| 00:01.26 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:12.33 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 00:34.42 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tglvrzgkqveobwsn) |
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| 01:10.17 | *** join/#asterisk jetlag (~jetlag@pool-71-168-254-144.cmdnnj.east.verizon.net) |
| 01:19.23 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 01:38.42 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 01:39.55 | sawgood | Hi: if I wanted voicemail.conf to do two externnotify=/etc/path, could I put one on each line (I've done that and I do not think it is working) |
| 01:40.15 | sawgood | externnotify=dothis |
| 01:40.20 | sawgood | externnotify=dothat |
| 01:40.28 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 01:44.33 | sawgood | nevermind ... I re-worked things and its ok now (sorry) |
| 01:48.53 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:18.57 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
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| 02:47.11 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 02:57.21 | LiuYan | hmm, AMI HTTP port changed to 8080 from 8088 since 1.8.11 ? |
| 03:05.39 | p3nguin | Change it to whatever you like. |
| 03:08.08 | jdoe | -1. |
| 03:08.22 | LiuYan | thanks, just curious about this change |
| 03:12.24 | p3nguin | It looks like 5038 by default for me. |
| 03:13.05 | p3nguin | Oh, wait, no... |
| 03:13.06 | LiuYan | 5038 is the TCP port |
| 03:13.25 | p3nguin | My bad... http.conf, not manager.conf |
| 03:14.33 | p3nguin | It defaults to 8088 in 1.8.10.1 |
| 03:14.45 | p3nguin | I don't have 1.8.11.0 yet. |
| 03:24.42 | LiuYan | found the change from source code => http://svnview.digium.com/svn/asterisk/tags/1.8.11.0/main/http.c?revision=360776&view=markup |
| 04:10.45 | *** join/#asterisk Yourname` (~yourname@unaffiliated/yourname/x-837320) |
| 04:11.42 | Yourname` | Hello. I remember Aastra 57i, and Polycom IP-330 being the good ones with Asterisk, auto-provisioning, scripts, PoE, etc in 2006-2007. Are there any better ones in the market today? |
| 04:12.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 04:25.09 | [TK]D-Fender | Yourname`: 330 is entry level, and technically discontinued. |
| 04:25.32 | Yourname` | So what's new and good like they used to be, now? |
| 04:25.38 | [TK]D-Fender | Yourname`: Poly is quality king. Aastra wins on soft-keys most of the time, but sacrificed build quality |
| 04:25.47 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 04:26.03 | [TK]D-Fender | Linksys is the international value backup option. |
| 04:26.22 | [TK]D-Fender | Snom is an odd duck... usually too pricey to find its right place vs the rest |
| 04:27.37 | Yourname` | Right. So, what's good now? Things like Polycom 331, etc? |
| 04:28.10 | [TK]D-Fender | Current models |
| 04:28.50 | p3nguin | I don't hate the 550/650. |
| 04:29.13 | p3nguin | I installed some of those a few months ago. |
| 04:30.07 | [TK]D-Fender | 550 is another odd duck. No expansion like the 650, and too pricey. the 450 invades nicely at the "make them feel important" level and is space saving. |
| 04:30.26 | [TK]D-Fender | Almost anywhere I'd look at a 331 I'd put a 335 instead. |
| 04:30.40 | [TK]D-Fender | 321 is the uber budget phone where wiring is made to match |
| 04:33.40 | Yourname` | Ok, I'll check 335 out. Thanks [TK]D-Fender! |
| 04:33.54 | Yourname` | I liked the Aastra 57i back then, is it still available? |
| 04:34.07 | *** join/#asterisk nixon (~Nixon@wikimedia/Nixon) |
| 04:38.11 | [TK]D-Fender | Yourname`: I believe so... I had plenty of gripes with mine though.... it's a matter of taste |
| 04:38.41 | *** part/#asterisk nixon (~Nixon@wikimedia/Nixon) |
| 04:39.08 | Yourname` | Great man thanks |
| 04:42.18 | Yourname` | Looks like it's not called 6757i.. |
| 04:42.23 | Yourname` | now* |
| 04:42.55 | *** join/#asterisk cstachris (~chrismylo@202.182.147.96) |
| 04:47.37 | Yourname` | grrr can't find that little expansion slot for it |
| 04:57.42 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
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| 06:28.02 | urvg4 | [2012-03-30 07:22:59] WARNING[23678]: res_musiconhold.c:659 monmp3thread: poll() failed: Interrupted system call |
| 06:28.02 | urvg4 | hi,please how I fix this without stopping moh module? |
| 06:56.06 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 07:06.40 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 07:07.43 | *** join/#asterisk BeeBuu (3afc49a4@gateway/web/freenode/session) |
| 07:08.32 | MarKsaitis | does asterisknow include asterisk 10.3? |
| 07:10.09 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 07:11.18 | BeeBuu | hi,anyone tell me what's the mean when i type "manager show connected" in CLI? |
| 07:12.04 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 07:12.25 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
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| 07:14.25 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 07:14.44 | kaldemar | BeeBuu: "core show help manager show connected" |
| 07:15.18 | kaldemar | MarKsaitis: doubt it. it usually does not have the most recent versions. |
| 07:15.39 | MarKsaitis | why not =))))) it's the easiest way to preview |
| 07:16.28 | kaldemar | nope. 1.7 has 1.4 or some 1.6 and 2.0 beta has 1.8. |
| 07:17.22 | kaldemar | easiest? a bit overkill to install a whole distro to preview some software. you can install/uninstall a version of asterisk that uses your current configs with very little effort. |
| 07:21.04 | BeeBuu | <PROTECTED> |
| 07:21.17 | BeeBuu | what's this mean? |
| 07:21.18 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:24.05 | BeeBuu | Elapsed ? what's it? |
| 07:24.26 | kaldemar | is the concept of elapsed time familiar to you? |
| 07:25.18 | kaldemar | it means the amount of seconds the user has been connected to the manager interface. |
| 07:25.30 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:28.23 | MarKsaitis | kaldemar, 2.0 beta asterisknow you mean has 1.8 asterisk? where do I download it from then |
| 07:30.00 | *** join/#asterisk smarac (~drazen@ghost.verat.net) |
| 07:30.28 | smarac | hello |
| 07:30.29 | kaldemar | MarKsaitis: http://www.asterisk.org/asterisknow |
| 07:30.42 | MarKsaitis | thanx, already downloading |
| 07:31.07 | smarac | i'm looking for a way to limit calls to a certain crediti limit |
| 07:31.27 | smarac | is there a way to implement this |
| 07:32.31 | smarac | ? |
| 07:34.15 | kaldemar | smarac: serach for calling card or prepaid |
| 07:35.38 | smarac | thanks |
| 07:35.51 | smarac | searching |
| 07:36.06 | smarac | but i need to implement this for postpaid service also |
| 07:37.19 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:39.23 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 07:40.09 | *** join/#asterisk timahvo1 (~rogue@197.176.86.104) |
| 07:41.05 | MarKsaitis | so whats so much different between 1.8 and 10.3? |
| 07:41.52 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:41.54 | schmidts | good morning |
| 07:43.56 | kaldemar | MarKsaitis: for a brief listing: http://svn.digium.com/svn/asterisk/tags/10.3.0/CHANGES |
| 07:44.12 | MarKsaitis | thanx |
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| 08:11.30 | *** join/#asterisk awk (~awk@trancehouse.co.za) |
| 08:11.59 | awk | hi, this may sound dum to most, but I have a windows DHCP server, handing out option 66 for snom http://ip/snom/{mac}.html |
| 08:12.13 | awk | now I want to provision polycom too with just the IP, what option can I use? |
