| 00:05.24 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:16.16 | p3nguin | Hmm. That was easy. |
| 00:19.59 | *** part/#asterisk Thedr (~Thedr@59.191.225.49) |
| 00:21.22 | p3nguin | Is there any difference if I use DumpChan() before the line is answered or after it's answered? |
| 00:24.40 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 00:33.04 | Dovid | p3nguin: Why don't you try it and see ? |
| 00:35.41 | *** join/#asterisk Canabinoide (~Fred@187.120.134.214) |
| 00:35.44 | jc319 | kaldemar: I've been looking at this group functions - do you mean one can write their own bandwith check/channel switching program (or module if that's the correct terminology)? https://www.asterisk.org/astdocs/api/func__dialgroup_8c.html I've read that this feature is new to Lync, therefore I won't be surprised if there's no such packages for Asterisk yet... |
| 00:39.04 | jc319 | hey p3nguin, got my first external call today using TCP/5060 with Windows SIP client connecting from a remote site. I also found a firewall check page on an ITSP site - TCP passes, UDP fails. This falls in line with my theory that something is wrong with UDP filtering on the router, I'll try to obtain another one just to test. |
| 00:40.50 | pabelanger | jc319: nothing inherent to asterisk, but you could write something via the Dialplan or FastAGI to handle this. I'm not sure how dynamic it would be |
| 00:41.25 | pabelanger | maybe TimeOfDay routing, even FollowMe() |
| 00:44.06 | pabelanger | However, not sure why QoS would not be enough to ensure you have enough bandwidth; I guess another layer ontop of it |
| 00:56.39 | luckman212 | can anyone tell me the difference between: canreinvite=yes, directmedia=yes, directrtpsetup=yes |
| 00:59.58 | p3nguin | canreinvite is the old option, directmedia is the new option. |
| 01:00.28 | luckman212 | k. so having them both in there would = BAD |
| 01:00.46 | luckman212 | how about directrtpsetup? is that still "experimental"? (using 1.8svn) |
| 01:12.55 | jc319 | pabelanger: I have been looking into this tonight, from what I gather: QoS is good and it is all you need ONLY IF you can somehow guarantee that voip traffic from A to B will have enough WAN bandwith at all times including peak times. Should anybody start hogging bandwith, downloading heavy files FTP/HTTP/torrent etc. this normally would result in low quality of service for voip but QoS |
| 01:12.55 | jc319 | policing kicks in and fixes the situation - actually prevents it from happening in the first place. However, if it is beyond your control -say the bandwith hogging is happening outside your network- then QoS cannot do anything to improve the service. If there's an additional CAC layer implemented, then the application became situation aware and reroutes the call to more expensive but butter |
| 01:12.55 | jc319 | quality PSTN, resulting in higher cost, but less complaints. |
| 01:15.18 | *** join/#asterisk el3slave (~email@ip68-4-133-145.oc.oc.cox.net) |
| 01:17.34 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 01:25.09 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:36.37 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 01:43.20 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 01:51.38 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.26.144.adsl.dynamic.totbb.net) |
| 01:55.21 | BugKhaM | I'm using the cmd Dial with the "L(7200000)" Option to dial the Local context and , within the local context, I am using the "L(10000)" option to dial through SIP channel. |
| 01:55.51 | *** join/#asterisk sourcode (~code@ppp-58-8-124-61.revip2.asianet.co.th) |
| 01:55.55 | BugKhaM | the call isn't disconnected at 10th sec, is it normal? |
| 02:22.09 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 02:29.27 | *** join/#asterisk tengulre (~tengulre@125.71.208.16) |
| 02:29.56 | tengulre | how many request nubers per second in asterisk? |
| 02:33.04 | tengulre | SIP request |
| 02:34.57 | tengulre | anybody here? |
| 02:35.58 | din3sh | wot do u mean? |
| 02:37.40 | tengulre | din3sh, I want to knonw. asterisk as sip proxyserver's performances. |
| 02:38.30 | din3sh | that would logically depend on ur hardware/processor i guess!? |
| 02:39.30 | tengulre | unusually? |
| 02:40.52 | florz | about the same performance as asterisk as a lawn mower |
| 02:46.55 | din3sh | lol |
| 02:49.02 | din3sh | tengulre:sorry man, I dont really have a clue |
| 02:49.08 | coppice | nice subtle reference to the grass cutting algorithms in the DSP code |
| 02:49.47 | florz | *lol* |
| 02:54.45 | *** join/#asterisk lulzsec (mouse@2001:b18:4059:0:f890:613a:1502:dcd0) |
| 02:55.13 | jc319 | Slow connection time issue - I had the same today. I made some test calls, Asterisk/SIP to mobile, it connected in 52 seconds. Some more tests, there was one or two ~10 seconds and others were in 44-52 seconds range. Apart from server's CPU load what else can cause this? |
| 02:59.38 | tengulre | jc319,tks for answer. my question is , how many sip request number in asterisk per second, in a same time, max request numbers. |
| 03:09.51 | jc319 | tengulre: I have no idea but this post ( http://lists.atlaug.com/pipermail/aaug/2011-January/001137.html ) says Asterisk cannot cope with more than 15-20 per second. |
| 03:12.09 | russellb | that is way off |
| 03:12.38 | russellb | it depends on a bunch of factors, but there is no inherit limit, i've seen plenty of tests doing hundreds of call setups per second in recent versions |
| 03:13.03 | *** join/#asterisk lulzsec (mouse@2001:b18:4059:0:f890:613a:1502:dcd0) |
| 03:13.40 | pabelanger | tengulre: Also, Asterisk is not a SIP proxy |
| 03:13.52 | russellb | hi pabelanger ! |
| 03:14.13 | pabelanger | waves at russellb |
| 03:14.43 | pabelanger | I foresee good times at astricon |
| 03:15.18 | russellb | orly? |
| 03:15.36 | WiretapWork_ | jc319, dialplan setup issues? |
| 03:15.39 | WiretapWork_ | i.e. timeouts |
| 03:16.04 | WiretapWork_ | tengulre, 'how long is a piece of string' |
| 03:17.55 | jc319 | WiretapSeven: This happened during my grand opening of external connection to home Asterisk. I wonder if it's about the extra network complexity or just a temp load on the server. Dialplan is fairly basic already, but before the next test, if I setup all timeouts to say 3 seconds (and there'll be only one or just a few extensions) then I should not see anything more than 20 seconds, right? |
| 03:17.55 | jc319 | If it goes beyond that I'll think it's the network (bad setup). |
| 03:18.17 | WiretapWork_ | potentially |
| 03:18.29 | WiretapWork_ | gotta remember that you can have termination delays with mobile carriers while it finds the phone too |
| 03:19.52 | *** join/#asterisk dorphalsig (c86ac9f2@gateway/web/freenode/ip.200.106.201.242) |
| 03:19.56 | dorphalsig | Hello |
| 03:20.04 | WiretapWork_ | ~ask |
| 03:20.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 03:20.14 | dorphalsig | I have a multi-tier asterisk cluster (about 5 server via IAX) |
| 03:20.18 | jc319 | Is there any way to log termination delay on Asterisk? Like a basic script checking timestamps when request sent & reply received, calculating difference and logging (it won't be bad if emailed this too). |
| 03:20.33 | dorphalsig | and I need somebody to help me upgrade it and tweak it |
| 03:20.39 | WiretapWork_ | jc319, you can hear it, and you can see it in the logs |
| 03:20.45 | dorphalsig | for a price of course |
| 03:21.31 | jc319 | How can I hear it? When I press 'dial' on softphone nothing happens until it finalizes the connection and produces the ring sound |
| 03:21.32 | *** join/#asterisk gruvfunk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
| 03:22.27 | jc319 | (and there's so much flowing through logs, mostly in a different language so a simple time in > time out = difference calculation would help a lot...) |
| 03:23.52 | WiretapWork_ | usually there's a change in sound at each handoff |
| 03:24.04 | jc319 | I keep testing, I think 7 seconds is the standard connection time, maybe it was a one-off issue. |
| 03:24.05 | WiretapWork_ | and the logs are not in that foreign a language |
| 03:24.15 | WiretapWork_ | set verbose to 4, set debug to off |
| 03:24.19 | WiretapWork_ | watch the logs |
| 03:24.25 | WiretapWork_ | they'll stop whjile it waits for things |
| 03:24.56 | jc319 | OK testing now |
| 03:25.52 | dorphalsig | Hello. I need some help upgrading properly an asterisk system |
| 03:26.09 | WiretapWork_ | dorphalsig, everyone heard you the firsttime |
| 03:26.14 | WiretapWork_ | if you didn't get a response, wait patiently |
| 03:27.09 | dorphalsig | WiretapWork_: a strategic reinforcement is useful once in a while :) as long as it doesnt flood |
| 03:27.12 | dorphalsig | :P |
| 03:30.39 | *** join/#asterisk linuxgecko (~playgroun@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
| 03:31.12 | linuxgecko | if all goes well, i may have solved my issue with @#$@#$@# iksemel |
| 03:32.14 | jc319 | WiretapWork_: Thanks, sip debug off helped a lot, it produces this message when it connects >> -- SIP/voipms_gbiz-00000033 is making progress passing it to SIP/001A6CA3696C-PB1-00000032 |
| 03:32.31 | WiretapWork_ | yep |
| 03:32.44 | WiretapWork_ | anything before that is local |
| 03:33.10 | linuxgecko | jc319: sip set debug on is useful when it's needed, and a hinderance when it's not :) |
| 03:33.28 | jc319 | Is it possible to have timestamps on each CLI output line? At the moment it produces timestamps at the top of each group (e.g. calling this number 'group' consists of dialplan exten lines + sip rtp cos mark msg + 'called xxx' + made progress passing + ...) |
| 03:34.48 | jc319 | If I can get timestamps maybe I can script it to do a test call every 15 minutes for 24 hours, then sit down and investigate the times when it takes 50 seconds to ring other party. |
| 03:34.59 | linuxgecko | FIANLLY!!!!!!! i have my connection connected :) |
| 03:35.29 | jc319 | congrats! what connection BTW? |
| 03:35.52 | din3sh | what was your issue? |
| 03:36.06 | linuxgecko | jc319: jabber connection for gtalk /gvoice calling :) everything except the jabber connection has worked for ages. |
| 03:37.20 | linuxgecko | din3sh: %@#$%# distro package of iksemel didn't have gnutls and it was not obvious that i also didnt have the openssl headers either.. took a fine-toothed grep on a tee of the ./configures to find all the issues. |
| 03:38.04 | din3sh | how much time you spent trying to fix that? |
| 03:41.48 | jc319 | I think I figured why SIP didn't work on UDP just not proven yet... My latest theory is this routers embedded voip features still run in the background passively, even if all voip stuff is turned off. Draytek voip integrated router... |
| 03:42.20 | *** join/#asterisk mKn0wt (~Taisigue@190.181.162.27) |
| 03:44.20 | dorphalsig | Hello. I need a consultant to help me upgrade properly an asterisk system and dimension correctly the call queues associated with it |
| 03:46.05 | linuxgecko | din3sh: nearly 3 days, most of it just not paying a close enough eye, and expecting things to "Just Work (TM)" |
| 03:48.25 | Dovid | dorphalsig: Try the asterisk biz list |
| 03:48.31 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-kipmaidfcwmekoto) |
| 03:48.41 | Dovid | Alex Bashelov seems to know what he is doing |
| 03:48.47 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-tudpdvfjlldkcwrm) |
| 03:49.51 | dorphalsig | Dovid: Yeah, it looks like I'll have to do that, but I'd really prefer to locate somebody here in IRC. You know to be able to have quick responses and stuff |
| 03:52.25 | gruvfunk | greetz |
| 03:53.01 | gruvfunk | can anyone here recommend a SIP provider with Toll Free numbers in Europe (Italy, Spain, France, Germany, UK) |
| 03:54.26 | jc319 | gruvfunk voip.ms is great, never had any down time or anything all my time with them. |
| 03:54.30 | jc319 | Almost 72 hours now. |
| 03:55.28 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 03:55.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:55.37 | p3nguin | In the few years that I have used them, I think I have had only one outage. |
| 03:56.13 | gruvfunk | jc319: I use voip.ms for US and Canada DID's, but Europe?? |
| 03:56.15 | jc319 | prices are good & I have been stress testing support department they seem to be customer friendly. |
| 03:59.34 | jc319 | gruvfunk: I'm in the UK, call quality is really good, made some test calls local & international and I'm happy with results. Here's is ping results: |
| 03:59.42 | jc319 | --- london.voip.ms ping statistics --- 7 packets transmitted, 7 received, 0% packet loss, time 5998ms rtt min/avg/max/mdev = 14.275/14.603/14.804/0.218 ms |
| 04:01.17 | gruvfunk | jc319: what kind of DID you do you have from voip.ms? |
| 04:01.33 | jc319 | Among many features in my todo list I have one minor issue that when I call UK numbers my caller ID seems out of standard. It is being processed as a US number I think, I am hoping that is something fixable but didn't get to it yet. Don't know if that'll be a dealbreaker for you, if not fixable. |
| 04:01.36 | gruvfunk | i'm seeking a Toll-Free or Freephone 800 in UK |
| 04:02.03 | gruvfunk | (among other countries) |
| 04:03.36 | jc319 | I have two London local numbers for intended production use but they take 1 business day to provision (voip.ms orders from a 3rd party apparently) and being so excited I couldn't wait 2 days and got two other numbers to test 1 US # and 1 iNum which is tiring to type. |
| 04:03.48 | linuxgecko | just did an utterly silly proof of concept test with my new asterisk-powered gogle-voice system :) |
| 04:04.51 | linuxgecko | i called my cell # from my cellphone, using sipdroid, and my google voice acct :) |
| 04:05.10 | WiretapWork_ | jc319, sign up for ISN |
| 04:05.41 | WiretapWork_ | then dial 020995800*404 |
| 04:05.45 | WiretapWork_ | err |
| 04:05.50 | WiretapWork_ | *1410 |
| 04:06.01 | WiretapWork_ | why the shit did I type 404 |
| 04:06.37 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 04:06.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:06.47 | p3nguin | What's on that extension? |
| 04:07.09 | p3nguin | person, recording? |
| 04:07.38 | jc319 | WiretapSeven: Does ISN stand for ITAD Subscriber Number |
| 04:07.52 | p3nguin | I have to figure out how to make calls via ISN before I can call it, or I'd simply call it and find out. |
| 04:09.05 | jc319 | hmm it appears so, found a cookbook here http://www.freenum.org/cookbook/ |
| 04:12.22 | jc319 | gruvfunk: Last time I checked they provided 0808 - not 0800. However it's the same in terms of billing (free). Some people may not know this and refrain from calling your # though. The same thing with 020 / 0203 misconceptions http://en.wikipedia.org/wiki/UK_telephone_code_misconceptions |
| 04:12.49 | WiretapWork_ | p3nguin, the it calls one of our queues, but essentially it is hold music and if you're lucky someone might answer :P |
| 04:13.13 | gruvfunk | jc319: thanks! |
| 04:14.18 | p3nguin | oh |
| 04:14.38 | p3nguin | I guess I'll configure my system to call ISN eventually. |
| 04:19.08 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 04:20.22 | linuxgecko | WiretapWork_: beacuse 404 is a common error number stuck in your hear??:) |
| 04:20.28 | linuxgecko | s/hear/head/ |
| 04:21.19 | *** join/#asterisk pcrane (~openbts@13.240.69.111.dynamic.snap.net.nz) |
| 04:21.25 | WiretapWork_ | linuxgecko, all my agents have extensions that are HTTP error codes :P |
| 04:21.34 | WiretapWork_ | little bit of humour for the ISP industry |
| 04:21.41 | WiretapWork_ | my partner chose 418 though |
| 04:21.45 | WiretapWork_ | apparently he's a teapot |
| 04:24.05 | pcrane | I've got a question about DUNDi and SIP MESSAGEs. I've managed to get local delivery of messages using russell's messaging branch, but would like to get them routed via DUNDi. I have calls working via DUNDi (and SIP trunks). Does anyone know much about this? Or am I going to have to figure it out for myself? |
| 04:28.12 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 04:35.06 | gruvfunk | jc319: thanks again, I never realized voip.ms had so many International Toll Free DID's available |
| 04:35.26 | gruvfunk | Now I just need Toll Free DID's in Germany and Italy -- anyone ?? |
| 04:39.29 | tzanger | gruvfunk: yeah voip.ms is really taking off nicely |
| 04:39.33 | sawgood | toll free in another country? |
| 04:39.38 | sawgood | well, outside the USA anyways? |
| 04:39.45 | tzanger | I just switched my international to them due to a policy change at unlimitel |
| 04:40.01 | gruvfunk | sawgood: correct, toll free for callers in that country |
| 04:40.14 | p3nguin | wiretapwork_: That number is "not in service." |
| 04:40.23 | sawgood | Oh .. in that country ... not for a USA caller to call the toll free number in "Germany" for example |
| 04:40.29 | p3nguin | That didn't take too long to set up ISN outbound calling. |
| 04:40.30 | WiretapWork_ | you dialled 995, its 955 :P |
| 04:40.31 | gruvfunk | right |
| 04:40.37 | p3nguin | hmm |
| 04:40.39 | sawgood | nice ... |
| 04:41.00 | p3nguin | Me mistype a crazy phone number?! NEVAR! |
| 04:41.08 | WiretapWork_ | lol |
| 04:41.26 | p3nguin | I just entered what you gave me. |
| 04:41.30 | sawgood | I have this new laptop with a 17" screen and it is really neat (from my 13" old one) |
| 04:41.32 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 04:41.33 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-caixybgtujchleyg) |
| 04:41.42 | WiretapWork_ | I mistyped, oops |
| 04:41.45 | sawgood | 1440x900 resolution is nice |
| 04:41.53 | p3nguin | I'll try again. |
| 04:42.02 | WiretapWork_ | given that is going on my business cards I should probably make sure I have it right :P |
| 04:42.59 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 04:43.25 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 04:43.29 | WiretapWork_ | p3nguin, call reasonably clear? |
