| 00:02.56 | angryuser | mandd: at least sip show peers with qualify on ;) |
| 00:09.13 | Servergod | sry, pastebin had bad info here is update http://pastebin.com/mfb9b716 |
| 00:09.16 | *** join/#asterisk moy (n=moy@189.169.88.136) |
| 00:09.57 | *** join/#asterisk macros73_ (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 00:14.13 | TJNII | Servergod: you have a firewall? |
| 00:15.29 | Servergod | customer premise equipment m0n0wall 1.3b9 not using the SIP ALG it mangles the ports, NAT and Rules are set |
| 00:15.41 | ManxPower | Servergod: we really can't support FreePBX/Trixbox configs |
| 00:16.41 | ManxPower | and stop masking the damn IPs, we can't even tell if the localnet= is correct for your network |
| 00:17.44 | Servergod | thanks, have a good day. |
| 00:17.53 | *** part/#asterisk Servergod (i=Servergo@70.97.157.6) |
| 00:18.02 | macros73_ | think you hurt his feelings |
| 00:18.12 | ManxPower | sip_nat.conf, indeed! |
| 00:18.39 | *** join/#asterisk Servergod (i=Servergo@70.97.157.6) |
| 00:19.33 | Servergod | thought i would ask here, trixboxppl dont know what a as5350 is, or what a dial-peer is, ppl in this chan are better educated. |
| 00:22.34 | ManxPower | Servergod: yeah, but we can't read their config files. |
| 00:23.05 | ManxPower | or more correctly, most of us are unwilling to spend 10x the work to understand the Trixbox config files. |
| 00:23.26 | ManxPower | Servergod: that is what you get when you use a system only newbies use. |
| 00:24.28 | angryuser | <ManxPower> some gui's are nice, like elastix |
| 00:24.29 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:25.07 | Servergod | that's cool, i can dig it. I've just had better luck here. LOL, not my box, theirs, i'm just the ITSP, i usually use like a 2651xm with a pri card to a pbx then handle the sip side, this is a sales deal! hate salesppl |
| 00:25.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 00:25.17 | angryuser | but i admit, i you have a bug, very difficult to find |
| 00:28.35 | Servergod | i updated the pastebin, with eth0 config and filled in the local net stuff for the firewall. gonna camp fer a bit and check the trix chan. if u get the hankerin' http://pastebin.com/m1cbb7b5c |
| 00:32.57 | ManxPower | This channel is not second level Trixbox support |
| 00:38.36 | Servergod | well, i'm out, thanks anyways, angryuser: thanks for looking and your time. ManxPower, either you are in a bad mood or are just a dick. Peace. |
| 00:39.38 | *** part/#asterisk Servergod (i=Servergo@70.97.157.6) |
| 00:40.54 | jaytee | ManxPower, are you dick? :-) |
| 00:41.25 | jaytee | cuz like, that wasn't my first impression |
| 00:42.00 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 00:42.49 | jaytee | Servergod> that's cool, i can dig it. <- He can dig it! She can dig it! We can dig it! Doo, dee wop! Bop, bop bop, doo, dee wop!!! |
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| 01:27.04 | mandd | both numbers work fine, in out, but one problem, all incoming calls on both numbers are going to fax |
| 01:27.05 | mandd | http://pastebin.com/m6103c70b |
| 01:27.19 | mandd | while one shouldgo to office, and one to fax |
| 01:27.26 | mandd | what am I doing wrong? |
| 01:28.30 | *** join/#asterisk gcarrillog (n=gcarrill@201.151.84.240) |
| 01:29.23 | mandd | grr, actually fax can not dial out |
| 01:30.26 | jaytee | remove the ; from the include => outbound-fax that's your problem with not being able to fax outbound |
| 01:30.56 | mandd | ah |
| 01:30.56 | mandd | yeah |
| 01:30.58 | mandd | bah |
| 01:31.12 | mandd | and as for incoming calls? |
| 01:31.24 | mandd | how do I make sure that one goes to fax and the other to office? |
| 01:31.39 | jaytee | just a sec |
| 01:31.46 | mandd | thank :) |
| 01:31.50 | mandd | thanks* |
| 01:32.59 | jaytee | does your inbound fax work? |
| 01:33.09 | mandd | yep, on both numbers |
