31 | jpmcallister | Is anyone having problems with Zoiper and IAX? When I made a call to a IAX peer connected via Zoiper I'm receiving that error: __auto_congest: Auto-congesting call due to slow response |
| 19:17.13 | thegoat | i am in the process of replacing my linux box with something quieter, and thinking about going to asterisk 1.8. will i be able to just pull my existing config over, or am i going to need to make massive changes? |
| 19:17.48 | Qwell | thegoat: You'll need to read the various UPGRADE.txt files in the versions you're moving over. |
| 19:22.24 | *** join/#asterisk tasca (~tasca@189.4.108.113) |
| 19:23.03 | *** part/#asterisk jpmcallister (~jpmcallis@200.242.28.231) |
| 19:25.04 | p3nguin | Hmm, I found a bug in 1.4.38 documentation. |
| 19:25.47 | leifmadsen | lies |
| 19:26.06 | n1gak | is shocked. SHOCKED I say to find errata in this documentation. |
| 19:26.15 | p3nguin | doc/localchannel.txt: line 247: exten => 201,n(hangup(),Hangup() |
| 19:26.33 | p3nguin | That makes the label "hangup(" |
| 19:26.45 | p3nguin | if it even works at all, that is. |
| 19:26.45 | leifmadsen | right |
| 19:27.43 | p3nguin | Just something I noticed. Carry on. |
| 19:31.24 | *** join/#asterisk ManWithNoName_ (~ManWithNo@200.242.28.231) |
| 19:32.25 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 19:48.33 | *** join/#asterisk Godfather_ (~godfather@79.109.251.250.dyn.user.ono.com) |
| 19:57.17 | lordvadr | Is there a way I can get the cdr_odbc to store in two different databases? |
| 20:02.56 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 20:19.46 | pabelanger | lordvadr: cdr_adaptive can |
| 20:19.50 | leifmadsen | lordvadr: just use cdr_adaptive_odbc |
| 20:19.51 | leifmadsen | pabelanger: jinx! |
| 20:20.35 | lordvadr | perfect, thank you both. I didn't even know that existed |
| 20:20.44 | leifmadsen | it's kind of a big deal |
| 20:21.24 | lordvadr | I just started using the cdr_odbc a few months ago and now that I have multiple servers, I'd like them to cross-log. I couldn't believe there wasn't a way. |
| 20:26.19 | devmod | If an endpoint hangs up during the execution of an AGI script, is the channel supposed to be destroyed immediately ? |
| 20:27.05 | p3nguin | I would think not, since the hangup extension is to be run. |
| 20:28.33 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 20:29.17 | devmod | p3nguin, would AGI get a HANGUP at least to be notified of the endpoint hangup? |
| 20:30.58 | p3nguin | I would expect something like that has to happen because extension h has to be run when a hangup by a device occurs. I don't have working knowledge of how it all happens underneath. |
| 20:31.31 | devmod | p3nguin, oh ok |
| 20:41.30 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 20:48.25 | *** join/#asterisk umay (~chris@184-99-227-195.hlrn.qwest.net) |
| 20:59.19 | *** join/#asterisk n0tk (~n0tk@216.160.42.30) |
| 21:03.07 | *** join/#asterisk kl4m (~kelam@199.21.216.30) |
| 21:03.34 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.13.166) |
| 21:12.41 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:17.36 | *** join/#asterisk Godfather_ (~godfather@241.Red-81-35-89.dynamicIP.rima-tde.net) |
| 21:17.59 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 21:25.32 | kl4m | Hi, I have a problem with calls with a particular path |
| 21:26.13 | kl4m | Incoming DAHDI call at office #1 -> Dial(IAX2/Office #2) -> Answer() -> Dial(SIP/Polycom phone #1) Polycom #1 transfers to another extension -> Dial(IAX2/back to office #1) -> Dial(SIP/That other extension). |
| 21:27.04 | kl4m | The problem is that if the call is not answer()ed and sent straight to Polycom#1, the call is fine, but if it's answer()ed before (for welcome message, menus, etc), the transfered call drops after a couple of seconds. SIP and IAX debugs show nothing unusual. Any idea what bug or special condition I could be triggering by answering it? |
| 21:28.16 | kl4m | I tried disabling or enabling the IAX transfer=yes/no settting and the sip reinvite but those calls still drop. Other calls with this setup are fine however I set up transfers and reinvites. |
| 21:29.59 | kl4m | For now I'm applying the "if it hurts DON'T DO IT" solution and skipping the menus :) |
| 21:32.58 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:39.44 | *** join/#asterisk thelusiv (~thelusiv@99-123-36-77.lightspeed.gnvlsc.sbcglobal.net) |
| 21:41.24 | *** part/#asterisk thelusiv (~thelusiv@99-123-36-77.lightspeed.gnvlsc.sbcglobal.net) |
| 21:50.14 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.7.212.69) |
| 21:56.44 | *** join/#asterisk [T]ank (~T]ank@206.71.78.158) |
| 22:00.09 | *** join/#asterisk Godfather_ (~godfather@241.Red-81-35-89.dynamicIP.rima-tde.net) |
| 22:00.09 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:00.09 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.13.166) |
| 22:00.09 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 22:00.09 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 22:00.09 | *** join/#asterisk ManWithNoName_ (~ManWithNo@200.242.28.231) |
| 22:00.09 | *** join/#asterisk thegoat (~thegoat@c-71-224-170-221.hsd1.pa.comcast.net) |
| 22:00.09 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 22:00.09 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 22:00.09 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 22:00.09 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 22:00.09 | *** join/#asterisk b_d (~brian@mail.bsg.bm) |
| 22:00.09 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 22:00.09 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:00.10 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 22:00.10 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 22:00.10 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 22:00.10 | *** join/#asterisk n1gak (~lawrence@189.162.186.156) |