| 04:43.49 | p3nguin | Yeah, your buddy must have a good quality phone. |
| 04:44.07 | WiretapWork_ | SPA922 |
| 04:44.29 | p3nguin | I was going to listen to the music, and then Chris answered. |
| 04:44.41 | WiretapWork_ | hehehe |
| 04:44.44 | p3nguin | :/ |
| 04:44.56 | sawgood | Where in Asterisk (1.8.4.2) would I setup what gets 'written' to /var/log/asterisk/cdr-csv/Master.csv |
| 04:45.01 | WiretapWork_ | erm, I don't have anything set up that is hold music only IIRC |
| 04:45.21 | WiretapWork_ | I actually didn't think he was at his desk |
| 04:45.32 | p3nguin | When you said no one was likely to answer, that's sorta what I expected. |
| 04:45.44 | p3nguin | Ah well, no harm done. |
| 04:46.27 | drmessano | What sort of connector does the cloud use and where can I get a crimp tool? |
| 04:46.59 | p3nguin | sawgood: Perhaps /etc/asterisk/cdr_custom.conf |
| 04:47.10 | sawgood | p3nguin: thank you ... reading now |
| 04:47.58 | p3nguin | wiretapwork_: I guess considering that was purely sip to sip, it should be pretty good quality. We didn't have to pass through any telco switches or anything like that. |
| 04:48.20 | WiretapWork_ | p3nguin, my PBX sits on the other end of a residential ADSL link |
| 04:50.17 | *** join/#asterisk xofapcom (~chatzilla@118.96.106.27) |
| 04:50.40 | p3nguin | It certainly wasn't mangling the media. |
| 04:51.24 | WiretapWork_ | this is what happens when your ISP respects DSCP :P |
| 04:51.34 | WiretapWork_ | I run another PBX for my dayjob here |
| 04:51.37 | WiretapWork_ | also down an ADSL link |
| 04:51.41 | WiretapWork_ | with similar levels of saturation |
| 04:51.47 | WiretapWork_ | inbound voice sounds like you're under water |
| 04:51.50 | WiretapWork_ | outbound is crystal |
| 04:52.22 | *** join/#asterisk irroot (~irroot@dsl-185-122-97.dynamic.wa.co.za) |
| 04:53.01 | p3nguin | Maybe that's why he didn't understand what I was saying at first. Either that, or the idea was so strange that he couldn't figure it out. |
| 04:53.23 | irroot | top 'o the mornin |
| 04:53.59 | WiretapWork_ | I'm asking him now |
| 04:54.02 | WiretapWork_ | nope |
| 04:54.08 | WiretapWork_ | I think it was an unexpected accent :P |
| 04:54.17 | p3nguin | That could do it, too. |
| 04:54.44 | WiretapWork_ | he said the voice qual was perfect, which I'm glad about |
| 05:00.19 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 05:00.42 | chrisjunkie | right, where's Rob :P |
| 05:00.48 | p3nguin | Here! |
| 05:00.59 | chrisjunkie | haha twas me who answered your call |
| 05:01.35 | p3nguin | wiretapwork_ said no one would answer if I called to test the number. :) |
| 05:01.44 | WiretapWork_ | I said it was unlikely :P |
| 05:01.44 | p3nguin | He underestimated you. |
| 05:01.54 | WiretapWork_ | I think my exact words were 'if youre lucky maybe a person' |
| 05:01.55 | WiretapWork_ | :P |
| 05:02.01 | p3nguin | I was lucky! |
| 05:02.21 | p3nguin | That was my first successful ISN call. |
| 05:02.30 | p3nguin | The first actual call was to a wrong number. |
| 05:02.45 | WiretapWork_ | I believe leifmadsen has a fun song you can hear if you call his ISN test number :P |
| 05:02.45 | p3nguin | (through no fault of my own, might I add) |
| 05:02.56 | p3nguin | Is it his Polycom song? |
| 05:02.59 | WiretapWork_ | yeh |
| 05:03.24 | p3nguin | I called it through a regular SIP URI before. |
| 05:03.29 | WiretapWork_ | ah |
| 05:04.24 | WiretapWork_ | I haven't got SIP URI dialling enabled, as I don't have SIP URI inbound allowed either |
| 05:04.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 05:07.47 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:09.45 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 05:11.06 | *** join/#asterisk xofapcom (~xofap@118.96.106.27) |
| 05:12.20 | WiretapWork_ | outtahere |
| 05:13.48 | irroot | cheers |
| 05:52.16 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:57.50 | *** join/#asterisk PhoenixMage (~Phoenix@ncao.vtcif.telstra.com.au) |
| 05:58.29 | PhoenixMage | Hi guys, are there issues with the 7975G SIP 9.2 firmware and asterisk? I have had a search but cant find anything definitive |
| 05:59.55 | sawgood | PhoenixMage: have you gotten one to register? |
| 06:00.37 | WiretapSeven | PhoenixMage, whats up |
| 06:00.41 | WiretapSeven | I can help you |
| 06:00.54 | WiretapSeven | but you'll need to clarify your question |
| 06:01.42 | sawgood | my question is: "how do I add" fields to the Master.csv file in /var/log/asterisk/cdr-csv/Master.csv |
| 06:01.43 | PhoenixMage | Cant seem to get the phone to register... I am running astlinux 0.7.7-1.8.3 |
| 06:01.54 | PhoenixMage | Softphone works fine |
| 06:02.00 | WiretapSeven | PhoenixMage, you will need to enable TCP globally, and then enable TCP for the peer |
| 06:02.13 | sawgood | TCP really neat ... |
| 06:02.18 | WiretapSeven | Cisco UC phones require TCP to connect |
| 06:02.28 | WiretapSeven | PhoenixMage, also, you need to 'USECALLMANAGER' for all your lines |
| 06:02.40 | PhoenixMage | WiretapSeven: Thanks buddy will give it a go when I get home |
| 06:02.48 | WiretapSeven | PhoenixMage, incoming link |
| 06:02.57 | WiretapSeven | http://www.wiretap.net.nz/asterisk-stuff/cisco-unified-ip-phones-on-asterisk/ |
| 06:03.01 | WiretapSeven | its incomplete |
| 06:03.06 | WiretapSeven | but contains enough to get the phone up and running |
| 06:03.17 | PhoenixMage | WiretapSeven: Kiwi huh? |
| 06:03.24 | WiretapSeven | yep |
| 06:03.39 | PhoenixMage | do much in the security scene? |
| 06:03.50 | WiretapSeven | enough |
| 06:04.10 | WiretapSeven | I'm more of a RS/V guy but I like my ASAs :P |
| 06:04.56 | sawgood | I like building my own wire/cable tap boxes (passive ones) |
| 06:04.58 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:05.55 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:05.57 | schmidts | good morning |
| 06:08.39 | kleszcz | morning |
| 06:09.13 | irroot | ok who has been having problems with MWI notification on 1.8 ?? please see ASTERISK-18002 ASTERISK-17866 |
| 06:12.56 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 06:19.32 | PhoenixMage | I wish the wireless 7925 had a SIP firmware :-/ |
| 06:21.50 | irroot | http://baldy.posterous.com/more-madness <- Al-Gegra Agent arrested in Johannesburg |
| 06:21.59 | p3nguin | Several people wish that. |
| 06:22.00 | irroot | http://baldy.posterous.com/more-madness <- Al-Gebra Agent arrested in Johannesburg |
| 06:22.55 | p3nguin | If you don't have to use Asterisk 1.8, chan_sccp-b works pretty well with the 7925. |
| 06:24.40 | PhoenixMage | p3nguin: 1.8 doesnt have sccp support? |
| 06:24.50 | p3nguin | None that's worth a shit. |
| 06:24.52 | WiretapSeven | does |
| 06:24.55 | WiretapSeven | but poor |
| 06:24.58 | WiretapSeven | chan_skinny is rubbish |
| 06:25.07 | WiretapSeven | chan_sccp-b is not supported yet cause the devs are lazy |
| 06:25.16 | p3nguin | chan_skinny barely works in it, and I can't get chan_sccp-b to build against it. |
| 06:25.36 | WiretapSeven | apparently there's a special non-released dev branch that compiles against it |
| 06:25.40 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 06:26.04 | p3nguin | I saw they added 1.8.0 to the supported versions, but the downloads page does not show it. |
| 06:26.48 | PhoenixMage | oic |
| 06:26.57 | WiretapSeven | that and we're up to 1.8.4.1 now |
| 06:26.58 | p3nguin | I'll stick to what's working for as long as I can possibly stand it. |
| 06:27.02 | PhoenixMage | chan_sccp-b is more fuly featured? |
| 06:27.14 | WiretapSeven | chan_sccp-b supports nearly all the cisco features |
| 06:27.33 | PhoenixMage | oic |
| 06:27.35 | p3nguin | I don't know what all chan_skinny is supposed to have, since it almost doesn't work. |
| 06:28.00 | WiretapSeven | basically nothing |
| 06:28.09 | WiretapSeven | its only feature is that you can connect phones to it |
| 06:28.27 | p3nguin | That's all I managed to get working. |
| 06:29.07 | PhoenixMage | I am using 1.8.3 as the astlinux site said there are issues with 1.8.4 and 79xx registration |
| 06:29.39 | p3nguin | I saw mention of that in here the other day. |
| 06:29.47 | WiretapSeven | PhoenixMage, applying Gareth's patch, as on my website, will fix any issues with 79xx and 1.8.4 |
| 06:30.01 | PhoenixMage | Funny since I had issues with 79xx registration on 1.8.3 which I will hopefulyl resolve with WiretapSeven's page this evening |
| 06:30.04 | WiretapSeven | I have a couple of 7912s, and a 7970 registered to * 1.8.4.1 |
| 06:30.11 | PhoenixMage | cool |
| 06:30.19 | WiretapSeven | the patch you definitely want btw |
| 06:30.24 | WiretapSeven | BLF is a great feature |
| 06:30.45 | PhoenixMage | Not even sure what that is, new to the voip world |
| 06:30.47 | p3nguin | That's one of the things I like about my sccp. |
| 06:31.00 | p3nguin | indicates someone is "on the phone" or not. |
| 06:31.01 | WiretapSeven | come to think of it, I also have a bunch of 7911s registered to 1.8.4.1 at $dayJob |
| 06:31.30 | PhoenixMage | ah ok |
| 06:31.32 | WiretapSeven | p3nguin, I get ringing, outbound, inbound and offline notifications |
| 06:31.36 | PhoenixMage | sounds like a useful feature |
| 06:31.52 | PhoenixMage | Can you register an extension to multiple phones? |
| 06:32.03 | WiretapSeven | a peer, you mean |
| 06:32.08 | p3nguin | No, because that's not how things work. |
| 06:32.10 | PhoenixMage | yes sorry |
| 06:32.10 | WiretapSeven | no, SLA is not supported |
| 06:32.25 | WiretapSeven | if you have multiple peers assigned to the same user that will work |
| 06:32.39 | WiretapSeven | but SLA is ancient stuff from the days of POTS PBX |
| 06:32.46 | PhoenixMage | Was just thinking it would be nice if someone tried to call my deskphone and my iphone would ring |
| 06:32.49 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 06:32.58 | PhoenixMage | and I could answer on either |
| 06:33.07 | WiretapSeven | PhoenixMage, see above about multiple peers on same user |
| 06:33.17 | PhoenixMage | thanks again |
| 06:33.17 | WiretapSeven | you could also use ringgroups |
| 06:33.24 | p3nguin | You can always create a peer for both phones. |
| 06:33.25 | PhoenixMage | I miss helpful freenode chans :) |
| 06:33.36 | WiretapSeven | I think SCCP-B supports multiple registrations for one line though |
| 06:33.44 | WiretapSeven | in the SLA behaviour |
| 06:33.48 | p3nguin | It does. |
| 06:33.49 | PhoenixMage | Will look into it, thanks guys |
| 06:33.55 | kaldemar | PhoenixMage: you can make an extension dial multiple devices, but the one that does not answer gets an unanswered call on it. |
| 06:34.11 | WiretapSeven | p3nguin, does it support multi-level priority presence? |
| 06:34.39 | p3nguin | I don't think so. You'd have to use ring groups and build that into dial plan. |
| 06:34.58 | WiretapSeven | eh |
| 06:35.05 | WiretapSeven | MLPP is differing call priorities |
| 06:35.20 | WiretapSeven | i.e standard, important, critical, etc |
| 06:35.45 | p3nguin | Perhaps I don't know what it does. Based on the name of the technology, I would expect it can be built with dial plan. |
| 06:36.09 | WiretapSeven | it would, however its an SCCP feature supported by the colour phones |
| 06:36.29 | WiretapSeven | allows differing priority levels to be handled differently at the handset in terms of alerting/takeover |
| 06:36.49 | WiretapSeven | (the top priority level will disconnect lower priority calls to come through, its a military-intended feature) |
| 06:38.30 | p3nguin | I'm only running legacy phones, so if they don't support the feature, I haven't tried to get it working with the channel driver. |
| 06:39.00 | *** join/#asterisk xofapcom (~xofap@118.96.106.27) |
| 06:39.04 | PhoenixMage | So does chan_sccp-b have more features then sip? |
| 06:39.13 | p3nguin | more than sip, yes |
| 06:39.30 | p3nguin | SIP on these phones, that is. |
| 06:39.46 | p3nguin | SIP on other phones, that's hard for me to say. There are lots of SIP features. |
| 06:40.06 | PhoenixMage | I have 2 or 3 if I pickup the one on my desk 7975G's and one 7925 |
| 06:40.10 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 06:47.23 | WiretapSeven | PhoenixMage, you must have deep pockets to afford a 7975 :P |
| 06:47.28 | WiretapSeven | they cost as much as my bloody 1841 |
| 06:53.43 | irroot | renames channel to asterisk-cisco :P |
| 06:55.59 | PhoenixMage | WiretapSeven: It was a gift :) |
| 06:56.38 | *** join/#asterisk Tim_Toady (~moi@178.128.143.44.dsl.dyn.forthnet.gr) |
| 06:56.57 | PhoenixMage | anyway, better head off, thanks for everything guys, may drop in from home later |
| 06:57.09 | *** part/#asterisk PhoenixMage (~Phoenix@ncao.vtcif.telstra.com.au) |
| 06:57.58 | WiretapSeven | irroot, cisco phones are popular for businesses |
| 06:58.17 | irroot | yeah it appears not so much here though |
| 07:11.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:15.27 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 07:25.24 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:29.11 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:39.57 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 07:41.23 | zkn | hi, is it possible to reload cli_permissions.conf through some module reload for example or is asterisk restart unavoidable there? |
| 07:53.03 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 07:55.30 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-87-253.zg3.cable.xnet.hr) |
| 07:59.53 | *** join/#asterisk Sertys (~sertys@vps121.webconnect.bg) |
| 08:02.12 | *** join/#asterisk wonderworld (~ww@port-92-201-82-36.dynamic.qsc.de) |
| 08:03.19 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 08:05.48 | l2trace99 | anyone know how I can reset the inbound leg of a sip call ? |
| 08:06.24 | *** join/#asterisk knorkeknie (~hans@p5496D489.dip.t-dialin.net) |
| 08:13.04 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 08:18.00 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 08:24.16 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:29.53 | zkn | aah got it |
| 08:34.39 | hetii | irroot, hi :) |
| 08:34.49 | irroot | hi there |
| 08:36.45 | hetii | irroot, i have such issue, when i send mail to my server that are transformed by faxmail application to fax, the endpoint recive it but when it is a mail with html tags, the result is a two page, the first one looks right and the secong page is a html content |
| 08:37.13 | *** join/#asterisk MariusAgon (~aa@79.142.116.89) |
| 08:37.16 | hetii | is it possible to set somehow to don`t attache the html content ? |
| 08:38.24 | irroot | that is why i do not do mail to fax :P its always tricky |
| 08:38.50 | irroot | use a script wrapper to pull and dump the html ?? |
| 08:44.51 | hetii | if its only way |
| 08:45.10 | hetii | you know i just test now what i can do with it :) |
| 08:46.47 | hetii | other question is where can i set some flag thati will got mail about each attempts, no i just got only when all attempts was done. |
| 08:52.17 | irroot | you can do that in the dialplan use a system command or agi script ?? |
| 08:56.00 | hetii | im talk about hylafax |
| 09:01.18 | hetii | so * do call to my IAX modem on this case and hylafax start process it. Then when file try 12 time deliver fax to endpoint and after that send mail |
| 09:02.10 | hetii | imho its more hylafax part then * |
| 09:02.24 | irroot | ah the hylafax is bit beyond me simple support |
| 09:02.33 | irroot | i get it running and leave it |
| 09:03.35 | *** join/#asterisk Tim_Toady (~moi@178.128.143.44.dsl.dyn.forthnet.gr) |
| 09:05.45 | *** join/#asterisk din3sh (~din3sh@41.136.100.32) |
| 09:07.10 | pecenipicek | irroot, i've figured the problems i've had few days ago with getting the distrotech-customers-1.8 stuff to compile. never ever copy stuff from windows when you're not sure your thrice damned scp client will convert the files properly. |
| 09:07.15 | hetii | yep, i know what you mean some years ago i work with it set up and forgot that even exist, now i do everything from scrath and i realize that i almost forget everything :) |
| 09:11.49 | irroot | lol @ pecenipicek indeed |
| 09:12.01 | irroot | rule no 1 dont use windows :P |
| 09:12.09 | pecenipicek | hah. |
| 09:12.17 | pecenipicek | windows good for what i need. |
| 09:12.37 | hetii | windows is not good for anything expect games :) |
| 09:12.43 | irroot | refrains from comment |
| 09:13.21 | pecenipicek | hetii, 3D work counts in a windows bonus as well. |
| 09:13.23 | pecenipicek | :p |
| 09:13.36 | *** join/#asterisk ketema (~ketema@ketema.net) |
| 09:13.46 | pecenipicek | but yeah, if the stuff i do could be done proper on linux, i'd have switched 100% years ago.ž |
| 09:13.56 | pecenipicek | alas, it is not so. |
| 09:14.30 | irroot | windows is good for facebook and patience :P improves receptionist retention |
| 09:15.05 | pecenipicek | until wine can run the apps i need properly, i'll be staying on windozer :p |
| 09:16.45 | cneb3000 | irroot: lol :) |
| 09:17.21 | jc319 | I think linux/*nix clones still take a lot of time to work with - probably not if all you need is 1 browser, 1 mail client, 1 office suit but if you need several new applications every day, it takes a lot of system administration time to keep things going. packaging systems help a lot but it's still not totally solved, yet. |
| 09:17.38 | irroot | pecenipicek give me couple seconds im commiting the MWI fixes to my branch will need em to have mwi work on 1.8 |
| 09:17.47 | pecenipicek | dont need mwi. |
| 09:17.56 | irroot | ah ok then |
| 09:18.16 | pecenipicek | but there appears to be a problem with res_musiconhold.so |
| 09:18.27 | pecenipicek | or my config files. |
| 09:18.28 | pecenipicek | sec. |
| 09:18.33 | irroot | was not me :P |
| 09:19.16 | pecenipicek | derp. |
| 09:19.37 | irroot | im a bad bugger the MWI patch had techies scratching there heads on site yesterday had a fix in place over nifght so was all working this AM now they confused more |