| 08:12.26 | awk | Polycom is using TFTP so I just need to hand out the ip address |
| 08:13.38 | *** join/#asterisk nutxase (~nutxase@unix-02.lair.co.za) |
| 08:13.43 | nutxase | anyone here who can help with Dadhi? |
| 08:14.23 | *** join/#asterisk kieppie (~jaco@ip-58-28-154-35.static-xdsl.xnet.co.nz) |
| 08:14.36 | kieppie | hi guys. |
| 08:15.00 | kaldemar | ~ask |
| 08:15.00 | 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. |
| 08:15.36 | nutxase | when i call my main pstn number it rings in my ear but doesnt hit the asterisk box and if i run dahdi show status it says it is ok |
| 08:16.31 | kieppie | I've gotten myself an overseas trunk provider to connect to so that friends & family there can call me cheaply, and I'm making calls out over it ok. unfortunately I'm unable to receive inbound calls, and we suspect it may a timeout issue on the remote end. is there anything I can set on my trunk registration that could influence that setup more favourably? |
| 08:19.47 | *** join/#asterisk singler (~singler@beta.kirneh.eu) |
| 08:19.59 | kaldemar | nutxase: what technology are you using? |
| 08:20.12 | kaldemar | nutxase: as in analog/BRI/PRI |
| 08:20.38 | nutxase | BRI |
| 08:21.29 | *** join/#asterisk singler (~singler@beta.kirneh.eu) |
| 08:21.45 | kaldemar | nutxase: pastebin a CLI output of a call with verbosity and pri debug. |
| 08:22.30 | nutxase | it wont let me i get this error No PRI running on span 1 |
| 08:22.42 | nutxase | when i do pri show spans i dont see anything |
| 08:23.27 | nutxase | cli wont show an error when i call as the call does not even hit the pbx |
| 08:28.42 | kaldemar | do you have libpri installed? |
| 08:28.52 | nutxase | yes |
| 08:30.33 | kaldemar | interesting. how did you configure the card and channels in DAHDI and chan_dahdi? use a pastebin |
| 08:31.06 | nutxase | i did dahdi_genconf |
| 08:31.20 | nutxase | http://pastebin.com/HnYgHQZE |
| 08:38.01 | *** join/#asterisk flyingbull (~Adium@cpe-065-190-158-078.nc.res.rr.com) |
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| 08:47.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:47.19 | Chainsaw | http://www.asterisk.org/node/51817 <- So is it out or is it not out? 10.3.0? |
| 08:49.15 | kaldemar | seems to be. |
| 08:49.40 | *** join/#asterisk din3sh (~din3sh@41.212.203.202) |
| 08:50.00 | kaldemar | seems i'm not authorized to view the secret node you pasted. the 10.3.0 tarball is on the downloads server though. |
| 08:51.51 | Chainsaw | kaldemar: Well yes. I just like to know what the changes have been. |
| 08:52.16 | Chainsaw | kaldemar: In case of a license change, signing over your first born, etc. You never know. |
| 08:55.53 | *** join/#asterisk jsjc (~Adium@51.Red-79-146-23.dynamicIP.rima-tde.net) |
| 09:01.11 | kaldemar | http://svn.digium.com/svn/asterisk/tags/10.3.0/ :P |
| 09:04.39 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 09:07.57 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 09:08.03 | Chainsaw | kaldemar: Not the level of detail I was looking for. |
| 09:13.57 | FlashDeluxe | hi! I got 40 sip phones connected to asterisk 1.8.1 witch is connected to a ptp with 40 extensions and i configured the callerids in sip.conf. But if i want to dial out, the callerid "0" from the main office is taken by every sip phone :( How can i force asterisk to take the callerid i configured in sip.conf? |
| 09:15.50 | kaldemar | FlashDeluxe: your asterisk sends out what you have in sip.conf unless you set something else in dialplan. that's all you can do. |
| 09:16.32 | kaldemar | or maybe i didn't quite understand what you meant by "take". |
| 09:16.35 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f82a.pool.mediaWays.net) |
| 09:17.34 | *** join/#asterisk wolfy (~lonewolf@fedora/wolfy) |
| 09:19.10 | FlashDeluxe | klademar: i guess you understood me right meant that the 12345678-0 is seen by the called party, althoug i configured the 12345678-77 in sip.conf |
| 09:23.31 | FlashDeluxe | i have to add, that if someone calls the 12345678-77 he gets connected with the right phone |
| 09:32.20 | *** join/#asterisk bn-7bc (bjarne@cl-1271.sto-01.se.sixxs.net) |
| 09:32.28 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 09:37.36 | *** join/#asterisk nutxase (~nutxase@unix-02.lair.co.za) |
| 09:38.09 | nutxase | anyone see the problem where an incoming dahdi call rings the phone but when u answer no one is there and the person on the other side is still hearing ringing? |
| 09:38.24 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 09:45.58 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 09:52.26 | nutxase | any ideas? |
| 09:53.47 | FlashDeluxe | nutxase: logs? |
| 09:55.50 | nutxase | http://pastebin.com/TK41VS94 |
| 09:56.01 | nutxase | shows answered but it isnt answered on the other side :/ |
| 09:59.48 | nutxase | i can not figure it out :/ |
| 10:00.40 | FlashDeluxe | are there any log from the incoming side? |
| 10:01.25 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 10:01.27 | nutxase | that is the incoming side |
| 10:01.50 | nutxase | if i call off my cellular phone for eg it keeps ringing in my ear so asterisk is not sending the answer signal to the other side :/ |
| 10:05.26 | nutxase | any ideas? |
| 10:05.55 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 10:15.14 | din3sh | anyone tried digium phones yet? |
| 10:20.37 | Chainsaw | They look like Polycom OEMs to me. |
| 10:22.05 | nutxase | anyone know which version of 1.8 is the most stable? |
| 10:29.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:31.15 | Chainsaw | nutxase: The latest release, generally. So that should be 1.8.11.0 |
| 10:31.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:31.57 | j0b | Chainsaw: like you nick |
| 10:32.19 | j0b | when i was younger i was a big an of "the texas chansaw massacre" -movies |
| 10:32.25 | j0b | big fan |
| 10:33.11 | Chainsaw | smiles |
| 10:34.04 | j0b | "chansaw massacre" |
| 10:34.07 | j0b | nice typo;) |
| 10:34.16 | j0b | anyways |
| 10:41.24 | Chainsaw | It's a reference to the computer game Doom II rather than the movie though :) |
| 10:43.37 | *** join/#asterisk PBXman (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 10:43.47 | PBXman | hello |
| 10:45.41 | Chainsaw | Good morning PBXman. |
| 10:50.15 | j0b | Chainsaw: nice. tested doom 3? |
| 10:50.40 | Chainsaw | j0b: Control issues. |
| 10:50.53 | j0b | hm |
| 10:51.00 | Chainsaw | j0b: If a monster appears out of the dark, I want to shoot it. Not go *bonk* with the flashlight. |
| 10:51.10 | j0b | anyways, i get a little nostalgic talking about old games |
| 10:51.20 | Chainsaw | j0b: And I have to carry around a flashlight, because everything is *DARK*. |
| 10:51.25 | j0b | hehe ok |
| 10:51.31 | Chainsaw | Good intro though. I liked that. |
| 10:51.55 | j0b | day of tentacle and that shit |
| 10:51.58 | Chainsaw | The best game ever made remains UT2004. 32 skilled bots on ONS-Torlan. |
| 10:51.59 | j0b | 11th hour |
| 10:52.00 | j0b | myst |
| 10:52.19 | j0b | no idea what youre talking about there:p |
| 10:52.31 | Chainsaw | Too new for you? |
| 10:52.35 | j0b | yeah |
| 10:52.41 | j0b | sort of |
| 10:52.49 | j0b | well i play bf3 from time to time |
| 10:52.53 | Chainsaw | Yet you know Doom III. *peers at you suspiciously* |
| 10:53.02 | j0b | i know it exists:p |
| 10:55.25 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 10:55.55 | nutxase | anyone who can help me ill appreciate it forever |
| 10:56.33 | j0b | nutxase: i can try |
| 10:56.44 | nutxase | thanks :) |
| 10:57.12 | nutxase | when an incoming call comes via dahdi it rings and when i answer the cli shows answer but on the other side it still rings |