| 22:00.10 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 22:00.10 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 22:00.10 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
| 22:00.10 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 22:00.10 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:00.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 22:00.10 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-lllbbbrrkwptbrkl) |
| 22:00.10 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 22:00.10 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 22:00.10 | *** join/#asterisk [netman] (~netman@110.Red-83-55-245.dynamicIP.rima-tde.net) |
| 22:00.10 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 22:00.10 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 22:00.10 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-169.telkomadsl.co.za) |
| 22:00.10 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 22:00.10 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 22:00.10 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 22:00.10 | *** join/#asterisk m_tadeu (~quassel@89.181.100.210) |
| 22:00.10 | *** join/#asterisk haakonnes (~haakonnes@137.174.34.95.customer.cdi.no) |
| 22:00.10 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 22:00.10 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:00.10 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 22:00.10 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 22:00.10 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 22:00.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:00.11 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 22:00.11 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 22:00.11 | *** join/#asterisk n0cturnal (~n0c@2403:dc00:ffff:fffc:226:18ff:fe3a:3bad) |
| 22:00.11 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 22:00.11 | *** mode/#asterisk [+o putnopvut] by niven.freenode.net |
| 22:00.11 | *** join/#asterisk rcsheets (~rcsheets@madeleine.picosecond.org) |
| 22:00.11 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 22:00.11 | *** join/#asterisk nightwalk (~null@daimon.vixel.org) |
| 22:00.11 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 22:00.11 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 22:00.11 | *** join/#asterisk Kobaz (~kobaz@64.27.7.21) |
| 22:00.11 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 22:00.11 | *** join/#asterisk brunner (~Chris@35-137.175-24.bham.res.rr.com) |
| 22:00.11 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 22:00.11 | *** join/#asterisk the_5th_wheel (~edd@webster.cybertek.co.za) |
| 22:00.11 | *** join/#asterisk jdoe (jdoe@665.be) |
| 22:00.11 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 22:00.11 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:00.11 | *** join/#asterisk akoma1s (~quassel@unaffiliated/akoma1s) |
| 22:00.11 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 22:00.11 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:00.11 | *** join/#asterisk mac-mini_ (~mac-mini@unaffiliated/macmini/x-648924) |
| 22:00.11 | *** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net) |
| 22:00.11 | *** join/#asterisk p_nguin (~pwne@cl-587.chi-02.us.sixxs.net) |
| 22:00.11 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 22:00.11 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 22:00.12 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 22:00.12 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 22:00.12 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 22:00.12 | *** join/#asterisk Silicium (~marco@mail.2am.ch) |
| 22:00.12 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 22:00.12 | *** join/#asterisk carrar (~tim@2604:5000:11:1::3) |
| 22:00.12 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 22:00.12 | *** join/#asterisk bigon (bigon@ubuntu/member/bigon) |
| 22:00.12 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 22:00.12 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 22:00.12 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 22:00.12 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 22:00.12 | *** join/#asterisk kleszcz (tick@80.54.23.230) |
| 22:00.12 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 22:00.12 | *** join/#asterisk Leppers (~Leppers@69-161-26-211.static.acsalaska.net) |
| 22:00.12 | *** join/#asterisk brookshire (mbrooks@hijacked.us) |
| 22:00.12 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:00.12 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 22:00.12 | *** join/#asterisk irrwitzer (~jjj@unaffiliated/irrwitzer) |
| 22:00.12 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 22:00.12 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 22:00.12 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 22:00.12 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 22:00.12 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 22:00.12 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 22:00.12 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 22:00.12 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 22:00.13 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 22:00.13 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 22:00.13 | *** join/#asterisk zleslie (~zleslie@li73-104.members.linode.com) |
| 22:00.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:00.13 | *** join/#asterisk blee (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 22:00.13 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 22:00.13 | *** join/#asterisk prudhvi (~nobody@nullpointer.in) |
| 22:00.13 | *** join/#asterisk nickaugust (~nick@li181-40.members.linode.com) |
| 22:00.13 | *** join/#asterisk ThoMe (tm@tm.muc.de) |
| 22:00.13 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 22:00.13 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 22:00.13 | *** join/#asterisk Corydon76-home (four@c-69-137-80-31.hsd1.tn.comcast.net) |