| 09:19.51 | irroot | all they need to do is google / look at JIRA |
| 09:19.56 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-csmncavpiwklplml) |
| 09:19.59 | pecenipicek | heheh |
| 09:21.11 | pecenipicek | let us see now if these silly p |
| 09:21.16 | pecenipicek | *phones shall work. |
| 09:21.22 | pecenipicek | oh. damn. need to regen certs. |
| 09:21.27 | pecenipicek | derpity derp derp derp. |
| 09:28.16 | jc319 | Someone is trying to register with my server with the wrong password every 10 minutes for the last 12 hours or so, initially it seemed cute and flattering to see someone trying to crack my little server but not any more. Is there a way to see what password they are trying? |
| 09:29.44 | florz | SIP? no |
| 09:29.49 | WiretapSeven | jc319, time for fail2ban |
| 09:30.08 | florz | fail2ban? wtf? |
| 09:32.09 | jc319 | WiretapSeven: Thanks checking out |
| 09:32.18 | WiretapSeven | be warned |
| 09:32.21 | WiretapSeven | fail2ban is a hack |
| 09:32.21 | cneb3000 | jc319: You MIGHT be able to catch it in a sip trace?... |
| 09:32.22 | WiretapSeven | but it does work |
| 09:32.35 | florz | cneb3000: no |
| 09:32.54 | florz | and fail2ban doesn't work and is completely pointless in this case in particular |
| 09:33.08 | ectospasm | No, the password will be MD5 hashed |
| 09:33.12 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-84-36.zg3.cable.xnet.hr) |
| 09:33.36 | pecenipicek | irroot, did you folks get anything of a similar in your logs? |
| 09:33.37 | pecenipicek | [Jun 15 11:33:13] WARNING[12439]: chan_sip.c:3346 __sip_xmit: sip_xmit of 0x24b4d00 (len 587) to 192.168.1.246:5062 returned -2: Success |
| 09:33.43 | jc319 | "Asterisk is an open source VOIP PBX. If you have your asterisk exposed to the Internet, you may see people bruteforcing for usernames and passwords; apart from the obvious security risks, this often occurs at a high rate, causing high CPU and bandwidth usage." This must be software introduction standard - very clear. |
| 09:33.58 | jc319 | florz: Why does it not work? |
| 09:34.05 | *** part/#asterisk cneb3000 (~Ben@87.127.15.113) |
| 09:34.15 | jc319 | (and yes it is SIP, Asterisk 1.8.4.2, CentOS altest) |
| 09:34.23 | jc319 | s/altest/latest/ |
| 09:34.53 | *** join/#asterisk frawd (~francois@132.Red-81-38-142.dynamicIP.rima-tde.net) |
| 09:34.54 | florz | jc319: because it uses authentication failures as a form of authentication |
| 09:34.59 | ectospasm | jc319: are all the connections coming from the same IP address (or block of addresses)? |
| 09:37.39 | jc319 | ectospasm: same IP, seems like automated (or someone doesn't have anything else to do) attempts are precisely at hh:m4:48 (sometimes he's even early and comes at 47th second, but mostly 48th). |
| 09:38.54 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-92.dsl.net.pk) |
| 09:39.23 | ectospasm | jc319: so create a firewall rule at your router to drop packets from that address. Beware, that may become very much like whack-a-mole if you're not careful, as they hop to other addresses. |
| 09:39.35 | florz | gee |
| 09:39.51 | florz | why would you do that? |
| 09:40.09 | mechbangirc | what is the difference between a sip extension and a sip trunk? |
| 09:40.30 | pecenipicek | extension is just a number/endpoint, trunk carries bulk traffic? |
| 09:40.36 | florz | it's completely pointless to spend any time on this--just let them do their "cracking" |
| 09:40.37 | pecenipicek | if i have my definitions right. |
| 09:40.37 | ectospasm | mechbangirc: yes. An extension is typically an endpoint, and a trunk is a connection to a provider |
| 09:40.59 | pecenipicek | what ectospasm said. |
| 09:41.20 | florz | as long as it's not impacting performance, it's something between a waste of effort and dangerous to do anything about it |
| 09:41.25 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 09:41.42 | ectospasm | florz: until they succeed in cracking the password. Making no assumptions about their relative strengths |
| 09:42.12 | mechbangirc | can I send multiple calls to an extension as it has an option something like call-limit? |
| 09:43.05 | florz | ectospasm: Well ... just use proper passwords? It's pretty idiotic to build an unreliable and dangerous workaround for weak passwords, isn't it? |
| 09:43.41 | ectospasm | jc319: another, effective way to mitigate this problem is to change the port on which Asterisk is listening for SIP connections, from 5060 to something else. That shouldn't be dangerous to set, but it will take work to configure all clients to use the new port. |
| 09:43.49 | florz | I mean, that's the point of passwords: Discerning who is allowed and who is not allowed to use a service |
| 09:44.02 | florz | and they are extremely good at that |
| 09:44.12 | florz | fail2ban is not |
| 09:44.28 | ectospasm | florz: right, but if the password is only a four-digit numeric code, it's not strong enough. |
| 09:44.49 | ectospasm | ...like I said, I make no assumptions about the relative strengths of the passwords here. |
| 09:44.58 | florz | ectospasm: well, that's why you don't use four-digit numeric codes as passwords? |
| 09:45.21 | pecenipicek | oh wth? |
| 09:45.24 | mechbangirc | and how does asterisk differentiate b/w an extension and a provider |
| 09:45.36 | jc319 | ectospasm: as an immediate solution I already did that now but as you know it is fairly easy to portscan and discover the service port so in the long run it will help only by keeping too-lazy-to-portscan bruteforcers away. |
| 09:45.39 | ectospasm | jc319: florz makes a good point, make sure your SIP passwords are strong |
| 09:45.44 | pecenipicek | okay, call from one phone to the other, when i hang up on the other, it doesnt kill the call. |
| 09:46.12 | florz | and just forget about firewalling or any of that |
| 09:46.15 | ectospasm | jc319: well, yes, but they won't be able to identify the SIP port unless they do a CONNECT scan, which should be blocked at the firewall |
| 09:46.34 | ectospasm | ...unless they just happen to guess the correct new port |
| 09:46.47 | florz | ports aren't passwords either |
| 09:46.51 | florz | passwords are passwords |
| 09:47.14 | ectospasm | florz: no, but they can remove most of the extraneous traffic |
| 09:47.44 | florz | well, he said something about "every ten minutes" ... which extraneous traffic?! |
| 09:48.10 | mechbangirc | ok now i got it asterisk always send calls to provider and in case of extension it would send the call to itself (self domain) right? |
| 09:48.53 | ectospasm | mechbangirc: no, not necessarily. It depends on how you've defined your dialplan |
| 09:49.25 | ectospasm | florz: I used "extraneous" as a synonym for "unwanted" |
| 09:50.18 | mechbangirc | ectospasm: I am talking about internal working. does it make sense cause i am just guessing |
| 09:51.04 | ectospasm | mechbangirc: in either case, you use the Dial() application, the destination determines whether it's an internal or external call |
| 09:51.32 | jc319 | Ok I will postpone looking into fail2ban (however I have some more questions coming next, trying to figure why do you think it won't work - it seems like it can work). I will use a non-standard port and see if this works with external clients well if it does I'll keep it that way. I will also increase password length and randomize them so one question: |
| 09:52.03 | jc319 | Two quotes from web /random sources: "In fact, I have just used the data to recalculate the Digest response according to RFC 2617 (as used in RFC 3261), and I can confirm that the SPA-3201 truncates the password to the first 40 characters." && "An Aastra9133i can take at least a 36-character password, but the cisco craps out (can't authenticate)". So what is a safe length for SIP passwords |
| 09:52.04 | jc319 | that won't cause device troubles but would 'just work'. 35-chars? |
| 09:52.12 | ectospasm | jc319: make sure you use alphanumeric and non-alphanumeric characters in the password (beware of using passwords with ';') |
| 09:53.01 | jc319 | Will do. How long would work with 7960 and 'most of the devices out there'? |
| 09:53.15 | ectospasm | jc319: anything above 20 characters is probably overkill. The difference in cracking time is thousands of years to hundreds of thousands... (-; |
| 09:53.27 | ectospasm | (or more) |
| 09:53.38 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 09:53.39 | ectospasm | assuming truly random passwords |
| 09:53.50 | florz | yeah, 20 characters random alphanumeric is probably a good length |
| 09:54.01 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 09:54.23 | florz | a bit shorter may be sufficient if you need to type them manually or something |
| 09:54.32 | jc319 | OK then, since I am never gonna manually type them, 30 chars would be safe and good, cool. |
| 09:55.45 | ectospasm | jc319: you can even set md5secret to hash your passwords so it makes it even that much more unlikely that it can be cracked. |
| 09:55.58 | ectospasm | jc319: remember that no security is absolute, no security is foolproof |
| 09:56.09 | ectospasm | ...and perfect is the enemy of good enough |
| 09:56.31 | florz | 20 chars password is foolproof, as far as remote cracking is concerned |
| 09:57.03 | florz | and md5secret doesn't add any security at all when passwords aren't shared with other services |
| 09:57.21 | mechbangirc | ectospasm: why cant I give host=xxx.xxx.xxx.xxx for an extension definition in sip.conf |
| 09:57.38 | ectospasm | mechbangirc: you should be able to, why do you say it can't? |
| 09:58.00 | mechbangirc | I read host=dynamic for extensions |
| 09:58.40 | mechbangirc | should i try? |
| 09:59.03 | ectospasm | that's for endpoints that get their IP address via DHCP and may not have the same IP from lease to lease |
| 09:59.17 | ectospasm | that, or remote endpoints that may not always have the same IP address |
| 09:59.39 | ectospasm | mechbangirc: it never hurts to try anything out in Asterisk, within reason of course (-; |
| 09:59.44 | jc319 | ectospasm: Yes that's sure, I just want to make it as secure as practical. I'll test 30 char passwords if it works, I'll keep it. The same with using a non-standard high port number. However I still think at some point in time someone might discover the port and start cracking, if they're lucky they will be successfull in the end. To help prevent that, I want to see if there is anything I |
| 09:59.44 | jc319 | can do that takes 10-20 minutes to set up when I set up the server. I am willing to spend another say 20 minutes to increase security greatly. |
| 10:00.56 | ectospasm | jc319: don't expect open source Asterisk to be a turnkey solution. It took me two weeks to get a dial tone, and then another week to make and receive phone calls. That was pre-1.0 days |
| 10:01.05 | florz | jc319: a random 20-char password is impossible to crack |
| 10:01.29 | pecenipicek | jc319, you are using TLS+SRTP then for SIP phones, amirite? |
| 10:01.29 | ectospasm | I wouldn't expect to have a fully functioning Asterisk system within an hour, even though I'm pretty well versed in this stuff. |
| 10:01.44 | florz | jc319: just forget about adding any protection in addition, you can't get better than uncrackable |
| 10:02.03 | irroot | there was a article on /. recently where GPU's are bruteforcing passwords in "record" times |
| 10:02.03 | jc319 | How can you say that, it is 'random' what if they're lucky. This is like the 14 guys serving lifetime all over the world for falling into the 1% range in DNA tests. |
| 10:02.21 | florz | jc319: if you want to worry, worry about vulnerabilities in asterisk, not about cracked 20-char passwords |
| 10:02.55 | pecenipicek | or vulnerabilities in SIP. |
| 10:03.10 | ectospasm | irroot: can't bruteforce a password if you wait ten minutes between attempts. |
| 10:03.18 | jc319 | pecenipicek: No it's not just a basic setup, but it's in my todo list once I finish my features wishlist. I only noticed this supposedly bruteforce attempt and focused on this now |
| 10:03.37 | pecenipicek | at the very least get TLS set up. |
| 10:03.45 | jc319 | florz: Yeah that makes sense, okay maybe 30char pwd is all I need..... |
| 10:03.46 | mechbangirc | ectospasm: it works, so now no other IP can register with this username/secret to my server right? |
| 10:04.06 | ectospasm | mechbangirc: no, you want permit/deny, contactpermit/contactdeny |
| 10:04.14 | ectospasm | mechbangirc: see the annotated sip.conf.sample |
| 10:04.54 | mechbangirc | let me |
| 10:05.09 | ectospasm | jc319: you may be served well by permit/deny/|contactpermit/contactdeny as well. |
| 10:05.15 | ectospasm | s/well// |
| 10:05.21 | florz | jc319: Yes, but it's so unlikely that in every other context you would call the same probability "impossible". And there are far greater risks that you don't do anything about (being run over by a car twice a day, for example) |
| 10:06.23 | jc319 | florz: Yes yes got the point now, climbing out of my needless worry pit, cheers. |
| 10:06.30 | florz | good! :-) |
| 10:07.27 | ectospasm | jc319: and join asterisk-announce so you can be notified of any security updates to Asterisk |
| 10:07.44 | ectospasm | ...among regular, non-security updates and related announcements |
| 10:07.55 | jc319 | ectospasm: Thanks noted that in my notes>security section, I will have a look along with fail2safe and buddies if my new long random passwd + non-std port security feels not enough :) |
| 10:08.24 | mechbangirc | why contactpermit/contactdeny instead of permit/deny can a phone change its IP while still being in registered state? |
| 10:09.11 | ectospasm | mechbangirc: contact* stuff is for registration |
| 10:09.15 | jc319 | ectospasm: would not bugtraq serve the same purpose, albeit with a little delay perhaps? |
| 10:09.48 | ectospasm | jc319: I dunno how delayed bugtraq/Mitre's CVE would be |
| 10:10.09 | ectospasm | it's a low traffic list |
| 10:10.27 | jc319 | oh and before sealing the topic, I also found these on google before florz let me out :D Additional solutions: 1) For those who may want a bit of additional security, this thread on iptables rate limiting. 2) You may also want to consider adding Asterisk security through geographic IP address restriction. |
| 10:10.57 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 10:11.30 | mechbangirc | ectospasm: can you tell me a use case where some one wants to use different ip/mask for (contactpermit/contactdeny) and (permit/deny)? |
| 10:11.52 | ectospasm | mechbangirc: no |
| 10:12.27 | ectospasm | mechbangirc: permit/deny is for sending calls, contact* is for registering endpoints |
| 10:13.01 | mechbangirc | ectospasm: so which one you think I should use? or tell me should I use both |
| 10:13.37 | mechbangirc | I am referring to extensions btw not trunks |
| 10:14.06 | ectospasm | mechbangirc: read the documentation and decide which one works best for your situation. If you configure extensions to be able to send calls withouth registering, then permit/deny is your only option |
| 10:14.53 | mechbangirc | ectospasm: ok now I got it. thanks |
| 10:16.09 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-219.telkomadsl.co.za) |
| 10:16.32 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 10:16.49 | jacc0 | hi all!! |
| 10:17.22 | jkroon | using chan_dahdi, does the ISDN (PRI, E1 specifically) negotiate the companding, ie, ulaw vs alaw or is this something I need to configure on * ? |
| 10:18.05 | ectospasm | jkroon: usually you don't worry about the companding on PRI, it's configured in DAHDI or Asterisk (can't remember exactly which) |
| 10:18.42 | ectospasm | Asterisk will take the G.711 and convert it to signed linear for its internal processing, and perform transcoding to other codecs as necessary |
| 10:19.13 | ectospasm | ...same as it does for analog |
| 10:19.14 | jkroon | ectospasm, in three years this is the first time i've had to worry about. |
| 10:19.39 | ectospasm | why is it an issue now? |
| 10:20.09 | jkroon | new installation. |
| 10:20.41 | jkroon | audio sounds crap, and as a test I've done a recording from one of my SIP channels to ulaw, renamed the file to .alaw and the distortion effect sounds about similar. |
| 10:21.21 | ectospasm | what version of Asterisk? |
| 10:21.21 | jkroon | 1.6.2.17.3 |
| 10:21.46 | ectospasm | so audio quality is poor, on both the PRI and the SIP side? |
| 10:22.26 | jkroon | don't know about the PRI side, but definitely on the SIP side. (ie, on the SIP side you can definitely hear distortion, i haven't been on the other end of the link recently) |
| 10:22.50 | WIMPy | jkroon: It needs to be configured. But thinking about it, I've no clue, where it can be done. |
| 10:25.15 | ectospasm | WIMPy: jkroon: there doesn't look like any options for companding in the DAHDI drivers, and I don't know of anywhere in Asterisk that it can be set. |
| 10:26.00 | WIMPy | I can't find anything, either. |
| 10:26.09 | ectospasm | jkroon: can you test the network link for noise? Also, is there any resource contention on this system? |
| 10:26.10 | jkroon | chan_dahdi.c seems to refer to companding ... so i'll check. |
| 10:26.24 | WIMPy | So it must be implied by some other setting. |
| 10:26.31 | jkroon | ectospasm, bringing in the PRI using either a epigy gw or a quintim gw solves the noise. |
| 10:26.34 | ectospasm | jkroon: it does, but it's not an option exposed by any driver parameters |
| 10:26.37 | irroot | jkroon there is a alaw overide options when loading the drivers |
| 10:26.51 | ectospasm | irroot: but not for PRI |
| 10:27.01 | ectospasm | irroot: wait, lemme look again |
| 10:27.17 | ectospasm | irroot: yeah, not for wct4xxp or wcte12xp |
| 10:27.18 | jkroon | no there is not, modinfo wcb4xxp agrees with you. |
| 10:27.49 | irroot | yeah indeed |