| 10:57.32 | bulkorok | hi... I want to set a global variable (defined in [globals]) everytime when an extension is hit with var + 1... I made this now: same => n,Set(VAR=$[${VAR}+1]) but the var is not incremented... any ideas?! |
| 10:57.36 | nutxase | even if i add s,1,Answer doesn make any diff |
| 10:58.03 | *** join/#asterisk krotos (~d3v1l@host124-100-dynamic.15-87-r.retail.telecomitalia.it) |
| 10:58.07 | krotos | hi all ;) |
| 10:59.53 | *** join/#asterisk simonmx (~simonmox@80-45-44-37.static.dsl.as9105.com) |
| 11:00.27 | simonmx | Can anyone remind me how to force unload a module from the CLI? |
| 11:00.44 | krotos | module unload <modulename> |
| 11:00.59 | simonmx | Thanks korotos, but is there a force option? |
| 11:02.30 | kaldemar | module unload -f <module> |
| 11:02.46 | kaldemar | may not exist in your version, at least 10 has the -f option. |
| 11:03.15 | kaldemar | there's also -h. see "core show help module unload" |
| 11:04.58 | nutxase | any ideas? |
| 11:05.12 | elred_ | nutxase, what driver are you using ? |
| 11:05.22 | elred_ | chan_dahdi ? chan_mISDN ? |
| 11:05.28 | elred_ | chan_zaptel ? |
| 11:05.33 | elred_ | mISDN package ? |
| 11:05.37 | elred_ | it look like a driver problem to me |
| 11:05.59 | elred_ | also |
| 11:06.01 | elred_ | not only answer |
| 11:06.04 | elred_ | but play something |
| 11:06.13 | kaldemar | bulkorok: you're incrementing a channel variable, not a global one. Noop(${INC(GLOBAL(VAR))}) or Set(GLOBAL(VAR)=$[${VAR} + 1]) |
| 11:06.14 | elred_ | to try |
| 11:06.46 | nutxase | chan_dahdi |
| 11:07.01 | nutxase | tried playing something and does same thing |
| 11:07.15 | nutxase | even re-compiled dahdi |
| 11:09.51 | krotos | kaldemar: have you ever had a strange problem with fax -spandsp in udptl? i'have a packet wit wrong ip header : UDPTL (SIP/0510988299): packet to 130.2.0.0:5004 (type 0, seq 55, len 25) |
| 11:10.11 | kaldemar | krotos: no. |
| 11:10.14 | krotos | 130.2.0.0 is not my ip..i think is a problem on the router that does nat |
| 11:10.25 | krotos | fucking endian firewall.. |
| 11:21.29 | nutxase | fucking hate Dahdi.............. |
| 11:22.29 | elred_ | nutxase, check the configuration of dahdi |
| 11:22.31 | schmidts | nutxase use zaptel instead |
| 11:22.33 | nutxase | please can someone assist me ill be so gratefullll |
| 11:22.39 | elred_ | you may have reversed some NT / TE or something like that |
| 11:23.58 | nutxase | where do i change it |
| 11:24.38 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 11:25.08 | Dovid | random OT question. whats a smaller way of saying "how long something is good for" ? |
| 11:31.30 | nutxase | any ideas guys/ |
| 11:32.01 | *** join/#asterisk oej (~olle@2001:df8:0:16:1ca5:cdb:ec9f:7257) |
| 11:37.55 | nickfennell | Dovid, lifetime |
| 11:39.49 | simonmx | Thanks kaldemar for the unload -f, it worked. |
| 11:39.51 | nutxase | has anyone seen this dahid error |
| 11:39.58 | nutxase | has anyone seen this dahdi error |
| 11:40.39 | Chainsaw | nutxase: Which one? |
| 11:41.08 | nutxase | when i call in it rings so i answer and it still rings in the callers ear even though asterisk says dahdi answered |
| 11:41.52 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:42.49 | Chainsaw | nutxase: And this is an analog line? |
| 11:42.54 | nutxase | nope bri |
| 11:43.17 | Chainsaw | nutxase: I'm amazed you've managed to break call signalling to that degree. |
| 11:43.52 | WIMPy | Yes, sounds hard to achieve. |
| 11:44.17 | Chainsaw | WIMPy: On an analog line I could do it. |
| 11:44.26 | nutxase | how though all ive done is run dahdi_genconf |
| 11:44.47 | Chainsaw | WIMPy: But on BRI, to have the circuit up and fail to accept a call, without getting a drop on the other side? |
| 11:45.41 | WIMPy | Pretty impossible. But dahdi is always good for interesting behavoiur. |
| 11:46.28 | nutxase | and cant use misdn because i get |
| 11:46.29 | nutxase | [2012-03-30 13:46:24] ERROR[10789]: chan_misdn.c:11196 load_module: Unable to initialize mISDN |
| 11:48.16 | WIMPy | Which mISDN? |
| 11:48.57 | WIMPy | chan_misdn only works with the old mISDN. |
| 11:49.12 | WIMPy | For the current version you need LCR. |
| 11:49.16 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 11:49.31 | nutxase | oh so i have to use dahdi for bri |
| 11:50.01 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 11:50.31 | WIMPy | You most probably don't have to. |
| 11:51.02 | nutxase | what else can i use? |
| 11:51.29 | WIMPy | It's all explained in the link I posted. |
| 11:52.39 | elred_ | nutxase, you should get it fine with LCR |
| 11:53.18 | nutxase | ok cool :) |
| 11:53.31 | nutxase | got dahdi working :D:D:D:D:D:D had to be euroisdn |
| 11:53.42 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:53.45 | Chainsaw | nutxase: Using the correct signalling protocol does help, yes. |
| 11:53.54 | nutxase | lol |
| 11:55.45 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-17-164.bois.qwest.net) |
| 11:59.16 | *** join/#asterisk slingr (~phunguy@will.one.day.hack-the-pla.net) |
| 12:08.13 | *** join/#asterisk leofishman (~leofishma@186.18.37.215) |
| 12:08.29 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 12:10.15 | *** join/#asterisk Syco_ (53f4d1f2@gateway/web/freenode/ip.83.244.209.242) |
| 12:10.41 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:10.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:13.22 | leofishman | hi |
| 12:14.15 | leofishman | Is there are any one here that can help me with an asterisk that is not running? |
| 12:15.25 | leofishman | I get a core dump |
| 12:15.31 | leofishman | <PROTECTED> |
| 12:15.31 | leofishman | <PROTECTED> |
| 12:15.32 | leofishman | <PROTECTED> |
| 12:15.32 | leofishman | <PROTECTED> |
| 12:15.32 | leofishman | <PROTECTED> |
| 12:15.32 | leofishman | <PROTECTED> |
| 12:15.32 | leofishman | Segmentation fault |
| 12:17.44 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:24.09 | Kobaz | sweet |
| 12:24.14 | Kobaz | my developer talk is submitted! |
| 12:24.23 | Kobaz | i didn't miss the deadline this year!! |
| 12:24.43 | Kobaz | leifmadsen: high five |
| 12:24.54 | leifmadsen | Kobaz: huzzuh! |
| 12:25.01 | leifmadsen | I haven't even thought about doing a talk .... |
| 12:25.26 | leifmadsen | I might just bring along some talks and let Bryan know I'm available as a filler |
| 12:25.45 | Kobaz | now we just need to make sure that some new code gets into trunk, so it's available in asterisk 111 |
| 12:25.54 | leifmadsen | heh |
| 12:25.59 | Kobaz | who do i need to pay off? |
| 12:26.06 | leifmadsen | Kobaz: Asterisk 111? :) |
| 12:26.11 | Kobaz | yeah sure |
| 12:26.18 | Kobaz | whatever number piques your fancy |
| 12:26.27 | Kobaz | maybe even asterisk 11 |
| 12:26.27 | leifmadsen | heh, you have stuff for Asterisk 11 now right? |
| 12:26.31 | Kobaz | yeah |
| 12:26.41 | leifmadsen | hmmm, well May is the announcement for feature freeze |
| 12:26.47 | Kobaz | Who do I need to send brownies and beer to |
| 12:26.50 | leifmadsen | and I think June is feature freeze month |
| 12:26.56 | Kobaz | yeap |
| 12:26.58 | Kobaz | plenty of time |
| 12:27.02 | Kobaz | the code is all done |
| 12:27.09 | leifmadsen | Kobaz: good question... you could try piquing russellb's interest with beer :) |
| 12:27.13 | Kobaz | except for whatever teaks you guys come up with |
| 12:27.14 | leifmadsen | Kobaz: I hear you though |
| 12:27.27 | leifmadsen | Kobaz: I don't work at Digium anymore... so.... "you guys" is probably not me :) |
| 12:27.33 | Kobaz | oh |
| 12:27.35 | Kobaz | srsly? |
| 12:27.36 | Kobaz | that's sad |
| 12:27.40 | leifmadsen | since January |