| 22:00.13 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 22:00.13 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:00.13 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 22:00.13 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 22:00.13 | *** mode/#asterisk [+o Corydon76-home] by niven.freenode.net |
| 22:00.40 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:00.42 | *** join/#asterisk ZX81_ (~ZX81_@202.49.106.158) |
| 22:00.45 | ZX81_ | are the mailing lists up? |
| 22:00.46 | ZX81_ | sent email this morning with subject "PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI" |
| 22:00.46 | [T]ank | i am transfering a call from my asterisk server on an extension on an avaya pbx. When i transfer from asterisk to avaya, i do not hear music on hold... when i dial the extension directly from the avaya, i do hear music on hold. The call works perfectly othersied. any suggestions? Both servers behind nat so no nat routing happening. |
| 22:00.47 | ferdna | i have a polycom phone that is forwarding my calls and the caller id displays my own phone number, is there anyway to forward calls with the caller number? |
| 22:00.47 | Naikrovek | are you transferring |
| 22:00.50 | WIMPy | ZX81_: Do you receive anything at all? |
| 22:01.00 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 22:01.15 | Naikrovek | ferdna: if you're transferring, a blind transfer will keep the callerid intact |
| 22:01.26 | WIMPy | ferdna: Ask the one you're forwarding via. Asterisk will always try to do so. |
| 22:01.36 | ZX81_ | WIMPy: looks like last message I got was 9 hours ago |
| 22:01.41 | ZX81_ | [asterisk-users] Call sip:user@domain.com? |
| 22:01.51 | ZX81_ | 6 minutes before midnight |
| 22:01.54 | Qwell | ZX81_: I can confirm that |
| 22:01.58 | Qwell | the 9 hours thing |
| 22:02.14 | WIMPy | ZX81_: Err, on the interface |
| 22:02.23 | ZX81_ | hmm ok, guess it's just teething with the new servers |
| 22:02.55 | Qwell | it may be queuing stuff. I recently got messages to svn-commits, from hours ago |
| 22:03.03 | ZX81_ | sweet |
| 22:03.08 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:03.21 | [T]ank | mass quit? wow |
| 22:03.39 | Naikrovek | netsplit |
| 22:05.07 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 22:05.49 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 22:06.20 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:06.39 | *** join/#asterisk Woody4286 (~Woody2143@machine76.Level3.com) |
| 22:07.31 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.13.166) |
| 22:12.32 | timholum1 | does anyone know of a way to do a test call for a video call? |
| 22:12.40 | *** join/#asterisk tasca (~tasca@189.73.91.88) |
| 22:13.41 | *** join/#asterisk ZX81_ (~ZX81_@202.49.106.158) |
| 22:15.17 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 22:17.07 | ferdna | Naikrovek, i am just pressing the forward button on the phone... |
| 22:17.58 | Naikrovek | k |
| 22:18.01 | Naikrovek | dunno then |
| 22:19.58 | [T]ank | i am transfering a call from my asterisk server on an extension on an avaya pbx. When i transfer from asterisk to avaya, i do not hear music on hold... when i dial the extension directly from the avaya, i do hear music on hold. The call works perfectly othersied. any suggestions? Both servers behind nat so no nat routing happening. |
| 22:20.12 | *** join/#asterisk digilink (~digilink@irc.stephennet.net) |
| 22:20.31 | [T]ank | i have options tT set as wekk |
| 22:20.35 | [T]ank | as well |
| 22:21.03 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 22:22.11 | Naikrovek | aw hell SGU is cancelled |
| 22:22.15 | Naikrovek | ^ offtopic |
| 22:22.27 | Naikrovek | is bummed out, now. |
| 22:25.08 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 22:26.06 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 22:27.14 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 22:34.50 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-plylsxwyzzbbkagf) |
| 22:34.51 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 22:39.54 | *** join/#asterisk jthurman42 (~jthurman@66.154.131.25) |
| 22:42.36 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 22:46.17 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 22:48.35 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 22:49.38 | *** join/#asterisk tasca (~tasca@189.26.130.229.dynamic.adsl.gvt.net.br) |
| 22:56.32 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 23:00.52 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn110.78-98-245.t-com.sk) |
| 23:01.22 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 23:03.57 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 23:08.23 | citywok | is there a way to play back the audio stream of a call from a pcap? |
| 23:08.29 | [T]ank | anyone here familiar with fail2ban? I have installed it and it does not seem to be doing what is should |
| 23:08.43 | citywok | yes, i followed the config page on voip-info and it works fine. |
| 23:08.52 | p3nguin | What have you done so far? |
| 23:10.26 | [T]ank | i think i may have found the issue... i was following the same page, but i missed changing the time format. |
| 23:10.48 | [T]ank | changing that, then retrying. |
| 23:17.47 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-233-75-73.mia.bellsouth.net) |
| 23:17.59 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 23:24.18 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 23:31.30 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 23:32.26 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 23:33.14 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:33.59 | *** join/#asterisk digilink (~digilink@irc.stephennet.net) |
| 23:40.16 | *** join/#asterisk b_d (~brian@mail.bsg.bm) |
| 23:41.09 | b_d | anyone good with phpagi here? |
| 23:51.58 | *** join/#asterisk fofware (~Fabian@host134.190-31-18.telecom.net.ar) |
| 23:57.21 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