| 10:27.54 | jkroon | no, no companding option in chan_dahdi either. |
| 10:28.28 | irroot | is it maybe spermcount ??? aka telkom |
| 10:29.09 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 10:29.15 | ectospasm | snorts his milk |
| 10:29.20 | ectospasm | spermcount? |
| 10:29.40 | irroot | ectospasm our local telco monopoly |
| 10:29.44 | irroot | telkom |
| 10:29.46 | WIMPy | Great. Now I feel like I don;t know why dahdi ever worked. |
| 10:30.05 | irroot | tel = count work out the reset :P |
| 10:30.14 | jkroon | WIMPy, as far as I can tell PRI actually notifies the remote peer of the companding in use. |
| 10:30.35 | jkroon | has a nasty feeling telscum is telling me ulaw but is in fact sending me alaw. or the other way round ... |
| 10:31.11 | ectospasm | jkroon: you might be able to see that in the PRI debug (but I dunno, I've never looked for that there) |
| 10:31.12 | WIMPy | jkroon: There can be a LLC IE, but that's optional and probably rare. |
| 10:31.21 | WIMPy | But certainly only on a per call basis. |
| 10:31.38 | jkroon | ectospasm, that's the plan, but I still need a way to override whatever is negotiated in order to test. |
| 10:32.37 | ectospasm | jkroon: you can try overriding it in the DAHDI source, but that is not supported (-; It can't hurt, just make backups! |
| 10:33.43 | irroot | #define AST_LAW(p) (((p)->law == DAHDI_LAW_ALAW) ? AST_FORMAT_ALAW : AST_FORMAT_ULAW) |
| 10:33.50 | WIMPy | According to redfone samples, you can configure alaw=<channellis> in chan_dahdi.conf. |
| 10:33.52 | irroot | swap em arround in chan_dahdi |
| 10:34.17 | irroot | WIMPy seems to have more elegant solution |
| 10:35.07 | WIMPy | I don't see that documented in the chan_dahdi.conf sample, however. |
| 10:35.29 | irroot | WIMPy cant find it in the source search for "alaw" |
| 10:36.21 | WIMPy | This is scary. |
| 10:37.06 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 10:37.39 | jkroon | dahdi show channel shows a default law option. |
| 10:40.32 | ectospasm | jkroon: try the alaw= option in chan_dahdi.conf like WIMPy suggested. It won't hurt to try it |
| 10:40.44 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:40.55 | jkroon | there is no alaw= option that I can find. |
| 10:42.15 | ectospasm | jkroon: if it's there, it's undocumented. |
| 10:42.28 | ectospasm | ...at worst chan_dahdi won't load with it set |
| 10:42.42 | ectospasm | jkroon: you are during a maintenance window now, right? |
| 10:44.27 | jkroon | ectospasm, no, busy preparing for that window. |
| 10:44.50 | irroot | jkroon you have a multi port card ?? maybe do a back to back loopback test ?? |
| 10:45.15 | ectospasm | irroot: no need to back to back, when a patlooptest can be performed |
| 10:45.35 | ectospasm | oh, wait, this is BRI, nevermind |
| 10:45.45 | ectospasm | back to back test is the only option right now. |
| 10:48.25 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 10:52.57 | Cadey | Hi guys, we have open sourced a project our c# dev team created for our asterisk box. Its a AMI proxy with features we needed as a business (call centre enviroment). It has 3 additional features which are tailored to a call centre (sales) enviroment. Extension Monitor, Line monitors and Call stats. Take a look if you run windows servers at your site which you could use to deploy this suite. |
| 10:53.02 | Cadey | url : http://amiproxy.codeplex.com/ |
| 10:54.50 | jkroon | ectospasm, it's PRI (E1) |
| 10:55.09 | WIMPy | It seems to go to dahdi/system.conf alaw=, ulaw= and deflaw=. |
| 10:56.18 | ectospasm | jkroon: oh, then a patlooptest is a viable test |
| 10:56.54 | irroot | thx wimpy filed in the memory bank can be usefull |
| 10:57.52 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 10:57.54 | ectospasm | irroot, WIMPy, jkroon: yep, the alaw/mulaw settings are in the annotated system.conf that comes with DAHDI |
| 10:58.15 | ectospasm | it's rarely used here, which is why I didn't immediately think of it |
| 11:00.07 | irroot | cant remember see it |
| 11:00.42 | jkroon | gotcha! |
| 11:00.44 | jkroon | thanks! |
| 11:00.45 | irroot | i use alawoveride for TDM so no need and ill be putting down a beer+ that telkom is screwing jkroon |
| 11:02.51 | jkroon | irroot, i won't take that bet. because I firmly believe you're right. |
| 11:03.47 | jkroon | alawoverride is not on option on any ISDN links. |
| 11:04.25 | jkroon | it only applies to tdm kit, and it's been deprecated: companding:Change the companding to "auto" or "alaw" or "ulaw". Auto (default) will set everything to ulaw unless a BRI module is installed. It will use alaw in that case (charp) |
| 11:05.00 | WIMPy | I've always wanted to know which options can be used with what interfaces. |
| 11:05.12 | irroot | jkroon im here in randburg can prolly scare up some resources ... yeah i have not seen your problem yet suspect i might had a BRI line the other day that was screwed |
| 11:05.18 | jkroon | WIMPy, use the "modinfo" command. |
| 11:05.49 | jkroon | hehe, it works perfectly against a Siemans PABX, an Epigy and a Quintim. |
| 11:05.50 | ectospasm | WIMPy: there's also the /sys/module/<driver>/parameters/* list, but the driver needs to be loaded |
| 11:05.52 | WIMPy | jkroon: For chan_dahdi.conf that is. |
| 11:05.54 | jkroon | problem is likely on * |
| 11:05.56 | irroot | for some reason they installed a nt-2 not nt-2a with S bus disabled |
| 11:06.04 | jkroon | ectospasm, that requires you to have the module loaded already :p |
| 11:06.14 | ectospasm | jkroon: yep, I said that (-; |
| 11:06.52 | ectospasm | jkroon: just know that Asterisk 1.6.2 is deprecated, and will only receive security updates through April 21, 2012, at which point it reaches its end of life. |
| 11:07.31 | ectospasm | ...so if you've identified a bug in Asterisk, it won't be fixed unless it's considered a security vulnerability (not likely with DAHDI) |
| 11:07.43 | ariel_ | I really hate that word deprecated, When the product is still available till it's end of life date. |
| 11:08.17 | irroot | deprecated = developers dont care or want to care about maintaining it .... |
| 11:08.21 | ectospasm | ariel_: it means it's still available, but most problems with it will not be fixed. What other word do you suggest? |
| 11:08.35 | ectospasm | the version schedule is on the wiki |
| 11:08.40 | ectospasm | (wiki.asterisk.org) |
| 11:08.43 | ariel_ | I know what it means, I did not say that, I just said I don't like the word |
| 11:09.10 | irroot | yeah but its lot better than the alternative :P |
| 11:09.11 | ectospasm | ariel_: why not? Do you have a better word? |
| 11:09.21 | irroot | FUBAR ?? |
| 11:09.38 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-nbqdehdngrkzjyff) |
| 11:10.12 | ectospasm | irroot: I prefer SOL |
| 11:10.34 | ariel_ | In the process of authoring computer software, its standards or documentation, deprecation is a status applied to software features to indicate that they should be avoided, |
| 11:10.45 | jkroon | ectospasm, trust me - currently using * 1.6.2 is causing me fewer headaches than 1.8. the less digium changes the better. so security updates only is awesome news for me. |
| 11:10.56 | ariel_ | this is not software that should be avoided yet |
| 11:11.07 | ariel_ | so the term is not applicable |
| 11:11.30 | ariel_ | and most of all our servers over 400 plus none are on 1.8 yet |
| 11:11.48 | ariel_ | no plans on it moving for some time as we develop and test before any move |
| 11:11.50 | ectospasm | won't be able to get support for it from Digium going forward. But you may not find that necessary. |
| 11:12.06 | ariel_ | actually we do |
| 11:12.13 | ariel_ | but that is a complete other story |
| 11:12.18 | ariel_ | it's not end of life yet |
| 11:12.40 | ectospasm | No, but we're encouraging all of our customers to begin their migration plans if they haven't already. |
| 11:13.27 | ariel_ | yes well encourage but it's not dropped |
| 11:14.09 | ariel_ | there is a big difference we have contacts with digium for support and they have always supported us on the software side if we have issues. |
| 11:14.10 | ectospasm | at least, engineering support for earlier versions of Asterisk is totally gone |
| 11:14.20 | ariel_ | 1.4 is lts |
| 11:14.25 | ectospasm | yes, support will help as best we can |
| 11:14.48 | ectospasm | ariel_: and 1.4 is deprecated as well. Went into security-fix only same time Asterisk 1.6.2 did. |
| 11:15.03 | ariel_ | I don't have issues with security fix only |
| 11:15.13 | ectospasm | No engineering support is a big deal, if you uncover non-security-related bugs. |
| 11:15.16 | ariel_ | it's a very bad word to be using on your own product |
| 11:15.50 | ectospasm | do you suggest a better one? |
| 11:16.14 | ariel_ | Like I said we have lots of asterisk setup and have no plan on any upgrade to them unless something comes up that we need to upgrade it |
| 11:16.39 | irroot | i self maintained 1.4 for a long time there is large community support opensource is a strange animal you welcome to fork it and run with it too |
| 11:16.55 | ariel_ | You don't see Cisco, Microsoft say those terms even when they no longer support the product |
| 11:17.07 | irroot | ariel_ beg to differ |
| 11:17.23 | ariel_ | irroot: we still have our own version off the 1.09 setup |
| 11:17.23 | irroot | microsoft has EOL on all products |
| 11:17.34 | ectospasm | I think you're reading too much into the term "deprecated." It's a succinct description of the versions' statuses |
| 11:17.38 | irroot | and they brutal with it |
| 11:17.54 | irroot | look at things like frontpage people invested heavily in it |
| 11:17.56 | ectospasm | deprecation is the phase between full support and EOL |
| 11:18.09 | irroot | we not allowed to distribute the apache mods any longer |
| 11:18.17 | ariel_ | ectospasm: it's a word that sound negative it should not be in normal use. |
| 11:19.39 | ectospasm | ...your opinion, but it wraps up the idea into one term. Again, I've asked this at least three times, do you have a better suggestion for a term we should use? It's not an offical term. The official term is "Security Fix Only", but the versions page mentions deprecated too |
| 11:20.03 | ariel_ | ectospasm: actually I don't but I will think of it and come up with one later. |
| 11:21.53 | ariel_ | bbl heading to the office. I will get back to you on that term. I also think it was a very bad move when they switched to adding core to the front of most command, it's a really bad move. |
| 11:22.47 | ectospasm | ariel_: all the core commands are those that exist when no other modules are loaded. It was a way to preface those commands in Asterisk proper. It would have been too unwieldy otherwise. |
| 11:23.19 | WIMPy | thinks it's a cleaner layout indeed. |
| 11:25.03 | irroot | i use aliases :P avoids having to retrain techies / worse checking all scripts |
| 11:26.33 | ectospasm | heh, my living the past four years has been about adapting to all the changes |
| 11:31.53 | *** join/#asterisk din3sh (~din3sh@41.136.100.32) |
| 11:33.58 | DND | hi guys. im having problem loading chan_oss.so |
| 11:34.02 | DND | it just says failed |
| 11:34.41 | DND | also im not sure what are the requirements |
| 11:35.14 | WIMPy | Are you sure, you want to use OSS? |
| 11:35.25 | DND | i wanted to dial from the cli |
| 11:35.51 | DND | as of now the server is at another office and i wanted to test the calling capabilities |
| 11:36.07 | WIMPy | chan_alsa or chan_console might be a better choice. |
| 11:36.23 | DND | which do you recommend? |
| 11:36.43 | WIMPy | uses chan_alsa |
| 11:37.55 | DND | hmm |
| 11:38.14 | DND | seems i will need a recompile |
| 11:38.40 | DND | but im using *now |
| 11:40.05 | DND | oohh its called asterisk-alsa |
| 11:40.22 | irroot | hehe im still using chan_oss with alsa compat ... :-S |
| 11:41.48 | DND | is there any tutorial on how to use alsa? |
| 11:42.23 | DND | or right after loading, i can use the dial command? |
| 11:43.56 | WIMPy | You only need to make sure, it's git the right devices. |
| 11:44.17 | leifmadsen | p3nguin: 7659*460 |
| 11:44.23 | leifmadsen | (poly*460) |
| 11:44.50 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 11:49.14 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:7c23:92b3:7e50:df74) |
| 11:49.48 | MariusAgon | When someone calls in my queue, periodic announcment starts, but since it's playing the call isn't passed through the agents. How can i solve it? |
| 12:03.37 | DND | WIMPy, it seems it still cannot load alsa |
| 12:03.38 | DND | :( |
| 12:04.17 | WIMPy | Do you have a working sound system then? |
| 12:04.31 | DND | no |
| 12:04.42 | DND | i have only the on board sound card :D |
| 12:06.20 | irroot | im using snd_dummy |
| 12:06.30 | irroot | cant hear anything but it works |
| 12:06.39 | irroot | modprobe snd_dummy |
| 12:06.49 | DND | i just need to dial to my mobile to make sure the line is working |
| 12:08.01 | DND | anyway. i'll just have to resort to plan B |
| 12:10.07 | irroot | DND the 16lb hammer ? |
| 12:10.25 | DND | no.. setting up vpn :D |
| 12:10.36 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-osnnjalpscuqopri) |
| 12:11.48 | pecenipicek | go-go minimalist asterisk! |
| 12:12.42 | *** join/#asterisk luckman212_ (~irc@2001:470:1f07:1225:7c23:92b3:7e50:df74) |
| 12:13.01 | *** join/#asterisk wesphillips (~wphill04@137.237.194.192) |
| 12:13.16 | pecenipicek | btw, irroot, thank you for linking me to your version, the SRTP compat stuff saved my ass. |
| 12:15.27 | irroot | awesome thx for the feedback ... perhaps go to https://reviewboard.asterisk.org/r/1173/ and post there and https://issues.asterisk.org/jira/browse/ASTERISK-17895 |
| 12:34.02 | luckman212 | anyone know how to make a polycom phone use a specific RTP port for media? |
| 12:34.51 | luckman212 | i've tried tcpIpApp.port.rtp.forceSend= as well as nat.mediaPortStart= in my XML configs but neither one seems to do anything, phone still uses port 2222 (default) |
| 12:39.15 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:42.59 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:45.04 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:48.41 | luckman212 | nobody? |
| 12:50.07 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 12:50.28 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 12:53.43 | *** join/#asterisk din3sh (~din3sh@41.136.100.32) |
| 12:55.28 | psilikon | luckman212, what about '.mediaPortRangeStart'? Maybe. |
| 12:56.18 | luckman212 | psilikon: that's not defined |
| 12:57.03 | psilikon | luckman212, I've had issues with webgui configuration changes overriding my xml configs on polycoms... |
| 12:57.26 | luckman212 | psilikon: yes me too, I have totally disabled the web interface on my phone |
| 12:57.35 | luckman212 | & deleted the overrides |
| 12:58.07 | psilikon | luckman212, how do you delete the overrides? By clearing local config? |
| 12:58.53 | luckman212 | i deleted <macaddress>-phone.cfg from my config FTP server |
| 12:59.07 | luckman212 | do I have to do more? |
| 12:59.59 | *** join/#asterisk blee (~blee@17.213.119.70.cfl.res.rr.com) |
| 13:00.30 | blee | Hi all, I was hoping someone could help me. I am having strange issues with asterisk not responding to certain SIP packets, namely 200OKs |
| 13:01.10 | psilikon | luckman212, Try to: reset local config, reset devices settings and maybe even format file system from the advanced settings menu on the phone. Then set the rtp directive in you <mac>.cfg file and reboot the phone. |
| 13:01.52 | psilikon | luckman212, actually I think the phone will reboot while performing each step from above. |
| 13:02.04 | irroot | blee why would asterisk respond to a valid response ?? other than turning retransmit off ?? |
| 13:02.25 | blee | it should ack the 200ok |
| 13:02.28 | blee | but it doesnt |
| 13:02.31 | *** join/#asterisk nettie (~nettie@stewie.freax.it) |
| 13:02.32 | blee | so the call keeps ringing and goes to voicemail |
| 13:02.50 | irroot | pastebin a trace please blee |
| 13:02.58 | blee | okay bear with me |
| 13:03.11 | psilikon | luckman212, I have some 601s and 650s... I love 'em but those web overrides have messed with my patience in the past. |
| 13:04.01 | blee | irroot: i have a pcap, one sec |
| 13:05.09 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 13:05.21 | luckman212 | yeah I am tearing my hair out just trying to get the phone to use the right port range for RTP, which I *swear* I had working before |
| 13:05.35 | mandla | irroot, did you get my mail? |
| 13:06.15 | irroot | indeed thx today is a bad day people all over litrally durban and all ... |
| 13:06.24 | irroot | expect a reply soonest |
| 13:06.34 | psilikon | luckman212, in my 601's sip.cfg I have tcpIpApp.port.rtp.mediaPortRangeStart="" |
| 13:06.52 | luckman212 | psilikon: that will cause it to just use the default (2222) |
| 13:08.27 | *** join/#asterisk heise2k (~rheise@static-108-16-123-66.phlapa.fios.verizon.net) |
| 13:13.02 | psilikon | luckman212, The 601 admin manual says: If set to Null, the value 2222 will be used for the first allo-cated RTP port, otherwise, the specified port will be used. |
| 13:13.33 | psilikon | luckman212, Then there is this: tcpIpApp.port.rtp.forceSend |
| 13:15.22 | luckman212 | yes I have the same thing here, but for some reason its not using the parameters I set :( |
| 13:15.58 | psilikon | luckman212, try to erase the local config on the phone. |
| 13:17.42 | jaytee | if you've got ftp provisioning working, then do a file system format on the phone to wipe out any setting set in the web interface. |
| 13:19.34 | psilikon | luckman212, I am confident that what jaytee recommended will solve you issue. |
| 13:19.38 | psilikon | your |
| 13:19.54 | luckman212 | alright.. will try that, right now I'm cleaning up some formatting in my xml files |