| 12:27.48 | leifmadsen | I work at CoreDial now full time |
| 12:28.50 | Kobaz | interesting interesting |
| 12:29.12 | leifmadsen | good chance I'll still be at AstriCon though |
| 12:29.17 | Kobaz | good good |
| 12:29.36 | Kobaz | it'll be nice in Atlanta in October |
| 12:29.47 | *** join/#asterisk albertoandrade (~albertoan@177.16.186.99) |
| 12:29.59 | leifmadsen | but ya... hmmm, not sure who you can bug, other than maybe malcolmd to make sure it gets on his list of, "needs to be reviewed and merged before Asterisk 11 deadline" |
| 12:30.02 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:30.11 | leifmadsen | that's probably your best bet to make sure someone is assigned to review your code |
| 12:30.20 | leifmadsen | Kobaz: I don't know... I've been to Atlanta :) |
| 12:30.43 | Kobaz | yeah |
| 12:30.48 | Kobaz | i got I think one person so far |
| 12:31.00 | Kobaz | i need to get some attention on the other reviews |
| 12:31.07 | leifmadsen | true |
| 12:31.11 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:32.41 | Kobaz | coredial has a 212 number |
| 12:32.49 | Kobaz | are you telecommuting, or working in nyc? |
| 12:33.04 | Kobaz | oh wait, that's a 5 |
| 12:33.05 | Kobaz | 215 |
| 12:33.12 | leifmadsen | Kobaz: CoreDial is in Blue Bell, PA, and I'm telecommuting like I always have :) |
| 12:33.26 | leifmadsen | I use google hangouts a lot |
| 12:33.34 | Kobaz | i've heard it's cool |
| 12:33.37 | leifmadsen | we use it for meetings and other such things |
| 12:34.03 | leifmadsen | Kobaz: join up, there is usually a group of us who join every day around lunch time or earlier and just leave it running on mute |
| 12:34.10 | leifmadsen | since a lot of us telecommunte |
| 12:34.17 | Kobaz | ah okay |
| 12:34.17 | leifmadsen | telecommute* heh |
| 12:34.32 | leifmadsen | just make sure Russell, or myself are in your circles and you should see it |
| 12:35.41 | *** part/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 12:35.47 | Kobaz | k |
| 12:35.48 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:36.00 | Kobaz | i don't even have this new fangled google+ |
| 12:36.30 | leifmadsen | heh |
| 12:36.34 | leifmadsen | pretty easy to obtain |
| 12:38.22 | Kobaz | i'm sure it's a click and a clack and sign you life away and poof there you go |
| 12:38.29 | leifmadsen | indeed :) |
| 12:41.04 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 12:41.52 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
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| 12:57.41 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 13:05.33 | *** join/#asterisk serafie (~erin@nat/digium/x-hnooxiriovpujahh) |
| 13:07.09 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-wiyehcxsdorkkvxt) |
| 13:07.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:07.29 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 13:08.00 | *** join/#asterisk fnwhat (~fnwhat@74.203.128.226) |
| 13:08.07 | McBoingBo | trying to put someone's phone into "vacation" mode I guess, by sending Dial straight to Voicemail, can I do that? |
| 13:10.02 | McBoingBo | nm stupid question lol |
| 13:10.17 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 13:10.57 | sereal-work | Hello, i'm trying to setup a fax service with asterisk. Some faxes appear to be timing out, and I have been told that on the fax machine end they are consistantly redialing. |
| 13:11.04 | sereal-work | Does anyone know of any solution to this? |
| 13:11.15 | sereal-work | faxes connected to voip appear to be working. |
| 13:11.28 | sereal-work | it's the pstn ones that seem to be having the problme |
| 13:12.49 | sereal-work | res_fax.so Generic FAX Applications 1 |
| 13:12.49 | sereal-work | res_fax_spandsp.so Spandsp G.711 and T.38 FAX Technologies 140 |
| 13:14.12 | fnwhat | I am getting one way audio after hold with Asterisk 1.6.2.9-2 sip trunk to CUCM/8.6(2a). Asterisk reports call still on hold after release. I have debugs and dumps of everything if somebody could lend me a hand. |
| 13:14.35 | *** join/#asterisk leofishman (~leofishma@200.80.231.178) |
| 13:15.10 | albertoandrade | fnwhat: you're over NAT? |
| 13:16.13 | fnwhat | albertoandrade, yup. Its the painful part. my PSTN trunk goes back out through an ASA. I have nat=yes set, and a local and external defined. |
| 13:16.49 | fnwhat | I also set canreinvite=no , and I can see p2p doing its thing. |
| 13:19.22 | fnwhat | Its confusing becuase I think if it was a NAT issue then asterisk would be reporting the call is not on hold, but it never takes the call off hold. sip show channels says the call is still on hold.... |
| 13:20.02 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-126.telkomadsl.co.za) |
| 13:21.17 | albertoandrade | fnwhat: 'rtp debug' show's one way rtp flow? |
| 13:21.24 | fnwhat | correct |
| 13:22.06 | albertoandrade | so..i really think that's a NAT issue |
| 13:22.15 | fnwhat | ok |
| 13:22.27 | fnwhat | so the ASA has dumped the UDP stream? |
| 13:22.32 | albertoandrade | your peer are using 'nat=yes' ? |
| 13:22.36 | fnwhat | correct |
| 13:24.33 | albertoandrade | are you using a DSL modem? |
| 13:24.41 | fnwhat | no |
| 13:25.11 | fnwhat | I have a 1g uplink at a datecenter |
| 13:26.34 | fnwhat | I went to the datacenter, plugged a phone in there, tried it, same issues. |
| 13:26.40 | albertoandrade | Do a "show ip nat translation" and look for values for "udp-port" and "port-timeout udp" |
| 13:26.46 | fnwhat | rgr |
| 13:27.11 | fnwhat | brb |
| 13:28.52 | albertoandrade | fnwhat: hm?! |
| 13:30.17 | *** join/#asterisk fnwhat (~fnwhat@96.45.117.68) |
| 13:35.23 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:36.18 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:37.04 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 13:38.25 | fnwhat | albertoandrade: ok I can see the UDP PAT, everything looks good |
| 13:38.28 | wolfy | http://www.youtube.com/watch?feature=player_embedded&v=Q5M3spJLRy8 |
| 13:38.31 | wolfy | sorry |
| 13:41.05 | fnwhat | albertoandre: http://pastebin.com/9gyP9XFG |
| 13:42.15 | fnwhat | address of my phone is 192.168.47.114 address of the cucm 10.110.9.13, asterisk is .11 |
| 13:45.55 | albertoandrade | fnwhat: can you pastebin 'show ip nat translation' (saving your confidential information) ? |
| 13:46.58 | fnwhat | albertoandrade, there is no show ip nat translation on an ASA. Close as I can get is show xlate detail, and I pasted that above |
| 13:47.56 | albertoandrade | ok...we just need to know udp-port and port-timeout values |
| 13:48.01 | fnwhat | rgr |
| 13:48.08 | fnwhat | let me see if I can pull that |
| 13:48.20 | fnwhat | udp port is listed in the dump |
| 13:48.34 | fnwhat | its after the / |
| 13:48.46 | fnwhat | timeout, I will see what I can dig up |
| 13:51.50 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 13:53.43 | fnwhat | albertoandrade, check pm |
| 13:55.39 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 13:55.41 | cusco | hi folks |
| 13:56.24 | cusco | in chat_dahdi.conf I can set a var that applies to all channels going trough dahdi, right? |
| 13:56.52 | cusco | in voip info I see an example: setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep |
| 13:57.54 | cusco | so that means I can use: Set(TIMEOUT(absolute)=3600) |
| 13:58.19 | cusco | to make every call going trough dahdi last only 1hour, right? |
| 13:58.47 | [TK]D-Fender | cusco, I know sip.conf supports it, but am unsure if the other channel drivers suppotr it as well. A while back I knew several did not.... |
| 13:59.07 | albertoandrade | fnwhat: have you 'keep-alive' prmetr on your peers?! |