| 01:33.16 | mandd | while it should only be on one |
| 01:33.30 | mandd | the other one should ring office |
| 01:34.28 | TJNII | Pastebin what the CLI says with vverbose=5 when you call the office number |
| 01:34.45 | mandd | ok, one sec |
| 01:34.48 | jaytee | well you've got globals set for m and b for mandd and alex but nothing to match against in your [office] context. How are calls coming in? FXO? |
| 01:35.05 | mandd | both are voip account |
| 01:35.08 | mandd | accounts* |
| 01:35.40 | mandd | there is a global for fax as well |
| 01:37.12 | jaytee | forget what I said about the [office] context not having a match. I missed it. it's there and looks ok to me. |
| 01:39.23 | mandd | TJNII http://pastebin.com/m2807a1f6 |
| 01:40.30 | mandd | err, with both numbers http://pastebin.com/m7b805e71 |
| 01:40.39 | TJNII | Well that doesn't help much now des it..... |
| 01:41.15 | mandd | well, it looks like it is comign from a same number |
| 01:41.19 | mandd | but I dial different one |
| 01:41.30 | TJNII | Yes, it does. |
| 01:41.54 | TJNII | Same provider? |
| 01:42.01 | mandd | yes, different accounts |
| 01:42.05 | TJNII | remembers seeing this bug before |
| 01:42.07 | mandd | log-ins/numbers |
| 01:42.35 | TJNII | I've seen * mess this up before |
| 01:42.52 | TJNII | Since the calls come from the same host it sends all calls to the last registered account |
| 01:43.08 | mandd | it's likely me who's messing up |
| 01:43.22 | mandd | hmm, anythign I can do? |
| 01:43.24 | TJNII | Does your ITSP send a number or anything? I know Broadvoice sends the phone number |
| 01:43.51 | mandd | yep |
| 01:44.15 | jaytee | which version of *? |
| 01:44.23 | TJNII | When I had this problem I had an exten that keyed off that number and then goto'd the correct extension |
| 01:44.25 | TJNII | jaytee: 1.4 |
| 01:44.55 | TJNII | I don't know if I saw it with 1.2 |
| 01:45.05 | mandd | Asterisk 1.4.20.1 on freebsd |
| 01:45.08 | jaytee | do a SIP SET DEBUG IP "Provider IP address" and do another test. |
| 01:45.43 | jaytee | with both numbers |
| 01:49.08 | mandd | http://pastebin.com/m5092677d |
| 01:50.47 | mandd | 2nd number http://pastebin.com/m2ba1f3a8 |
| 01:53.17 | mandd | hmm, i'll go google a bit |
| 01:54.47 | jaytee | um, you're at 76.66.164.250 as an IP? |
| 01:56.51 | jaytee | the Contact field is identical in both calls, s@76.66.164.250 |
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| 02:07.26 | jaytee | manda, you need to read page 99 of the book and fix your SIP.CONF file. refer to the To: section of each of the calls for you sip accounts to match against. |
| 02:07.33 | jaytee | ~book |
| 02:07.33 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 03:26.58 | nephfl | hello im trying to load trixbox on a gigabyte motherboard and it is freezing at loading the zap modules can someone help me out? |
| 03:27.22 | nephfl | ive tried manually setting the irqs for pci and disabled other integrated device |
| 03:31.09 | nephfl | hello? |
| 03:32.16 | fakhir | ~trixbox |
| 03:32.17 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
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| 03:59.02 | macros73 | how do I configure an aastra 480i to pass-through g729 for external calls but use g711 for internal calls? |
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| 04:06.18 | bbryant | ~itsp |
| 04:06.20 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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 |
| 04:06.31 | bbryant | ~itsplist-us |
| 04:06.32 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:19.24 | *** join/#asterisk mattwj2005 (n=Matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 04:19.32 | mattwj2005 | hi everyone |
| 04:19.44 | mattwj2005 | I am having a problem getting my incoming calls to work |
| 04:24.57 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 04:26.58 | [TK]D-Fender | mattwj2005: Where are your calls coming from? |