| 13:23.31 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:23.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:26.03 | *** join/#asterisk m_tadeu (~quassel@89-180-27-162.net.novis.pt) |
| 13:26.13 | nettie | Hi guys, I'm looking into asterisk clustering solutions for an IVR application, from what I read on the web looks like the suggested and most used setup is load-balancer->openser->dundi->asterisk_boxes this coupled of course with a replciated DB with n+1 slaves depending on the read load and n+1 app servers to handle the application logic if unix_odbc is too limited for the application. All the information I gathered are pretty old so I would like to know |
| 13:28.48 | leifmadsen | nettie: from what you've described it seems like you're on the right track |
| 13:29.02 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 13:29.09 | leifmadsen | depending on how large, you may or may not need all of those components |
| 13:32.03 | nettie | leifmadsen I see thanks for the info, I'm just trying to figure out what could be feasible technically most of the components are already avaialble in our infrastructure, I just need to glue them together and of course as you suggest figure out the needs to avoid additional efforts in terms of architecture design and maintainance |
| 13:33.47 | leifmadsen | ya, basically you'll want something like OpenSIPS for the distribution of the calls amongst the Asterisk boxes, then have your end points registered to OpenSIPS. After that, then the Asterisk boxes would be connected to a replicated database which would then access data via res_odbc and func_odbc to access the dynamic data. You make the dialplan as static as possible across the boxes. |
| 13:34.31 | nettie | loosk like the latest asterisk definitive guide has an interesting chapter on clustering I think I'll buy it |
| 13:34.45 | leifmadsen | if you need to get into things like voicemail MWI and things like that, you might need to look at scripts to trigger MWI to phones directly through the externnotify script in voicemail.conf, or make sure your OpenSIPS and Asterisk are all setup to play nice to pass those messages around. |
| 13:34.46 | nettie | leifmadsen thanks |
| 13:34.53 | leifmadsen | nettie: I wrote that chapter :) |
| 13:35.01 | leifmadsen | it's more an overview with pictures of topologies |
| 13:35.02 | nettie | leifmadsen: ouch :) |
| 13:35.36 | leifmadsen | check http://ofps.oreilly.com to check out if it does what you want, then buy a copy if you find it useful |
| 13:35.49 | nettie | thanks really. |
| 13:35.56 | leifmadsen | positive reviews on Amazon's site are always welcome :) |
| 13:36.59 | *** join/#asterisk Ruckman (~Ruckman@2001:470:7:e32::2) |
| 13:38.22 | nettie | This is "just" an IVR so I don't play to play with physical phone I think the probable customer will simply forward calls to us, sorry for the lack of informations but I have not much project visibility at this time. |
| 13:38.37 | leifmadsen | ya no worries have fun! |
| 13:38.45 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 13:38.59 | nettie | thanks again! |
| 13:44.11 | mandla | irroot, alright then man, Dankie. |
| 13:44.57 | *** join/#asterisk hc (~hc@pdpc/supporter/active/hc-e) |
| 13:45.17 | hc | hi. is there a way to let asterisk pass SIP INFO messages between two bridged phones? |
| 13:46.11 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:46.12 | hc | and/or has anyone successfully used cisco PAP2T's built in proprietary RTP cryptography together with asterisk? |
| 13:46.26 | irroot | :P Kena ka kgotso |
| 13:50.08 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 13:51.14 | mandla | irroot, you aint that bad. |
| 13:51.35 | irroot | hehe got some help ill be honnest |
| 13:55.49 | `mx | when will asteriskNow have 1.8 code? |
| 13:57.30 | mandla | mmmmh, irroot, the equivalent of zapata.conf is cha_dahdi.conf right? |
| 13:57.46 | irroot | chan_dahdi.conf indeed |
| 13:59.03 | mandla | irroot, now i have to configure channels in chan_dahdi.conf and make reference to it in extensions.conf, so i can be able to make out going calls? |
| 13:59.40 | blee | irroot: sorry I got distracted, can i give you a pcap file to pull up in wireshark? |
| 14:00.01 | irroot | blee i bit busy this side post it to pastebin |
| 14:00.08 | irroot | mandla yeah indeed |
| 14:00.38 | irroot | group the outside lines all together |
| 14:00.54 | irroot | then put the "extensions" in one by one |
| 14:00.57 | mandla | ok, will be in touch, let me do something. |
| 14:01.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:01.25 | irroot | so dialing DAHDI/g1/02711.... will call jozi |
| 14:01.43 | irroot | and DAHDI/20 calls exten 20 |
| 14:02.39 | pecenipicek | okay, if anyone can help me, i'd be grateful. how do i set up stuff dialplanwise, when i want to use a IAX line to connect two boxes, with RSA auth involved? what should the dial line look like? |
| 14:03.17 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:03.37 | pecenipicek | i currently got Dial(IAX2/<user>:${SECRET}@<server ip>/${EXTEN},30) and it fails because i didnt provide a pass apparently. |
| 14:03.38 | kaldemar | pecenipicek: iirc, business as usual but with encryption=yes in iax.conf. |
| 14:04.13 | blee | irroot: pm :D |
| 14:05.07 | pecenipicek | kaldemar, define business as usual, please :) |
| 14:05.20 | kaldemar | pecenipicek: Dial(IAX2/peer_defined_in_iaxconf/number) |
| 14:06.24 | Katty | gooooooooooooood morning! |
| 14:06.56 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:06.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:07.13 | pecenipicek | force encrypt should be dumped outta the window, amirite? |
| 14:07.35 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-181-35-206.red.bezeqint.net) |
| 14:09.42 | kaldemar | pecenipicek: only if you want to enable unencrypted calls too. |
| 14:09.54 | pecenipicek | no. |
| 14:09.55 | pecenipicek | okay. |
| 14:10.25 | pecenipicek | brb anyhow |
| 14:10.36 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:13.30 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:13.33 | Kobaz | http://www.freepatentsonline.com/y2002/0131402.html |
| 14:13.57 | Kobaz | looks like someone is trying to patent ip phone registration |
| 14:14.29 | irroot | kobaz a company who would love to patent a certain decidious fruit :P |
| 14:14.42 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 14:16.15 | timholum1 | I am wondering if there is a way to see if calls are encrypted? I am running Polycom SoundPoint 335 Using tcp for the connection, every once and a while I will see some things in my log's about tcp/tls which would be encryption I just dont know if it is about thoughts phones |
| 14:16.42 | pecenipicek | capture them packets when stuff is going around? |
| 14:16.48 | pecenipicek | calls/registrations/whatever |
| 14:17.30 | mandla | irroot, are you buzy? |
| 14:17.31 | timholum1 | so there is no "core show encrytion" or something of the sort |
| 14:18.02 | irroot | always but we friends and neighbours want some milk |
| 14:19.50 | *** join/#asterisk ipc9 (~rob@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 14:21.25 | mandla | irroot, i need to know exactly wat to put in chan_dahdi and in extensions.conf, so i could be able to call outside. |
| 14:21.47 | ipc9 | what would be the easiest way to pass music from line in on the machine running asterisk, to a voip call? |
| 14:22.03 | *** join/#asterisk din3sh (~din3sh@41.212.248.153) |
| 14:22.14 | sxpert | ipc9: there's the audio console |
| 14:22.55 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 14:23.34 | irroot | what ports you got pri and ana what numbers are they ill do a config for you |
| 14:23.45 | ipc9 | sxpert: ok, is that a web based config? or is there a command line/config file |
| 14:24.32 | sxpert | web based config ? |
| 14:24.53 | mandla | irroot, iv got BRI |
| 14:25.16 | irroot | same idea :P just need the ports |
| 14:25.22 | *** part/#asterisk mandla (~mandla@168.167.180.161) |
| 14:25.32 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 14:25.43 | ipc9 | sxpert: what should i google to know what you are recommending? |
| 14:25.55 | MariusAgon | Is it possible to call another agent in queue when announcment is playing? |
| 14:26.22 | sxpert | ipc9: http://www.voip-info.org/wiki/view/Asterisk+console+channels |
| 14:26.32 | mandla | irroot, XBUS-00/XPD-00: BRI_TE (3) Span 1 DAHDI-SYNC |
| 14:26.42 | ipc9 | sxpert: perfect, thank you. |
| 14:26.51 | mandla | irroot, does that help? |
| 14:26.54 | irroot | and analogue ? |
| 14:27.07 | irroot | lsdahdi ?? and email it perhaps |
| 14:27.15 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 14:27.30 | mandla | il pastebin it. |
| 14:27.51 | PhoenixMage | Hi guys, is there comprehensive documenation of the SEP<mac>.cnf.xml file? |
| 14:28.21 | pecenipicek | cisco phones? |
| 14:28.27 | pecenipicek | you wish :p |
| 14:28.27 | PhoenixMage | yerp |
| 14:28.39 | pecenipicek | try cisco's site, but i seriously doubt you'll have much luck. |
| 14:28.42 | pecenipicek | i know i didnt. -.- |
| 14:28.56 | PhoenixMage | I have, didnt find anything |
| 14:29.48 | mandla | irroot, http://pastebin.com/3wESrS5S |
| 14:30.54 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:30.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:31.18 | irroot | mandla looking good |
| 14:31.30 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 14:31.32 | mandla | irroot, is that all you needed? |
| 14:31.47 | irroot | so lines are 1/2 |
| 14:32.01 | irroot | phones are 18-43 |
| 14:32.23 | irroot | not 26/35 |
| 14:32.27 | pecenipicek | kaldemar, are register => blahblah@blah statements neccesary with that stuff for iax? |
| 14:33.00 | mandla | irroot, what do you mean? |
| 14:33.10 | irroot | see the outptu |
| 14:33.14 | irroot | output |
| 14:33.22 | irroot | is a map of all lines |
| 14:33.57 | irroot | 1-2 are a BRI active in asterisk |
| 14:34.22 | irroot | 13-14 are extensions to phones |
| 14:34.58 | mandla | yah but iv only two analog phones connected, yah true true, i get wat you mean now. yah its like that. |
| 14:35.18 | MariusAgon | Why everyone is ignoring me? :/ |
| 14:35.52 | mandla | MariusAgon, Yah i think its possible. |
| 14:36.12 | irroot | mandla now paste chan_dahdi.conf and extensions.conf |
| 14:36.20 | irroot | do it step by step |
| 14:36.36 | MariusAgon | I'm trying to find something about that, but failing entire day :/ |
| 14:36.46 | mandla | irroot, these are huge files. |
| 14:37.23 | irroot | mmm without the comments ?? |
| 14:37.55 | mandla | almost everything is commented out. |
| 14:38.09 | mandla | especially in chan_dahdi.conf |
| 14:38.58 | mandla | How do i copy the entire file? |
| 14:41.41 | Kobaz | is there a way to turn off the loop detection on dials? |
| 14:42.03 | Kobaz | dial(sip/localhost) Got SIP response 482 "Loop Detected" back from 127.0.0.1 |
| 14:42.10 | Kobaz | and it converts it into a local channel |
| 14:43.32 | Kobaz | maybe the 'i' option? |
| 14:44.13 | Kobaz | hmm, that drops the call |
| 14:46.32 | mandla | irroot, you still there? |
| 14:46.45 | irroot | indeed emailing you |
| 14:47.58 | irroot | its a live config you can use as template |
| 14:48.15 | *** join/#asterisk justnulling2 (~jnull@ool-4b7fd02a.static.optonline.net) |
| 14:48.58 | *** join/#asterisk dorphalsig (be939a80@gateway/web/freenode/ip.190.147.154.128) |
| 14:49.03 | dorphalsig | Hello |
| 14:49.57 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:50.23 | dorphalsig | If a call enters a queue and the agent who answers transfers it to another queue, how would that call appear in the CDR? |
| 14:50.54 | mandla | irroot, so i delete what i have in chan_dahdi.config and replace it with the one you sent? |
| 14:51.26 | Kobaz | ah, IAX2 allows loops |
| 14:51.47 | irroot | mandla look at it there extentions in there and a pri [group 1] change that to the correct signalling |
| 14:52.05 | irroot | also the contexts used may differ |
| 14:52.27 | WIMPy | Oh. Back to the first step again? |
| 14:52.31 | justnulling2 | keep getting 'Bridge technology softmix failed to setup bridge structure' any reason as to why softmix fails like that? |
| 14:52.37 | irroot | what i want is you to have the right setup here so we can get ot going |
| 14:53.19 | *** join/#asterisk cerberus_za (~coert@196-210-151-122.dynamic.isadsl.co.za) |
| 14:53.21 | dorphalsig | Hello If a call enters a queue and the agent who answers transfers it to another queue, how would that call appear in the CDR? |
| 14:55.24 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 14:55.50 | mandla | irroot, oh i thought, you had edited it looking at the info iv given you on pastebin. |
| 14:56.21 | irroot | yeah was going too but then got stuck with something |
| 14:57.18 | irroot | change the mailbox/context/channel and callerid to match your setup |
| 14:57.25 | irroot | for the extensions |
| 14:58.00 | PhoenixMage | its got me why thisdamn 7975 wont register :( |
| 14:58.09 | irroot | for the group change channels context and signalling |
| 14:58.20 | irroot | and check the other options against yours |
| 14:58.54 | PhoenixMage | cant even see anything in tcpdump |
| 14:59.18 | PhoenixMage | gets its SEP file from tftp gets the time from the ntp, then nothing :( |
| 15:01.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:02.11 | mandla | irroot, when a line starts with ; this means its commented out right? |
| 15:02.22 | irroot | yes |
| 15:03.39 | l2trace99 | anyone know of a way to reset the codec on an already established call leg ? |
| 15:04.23 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:04.48 | dorphalsig | Hello If a call enters a queue and the agent who answers transfers it to another queue, how would that call appear in the CDR? |
| 15:05.28 | dorphalsig | I mean, I have about 4 queues, but I want agents to forward callers to the appropiate queue if they call the wrong one |
| 15:05.35 | irroot | dorphalsig intresting question indeed |
| 15:05.47 | dorphalsig | how can I know how many calls were redirected? |
| 15:06.26 | irroot | dorphalsig check https://reviewboard.asterisk.org/r/1266/ |
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| 15:30.03 | mandla | irroot, now i think i understand |
| 15:30.17 | irroot | great once you get it its got |
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| 15:56.50 | Faustov | ~book |
| 15:56.50 | infobot | For more information about the Asterisk book, see ~thebook |
| 15:57.08 | Faustov | ~thebook |
| 15:57.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 15:57.08 | Faustov | ffs |
| 15:57.28 | leifmadsen | patience |
| 15:57.52 | Faustov | why not give the same info for ~book, but point to another alias |
| 15:57.54 | Qwell | who changed that, and why is there like no info now? O.o |
| 15:59.10 | russellb | infobot: forget book |
| 15:59.23 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 15:59.29 | russellb | ~book |
| 15:59.29 | infobot | For more information about the Asterisk book, see ~thebook |
| 15:59.45 | russellb | infobot: no, book is <reply> Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 15:59.45 | infobot | russellb: okay |
| 15:59.55 | russellb | there. |
| 16:00.01 | Faustov | high five |
| 16:00.59 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 16:06.06 | *** join/#asterisk disasterisk (48edd5a2@gateway/web/freenode/ip.72.237.213.162) |
| 16:06.17 | disasterisk | hi |
| 16:07.18 | disasterisk | how is everyone? |
| 16:09.08 | disasterisk | does anybody know how to prevent memory leaks in FastAGI? |
| 16:09.28 | leifmadsen | Qwell: I changed it |
| 16:09.37 | leifmadsen | because it was wrong, and I hate having the same information in 100 places |
| 16:10.04 | leifmadsen | and, afaik you can't "alias" or "symlink" the same data with infobot |
| 16:10.34 | leifmadsen | russellb: you just circumvented what I was trying to prevent though :) |
| 16:11.37 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 16:12.02 | Qwell | infobot: no, book is <reply> see thebook |
| 16:12.02 | infobot | Qwell: okay |
| 16:12.09 | Qwell | infobot: no, book is test |
| 16:12.09 | infobot | okay, Qwell |
| 16:12.11 | Qwell | ~book |
| 16:12.11 | infobot | somebody said book was test |
| 16:12.12 | russellb | heh, i was going to say that |
| 16:12.13 | Qwell | infobot: no, book is <reply> see thebook |
| 16:12.13 | infobot | okay, Qwell |
| 16:12.15 | Qwell | test |
| 16:12.17 | Qwell | err |
| 16:12.21 | russellb | infobot: no, book is |
| 16:12.34 | Qwell | ~book |
| 16:12.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 16:12.37 | Qwell | there we go |
| 16:12.48 | leifmadsen | eh? |
| 16:12.51 | leifmadsen | what did you use? |
| 16:12.53 | Qwell | <reply> see symlinkgoeshere |
| 16:13.01 | leifmadsen | ah |
| 16:13.04 | leifmadsen | well there we go then |
| 16:13.16 | Qwell | /msg infobot help redirection |
| 16:13.17 | Qwell | :) |
| 16:13.28 | leifmadsen | ya I did a /msg infobot help but there was 100 commands replied back |
| 16:13.34 | leifmadsen | goes to shower, eat lunch, then return to work |
| 16:15.28 | *** join/#asterisk JerJer (~jj@asterisk/original-h323-guy/JerJer) |
| 16:15.43 | *** join/#asterisk carrar (~tim@osburn.com) |
| 16:17.30 | dont_taze_me_bro | hello everyone |
| 16:18.11 | linuxgecko | if i wanted to run lots of low-powered opensuse server installs, what's the lowest ram i could get away with? |
| 16:18.32 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 16:18.50 | dont_taze_me_bro | linuxgecko: 64MB |
| 16:19.27 | otwieracz | Are you jokeing? |
| 16:19.33 | otwieracz | It will be not usable. |
| 16:19.55 | carrar | How do you know? |
| 16:19.55 | dont_taze_me_bro | linuxgecko: how many concurrent calls? |
| 16:20.42 | otwieracz | IMO 64MB isn't enough for pure linux. |
| 16:20.54 | Kobaz | 64KB is enough for everybody |
| 16:21.06 | carrar | 64bits! |
| 16:21.08 | JonathanRose | ports Asterisk to Commodore 64 Basic |