| 13:59.49 | cusco | hmm I was reading this on http://www.voip-info.org/wiki/view/chan_dahdi.conf - I was only wondering about the Set sintax |
| 14:00.01 | cusco | should I use setvar, or Set() or Set: |
| 14:00.46 | fnwhat | albertoandrade, no |
| 14:01.15 | albertoandrade | can you turn it on? |
| 14:01.19 | fnwhat | sure |
| 14:01.34 | *** join/#asterisk oldhack (~oldhack@cpe-24-28-23-78.austin.res.rr.com) |
| 14:06.50 | fnwhat | albertoandrade, no change |
| 14:07.57 | *** join/#asterisk ctaloi (~ctaloi@nat-66-218-1-193.usadatanet.com) |
| 14:09.03 | [TK]D-Fender | cusco, That does seem to be a straight-up copy of a reference sample config fromt he tarball, so I suppose I'd take it at face value that it works as advertised. Syntax should be "setvar=TIMEOUT(absolute)=3600" |
| 14:17.10 | cusco | thank you [TK]D-Fender |
| 14:19.46 | albertoandrade | fnwhat: sorry...i think i'm not able to help you |
| 14:20.29 | fnwhat | albertoandrade, no worries, its a tough one. Been working on it for a week |
| 14:20.59 | albertoandrade | so, good look.. ;-) |
| 14:21.09 | fnwhat | hehe, I'm gunna need it |
| 14:21.10 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 14:21.55 | fnwhat | It looks like a bug in asterisk, but what do I know. I just wonder what triggers asterisk to understand the call is no longer on hold.... |
| 14:23.11 | *** join/#asterisk ctaloi (~ctaloi@nat-66-218-1-193.usadatanet.com) |
| 14:23.14 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:23.20 | asteriskATmarmuD | hi guys, we are using asterisk 1.4.39.2. dahdi_test with pseudo device (dahdi_dummy) gets us "Worst: 99.991". should we consider buying a sangoma voice time usb? or are the results of dahdi_test good enough? will the results be worse if the system is under load (I just checked while the system was doing nothing) |
| 14:24.33 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:26.39 | [TK]D-Fender | asteriskATmarmuD, Maybe you should answer the simpler question : Do you have an actual PROBLEM? |
| 14:27.38 | kaldemar | asteriskATmarmuD: from "man dahdi_test": Values of 100% and 99.99% Are normally considered a definite pass. Values of 99.98% and 99.97% are probably OK as well. |
| 14:30.01 | onixx99 | Hi all, I am looking for someone that uses asterisk to decode a fast inband dtmf sequence. 50ms tone, 50ms silence. I need to know if this is even possible in asterisk 1.8. |
| 14:30.27 | asteriskATmarmuD | [TK]D-Fender: :) we experience some delays and call drops |
| 14:31.03 | asteriskATmarmuD | [TK]D-Fender: so this would be rather a network issue than a timing issue? |
| 14:32.04 | [TK]D-Fender | asteriskATmarmuD, You haven't given and details about the nature of the calls... |
| 14:33.25 | schmidts | onixx99 there are some well known problems and limitations to such fast dtmf tones, btw it depends on the used rtp codec but normally you will have rtp pakets every 20 ms so getting exactly 50ms will be hard |
| 14:33.46 | asteriskATmarmuD | [TK]D-Fender: one moment please, we are just starting our shift |
| 14:34.01 | *** join/#asterisk WindBack (~quassel@190.220.135.165) |
| 14:36.02 | onixx99 | schmidts : I have a wav recording of the tone sequence. With 1.4, I can even play it thru my cell phone from my computer speakers and it detects ;-)... but on 1.8, 4 digits out of 16 if i'm lucky and played directly rather than thru a wav recording over cell phone |
| 14:36.24 | WindBack | When I have Local channel executing a dialplan part because some sip peer forwarded the call, Is there any way to know the peer who previously forwareded the call from the dialplan? Any variable or function? |
| 14:37.05 | [TK]D-Fender | WindBack, ${BLINDTRANSFER} might hold it. |
| 14:37.14 | asteriskATmarmuD | [TK]D-Fender: we are outbound dialing and putting agents and callee into a meetme room. I don't really get what information you'd like to have. though I am willing to give you anything you need :) |
| 14:37.16 | [TK]D-Fender | WindBack, depending on the precise forwarding mechanism |
| 14:37.58 | WindBack | [TK]D-Fender: the mechanism is sip peer responding with 302 message |
| 14:38.18 | schmidts | onixx99 thats what i meant with well known limitations and problems, AFAIK oej had a dev branch for this problem but i dont know if its allready merged into the normal branch. maybe you should try asterisk 10 or svn trunk version and see if its better there |
| 14:38.21 | *** join/#asterisk myyrdin (~textual@74-93-34-252-Sarasota.hfc.comcastbusiness.net) |
| 14:40.40 | *** join/#asterisk timahvo1 (~rogue@197.176.86.104) |
| 14:42.00 | *** join/#asterisk timahvo1 (~rogue@197.176.86.104) |
| 14:42.11 | onixx99 | schmidts : thanks, I will give those a try. Lastly, how can I influence the frequency of rtp packets |
| 14:45.17 | WindBack | [TK]D-Fender: I have just verified that ${BLINDTRANSFER} doesn't store that info |
| 14:46.19 | [TK]D-Fender | WindBack, 302 REDIRECT isn't exactly a blind transfer so I suspected it might not. There may be no way of knowing |
| 14:47.51 | WindBack | [TK]D-Fender: what a pity.. However thanks |
| 14:52.11 | WindBack | [TK]D-Fender: is there any whay to disable 302 REDIRECT in asterisk. I mean.. I want to avoid the execution o dialplan by local channel when some peer ask with 302 |
| 14:53.15 | *** join/#asterisk Sconk (~krh@2a01:7e8:a0:1:c8fe:ff:fe00:babe) |
| 14:55.04 | bobb_WU | ok so i have a dialplan change that I can't test live so i would like someone to double check me if possible |
| 14:55.51 | bobb_WU | i have a _[5,9],1,Dial(.../${EXTEN},,o) |
| 14:56.03 | bobb_WU | that is what is currently running our relay to the ... sip trunk |
| 14:56.27 | bobb_WU | excuse me that's _[5,9]x. |
| 14:56.58 | bobb_WU | so can i do _[5,9]x.,n,GOTO(_[5,9]x.-${DIALSTATUS},1) |
| 14:57.17 | bobb_WU | then _[5,9]x.-CONGESTION,n,... ? |
| 14:58.11 | bobb_WU | i'm concerned about the "x.-C" part- the period next to the - |
| 15:01.25 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 15:02.43 | *** join/#asterisk bn-7bc (bjarne@cl-1271.sto-01.se.sixxs.net) |
| 15:04.05 | [TK]D-Fender | bobb_WU, You don't jump to a pattern |
| 15:04.53 | [TK]D-Fender | bobb_WU, Use labels instead of that rather nasty 1.0 style hold-over |
| 15:05.22 | bobb_WU | can you give me an example? i'm modifying someone else's dialplan and i'm a noob |
| 15:07.35 | [TK]D-Fender | exten => 1234,n,Goto(somewhere) |
| 15:07.49 | [TK]D-Fender | exten => 1234,n(somewhere),NoOp(Here am) |
| 15:11.11 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 15:12.54 | *** join/#asterisk ctaloi (~ctaloi@nat-66-218-1-193.usadatanet.com) |
| 15:13.06 | bobb_WU | ah i see |
| 15:13.24 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 15:14.22 | bobb_WU | thanks Fender |
| 15:22.16 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
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| 15:23.04 | skrusty | has anyone had a problem with linksys phones (spa 504g for example) where they fail to even try and register and output (via syslog) an error along the lines of "failed to bind to -> .... " ? |
| 15:24.28 | [TK]D-Fender | skrusty, You should be showing us logs, etc.... |
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| 15:32.52 | p3nguin | bobb_wu: _[5,9] pattern will match 5, comma, and 9. You probably want _[59] instead. |
| 15:34.16 | p3nguin | I doubt anyone dials a phone number which starts with a comma. |
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| 15:35.11 | bobb_WU | so how would i do this _[1-4,6-8]xxx ? |
| 15:35.46 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:36.55 | p3nguin | You don't do that because phone numbers don't start with commas. |