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| 04:28.34 | mattwj2005 | Teliax |
| 04:28.46 | mattwj2005 | well that is my provider |
| 04:28.55 | mattwj2005 | I am trying to call using skype |
| 04:33.25 | jblack | mattwj2005: What's the problem? |
| 04:34.34 | mattwj2005 | I can't receive incoming calls..... |
| 04:34.54 | [TK]D-Fender | mattwj2005: go to * CLI, enable SIP DEBUG, and pastebin an incoming call attempt |
| 04:34.56 | [TK]D-Fender | ~pb |
| 04:34.57 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:34.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 04:35.04 | jblack | Yeah, I got that part. You're not being specific enough. What does the log say in /var/log/asterisk ? |
| 04:35.15 | jblack | Or as [TK]D-Fender asks, pastebin would be nice. |
| 04:35.43 | mattwj2005 | it says this NOTICE[3402]: chan_iax2.c:7348 socket_process: Rejected connect attempt from 74.201.8.23, who was trying to reach 'phonenumber@' |
| 04:35.52 | mattwj2005 | phonenumber is my incoming number |
| 04:36.08 | mattwj2005 | let me check the log |
| 04:37.13 | [TK]D-Fender | mattwj2005: thats typically a context/exten match error. enable IAX2 debug and provide the pastebin, along with your iax2.con masking only passwords, and your extensions.conf |
| 04:44.29 | mattwj2005 | http://pastebin.com/m330f0163 |
| 04:44.41 | mattwj2005 | crap |
| 04:45.01 | mattwj2005 | I didn't hide my number in all cases |
| 04:45.02 | mattwj2005 | :( |
| 04:45.09 | jblack | You shoudln't hide your number at all. |
| 04:46.03 | mattwj2005 | ok |
| 04:46.20 | jblack | Yeah, I'd say it's context. |
| 04:46.41 | jblack | Show us your extensions.conf and your iax.conf, with all passwords replaced with XXXXXXX |
| 04:47.27 | jblack | Or, if you understand what a context is, set the context for the mattwj2005 account to the one that has the phone number as an extension. |
| 04:47.45 | mattwj2005 | yeah I understand context |
| 04:48.09 | jblack | Ok. it appears you're not setting the context for that account, and you don't have a default context set. |
| 04:50.35 | *** join/#asterisk MCooper (n=kmoore@64.110.169.173) |
| 04:50.44 | MCooper | Good morning fair channel..... |
| 04:52.27 | mattwj2005 | yeah it doesn't appear to be work |
| 04:52.35 | mattwj2005 | I verified the context is correct |
| 04:52.40 | mattwj2005 | can it be anything else? |
| 04:53.00 | jblack | did you tell asterisk to reload iax.conf, or restart iax? |
| 04:53.21 | mattwj2005 | I restarted asterisk |
| 04:55.33 | [TK]D-Fender | mattwj2005: pastebin your dialplan.... |
| 04:59.19 | jeev | wow fender |
| 04:59.31 | mattwj2005 | here is something weird I have noticed |
| 05:01.25 | mattwj2005 | in my web portal for my did there is an option to select the destination which is my username, the default context or no where |
| 05:01.38 | mattwj2005 | the errors I get change when I change this drop down |
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| 05:18.09 | Kyoshi | using realtime for sipusers and sippeers, would it be possible to map freetds as the driver rather than mysql (without having to use unixodbc)? example, extconfig.conf for odbc would be "sipusers => odbc,res_odbc_sipusers,sippeers" and mysql would be "sipusers => mysql,asterisk,sippeers" but "tds" and "freetds" dont seem to be valid driver names. any ideas? |
| 05:20.23 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 05:21.49 | Kyoshi | it must since cdr's can be written directly to tds db's |
| 05:23.04 | alancio | hey is anybody in here familiar with the snom m3? |
| 05:24.56 | Jameno123 | Is there any easy way to check if an extension exists in an AGI script? |
| 05:28.22 | Jameno123 | Also, i have a weird problem with the Dial Macro agent-side; the callerid info is the number that customer dialed into rather than the users callerid.... |