| 16:22.53 | paulc | sighs - Ah, Commodore 64.. with its funny floppy drive.. those were the days |
| 16:22.55 | *** join/#asterisk gruvfunk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
| 16:23.00 | carrar | otwieracz, setup a box with lots of memory, install everything you wanna run, and find out how much mem it's using |
| 16:23.15 | gruvfunk | greetz all |
| 16:23.47 | gruvfunk | I'm looking for a SIP provider with Toll Free DID's in Germany and Italy - if anyone can recommend a provider, please let me know. TIA |
| 16:23.51 | dont_taze_me_bro | greetings gruvfunk |
| 16:25.21 | linuxgecko | dont_taze_me_bro: probably one per? yeah i know, one monolithic server would probably work more efficiently, but i am proposing a plan for a friend, who runs an office of independdent agents, and i doubt they want me lumping all the credentials innto one file. |
| 16:25.47 | dont_taze_me_bro | linuxgecko: try the pogoplug |
| 16:26.05 | dont_taze_me_bro | put debia squeeze on it |
| 16:26.10 | dont_taze_me_bro | *debian |
| 16:28.11 | p3nguin | There's no reason you need to have all the configs in a single file just because you only have one centralized asterisk system for several offices. See #Include. |
| 16:30.48 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 16:33.16 | pigpen | So, what is the general census on a CDR reporting web app these days? |
| 16:33.36 | l2trace99 | will sip peers renegotiate codecs with re-INVITES ? |
| 16:33.54 | pigpen | I have been using the old Asterisk CDR (from Belgium) but php is now unhappy with it. |
| 16:34.06 | pigpen | l2trace99, if necessary. |
| 16:34.58 | l2trace99 | will dialplan vars persist ? |
| 16:35.46 | l2trace99 | I guess not |
| 16:35.52 | pigpen | yes as it is still part of the dialplan. |
| 16:36.28 | pigpen | well, I would think it will. To my understanding, asterisk is still in the loop, just the media stream is direct. |
| 16:37.19 | l2trace99 | but if at exten => _5XX,1,Set(booger=$[${booger} +1]) |
| 16:37.19 | l2trace99 | exten => _5XX,n,Dial(SIP/${EXTEN}) |
| 16:37.32 | l2trace99 | booger would never increment would it |
| 16:37.33 | l2trace99 | ? |
| 16:37.55 | l2trace99 | assuming syntax was correct |
| 16:38.18 | pigpen | sorry, I am still thinking on the "booger" part. |
| 16:38.33 | pigpen | Congested today? |
| 16:38.51 | l2trace99 | somewhat |
| 16:39.06 | pigpen | the re-invite does it on the dial, not the Set |
| 16:40.23 | l2trace99 | can pass values on Dial() with out altering the exten ? |
| 16:41.41 | pigpen | in your example, the variable "booger" is not called, why why would it care? |
| 16:41.53 | pigpen | And, I think I am not following you. |
| 16:42.05 | *** join/#asterisk irroot (~irroot@197.106.195.34) |
| 16:42.27 | l2trace99 | here is what I am trying to do |
| 16:43.25 | l2trace99 | I have to answer the call with 1 codec and then pass it to meetme with another |
| 16:43.48 | l2trace99 | the call is answered via an IVR |
| 16:44.19 | l2trace99 | transfer works awesome except the carrier will not support SIP REFER |
| 16:45.00 | pigpen | Then it sounds like your asterisk box will be in the middle doing the transcoding |
| 16:45.24 | l2trace99 | my thinking is if I can send reinvite and reneg the codec I would be good |
| 16:45.43 | l2trace99 | due to licensing I can't transcode |
| 16:46.07 | pigpen | depending on the codec. |
| 16:47.50 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 16:49.15 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 16:50.39 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 16:51.07 | linuxgecko | l2trace99: how do oyu accept one codec and pass it the call on as a different codec, w/o transcoding? |
| 16:51.42 | l2trace99 | by resetting the first call leg |
| 16:51.52 | l2trace99 | it can be done via TRANSFER() |
| 16:52.14 | l2trace99 | but in this case I can't use TRANSFER |
| 16:52.39 | *** join/#asterisk cVsup (~cVsup@189.107.225.148) |
| 16:53.07 | linuxgecko | l2trace99: so you want to actually give the call to the meetme, but ti can't use the same codec as the originating call, yes? |
| 16:53.15 | l2trace99 | yes |
| 16:54.00 | l2trace99 | call comes in g729 needs to goto meetme g711 |
| 16:54.20 | linuxgecko | do you NEED to use the untranscodable codec on the originating call, or could you use the meetme-frieeendly codec to begine with? |
| 16:54.33 | l2trace99 | yes |
| 16:54.44 | l2trace99 | g729 is required |
| 16:54.49 | l2trace99 | so is the ivr |
| 16:55.00 | l2trace99 | if i wasn't calling to answer() |
| 16:55.11 | l2trace99 | then I could just set the SIP_CODEC var |
| 16:55.55 | l2trace99 | I am wondering if reinvites will reneg codec selection |
| 16:56.19 | l2trace99 | and then I would have to detect that it was re invited |
| 16:56.37 | l2trace99 | ( providing that the carrier will play along ) |
| 16:57.44 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:00.11 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 17:02.34 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:04.08 | pigpen | g729 is free as long as you don't have it transcoding to a different codec. |
| 17:04.15 | pigpen | but, you need to buy it if you do., |
| 17:04.36 | pigpen | Now, unless you have a ton of simultaneous calls, why not just buy it? |
| 17:04.45 | p3nguin | l2trace99: If you're going to do MeetMe, you cannot do reinvites to get asterisk out of the stream -- asterisk will always be in the stream providing the MeetMe conference. |
| 17:05.33 | p3nguin | pigpen: You're not going to legally obtain the codec and license without paying for it, regardless if you are going to use it to transcode or not. |
| 17:06.13 | dont_taze_me_bro | p3nguin: aren't they just as good codecs for free? |
| 17:06.29 | pigpen | p3nguin, I thought I read in the license detail that you can use it for free, as long as you don't transcode it. |
| 17:06.34 | p3nguin | dont_taze_me_bro: There is no free G.729 codec. |
| 17:06.41 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 17:06.59 | pigpen | now, it has been about a year ago, and I have slept since then... |
| 17:07.05 | *** part/#asterisk ketema (~ketema@ketema.net) |
| 17:07.10 | dont_taze_me_bro | p3nguin: understood, but other codecs have better quality and can be transcoded for free |
| 17:07.14 | pigpen | either way, if he has it in your example, he can transcode. |
| 17:07.28 | p3nguin | pigpen: I agree with you assessment, but Digium certainly isn't going to give it to you at no cost just because you tell them you promise not to transcode. |
| 17:07.38 | gruvfunk | what's the best codec to be using right now for free? |
| 17:07.40 | p3nguin | (1154.45) <l2trace99> g729 is required |
| 17:07.56 | p3nguin | gruvfunk: What is your definition of best? |
| 17:08.13 | gruvfunk | voice quality vs. used bandwidth (been using ulaw for ages) |
| 17:08.43 | leifmadsen | gruvfunk: you could try ilbc or speex.... |
| 17:08.47 | leifmadsen | I just use ulaw, g722, or g729 |
| 17:08.55 | leifmadsen | (yes i know g729 isn't free) |
| 17:09.03 | p3nguin | G.729 is a good quality codec, saving a lot of bandwidth... but we're just discussing that it isn't free. |
| 17:09.11 | gruvfunk | yep |
| 17:09.24 | gruvfunk | leifmadsen: how's g722 compared to ulaw in quality? |
| 17:09.37 | gruvfunk | (in your opinion) |
| 17:10.31 | pigpen | p3nguin, "Each Asterisk server that you want to perform G.729 transformations will require its own license key. " Note the word, transformations |
| 17:10.37 | pigpen | I called digium about that. |
| 17:11.38 | Katty | sets up stand |
| 17:11.41 | Katty | free hugs. |
| 17:11.47 | pigpen | p3nguin, http://www.digium.com/en/products/g729codec.php |
| 17:11.48 | p3nguin | In the case where a person wishes to use g729 end to end and not transcode, while keeping Asterisk in the media path, will the g729 codec need to be present on Asterisk? |
| 17:12.04 | p3nguin | |
| 17:12.22 | pigpen | p3nguin, yeah, passthrough. |
| 17:12.35 | pigpen | ie: the call would not benefit from the features of asterisk...now I remember. |
| 17:13.04 | pigpen | So I was not completely wrong, but not completely right either. ;-) |
| 17:13.10 | p3nguin | How does one obtain the codec from Digium for free? The promise of not transcoding surely will not convince them. Do they provide a free non-transcoding codec? I doubt they do. |
| 17:13.51 | pigpen | you can install the codec, but to use it with typical usage, it need to be licensed. |
| 17:14.04 | sunfone | If you aren't transcoding it, then the license is being paid for by the endpoints - that is why you don't have to give Digium any money if it is a passthru |
| 17:14.17 | p3nguin | How will you legally obtain it? You're not addressing what I am asking. |
| 17:14.18 | pigpen | sunfone, exactly. |
| 17:14.20 | sunfone | if asterisk is transcoding then it is an endpoint itself |
| 17:15.02 | p3nguin | I'm telling you that Digium is not going to give it to you if you don't pay for it. |
| 17:15.02 | sunfone | and therefore needs a license |
| 17:15.11 | leifmadsen | gruvfunk: night and day? g722 is significantly better sounding than ulaw |
| 17:15.12 | pigpen | p3nguin, on all the * boxes we compile from scratch has the g729 ready to go. |
| 17:15.18 | sunfone | you don't even need it loaded to pass through calls |
| 17:15.19 | pigpen | part of the base I beleive. |
| 17:15.27 | gruvfunk | thx leifmadsen |
| 17:15.28 | p3nguin | I've never heard of such a thing. |
| 17:15.46 | leifmadsen | if everything is g729 ya you don't need the license unless you want to record audio |
| 17:15.54 | leifmadsen | (i.e voicemail) |
| 17:15.56 | pigpen | p3nguin, do: core show codecs you will seei it. |
| 17:16.39 | p3nguin | leifmadsen: My question was if the codec is required to be present (installed) on Asterisk if you're doing g729 end-to-end and never transcoding or recording. |
| 17:16.58 | p3nguin | I'm sure I'll see it, since I have the codec and license installed. |
| 17:17.08 | pigpen | I don't, and I see it. |
| 17:17.38 | p3nguin | Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. |
| 17:17.55 | p3nguin | It seems like that's a generic table. |
| 17:18.04 | sunfone | ya that just calculates transcode time |
| 17:18.29 | sunfone | although it will show a '-' if you don't have the module loaded |
| 17:18.36 | sunfone | (I think) |
| 17:18.36 | p3nguin | Try "core show translation recalc 10" |
| 17:19.03 | p3nguin | core show codecs doesn't show the translation time. |
| 17:19.44 | p3nguin | core show translation shows the times. If g729 is all - - - - - , then the codec is not present. |
| 17:19.56 | *** join/#asterisk golikwid|mac (~chrislees@64.45.192.156) |
| 17:21.24 | sunfone | that probably changed in a later version - in 1.4 core show translation shows '-' if it isn't loaded |
| 17:21.38 | sunfone | but it can still pass through G.729 calls |
| 17:22.29 | p3nguin | Okay, so you're saying that end-to-end g729 calling, without recording or transcoding, does not need to have the codec loaded at all. Do I read you correctly? |
| 17:22.39 | sunfone | yes |
| 17:22.53 | sunfone | because the codec isn't being exercised |
| 17:23.01 | sunfone | and the endpoints have already paid a royalty |
| 17:23.10 | p3nguin | This is contradictory to what pigpen said when I asked before, so I'm just trying to get the right story. |
| 17:23.18 | sunfone | :) |
| 17:23.31 | sunfone | I didn't read the whole thread... just sat down |
| 17:23.45 | drmessano | Asterisk supports G.729 passthru |
| 17:23.49 | p3nguin | It's hard for me to test things such as this when I do have the codec installed. |
| 17:23.51 | drmessano | Transcoding requires a license |
| 17:23.58 | sunfone | unload the module |
| 17:24.20 | p3nguin | Eh... *headsmack* |
| 17:24.27 | sunfone | I wonder if you can even record as long as it is saved in g.729 format? |
| 17:24.38 | sunfone | probably... |
| 17:24.50 | p3nguin | You'd have to transcode. |
| 17:24.57 | drmessano | The license isn't to USE G.729.. your endpoints have already paid for their licenses. Asterisk just passes media as it would any codec, especially the video |
| 17:25.17 | p3nguin | I believe it goes through slin when recording, no matter what. |
| 17:25.21 | drmessano | When you want to TRANSCODE is where Asterisk itself needs a license |
| 17:25.25 | sunfone | ah ya I think you are right |
| 17:25.38 | sunfone | (the slin bit) |
| 17:26.48 | sunfone | I dumped g.729 a while back anyway... if you have the bandwidth there doesn't seem to be a reason not to use g.711u |
| 17:26.53 | sunfone | sure sounds better |
| 17:27.17 | sunfone | are any of the ITSPs doing wideband yet? |
| 17:27.21 | drmessano | I use G.729 over 3G. ulaw is horrible |
| 17:27.32 | sunfone | then you don't have the bandiwdth ;) |
| 17:27.38 | drmessano | Duh |
| 17:28.15 | p3nguin | I doubt any ITSP cares about wideband. It would only be useful for calls that never leave their systems. Anything that hits the telco switch would be reduced back to ulaw anyway. |
| 17:28.24 | drmessano | I don't see a big push for it |
| 17:28.35 | sunfone | its one of those catch22 issues... like IPv6 :) |
| 17:28.48 | drmessano | Transcoding G.722 down to tin can and string is a waste |
| 17:28.52 | sunfone | no one will move to it until everyone is using it... |
| 17:29.04 | drmessano | That's irrelevant |
| 17:29.13 | drmessano | It won't make a difference on the PSTN |
| 17:29.31 | sunfone | we have to imagine that the traditional PSTN's days are numbered though |
| 17:29.38 | sunfone | just like IPv4 |
| 17:29.38 | drmessano | LOL |
| 17:29.44 | sunfone | there will be some kind of tipping point |
| 17:29.52 | p3nguin | But if the PSTN infrastructure ever changed, then we might have an opportunity to go HD on it. |
| 17:29.55 | drmessano | No, actually, the situation is worse |
| 17:30.45 | drmessano | People are now using mobile handsets, which makes even less of a case for wideband |
| 17:30.57 | sunfone | not for long! LTE will change that |
| 17:31.07 | drmessano | LOL |
| 17:31.07 | sunfone | 3G is so last year :) |
| 17:31.11 | p3nguin | Not every company is going to adopt LTE. |
| 17:31.31 | sunfone | maybe not LTE per se, but the bandwidth over whatever comes next is inevitable |
| 17:31.52 | irroot | LTE or 3G many providers here all of them are protectionist and block/shape/bar sip/rtp |
| 17:31.53 | sunfone | I run VoIP over my EVO |
| 17:31.55 | p3nguin | Just look at AT&T: Long live GSM! |
| 17:32.24 | drmessano | More bandwidth is not going to give way to higher quality calls |
| 17:32.37 | sunfone | why not? |
| 17:32.38 | p3nguin | It could provide the potential. |
| 17:32.46 | drmessano | There may be more room for Facebook and Youtube streaming, but carriers have NO reason to touch call quality |
| 17:32.53 | p3nguin | It's not going to happen, but it would be more of a possibility. |
| 17:32.59 | gruvfunk | Repost: I need a Toll Free DID in Italy and one in Germany - can anyone recommend an ITSP? |
| 17:33.08 | Dovid | hi. i have 65 G729 liscences: No translator path from alaw to g723 |
| 17:33.12 | Dovid | why would I ge tthat error ? |
| 17:33.19 | Dovid | i do not have G723 enabled |
| 17:33.35 | sunfone | I wouldn't be so quick to say so - Sprint (Clearwire) is already shaping SIP |
| 17:33.40 | drmessano | Because call quality is NOT an issue for most users.. they don't complain or care. |
| 17:33.52 | sunfone | On cell phones? I complain |
| 17:34.09 | sunfone | I think the public has become apathetic |
| 17:34.19 | Dovid | anyone know what would cause this ? http://pastebin.com/j35JqB1q |
| 17:34.33 | p3nguin | I think the public are a pushover, and they'll use whatever The Man tells them they will use. |
| 17:34.38 | sunfone | heh |
| 17:34.46 | drmessano | Seriously? Do you really think the compressed audio on most mobile handsets is something that gets carriers swamped with complaints? Hell no. It's MECCA for AT&T |
| 17:34.48 | sunfone | but if something shiny comes out, they will clamor for it |
| 17:35.23 | p3nguin | I think there's a term for that. |
| 17:35.30 | sunfone | drm: right, because they have become apathetic about it |
| 17:35.32 | kaldemar | Dovid: i think you just answered your own question. |
| 17:35.46 | drmessano | If you gave the average mobile user the option of 2 more MB of data a month or wideband calling.. They would take the measly 2MB of data |
| 17:35.56 | sunfone | Dovid - one of your endpoints is trying to use g.723 |
| 17:36.06 | p3nguin | The average user, yes. |
| 17:36.25 | sunfone | but who says the plans will even stay that way? Sprint's 4G plan is unlimited |
| 17:36.31 | p3nguin | We in the minority don't make enough of a difference in wanting wideband for it to happen. |
| 17:36.46 | sunfone | All it will take is a new handset that is entirely SIP |
| 17:36.53 | sunfone | over 4G of whatever kind |
| 17:36.57 | drmessano | You can sit here and wish on a SIP star, daydream about the day SIP URI calling bypasses PSTN numbers.. and G.722 or better is the default codec across the universe.. but it won't happen |
| 17:37.02 | sunfone | We will have the "pin drop" commercials again |
| 17:37.26 | drmessano | sunfone: What is the emoticon for a pin drop? There's your market |
| 17:37.44 | sunfone | not sure I get that :) |
| 17:38.05 | p3nguin | Look back and the days of the old pin drop commercials. Use that same technology right now, compared to what some of us have grown accustom to, and you can't even tell the pin exists. |
| 17:39.09 | drmessano | People walk into a mobile phone outlet and they want "iPhones" "Androids" "Apps" "3G" and "Angry Birds".. the only pin they care about dropping is part of some new Zynga game |
| 17:39.12 | sunfone | well the original pin drop commercials were touting Sprint's new fiber infrastructure |
| 17:40.01 | sunfone | drm: I totally disagree - people complain about call drops and bad audio... I switch conference calls to landlines frequently because of it |