| 15:37.36 | bobb_WU | i mean, how would you specify 1 through 4 and 6 through 8 without a comma? |
| 15:37.53 | ectospasm | bobb_WU: [1-46-8] |
| 15:38.05 | p3nguin | _[1-46-8] will match 1, 2, 3, 4, 6, 7, and 8. |
| 15:38.14 | bobb_WU | that just feels wrong lol |
| 15:38.26 | p3nguin | _[1-4,6-8] will match 1, 2, 3, 4, a comma, 6, 7, and 8. |
| 15:38.31 | ectospasm | no, remember it's a character class |
| 15:38.39 | p3nguin | which is wrong, because numbers don't start with comma. |
| 15:38.44 | *** join/#asterisk celord (~celord@186.32.25.13) |
| 15:39.38 | bobb_WU | ok commas removed from the dial plan patterns, thanks guys |
| 15:39.43 | bobb_WU | or gals |
| 15:42.29 | skrusty | seems to happen with sorry [TK]D-Fender: please see 27/03/2012 18:29,Info,192.168.10.202,[0:5060]->83.166.169.222:5060{528) |
| 15:42.29 | skrusty | 27/03/2012 18:29,Info,192.168.10.202,[0]Reg Addr Change(0) 0:0->53a6a9de:5060 |
| 15:42.29 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkdnd in Shared Held key list is not defined |
| 15:42.30 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkcfwd in Shared Held key list is not defined |
| 15:42.30 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkdnd in Shared Active key list is not defined |
| 15:42.30 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkcfwd in Shared Active key list is not defined |
| 15:42.31 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkdnd in Hold key list is not defined |
| 15:42.31 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkcfwd in Hold key list is not defined |
| 15:42.32 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkdnd in Releasing key list is not defined |
| 15:42.32 | skrusty | 27/03/2012 18:29,Error,192.168.10.202,chkcfwd in Releasing key list is not defined |
| 15:42.32 | skrusty | oh shit! |
| 15:42.34 | skrusty | sorry! :( |
| 15:42.55 | skrusty | http://pastebin.com/Qa2Dj07Z |
| 15:42.56 | skrusty | even :( |
| 15:43.59 | [TK]D-Fender | skrusty, 27/03/2012 18:29,Info,192.168.10.202,SCCPTCP:ERR:Can't bind to c0a80aca:5060 errno=49 <- Why am I seeing SCCP and a SIP port in the same message like that? |
| 15:44.17 | [TK]D-Fender | skrusty, And that pile of errors seems to indicate you have done a LOT of things wrong in your config |
| 15:44.52 | skrusty | to be honest, i reset the device and simply set the proxy, nat keepalive username and password... |
| 15:45.22 | skrusty | as to the SCCP and SIP in the same, i've no idea... |
| 15:46.43 | [TK]D-Fender | skrusty, I'd start showing all your configs, screenshots, etc for this.... |
| 15:47.07 | skrusty | will put it together, thanks for the assistance so far! |
| 15:47.39 | skrusty | jsust as a small point of note, it happens with the dsl router fails between wan links in quick succession |
| 15:47.51 | skrusty | using a 2820 from draytek |
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| 15:50.52 | Qwell | it's a...ssokol! |
| 15:51.54 | ssokol | Qwell: yep. just stopped in to see what's happening |
| 15:52.27 | Qwell | ssokol: Rome is just burning, nothing big. |
| 15:52.58 | ssokol | breaks out his fiddle... |
| 15:54.40 | leifmadsen | ssokol: ohai! |
| 15:55.17 | ssokol | leifmadsen: Leif lives! |
| 15:55.22 | leifmadsen | heck ya he does |
| 15:55.27 | ssokol | leifmadsen: How's the new gig working out? |
| 15:56.05 | ssokol | #join vuc |
| 15:56.17 | ssokol | Er, ok that doesn't seem to work in here... |
| 15:56.49 | sp00kz | / |
| 15:57.14 | leifmadsen | ssokol: pretty good actually! |
| 15:57.24 | [TK]D-Fender | <PROTECTED> |
| 16:01.21 | leifmadsen | ssokol: /join #vuc :) |
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| 17:13.37 | nsgn | silly question; registerattempts and registertimeout..where do they exist and/or if they dont exist by default where should i put them? |
| 17:14.17 | nsgn | i ask because they don't seem to be in my sip.conf, but perhaps they don't exist by default and i need to make them? |
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| 17:18.10 | p3nguin | ;registertimeout=20 ; retry registration calls every 20 seconds (default) |
| 17:18.13 | p3nguin | ;registerattempts=10 ; Number of registration attempts before we give up |
| 17:18.21 | p3nguin | These are found in the sample sip.conf. |
| 17:18.32 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 17:18.51 | drfreeze | I'm working with a new phone company and a PRI system |
| 17:19.26 | drfreeze | They claim they are sending the callerid name, but I am seeing a "" for the name |
| 17:20.59 | p3nguin | Who is sending CNAM? |
| 17:21.09 | p3nguin | And to whom are they sending it? |
| 17:21.38 | drfreeze | exten => _XXXX,n,NoOp(${CALLERID(all)}) |
| 17:21.39 | drfreeze | NoOp("DAHDI/i1/512XXXXXXXX-11", """ <512XXXXXXX>") in new stack |
| 17:22.07 | drfreeze | p3nguin: Logix is sending CNAM to Asterisk system |
| 17:22.19 | p3nguin | You're talking about an inbound call from your telco to your PBX? |
| 17:22.30 | drfreeze | Digium PRI, Asterisk |
| 17:22.31 | drfreeze | Yes |
| 17:23.16 | drfreeze | Asterisk 1.8.7.0 |
| 17:23.36 | p3nguin | Are you using callerid = asreceived or did you define something? |
| 17:23.55 | p3nguin | or maybe hidecalleridname=yes? |
| 17:24.29 | drfreeze | It's a very sparse extension.conf file |
| 17:24.38 | drfreeze | I assume the hidecalleridname would be in a config file |
| 17:24.55 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 17:25.18 | p3nguin | Yes, both of those settings are in chan_dahdi.conf. |
| 17:25.43 | nny | odd situation (using 1.8.6 currently) my sip show peers latency is showing 80-90s but pinging the devices manually show .90 to .80, is this possibly a display bug of some kind? |
| 17:26.24 | drfreeze | p3nguin: http://pastie.textmate.org/private/edmrwzn5yqfi0ozolyaowg |
| 17:27.24 | p3nguin | I, unfortunately, do not have an explanation for the absence of your CNAM. |
| 17:28.14 | Qwell | pri intense debug - do you see it coming through? |
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| 17:28.50 | p3nguin | I guess I need to get a PRI just so I can learn it and support it. :/ |
| 17:29.19 | anonymouz666 | p3nguin: what do you have? |
| 17:29.37 | p3nguin | I use, provide, and support VoIP only. |
| 17:30.08 | anonymouz666 | here we use VoIP, MFC/R2 and ISDN PRI |
| 17:30.47 | anonymouz666 | MFC/R2 is third world stuff, you probably would never care about it |
| 17:31.08 | anonymouz666 | unless you want to explore third world market :-) |
| 17:31.10 | p3nguin | I'm willing to learn any tech that I have available, but I just don't currently have any PRI. |
| 17:31.50 | drfreeze | Qwell: the pri set debug on is not working |
| 17:32.10 | drfreeze | it claims: No such command 'pri set debug on' (type 'core show help pri set' for other possible commands) |
| 17:32.28 | drfreeze | but there is not core pri |
| 17:33.04 | *** join/#asterisk bn-7bc (bjarne@cl-1271.sto-01.se.sixxs.net) |
| 17:33.05 | [TK]D-Fender | pri debug span 1 |
| 17:33.18 | drfreeze | ok, I see that now |
| 17:33.21 | drfreeze | [TK]D-Fender: thanks |
| 17:33.52 | WIMPy | pri set debug 2 span 1 |
| 17:35.55 | nny | anyone know of a reason why the sip show peers output for latency would be incorrect, for example showing .98 ms as 98 ms? |
| 17:36.50 | WIMPy | nny: Because you're comparing apples to pears. |
| 17:38.30 | nny | WIMPy: what is the latency shown in sip show peers? |
| 17:38.41 | p3nguin | I've never had a peer show a qualify time of less than 1 ms. |
| 17:39.04 | nny | 64 bytes from 10.1.2.87: icmp_seq=2 ttl=64 time=0.921 ms |
| 17:39.06 | nny | vs |
| 17:39.27 | Qwell | umm, sip show peers doesn't show ping latency |