| 05:28.42 | Jameno123 | pre-bridge, marco execution or whatever you want to call it |
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| 06:36.13 | bijit | slackboy is a bot? |
| 06:36.39 | bijit | sorry wrong irc |
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| 06:39.47 | danpwi | anyone here use random() in AEL? |
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| 07:58.23 | pcrane | I'm having some problems getting menuselect to compile for asterisk addons for 1.6 |
| 07:58.57 | pcrane | I'm on a mac, and it complains that I need to install ncurses. This has been done via fink |
| 07:59.47 | pcrane | the make menuselect works for asterisk itself... |
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| 08:31.07 | BBHoss | pcrane: do you have libncurses installed |
| 08:33.00 | pcrane | according to fink, yes |
| 08:33.12 | BBHoss | libterm? |
| 08:33.23 | tzafrir_laptop | is there a separate -dev or -devel package? |
| 08:33.49 | pcrane | there is for ncurses (which is installed by default) |
| 08:33.57 | pcrane | I can't find libterm using fink... |
| 08:34.51 | BBHoss | see if you can do port install asterisk or w/e and see what deps it dls |
| 08:38.07 | pcrane | hmm... |
| 08:38.14 | pcrane | is it on fink? |
| 08:39.10 | pcrane | aparently not |
| 08:39.45 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-151-207.neo.res.rr.com) |
| 08:39.49 | pcrane | the info on voip-wiki is strange... |
| 08:40.00 | pcrane | it tells you to download (among other things) popt |
| 08:40.07 | pcrane | then it tells you to do ./configure --prefix=/usr/local CFLAGS=-I/sw/include LDFLAGS=-L/sw/lib --without-popt |
| 08:41.25 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:43.35 | MCooper | Does anyone have a good blacklisting solution based on the number of calls from the same number over a period of time.. or maybe a link to one? |
| 08:44.42 | pcrane | MCooper: use the astdb to count the number of times a number calls, and use the caller id blacklist app |
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| 08:47.15 | MCooper | pcrane, is there a good source of information about the Askdb? |
| 08:48.16 | pcrane | http://www.voip-info.org/wiki/view/Asterisk+database |
| 08:48.36 | MCooper | pcrane, thank you.... |
| 08:48.39 | pcrane | np |
| 08:49.02 | pcrane | BBHoss: I don't think that'll help much |
| 08:49.10 | pcrane | cause asterisk's menuselect works fine |
| 08:49.29 | pcrane | asterisk-addons menuselect doesn't (complains about the lack of ncurses) |
| 08:52.02 | *** join/#asterisk cabbiepete (n=cabbiepe@host86-143-32-91.range86-143.btcentralplus.com) |
| 08:55.14 | pcrane | hmm... oddd |
| 08:55.24 | pcrane | I'll have another go at trying to figure it out later on |
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| 09:00.15 | brandon_1 | hi |
| 09:01.43 | brandon_1 | i'm using asterisk 1.2 with several Grandstream GXP-2000 phones |
| 09:02.15 | brandon_1 | and it's working ok |
| 09:03.37 | brandon_1 | incomming calls don't work sometimes (no sound) |
| 09:03.47 | brandon_1 | until i restart my asterisk server ( restart now ) |
| 09:04.42 | brandon_1 | does anybody knows why ?? |
| 09:06.38 | *** join/#asterisk friendly12345 (n=friendly@ppp59-167-145-230.lns4.mel6.internode.on.net) |
| 09:07.48 | x86 | dude asterisk 1.2 is no longer supported |
| 09:08.49 | tzafrir_laptop | brandon_1, please provide some clues |
| 09:09.18 | brandon_1 | i MUST use asterisk 1.2 |
| 09:09.21 | tzafrir_laptop | e.g: do you see anything in a CLI trace on an incoming call? |
| 09:09.34 | brandon_1 | because calls are dropping with newer versions than 1.2.13 |
| 09:09.55 | tzafrir_laptop | brandon_1, do you have a specific bug number for that? |
| 09:10.31 | brandon_1 | yes |
| 09:18.21 | tzafrir_laptop | brandon_1, two things to see: do you see anything in the CLI trace if you have 'verbose' set to at least 3? |