| 17:40.14 | sunfone | but there is no relief from it right now, because the bandwidth isn't there |
| 17:40.36 | sunfone | in fact the call drops on AT&T's network were caused IMO by the flood of new iphone users |
| 17:40.38 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 17:41.32 | p3nguin | and because AT&T wireless sucks. |
| 17:41.37 | drmessano | I completely disagree.. If someone like AT&T wanted to beef up the call quality with existing hardware, they could.. but there's no motivation to do so, and that B/W is better allocated to data |
| 17:42.22 | Dovid | sunfone: It keeps scrolling on the screen over and over. It's not just once. it seems very strange. how would i see who is trying that ? |
| 17:42.25 | sunfone | Not on cell phones - they are hampered by density and spectrum, because the current protocols don't make good use of it |
| 17:42.45 | jc319 | Is there a (semi-) standard key for voicemail, because I have *97 or *99 in two different configs, would like to keep the more std one if there's such a thing? |
| 17:43.02 | sunfone | Dovid: maybe tcpdump? |
| 17:43.19 | p3nguin | jc319: I prefer *86. *VM, that is. |
| 17:44.00 | jc319 | p3nguin: That's clever, it is also a very nice finger move * > 8 > 6 on the keypad, I'll go for it. |
| 17:44.05 | p3nguin | VM, VoiceMail... makes sense to me. |
| 17:44.14 | p3nguin | and to Verizon. |
| 17:44.37 | irroot | there is a asterisk company here in za www.shifteight.co.za |
| 17:44.37 | Dovid | sunfone: I am also getting: Unable to translate to format gsm, source format g729 |
| 17:44.49 | leifmadsen | I use 8500 because that's what my old Cisco phone used to use :) |
| 17:45.11 | irroot | i use 100 thats the vmail access code for largest cell provider |
| 17:45.26 | sunfone | Dovid: better make sure you don't have people registered you aren't expecting :) |
| 17:45.58 | jc319 | * 8 6 also sounds very nice, almost like do-re-mi. I don't know if it's me or is it really a beatiful sound. I love changing things with this pbx but perhaps I need to find some other hobbies.. |
| 17:45.58 | p3nguin | I use *86 to go to your own mailbox, use 9000 to go to VoiceMailMain, and use 9XXX to reach a mailbox of anyone (where XXX is their normal 3-digit exten). |
| 17:46.07 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 17:46.10 | p3nguin | irroot: Which provider? |
| 17:46.31 | irroot | vodacom now got vodaphone branding |
| 17:47.40 | p3nguin | *VM still makes the most sense to me. |
| 17:48.23 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 17:48.39 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 17:48.42 | *** join/#asterisk errr (~errr@fedora/errr) |
| 17:48.59 | errr | is there a way to show what version of zaptel I have (asterisk 1.2.x)? |
| 17:49.21 | p3nguin | I would check my package manager. |
| 17:49.43 | errr | its from source and there are dozens of version in /usr/local/src/zaptel |
| 17:49.52 | p3nguin | That's what you get when you refuse to use your package manager. |
| 17:49.57 | errr | no way to show whats running currently? |
| 17:51.16 | errr | p3nguin: and fwiw that would be a silly way to look anyway becuase that wouldnt tell you if someone installed a version from source or tossed up some binaries in its place. |
| 17:51.36 | p3nguin | That's why I don't let noobs manage my systems. |
| 17:51.47 | sunfone | errr: do a 'strings' on the binary module |
| 17:51.53 | Katty | throws things at Qwell |
| 17:52.01 | p3nguin | intercepts |
| 17:52.06 | Katty | :< |
| 17:52.07 | p3nguin | You could put his eye out! |
| 17:52.19 | Katty | what if they're 1s? |
| 17:52.21 | sunfone | does he only have one eye? |
| 17:52.27 | Katty | yes. he is a cyclops. |
| 17:52.34 | p3nguin | hahaha |
| 17:52.34 | sunfone | I thought so. |
| 17:52.55 | jc319 | p3nguin: Do you even use the on-device directory for 79x0s (SIP)? I've found this one >> "Open 79XX XML Directory". I will try that but would love to benefit from experience rather than trying one by one (there is several listed in voip-info.org web site) |
| 17:53.24 | p3nguin | jc319: When I used SIP, I did use the directory file. |
| 17:53.55 | *** part/#asterisk errr (~errr@fedora/errr) |
| 17:53.55 | jc319 | Is it limited to 32 entries |
| 17:54.19 | p3nguin | I didn't have 32 or try to add more, so I couldn't say. |
| 17:55.42 | Dovid | sunfone: I know I don't |
| 17:55.49 | Dovid | i restarted asterisk and the error went away |
| 17:59.55 | sunfone | Dovid: when you restarted asterisk you "disconnected" whoever was trying to place those calls |
| 18:00.21 | sunfone | they might come back! |
| 18:05.54 | irroot | got a eclipse happening soon |
| 18:05.57 | l2trace99 | p3nguin: I am not actually trying to get asterisk out of the stream as much as change the codec |
| 18:09.09 | Dovid | sunfone: they did not |
| 18:09.39 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 18:12.31 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 18:14.43 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.212) |
| 18:15.37 | bobb_WU | hey is there anyone who has experience with modems and asterisk? |
| 18:15.45 | bobb_WU | i have an atm on campus that isn't working and would like some advice |
| 18:16.11 | *** part/#asterisk l2trace99 (~jr@74.118.40.1) |
| 18:16.22 | jc319 | Is this command supposed to restart 7960? I found this in a post from 2006, is it depreciated? "sip notify cisco-check-cfg <peer>" [CLI responds sending NOTIFY of type 'cisco-check-cfg' to 'MAC' but nothing happens] |
| 18:16.55 | p3nguin | Depriciated? Like it lost all its monetary value? |
| 18:17.23 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 18:18.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:18.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:19.55 | p3nguin | Perhaps deprecated rather than depreciated... but I don't know why it would restart the phone. cisco-check-cfg doesn't indicate to me that it needs to restart anything. |
| 18:22.29 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
| 18:22.53 | hardwire | anybody have a preference on a windows sip/iax phone for a call center that needs to answer multiple lines and do simple holds/transfers? |
| 18:23.32 | hardwire | I'll probably start off with counterpath products. |
| 18:23.47 | citywok | What are teh chances i'll be able to restore a backup from my AA50 to AsteriskNOW? |
| 18:24.02 | p3nguin | Zoiper Classic doesn't fit the bill? |
| 18:24.20 | citywok | hardwire: we use zoiper web and it works fairly well |
| 18:24.30 | hardwire | citywok: don't you work for zoiper? |
| 18:24.32 | citywok | we just bought one license for it and embedded it in our agent interface |
| 18:24.39 | citywok | hardwire: no i work for a call center |
| 18:24.40 | hardwire | ah ok :) |
| 18:24.42 | hardwire | thats right |
| 18:24.47 | hardwire | we've butt heads before |
| 18:24.55 | citywok | the zoiper guys? |
| 18:24.59 | hardwire | you and I |
| 18:25.04 | hardwire | just had to remember |
| 18:25.07 | citywok | oh haha |
| 18:25.14 | citywok | i don't even remember it :P |
| 18:25.24 | hardwire | *sadface* |
| 18:25.51 | citywok | but yea, for zoiper web our experience has been pretty good, and the zoiper guys are generally pretty good at helping fix bugs in their software. |
| 18:26.17 | citywok | we helped them find and fix quite a few bugs in it and they ended up giving us the licenses for 3 way calling and some other feature we needed for free b/c of it |
| 18:26.23 | dont_taze_me_bro | is there any way to prevent UA registration? i constantly see these bots trying to register thousands of times |
| 18:26.25 | hardwire | I'm guessing zoiper web is easy to provision. |
| 18:26.42 | citywok | yea i use a php script to generate it per extension |
| 18:26.42 | dont_taze_me_bro | i dont even use the server for UAs |
| 18:26.53 | WIMPy | also likes zoiper, but zoiper doesn;t seem to like 10ms paket size. |
| 18:26.59 | hardwire | citywok: I was thinking something along that line. |
| 18:27.10 | hardwire | WIMPy: too cool for 20ms? |
| 18:27.20 | citywok | if you want my javascript provisioning i'll send it to you, but it was pretty much a rip off of their example with a few flags added in. |
| 18:27.27 | hardwire | ah ok |
| 18:27.29 | WIMPy | jepp ;-) |
| 18:27.31 | citywok | not all the flags i needed were in their docs so i had to ask (auto-answer) |
| 18:28.02 | citywok | oh, also it can not QoS the SIP stream, only the RTP stream via the javascript autoprovisioning. |
| 18:28.05 | hardwire | zoiper web is kinda spensive. |
| 18:28.07 | hardwire | hmm. |
| 18:28.09 | citywok | you have to set that in the gui |
| 18:28.10 | hardwire | I'll have to try it out |
| 18:28.24 | citywok | i think we paid a grand for it, which is really cheap /100 agents |
| 18:28.32 | jc319 | p3nguin: Yes depreciated or abandoned would be a better word maybe. I was looking into this reboot, apparently it is a bit more difficult than I thought, still possible with some 3rd party code though http://www.voipphreak.ca/2008/09/05/remotely-reboot-your-cisco-79xx-phone-even-with-asterisk-pbx/ I'll try later it might be useful for remote sites one day |
| 18:28.36 | hardwire | it's more expensive than free.. but yeh. |
| 18:28.56 | citywok | how many seats do you have? |
| 18:28.58 | hardwire | citywok: can you have local clients prioritize an audio device? |
| 18:29.07 | hardwire | citywok: 6 for the site I'm working on. |
| 18:29.14 | citywok | we went the web way b/c it was so cheap, otherwise clients with autoprovisinoing are like $50/ea |
| 18:29.22 | citywok | autoprovisioning* and autoanswer |
| 18:29.34 | citywok | yea, you can open the options and set the audio device |
| 18:29.39 | hardwire | yeh.. autoanswer won't be key thankfully. |
| 18:30.05 | hardwire | citywok: Will it remember it? |
| 18:30.41 | citywok | not sure |
| 18:30.44 | hardwire | it looks like a neat product. |
| 18:31.11 | hardwire | looks like we need a doze ISS server |
| 18:31.27 | hardwire | or maybe not.. you just need to put the cab somewhere |
| 18:31.38 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 18:31.56 | hardwire | err IIS :) |
| 18:35.42 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 18:37.08 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 18:46.19 | gandhijee | anyone have any with RingCentral? |
| 18:47.42 | linuxgecko | WIMPy: ... 10ms?? how is a time measurement a packet size? |
| 18:48.13 | *** part/#asterisk hdiogenes (~humberto@201.76.154.133.intranet.digi.com.br) |
| 18:49.50 | carrar | it's codec sample size |
| 18:49.54 | carrar | which is time |
| 18:50.45 | linuxgecko | ahhh.. |
| 18:51.03 | linuxgecko | i use sipdroid or ekiga for now |
| 18:51.06 | irroot | http://eclipse.slooh.com <- what is happening outside but its too cold ... so watching the stream |
| 18:52.46 | carrar | linuxgecho: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 18:52.49 | carrar | read that |
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| 18:56.15 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
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| 19:01.02 | irroot | cool im going outside soon to see the eclipse |
| 19:01.41 | *** join/#asterisk nny (~SM@174.107.223.14) |
| 19:01.54 | nny | well, i feel stupid |
| 19:02.44 | nny | so normally say i have 20 phones, i do exten => 1,1,Macro(something,1,SIP/1) exten => 2,1,Macro(something,2,SIP/2) etc etc |
| 19:03.23 | nny | can't i just do exten => [1-20],1,Macro(something,$EXTEN,SIP/{$EXTEN}) *syntax off* or something like that? |
| 19:03.41 | WIMPy | sure |
| 19:03.53 | WIMPy | What's that 2nd parameter? |
| 19:03.54 | nny | lol i've been wastin valuable bits and time for 4 years now |
| 19:04.10 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:04.31 | WIMPy | But it's ${EXTEN} in both cases, off course. |
| 19:04.51 | nny | WIMPy: for the dial macro (Dial($ARG2},20) etc |
| 19:05.15 | nny | yeah, was just guessing it out a bit, let me correct the syntax |
| 19:05.19 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 19:05.23 | nny | i also do the 1,2,3 etc for hints :S |
| 19:05.24 | WIMPy | Wll, the first argument if you don;t count the macro name. |
| 19:05.50 | WIMPy | Hints? |
| 19:06.09 | nny | oh the first argument is for voicemail, but I could just do Dial(SIP/$ARG1,20) |
| 19:06.24 | nny | er |
| 19:06.29 | nny | ${ARG1} |
| 19:06.33 | nny | and remove arg2 all together |
| 19:06.36 | WIMPy | Ah, yes. |
| 19:06.52 | nny | WIMPy: exten => 11,hint,SIP/11 |
| 19:07.02 | nny | WIMPy: essentially just a hint context for the sidecar etc |
| 19:07.29 | WIMPy | Ja, bit the hints aren't related to the macro. Tha's why I was asking. |
| 19:07.48 | *** join/#asterisk wwgd (~office@208.79.14.130) |
| 19:07.53 | WIMPy | And I think in theory you should be able to use a wildcard hint as well. |
| 19:08.13 | nny | WIMPy: oh no, yeah i am thinking seperate context changes for hints to avoid making 20 lines for 20 phones etc |
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| 19:13.49 | psilikon | citywok, Do you use Vicidial? |
| 19:14.03 | citywok | psilikon: no, we don't predictive dial |
| 19:14.22 | citywok | we're not that kind of call center |
| 19:15.59 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 19:25.41 | irroot | the eclipse is pretty freaky |
| 19:26.17 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 19:31.43 | gruvfunk | little help people (i'm in the USA) - how do you dial a UK toll free from within the UK? do you prefix a 0 to the 808 #? |
| 19:32.00 | WIMPy | yes |
| 19:32.13 | gruvfunk | thx, can I have the same response for France ? |
| 19:32.39 | WIMPy | Not sure there, but most probably the same. |
| 19:33.14 | jc319 | 0800<#> |
| 19:33.32 | jc319 | I'm in the UK but my phone thinks it's in the US I need to dial 0044 first :D |
| 19:33.43 | hardwire | gruvfunk: dialaroundworld.com helps me quite a bit. |
| 19:34.03 | WIMPy | Why do I think, it's not the phone? |
| 19:34.04 | gruvfunk | hardwire: thx |
| 19:36.37 | breardo | hello |
| 19:36.53 | sunfone | You might not be able to dial a toll free UK number (or France or any other country for that matter) from a US line... the scope of the toll free probably doesn't reach that far |
| 19:37.03 | jc319 | When I place calls, the caller ID comes up in the US format (1-xxx number), is this formatting/digits something I can change using asterisk or is it something that's configured from the ITSP? ITSP is voip.ms and says "you provide us your caller ID" in every web config screen but it does not show what I typed in in the config files. |
| 19:37.36 | breardo | trying to compile asterisk 1.8.4.2 with latest dahdi and libpri, and I get the error: "The PRI_MWI installation appears to be missing or broken, Either correct the installation or run confiure with --without-pri" |
| 19:37.36 | WIMPy | It parobably works, but might not be free. |
| 19:37.38 | jc319 | yeah I tried to call siemens today, an 0845 number, it does not connect with 0044845<no>. |
| 19:37.46 | nny | i am changing my dialplan to play a recording if a variable is set via key combo, then carry on. What's the proper way to do If, then, return? Does gotoif return to the dialplan after it's step? |
| 19:37.49 | breardo | libpri and dahdi/dahdi-tools installed fine |
| 19:38.04 | nny | er return to the line in the context it jumped from* |
| 19:40.46 | breardo | ok.. well, switching to libpri-1.4.12-beta3 corrected the errors |
| 19:40.47 | CaptainPants | [PRI-1] [Status: Closed] Incorrect "user information layer 1" representation - https://issues.asterisk.org/jira/browse/PRI-1 |
| 19:42.35 | Qwell | russellb: You broke it. |
| 19:42.39 | Qwell | :p |
| 19:42.40 | *** join/#asterisk linuxgecko (~gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
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| 19:43.25 | gruvfunk | anyone here currently in Spain? |
| 19:43.46 | russellb | Qwell: nice |
| 19:44.03 | russellb | i bet it's also broken if you talk about asterisk-1.8.4.2.tar.gz |
| 19:44.04 | CaptainPants | [ASTERISK-1] [Status: Closed] SIP re-invites failing with certain proxies - https://issues.asterisk.org/jira/browse/ASTERISK-1 |
| 19:44.10 | russellb | sure enough. |
| 19:48.54 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 19:51.29 | breardo | im trying to build astersk 1.8.4.2 with --with-dahdhi, and ./configure executes fine but there is no chan_dahdi.so being built... I dont see it come up when I run 'make' either... any clues? |
| 19:51.40 | *** join/#asterisk philippel_mac (~p_lindhei@74-92-226-109-Washington.hfc.comcastbusiness.net) |
| 19:51.53 | breardo | yet, everything that I see in config.log shows DAHDI as being successfully found/probed |
| 19:52.21 | breardo | err with --with-dahdi that is to say |
| 19:53.19 | breardo | i run 'make clean' and then 'make channels' and no chan_dahdi gets built or even attempted.. :/ |
| 19:55.07 | sunfone | breardo: does "make menuselect" show it? |
| 19:57.31 | breardo | no, its's XXX'd out |
| 19:57.54 | breardo | cant enable it either... |
| 20:01.53 | wdoekes2 | breardo: doesn't it show a "Depends on: ..." at the bottom? |
| 20:01.57 | sunfone | It should be telling you what is required |
| 20:02.33 | breardo | yes, it does.. i have those things installed already |
| 20:02.46 | breardo | i dont understand why it depends on SS7 though.. i dont have that installed |
| 20:03.34 | breardo | says it depends on dahdi(E), tonezone(E), res_smdi(M), pri(E), ss7(E) |
| 20:03.47 | breardo | i built with --with-pri --with-dahdi --with-tonezone |
| 20:04.43 | wdoekes2 | and what does it say there if you rerun ./configure without any --options? |
| 20:04.49 | breardo | one sec |
| 20:05.33 | breardo | same exact stuff |
| 20:05.40 | breardo | still cant enable it |
| 20:06.30 | wdoekes2 | odd that ss7 is in the depends list.. over here with 1.8-svn it's in "Can use:" |