| 17:39.51 | nny | that was my question above |
| 17:40.02 | WIMPy | nny: The latency from application to application. |
| 17:40.02 | p3nguin | One is an ICMP echo (ping) time, the other is qualify time (time for a response to an OPTIONS packet). |
| 17:40.29 | p3nguin | network layer vs. application layer |
| 17:40.37 | nny | 10.1.2.87 D 5060 OK (107 ms) is the sip show peers time. Assuming this is due to the deviuces/clients being used if it's that far up the OSI model |
| 17:40.43 | nny | p3nguin: thanks, that clears that up |
| 17:40.54 | p3nguin | 107 ms reponse to your systems OPTIONS |
| 17:41.23 | nny | yeah, so the peers (in this case cisco 7940) are slower to respond to the application layer based ping, got it |
| 17:41.32 | nny | just never seen it so high before on a local network |
| 17:41.34 | p3nguin | It's totally normal. |
| 17:42.00 | *** join/#asterisk mahaD (~chatzilla@123.238.108.122) |
| 17:42.05 | p3nguin | The SIP application on the Cisco phone could be THAT slow. :) |
| 17:42.08 | nny | my offsite SPA 962 is around 60 ms heh |
| 17:42.17 | nny | p3nguin: that's what I gathered heh, not my choice of phones |
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| 17:54.06 | irule | hi, may I get help seting up a spa3102? |
| 17:54.18 | p3nguin | ~ask |
| 17:54.18 | 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. |
| 17:56.19 | paulc | irule: what do you want to do with it? describe your setup scenario? |
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| 17:57.30 | irule | paulc I am about to install elastix on an old pc and have a single POTS line and I would like to use through the spa3102 |
| 17:58.50 | paulc | irule: fairly easily done then.. you setup both the FXO and FXS ports as separate peers in sip.conf, then some tweaking on the SPA3102 to send inbound FXO calls into Asterisk, and accept calls from Asterisk to send straight out on the FXO |
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| 18:28.50 | paulc | ~itsp-ca |
| 18:28.58 | paulc | ~itsp |
| 18:28.59 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:31.37 | WindBack | Is there any whay to disable 302 REDIRECT in asterisk? I want to avoid the execution o dialplan by local channel when some peer answers with 302 |
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| 18:34.07 | cloakable | ~itsplist-uk |
| 18:34.07 | infobot | itsplist-uk is, like, UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
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| 18:42.35 | pabelanger | Anybody provisioning polycom phones via HTTP? I'm curious what your apache config file looks like to allow the phones to upload log files |
| 18:42.58 | Naikrovek | I would also like to know that. |
| 18:43.07 | Naikrovek | I imagine those folks log with syslog, though. |
| 18:44.16 | pabelanger | Ya, I was thinking about the same thing. |
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| 18:50.30 | pabelanger | Naikrovek: okay, just figure it out. Had to enable mod_dav, and mod_dav_fs |
| 18:50.43 | Naikrovek | ah dav |
| 18:50.46 | Naikrovek | makes sense |
| 18:50.57 | Naikrovek | now do it with https |
| 18:51.00 | pabelanger | http://pastebin.com/bvSpbE21 |
| 18:51.07 | pabelanger | nods |
| 18:51.16 | pabelanger | need to read more up on it to check security |
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| 19:14.07 | onixx99 | guys, what would be the best way to see all changes made to asterisk between asterisk 1.8.9.3 and 1.8.10-rc1. I have determined that my dtmf decoding problem started in 1.8.10rc1 |
| 19:14.39 | Qwell | onixx99: The ChangeLog file stored in the tarball. |
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| 19:16.22 | onixx99 | Qwell : thanks I will go thru this now. :) starting in this version, a quick dtmf sequence inband just won't decode properly. out of 16 digits, 4 at the most go thru |
| 19:19.11 | onixx99 | Qwell : would you know why I can't find 1.8.9.3 "release" in the ChangeLog |
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| 19:19.53 | Qwell | just use .0 as your base |
| 19:21.43 | onixx99 | Qwell : okay, I started going up the file from .0, my first suspect is 2012-01-05 21:46 +0000 [r349672-349728], is there a way to grab a version joust before this change ? how would you proceed ? |
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| 19:39.11 | showy | hi, is a DL380 G7 with a digium T410 card a safe hardware configuration / |
| 19:39.22 | Qwell | I don't see why it wouldn't be |
| 19:39.24 | showy | ? |
| 19:39.40 | showy | thought the same |
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| 19:57.16 | jdoe | on a fresh upgrade to 10.3.0 I'm getting "Couldn't create astdb table: SQL logic error or missing database" on startup, and asterisk helpfully creates a 0-byte astdb. |
| 19:57.19 | jdoe | has anyone else seen that? |
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| 20:13.40 | fireman_biff | I'm trying to generate a csv-formatted CDR but I need ${CDR(billsec)} to have a fixed number of digits (left-padded with zeros) how can I do this? |
| 20:21.25 | akrohn | fireman_biff, probably can't do it from within asterisk. might need to be something you do at the time of report generation |
| 20:22.24 | fireman_biff | akrohn: alright, thats what i figured i'd end up doing... thanks |
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| 20:58.29 | drfreeze | I'm trying to catch pri debugging info |
| 20:58.43 | drfreeze | Any ideas on how to get the output directed to a file? |
| 21:02.01 | Deetz | I have fax detect on so I can receive faxes and email them. |
| 21:02.17 | Deetz | But on several outgoing calls it detects a CNG signal partway through a call and redirects/drops the call. |
| 21:02.55 | Deetz | I see a few people on google have had this problem but there's nothing else out there |
| 21:10.21 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 21:10.51 | drfreeze | Ok, I've got a PRI and the ISP is definitely providing a callerid |
| 21:10.53 | drfreeze | http://pastie.textmate.org/private/sj8c303sucva7ykyxvxeg |
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| 21:11.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:11.19 | drfreeze | But, that is not showing up in any of the CALLERID(<item>) returns |
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| 21:39.30 | BJD10 | Q: More of a switchvox question here. Is it possible using the Switchvox platform to do variable manipulations? e.g. DIAL(SIP/1${EXTEN}) type of thing. From what I can tell it looks like you have to build each route by hand and can only use a variable from the dropdown list after assigning it. |
| 21:40.35 | leifmadsen | BJD10: sounds about right, but not a lot of people in here are going to be switchvox users |
| 21:40.46 | leifmadsen | you could call Digium support (assuming you have support options) and ask them |
| 21:43.42 | BJD10 | leifmadsen: Thanks Lief, was thinking I would have to do that, but wanted to take a quick check to see if I was on the right assumptions. (first time working on the Switchvox appliance. Not use to being locked out of everything :/) |
| 21:44.39 | drfreeze | Does 1.8 have a bug in pulling callerid name out of the PRI packet? |
| 21:45.46 | leifmadsen | drfreeze: I doubt it |
| 21:46.00 | leifmadsen | did you check your PRI debug to see that you're actually getting CID? |
| 21:47.47 | drfreeze | leifmadsen: http://pastie.textmate.org/private/sj8c303sucva7ykyxvxeg |
| 21:47.50 | drfreeze | It's there |
| 21:48.14 | leifmadsen | ok |
| 21:48.59 | drfreeze | We are using 1.8.7.0 |
| 21:49.09 | drfreeze | not sure why it is not pulling it out |