| 09:18.26 | tzafrir_laptop | set verbose 3 |
| 09:18.50 | tzafrir_laptop | And if not, enable sip debug, and tell us what you see there |
| 09:19.47 | brandon_1 | nothing wrong |
| 09:19.53 | brandon_1 | call is coming in like always |
| 09:20.03 | brandon_1 | only no sound |
| 09:20.15 | brandon_1 | my problem only occurs when my provider is switching to a different ip address |
| 09:20.20 | *** join/#asterisk matrix1233 (n=Administ@196.203.43.109) |
| 09:20.58 | brandon_1 | for example: |
| 09:21.18 | brandon_1 | sip.sipnl.net is normally 83.98.222.5 |
| 09:21.31 | brandon_1 | but they sometimes switch to a different server |
| 09:21.41 | brandon_1 | then my problem comes back |
| 09:21.49 | brandon_1 | until i restart my asterisk server |
| 09:21.53 | brandon_1 | with restart now |
| 09:23.17 | brandon_1 | does anyone knows how to bypass this problem ??? |
| 09:33.04 | Maliuta | you could reload the sip config, that should cause a re-register, which might allow the relookup of the hostname |
| 09:33.34 | Maliuta | but I would be looking for a different provider if it happened all the time |
| 09:36.47 | tzafrir_laptop | brandon_1, hmm.... I'm not sure how to fix that. Does a 'sip reload' fix it? |
| 09:39.16 | brandon_1 | i don't know |
| 09:39.19 | brandon_1 | will try that |
| 09:39.22 | brandon_1 | thanks ! |
| 09:39.47 | brandon_1 | maybe a cron jobb that reload asterisk every hour ... |
| 09:40.24 | Maliuta | why reload the whole thing if you only need to do a sip reload |
| 09:40.32 | tzafrir_laptop | a cron job that checks for a change of IP address and then reloads? |
| 09:40.48 | brandon_1 | you're right Maliuta |
| 09:40.53 | tzafrir_laptop | getent hosts sip.sipnl.net |
| 09:40.54 | brandon_1 | just sip reload |
| 09:42.16 | brandon_1 | hmm maybe just sip reload every hour ... |
| 09:46.17 | steliosk | brandon_1 : looks like the Grandstream tries to re-register after the default timeout (1h) |
| 09:47.05 | steliosk | do you have canreinvite=no ? |
| 09:55.39 | brandon_1 | yes i have canreinvite=np |
| 09:56.17 | Maliuta | you might face some problems with dns caches though |
| 09:56.35 | Maliuta | if the result is coming from a cache and the providers have changed it |
| 09:57.02 | brandon_1 | yes that might possible Maliuta |
| 09:57.20 | Maliuta | if the provider is changing that on a regular basis I'd go elsewhere, there are better ways to do HA than having the resolution of your major service change all the time |
| 09:57.32 | brandon_1 | ok |
| 09:57.42 | brandon_1 | they changed it 3 times this week ... |
| 09:57.46 | Maliuta | any load balancer would take care of it, shit even a decent STONITH solution is better than that |
| 09:58.24 | Maliuta | I'd say they are either being way too cheap, or they don't have decent sysadmins in house |
| 09:58.35 | Maliuta | which is the same as a |
| 09:59.12 | brandon_1 | 6.95 excl. tax for 2 numbers |
| 09:59.19 | brandon_1 | EURO 6.95 excl. tax for 2 numbers |
| 10:01.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:05.40 | mort_gib | Morning... |
| 10:09.35 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 10:28.25 | mikkel | Is there any "standard" for dialplan layout, best practices. Ex. when making system menu, is it best to start all system menu with 9, * or #. Any links, ideas ? |
| 10:32.50 | mikkel | Is there no TRIM function, if the user entered "0001" I would like to trim the leading zeroes ? |
| 10:32.52 | mort_gib | Hej Mikkel |
| 10:34.01 | mikkel | mort_gib, Hey. Finally got the Sangoma card to work. Still can't get the phones working when connected directly. But found that Linksys 2101 and PAP2T is very cheap and makes life so much easier. |
| 10:38.54 | pputman- | mikkel, you wouldn't use a separate application to trim off numbers, you can use the {EXTEN} variable with a colon to cut off numbers off the variable. For example, exten => 001,1,Dial(sip/${EXTEN:2} |