| 20:06.54 | breardo | i thought so as well |
| 20:07.00 | breardo | im gonna install the ss7 lib just to see |
| 20:08.22 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:09.09 | breardo | nope |
| 20:09.16 | breardo | even with libss7 installed.. same stuff |
| 20:10.58 | Qwell | breardo: pastebin your config.log |
| 20:11.08 | breardo | okay.. it'll take a sec |
| 20:20.44 | breardo | pastebin.com/tZ2rnfv1 |
| 20:21.07 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 20:21.25 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:22.49 | Qwell | and what does menuselect say about it? |
| 20:25.47 | breardo | it says that it cant be installed |
| 20:25.51 | breardo | just has XXX and nothing else |
| 20:28.46 | Qwell | bleh. just make dist-clean |
| 20:30.16 | breardo | this is a fresh install |
| 20:30.28 | Qwell | do it anyways |
| 20:30.37 | breardo | done |
| 20:30.44 | Qwell | now configure, make menuselect |
| 20:30.52 | breardo | config with no options? |
| 20:31.02 | Qwell | yes |
| 20:31.05 | breardo | ok |
| 20:31.28 | breardo | done |
| 20:31.53 | breardo | want a fresh config.log ? |
| 20:32.43 | breardo | results are the same btw... no chan_dahdi available |
| 20:33.10 | Qwell | What versions of things did you install? |
| 20:33.33 | breardo | latest everything, except I had to install libpri-1.4.12-beta3 because 1.4.11.5 would give me errors during ./configure of asterisk |
| 20:33.35 | CaptainPants | [PRI-1] [Status: Closed] Incorrect "user information layer 1" representation - https://issues.asterisk.org/jira/browse/PRI-1 |
| 20:33.56 | Qwell | errors? |
| 20:34.24 | breardo | yeah it would complain about PRI_MWI (and others) not being available, and to install with --without-pri |
| 20:35.17 | breardo | want a config.log from that build? |
| 20:35.22 | Qwell | no |
| 20:35.24 | breardo | ok |
| 20:35.25 | Qwell | what other versions of what? |
| 20:36.09 | breardo | i installed dahdi from the dahdi-complete 2.4.1.2+2.4.1 |
| 20:36.30 | breardo | and i threw in the latest libss7 just to check |
| 20:36.55 | Qwell | There are a lot of people that use all of those versions of things together, and it works just fine... |
| 20:37.08 | breardo | not working here... this is on Debian Squeeze |
| 20:37.24 | breardo | i installed libpri first, then dahdi, then compiled asterisk.. |
| 20:38.02 | WIMPy | breardo: Do you have old versions of anything, perhaps installed as packet? |
| 20:38.29 | breardo | no.. this was a clean install of Debian Squeeze with only these packages downloaded as source and installed |
| 20:40.49 | breardo | i guess i'll reinstall and try again :| |
| 20:41.23 | mickecarlsson | breardo do you have kernel-headers installed? |
| 20:41.33 | breardo | yes and kernel-source |
| 20:41.38 | mickecarlsson | ok |
| 20:41.40 | breardo | couldnt compile without :) |
| 20:41.58 | *** join/#asterisk tamiel (~tamiel@ip-28.net-81-220-88.toulouse.rev.numericable.fr) |
| 20:42.41 | mickecarlsson | when you do make menuselect and go to channel drivers and the XXX for dahdi, what does it say on the bottom of the screen? |
| 20:43.10 | breardo | says it depends on dahdi(E), tonezone(E), res_smdi(M), pri(E), ss7(E) |
| 20:44.10 | mickecarlsson | I have a suggestion, but you probably wont like it |
| 20:44.15 | mickecarlsson | use centos |
| 20:44.21 | mickecarlsson | is running and ducking |
| 20:44.21 | breardo | i shouldnt have to |
| 20:44.43 | breardo | heh... yeah im not going to use CentOS. |
| 20:45.01 | mickecarlsson | Well, I said you would not like it :-) |
| 20:45.09 | breardo | haha |
| 20:45.25 | breardo | it should work fine here is all.... im not a centOS hater or anything |
| 20:45.26 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 20:46.18 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:46.23 | mickecarlsson | Well, check your packages: #define HAVE_DAHDI_VERSION 230 |
| 20:46.36 | mickecarlsson | Should that not be 241? |
| 20:46.40 | *** part/#asterisk nny (~SM@174.107.223.14) |
| 20:46.51 | mickecarlsson | is looking at the pastebin |
| 20:47.22 | UnixDev | Hi, I'm using asterisk 1.8, which we upgraded to from 1.6... but for some reason, inbound calls are being re-invited when I specifically have canreinvite=no and directmedia=no on the peer, this only seems to be a problem for inbound calls that get transferred, asterisk wants to re-invite every time it transfers and/or hold's the call...how can I stop this behavior? it did not happen in 1.6 or 1.4 |
| 20:48.35 | breardo | yeah thats weird eh |
| 20:48.40 | breardo | wtf |
| 20:49.22 | mickecarlsson | is looking at one of his config.logs |
| 20:50.02 | Qwell | That isn't how it works.. |
| 20:50.02 | mickecarlsson | Hmm, I have the same here: 230 |
| 20:50.09 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 20:50.10 | breardo | oh well |
| 20:50.12 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 20:50.15 | breardo | thought you were onto something :) haha |
| 20:51.19 | mickecarlsson | and you did make make install on dahdi? |
| 20:51.28 | breardo | yes |
| 20:51.37 | breardo | i have /usr/include/dahdi and kernel modules installed appropriately |
| 20:51.43 | mickecarlsson | before you run configure for asterisk? |
| 20:52.04 | breardo | yes.. libpri and dahdi were installed prior to the asterisk configuration |
| 20:52.14 | mickecarlsson | blames debian |
| 20:52.25 | breardo | classic |
| 20:52.29 | mickecarlsson | :-) |
| 20:52.39 | breardo | friggin linux :) you guys need to say screw this and move to FreeBSD exclusively :) |
| 20:53.18 | breardo | i'll probably try a fresh install of everything again tomorrow.. |
| 20:53.53 | mickecarlsson | Does is say depends on all the above or just depend on dahdi and tonezone? |
| 20:53.55 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:54.03 | breardo | funny how when i run ./configure with --with-dahdi, it doenst have any problems at all.. |
| 20:54.32 | breardo | says: Depends on: dahdi(E), tonezone(E), res_smdi(M), pri(E), ss7(E) |
| 20:54.34 | breardo | thats all |
| 20:54.40 | mickecarlsson | ?? |
| 20:54.49 | mickecarlsson | Mine say: |
| 20:54.56 | mickecarlsson | <PROTECTED> |
| 20:55.01 | mickecarlsson | Can use: res_smdi(M), pri(E), ss7(E), openr2(E) |
| 20:55.12 | breardo | none of my options have a "can use" field.. anywhere |
| 20:55.16 | breardo | i looked |
| 20:55.25 | breardo | what the fsck.. |
| 20:55.28 | mickecarlsson | Well, then you need all depends installed |
| 20:55.32 | Qwell | How exactly did you download Asterisk? |
| 20:55.42 | breardo | from downloads.digium.org on the asterisk.org website |
| 20:55.44 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:55.52 | breardo | or downloads.asterisk.org, whatever it is.. |
| 20:56.57 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:57.08 | Qwell | wait, you used --prefix? |
| 20:57.20 | breardo | originally yeah.. not during these tests though |
| 20:57.47 | Qwell | Then you didn't pastebin the correct config.log |
| 20:57.59 | Qwell | Reinstall everything. Don't use --prefix garbage. on any of it. |
| 20:58.11 | Qwell | ./configure; make; make install |
| 20:58.11 | breardo | i only used prefix in asterisk.. not anywhere else |
| 20:59.00 | breardo | adding or removing --prefix did not have any impact on this issue |
| 21:00.17 | mickecarlsson | Qwell what does the letters, in this case E, mean after the depends? (Depends on: dahdi(E), tonezone(E)) |
| 21:00.28 | mickecarlsson | And the M? |
| 21:00.31 | breardo | i presumed External vs Modules |
| 21:00.51 | breardo | External depends are outside the typical asterisk package... while M's are included..? |
| 21:00.51 | mickecarlsson | Ah, learned something new today |
| 21:00.56 | breardo | i presume anyways |
| 21:01.23 | Qwell | When you do the above, run make dist-clean in everything first. |
| 21:01.27 | breardo | i did |
| 21:02.00 | breardo | this time its downloading sounds.. didnt do that before |
| 21:04.13 | breardo | nope |
| 21:04.15 | breardo | no chan_dahdi |
| 21:04.29 | Qwell | I somehow doubt you reinstalled everything that quickly. |
| 21:04.33 | mickecarlsson | And it says depends on all items? |
| 21:04.38 | breardo | i did... i have a fast box |
| 21:04.55 | breardo | i only reinstalled asterisk... i didnt use --prefix or anything on dahdi or libpri |
| 21:05.26 | breardo | those were built with straight 'make; make install' |
| 21:06.43 | mickecarlsson | bedtime in Sweden, goodnight. |
| 21:07.19 | breardo | goodnight |
| 21:07.21 | breardo | thanks |
| 21:07.29 | UnixDev | anyone familiar with asterisk 1.8? |
| 21:08.12 | *** join/#asterisk dym (~patrick@netsplit.me) |
| 21:08.56 | dym | Hey all. Would it be possible to just rent a dedicated server, install asterisk on it and then use it for hold lines and call redirection using some random sip provider? |
| 21:09.20 | dym | Am i thinking totally out of phase? This should be easily doable, right? |
| 21:14.34 | UnixDev | dym: yes |
| 21:14.47 | *** join/#asterisk sam555 (~chatzilla@udp124488uds.hawaiiantel.net) |
| 21:14.51 | sam555 | hello all! |
| 21:14.51 | UnixDev | you could also do it with a cheap(er) vps |
| 21:15.21 | sam555 | trying to set up an asterisk pbx and I wanted to buy a panel to plug a bunch of phone lines into |
| 21:15.34 | sam555 | i wanted to do this without having to wire them in with a BIC panel |
| 21:15.41 | sam555 | i just wanted to plug the rj11 directly in |
| 21:15.43 | *** join/#asterisk wonderworld (~ww@port-92-201-85-236.dynamic.qsc.de) |
| 21:15.49 | sam555 | anyone know of such a panel and what it would be called? |
| 21:16.23 | breardo | a shitload of ATAs? I guess.. |
| 21:17.34 | sunfone | You want an RJ11 patch panel with an amphenol connector to a block, correct? |
| 21:17.52 | sunfone | I have one here gathering dust I could sell you :) |
| 21:18.01 | sunfone | I'll even through in the block ;) |
| 21:19.08 | sunfone | s/through/throw |
| 21:19.57 | sam555 | sunfone: sounds like it! Just didn't know the name |
| 21:20.23 | sam555 | sunfone: we were looking for 2, do you have a a image I could look at? |
| 21:24.46 | dym | UnixDev: awesome. thanks |
| 21:26.44 | WiretapWork_ | sam555, we use standard RJ45 patch panels and only connect pins 4 and 5 |
| 21:26.51 | WiretapWork_ | works fine with RJ11 jacks |
| 21:30.17 | sam555 | Wiretap k: I'm trying to get an image of this so I can purchase such |
| 21:30.39 | sam555 | the front of this panel excepts rj11 jacks and the back we wire the pires directly to the back of the panel (the non exposed side) |
| 21:30.56 | gruvfunk | Help: Anyone know an ITSP with Toll Free DID numbers in Italy and/or Germany? |
| 21:31.23 | WIMPy | What kind of toll free? |
| 21:31.25 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-094-216-162-118.pools.arcor-ip.net) |
| 21:31.58 | gruvfunk | Numero Verde (a number Italian citizens can call free of charge, billed back to the toll free number's owner) |
| 21:32.07 | WiretapWork_ | sam555, the front accepts modular jacks, from RJ11 through to RJ45 |
| 21:32.17 | WiretapWork_ | sam555, the back is a punchdown connect |
| 21:32.24 | WiretapWork_ | standard datacentre patch panel |
| 21:32.31 | WiretapWork_ | available from 16 to 48 port |
| 21:32.38 | sam555 | Wiretap gotcha! |
| 21:32.56 | WiretapWork_ | if you order an 'rj45 patch panel' you will recieve exactly that |
| 21:33.01 | gruvfunk | or in Germany: "Null-achthunderter Nummern" |
| 21:33.02 | sam555 | ok, wasn't sure if I could use the standard patch panel for rj45 when dealing with punching down phone lines and using rj11 |
| 21:33.51 | WIMPy | gruvfunk: Usually you get the 800 number for some service provider that just forwards it to some normal number. |
| 21:33.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 21:34.29 | gruvfunk | WIMPy: looking to avoid forwarding, would like true trunk to * |
| 21:35.49 | WIMPy | gruvfunk: nummerndirekt.de I don;t remember the other one ATM. |
| 21:37.05 | gruvfunk | thx WIMPy |
| 21:40.15 | WIMPy | Thinking about that, I might have seen something abot SIP at corazon as well. |
| 21:41.19 | jc319 | Any ideas why do I receive this warning? >> [Jun 15 22:40:22] WARNING[11211]: app_dial.c:2041 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 21:43.55 | *** join/#asterisk mclaro (~mclaro@207.204.232.112) |
| 21:49.32 | UnixDev | Hi, I'm using asterisk 1.8, which we upgraded to from 1.6... but for some reason, inbound calls are being re-invited when I specifically have canreinvite=no and directmedia=no on the peer, this only seems to be a problem for inbound calls that get transferred, asterisk wants to re-invite every time it transfers and/or hold's the call...how can I stop this behavior? it did not happen in 1.6 or 1.4 |
| 21:50.25 | *** join/#asterisk lulzsec (mouse@2001:b18:4059:0:fd44:9a7b:e679:4512) |
| 21:51.20 | lulzsec | good evening |
| 21:52.25 | WiretapWork_ | sam555, yep, standard panel will be absolutely fine |
| 21:52.38 | WiretapWork_ | sam555, we have one panel here that is half phonelines, half ethernet |
| 21:53.40 | sam555 | WiretapWork_: question: We have a sangoma card in our PBX box that has 24 wires for 12 pairs. |
| 21:54.09 | sam555 | We are trying to figure out how we should wire the 12 pairs so that it's "clean" and then connect that to the patch panel. |
| 21:54.30 | sam555 | The wires that come from the card are only like 1 foot long and we need the wires to go further |
| 21:54.31 | WiretapWork_ | does it terminate in 12 RJ45 or? |
| 21:54.35 | WiretapWork_ | ah, right |
| 21:54.44 | sam555 | they are just look wires right now |
| 21:54.45 | WiretapWork_ | you have two options I guess |
| 21:54.50 | sam555 | *loose |
| 21:55.11 | WiretapWork_ | you can punch them into a block and then run a 12 pair feeder or a bunch of cat5 away from that |
| 21:55.25 | WiretapWork_ | or you can use scotchlok connectors to extend the cable |
| 21:55.32 | WiretapWork_ | I personally would go with the block |
| 21:55.37 | WIMPy | gruvfunk: Portunity.de |
| 21:55.39 | WiretapWork_ | mount it to the back or side of the PBX |
| 21:56.00 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:56.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:56.19 | sam555 | would the 12 pair feeder be like an amphenol connector? |
| 21:56.48 | WiretapWork_ | eh? |
| 21:57.01 | sam555 | let me show you |
| 21:57.04 | WiretapWork_ | 12pair feeder is just going to be a teletrunk cable |
| 21:57.09 | WiretapWork_ | with 12 pairs in it |
| 21:57.18 | sam555 | i see |
| 21:57.41 | gruvfunk | thanks WIMPy |
| 21:58.47 | WiretapWork_ | http://onegoldensquare.com/images/uploads/2009/09/Step-Two.jpg <-- block |
| 21:58.49 | WIMPy | gruvfunk: That's still not the one, I was thinking about, but even with google, I can't recall the name :-( |
| 21:58.50 | WiretapWork_ | http://www.computercablestore.com/images/products/Comtran%20Corporation/0-CT3570.jpg <-- cable |
| 21:58.56 | WiretapWork_ | that's a 25 pair cable |
| 21:59.00 | WiretapWork_ | aka 'feeder' |
| 22:01.20 | *** join/#asterisk sondrove (~mclaro@190.183.222.194) |
| 22:05.04 | sam555 | k |
| 22:05.53 | sam555 | Wiretap we want to take something like this |
| 22:05.54 | sam555 | http://www.twacomm.com/catalog/ICC_25-pair-cable-assembly.htm?sid=6A2F43D1A68BDC218EE4054DB9492665 |
| 22:06.02 | sam555 | and plug it into possibly this |
| 22:06.16 | sam555 | http://www.twacomm.com/catalog/model_AT450.htm?sid=6A2F43D1A68BDC218EE4054DB9492665 |
| 22:06.28 | WiretapWork_ | sam555, is that the connector the sangoma card has? |
| 22:07.12 | WiretapWork_ | what you will want, in that case, is something that starts with one of those and terminates in bare wire at your patch panel |
| 22:08.04 | sam555 | no the sangoma card is a db 25 cable with a male end on the card and loose wires on the other end |
| 22:08.27 | WiretapWork_ | ah, I see |
| 22:08.50 | WiretapWork_ | so you want something that has a DB25 on one end and bare wires on the other, but is long enough to reach |
| 22:08.59 | WiretapWork_ | I'd suggest you may find yourself making that yourself |
| 22:18.41 | sam555 | WiretapWork_: yes, that's what were trying to figure out |
| 22:20.48 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 22:25.23 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 22:25.26 | cj | hey folks |
| 22:25.51 | WiretapWork_ | aloh |
| 22:26.12 | cj | asterisk is using its own address as the SDP / Connection Address. I would have expected it to use the calling phone's address. |
| 22:26.44 | cj | this is with direct rtp set to on |
| 22:27.22 | cj | directmedia=yes , directrtpsetup=yes |
| 22:27.36 | cj | are my expectations off? |
| 22:29.09 | dym | What are the strongest GUI's to asterisk nowadays? I remember back in the days there was FreePBX and stuff like that |
| 22:30.00 | cj | http://paste2.org/p/1472306 |
| 22:36.06 | sam555 | Wiretap thanks for the input!!! |
| 22:36.13 | sam555 | we have to go to lunch, but we'll look for you later |
| 22:36.31 | cj | looks like that was just the initial invite. the re-invite included the correct address. |
| 22:36.43 | *** join/#asterisk hdiogenes (~humberto@201.76.153.10.wimax.digizap.com.br) |
| 22:45.52 | linuxgecko | dym: there are other channels that focus on gui interfaces to asterisk, but IIRC, this channel sticks pretty rigidly to raw .conf file editing and the like. |
| 22:46.17 | dym | i just wanted to know whats new |
| 22:46.20 | dym | nothing specific |
| 22:52.33 | linuxgecko | i think there might be some things, but i think frepbx might be among the widest used gui's |
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| 23:18.15 | cusco | ~book |
| 23:18.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 23:21.22 | WiretapWork_ | I've never met a guy who likes to refer to himself in first person plural like sam555 does before |
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