| 21:55.50 | *** join/#asterisk albertoandrade (~albertoan@186.206.5.33) |
| 22:02.41 | *** join/#asterisk jsjc (~Adium@51.Red-79-146-23.dynamicIP.rima-tde.net) |
| 22:19.18 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 22:29.56 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176138838.dsl.bell.ca) |
| 22:30.25 | dijib | hey all can Asterisk recieve calls from Skype still or no? |
| 22:31.20 | russellb | dijib: nope, that product got killed |
| 22:32.11 | dijib | no hacked versions? |
| 22:34.07 | dijib | any other a/v stuff available? msn? teamspeak anything? |
| 22:34.22 | cloakable | sip, ptsn? |
| 22:34.31 | cloakable | *pstn |
| 22:36.00 | dijib | sip |
| 22:36.36 | oneadvent | doesn't nerdvittles have a solution for that still? |
| 22:36.37 | cloakable | iirc you can hook in googlevoice |
| 22:37.50 | oneadvent | http://nerdvittles.com/?p=784 |
| 22:38.21 | dijib | nerdvittleshttp://nerdvittles.com/index.php?p=784 |
| 22:39.14 | russellb | it's built into asterisk ... |
| 22:39.14 | lanning | there is this: http://www.skype.com/intl/en/business/skype-connect/ |
| 22:39.33 | oneadvent | that is the paid for version |
| 22:39.46 | lanning | yes |
| 22:40.37 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 22:40.58 | oneadvent | lanning: that is pretty cool. This was different than i thought it would be...so it basically makes skype a sip trunk? |
| 22:41.23 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 22:41.24 | lanning | there is that word again... "sip trunk" |
| 22:41.43 | lanning | but yes, you can buy channels for concurrent calls |
| 22:44.58 | oneadvent | lanning: can you enlighten me why you do not like "sip trunk"? (I like learning, not trying to call you out or anything) |
| 22:45.38 | lanning | ~siptrunk |
| 22:45.38 | infobot | somebody said siptrunk was something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 22:45.42 | russellb | it bugs the crap out of me when people complain about "sip trunk" in here |
| 22:45.50 | russellb | EVERYONE ELSE other than #asterisk is just fine with that term |
| 22:45.53 | russellb | infobot: forget siptrunk |
| 22:45.53 | infobot | russellb: i forgot siptrunk |
| 22:46.05 | _Corey_ | haha, I roll my eyes every time |
| 22:46.14 | russellb | it's a reasonable common term in this industryu |
| 22:46.32 | russellb | just because there isn't something called a trunk in sip.conf doesn't mean it doesn't exist |
| 22:46.33 | lanning | because it is hold over from "trunk lines" |
| 22:46.35 | russellb | </rant> |
| 22:46.41 | _Corey_ | I'd go as far as say it's ubiquitous |
| 22:46.55 | lanning | there is no difference between a sip channel and a sip trunk |
| 22:46.57 | russellb | _Corey_: yep |
| 22:47.03 | lanning | channels are dynamic |
| 22:47.03 | _Corey_ | try ordering sip origination from a carrier and not use the term... ;) |
| 22:47.13 | russellb | lanning: well one is an asterisk concept, and one is a common term used in the SIP industry in general. |
| 22:47.46 | lanning | so you configure sip trunks to your desk phone? |
| 22:48.13 | russellb | no. but a connection to a SIP termination provider is quite commonly referred to as a SIP trunk |
| 22:48.20 | _Corey_ | lanning: like to think of "sip trunks" as a non-technical term... |
| 22:48.22 | russellb | I didn't say everything SIP is referred to as a trunk |
| 22:48.36 | lanning | sip is sip is sip, I don't see any difference |
| 22:48.45 | russellb | between 2 servers, pretty darn common to call that a SIP trunk |
| 22:49.06 | lanning | and there is no difference between two servers and a server and a phone |
| 22:49.16 | _Corey_ | lanning: It is just SIP in reality... but you don't call up a carrier and tell them you'd like to order a sip UA |
| 22:49.39 | lanning | no, but you are the UA |
| 22:49.49 | lanning | you usually register with them |
| 22:50.51 | _Corey_ | lanning: It's just a generic term most people use to describe SIP communication between B2BUAs or other gateways |
| 22:51.30 | lanning | great an language context term |
| 22:51.47 | lanning | s/n// |
| 22:52.15 | dijib | can not use nerviddles skype in console? |
| 22:52.17 | dijib | sucks |
| 22:52.49 | _Corey_ | I'm about to get out of here, I was just amused by russelb and the perpetual "there's no such thing as a sip trunk" pedantic arguing |
| 22:53.00 | russellb | _Corey_: take it easy :) |
| 22:53.00 | _Corey_ | ;) |
| 22:53.07 | _Corey_ | adios |
| 22:55.15 | oneadvent | :/ sorry didn't mean to start something...ok i haven't lost my mind, sip trunk is an ok term...isn't that point of asterisk, that to asterisk all things appear equal? |
| 22:55.38 | oneadvent | also why didn't info bot change the n in language and context? |
| 22:56.25 | lanning | heh, the argument is that there is a business term "SIP Trunk" that technically is no different than any other SIP setup. |
| 22:57.08 | lanning | there is no difference in the protocol, no difference in the settings/configuration |
| 22:57.40 | oneadvent | :) ok |
| 22:58.22 | oneadvent | is it just me or do you read "Fields marked with an asterisk * are required." differently too? |
| 22:58.36 | lanning | the term is a hold over from physical trunk lines |
| 23:00.00 | lanning | marked with a phone call? :) |
| 23:00.36 | oneadvent | hehe i just see asterisk and * is not the first thing i think of lol |
| 23:01.03 | oneadvent | i get suspicious and wonder why they want my asterisk information. |
| 23:02.14 | *** join/#asterisk timahvo1 (~rogue@41.81.191.206) |
| 23:03.01 | lanning | I have issues with Togos receipts. Sandwiches are listed by number "#10". I keep missing it, because I skip it as a comment. :) |
| 23:04.26 | oneadvent | hahaha |
| 23:05.36 | p3nguin | Anyone here do PAP2 provisioning? |
| 23:05.45 | *** join/#asterisk Gokee2 (~gokee2@204-195-14-78.wavecable.com) |
| 23:06.05 | oneadvent | i do p3nguin |
| 23:06.27 | oneadvent | i use the linksys pap2 and then regular home phones |
| 23:06.30 | p3nguin | oneadvent: You're familiar with $MA, $MAU, etc... |
| 23:06.41 | oneadvent | :) i didn't change any of those things |
| 23:06.51 | p3nguin | What do you mean by that? |
| 23:07.08 | oneadvent | it was configured on the pap2 already |
| 23:07.10 | p3nguin | I'm not asking if you use a PAP2. |
| 23:07.17 | p3nguin | I'm asking about provisioning. |
| 23:08.32 | oneadvent | i'm sorry i misunderstood. I did not have any of those options changed when I setup my pap2... |
| 23:09.11 | p3nguin | Of course you wouldn't change it... I'm talking about provisioning from a provisioning server. |
| 23:10.39 | oneadvent | :) |
| 23:11.48 | p3nguin | The problem I was running into was using $MAU in the User ID field in the cfg file. I was curious if I was doing it wrong or if it doesn't work in certain areas. |
| 23:17.19 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 23:18.39 | *** join/#asterisk poptix (poptix@poptix.net) |
| 23:18.47 | poptix | arg, I give up |
| 23:18.55 | poptix | anyone familiar with polycom 550's? |
| 23:19.36 | poptix | it fetches the config just fine, verified config is right, etc. but when it tries to register asterisk says 401 unauthorized and sends a www-authenticate |
| 23:19.41 | poptix | which the polycom promptly ignores |
| 23:19.52 | poptix | it doesn't seem to be responding to the authentication request at all |
| 23:27.04 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:27.06 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
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| 23:57.36 | poptix | arrg, fucking comcast |
| 23:57.39 | *** part/#asterisk poptix (poptix@poptix.net) |