| 10:39.17 | pputman- | mikkel, or you can use set to set a different variable to {EXTEN:2} if you need |
| 10:40.01 | pputman- | + ) |
| 10:41.28 | mikkel | pputman, Problem is I don't know how many zeroes there is. |
| 10:41.52 | mikkel | pputman, Could use an external command. |
| 10:42.03 | pputman- | mikkel, you can use a wildcard pattern. |
| 10:43.58 | pputman- | mikkel, I'm not an expert on dialplan patterns, but there are patterns to say "match 1 or 0 of these numbers" I think the period does that, so maybe (I might be wrong you'll have to read up on it) exten _0.1 => dial(sip/1) |
| 10:44.10 | pputman- | sorry, s,1,dial(sip/1) rather |
| 10:45.14 | mikkel | pputman, It is not in the extension, it is in the Read command, the user presses "001#" and I record a message with that number, but would like it to be saved as "1" and "001" = "01" = "1" |
| 10:46.33 | pputman- | mikkel, hrm, not sure then |
| 10:53.49 | mort_gib | If I want to monitor if Asterisk is running, what would I do?? /var/run/asterisk.pid or a ps fax | grep asterisk ?? |
| 10:53.55 | mort_gib | Ideas?? |
| 10:59.00 | tzafrir_laptop | mort_gib, ps `cat /var/run/asterisk/asterisk.pid` |
| 10:59.15 | tzafrir_laptop | that is: if you know the PID file, you need to verify it |
| 10:59.43 | tzafrir_laptop | monitoring programs (e.g. monit) normally check the name of the executable |
| 11:00.03 | tzafrir_laptop | ls -l /proc/THAT_PID/exe |
| 11:00.09 | tzafrir_laptop | this should be asterisk |
| 11:00.26 | tzafrir_laptop | Or even the name of the process |
| 11:01.34 | mort_gib | tzafrir_laptop: I have noticed that the .ctl file goes stale, but I can trust .pid ?? |
| 11:01.57 | tzafrir_laptop | You shouldn't. You should verify it. I just told you how |
| 11:02.15 | tzafrir_laptop | Check that there is such a process, and that this process is asterisk |
| 11:02.36 | tzafrir_laptop | asterisk.ctl is a unix domain socket |
| 11:08.43 | mort_gib | tzafrir_laptop: Thanks, looking into monit now, looks good |
| 11:09.13 | tzafrir_laptop | mort_gib, what type of monitoring exactly do you need? |
| 11:09.13 | WhiteWolf | he was just using that as an example, it's fairly simplistic to implement yourself |
| 11:09.54 | mort_gib | I have some issues with one of my clients, * went down the other day causing much frustration |
| 11:10.00 | mort_gib | So i need to know |
| 11:10.48 | mort_gib | I need to know if the service is running, I would LIKE to do more checks, like wanrouter (Sangoma) zaptel but I haven't got time right now.... |
| 11:11.35 | mort_gib | -Like Citrix where you connect to the port ans request a logon.... |
| 11:11.35 | tzafrir_laptop | What would you like to know about zaptel? |
| 11:11.44 | mort_gib | Running... Simple |
| 11:11.59 | mort_gib | In my views a few things are missing in * |
| 11:12.50 | WhiteWolf | perhaps they're there -- just not as obvious as you think |
| 11:13.05 | tzafrir_laptop | define "running". |
| 11:13.21 | tzafrir_laptop | If "asterisk process is up", I gave you a test |
| 11:14.38 | tzafrir_laptop | and you already have ssh access (or manager interface access) |
| 11:17.58 | mort_gib | WhiteWolf: True, we can always learn... how do you verify syntax in conf files prior to restarting * |
| 11:18.49 | mort_gib | I have had problems with dropped lines, I suspect my Telco (Bri/ISDN2) lines but the logging is rather confusing.... |
| 11:18.53 | tzafrir_laptop | that's indeed a problem |
| 11:19.28 | WhiteWolf | mort_gib: you don't need to restart * to reload /most/ of the conf files |
| 11:19.35 | WhiteWolf | mort_gib: a simple module reload will reload all of the ones it cna |
| 11:19.43 | WhiteWolf | and in doing so, it will spout off the errors |
| 11:19.57 | mort_gib | So I have a client, that has had dropped lines for a while, and yesterday I made a small syntax error, which didn't prevent * from a restart |
| 11:20.25 | mort_gib | All extensions worked fine, except for the pickup where I had forgotten a ) |
| 11:20.44 | mort_gib | I think I saved (:x!) a bit fast and zapped that damn ) |
| 11:20.56 | WhiteWolf | mort_gib: i think you're missing my point |
| 11:21.12 | mort_gib | Trouble is * just crashed when they tried to use the pickup |
| 11:22.01 | mort_gib | I'm sure I do, but pfctl (OpenBSD) allows for a "dry run" and wont load until pf.conf makes sense |
| 11:23.08 | WhiteWolf | mort_gib: it shouldn't crash on a bad extensions file/extention, maybe submit a bug report |
| 11:23.19 | WhiteWolf | mort_gib: but in either case, there's very few reasons to restart * |
| 11:23.44 | mort_gib | Yeah, I know but things has gone a bit out of hand, and my client i majorly pissed off |
| 11:31.35 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 11:32.00 | mandh | hi |
| 11:32.25 | mandh | exten => s,3,PlayBack(welcome-music&/var/lib/asterisk/sounds/ar/callwaitingsecondmessage.wav) |
| 11:32.38 | mandh | <PROTECTED> |
| 11:32.38 | mandh | ch file or directory |
| 11:32.54 | mandh | so is the problem that wav format? |
| 11:33.03 | WhiteWolf | what asterisk version |
| 11:33.23 | mandh | Asterisk 1.4.10.1 |
| 11:33.48 | WhiteWolf | hm |
| 11:34.06 | WhiteWolf | you need to omit .wav |
| 11:34.24 | mandh | aha? |
| 11:34.40 | mandh | but i have not the same file with gsm |
| 11:34.46 | mandh | so what happens then? |
| 11:35.18 | WhiteWolf | i'm sure there's some nature of precedence order |
| 11:35.25 | WhiteWolf | but tbh, not sure |
| 11:36.20 | mandh | let us test then |
| 11:37.18 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 11:37.18 | mandh | by the way can i put like |
| 11:37.18 | mandh | s,3,playback(f1) |
| 11:37.24 | mandh | s,3,playback(f2) |
| 11:37.32 | mandh | with same priortey |
| 11:37.49 | mandh | with same priortey |
| 11:37.52 | mandh | sorry |
| 11:37.54 | matrix1233 | hello |
| 11:38.05 | Mike8861 | mandd: what do u mean priortey |
| 11:38.21 | WhiteWolf | mandd: you cannot |
| 11:38.28 | mandh | i mean s,3 and in next line s,3 |
| 11:38.32 | WhiteWolf | instead, do playback(f1&f2) |
| 11:38.47 | mandh | yah i make that |
| 11:39.00 | mandh | but the pervious problem appear |
| 11:39.38 | WhiteWolf | the problem appears because you have the extension on there |
| 11:40.28 | mandh | humm i omit the extnerison , but there is an call on zap , so if i make reload may the call dropped?!! |
| 11:40.49 | matrix1233 | any one can tel me whay i cant found the h263 codecs for video when i install asterisk |
| 11:41.21 | matrix1233 | ?? |
| 11:42.57 | WhiteWolf | mandd: it will not drop the zap call |
| 11:43.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:45.24 | mandh | WhiteWolf, :) it is ok now |
| 11:47.26 | WhiteWolf | good to hear |
| 11:48.27 | mandh | why u are wolf? mmm |
| 11:48.39 | mandh | snowwolf |
| 11:56.52 | WhiteWolf | hehe |
| 11:56.57 | WhiteWolf | dunno |
| 12:03.51 | mandh | WhiteWolf, i will change mandh to SnowWolf |
| 12:04.41 | *** join/#asterisk tinloaf (n=tinloaf@d90-135-181-75.cust.tele2.de) |
| 12:04.47 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 12:06.15 | tinloaf | hi. question: how do i archieve this: my asterisk box has an ISDN-card. i configured extensions.conf to ring some SIP phone if a certain MSN is called, but the problem is that i seem to have to answer() the call before my SIP-Phone rings |
| 12:06.38 | tinloaf | i don't want that because there is another isdn-phone in the network which should ring simultanously |
| 12:07.09 | tinloaf | so I want this dialplan: "1. ring the SIP-phone, 2. If the SIP-phone accepts the call, answer() the call on ISDN" |
| 12:07.14 | tinloaf | is this possible? |
| 12:13.20 | Strom_M | you're going about it the wrong way |
| 12:13.33 | tinloaf | Strom_M: so what's the right way? |
| 12:13.34 | mvanbaak | hhmm, you shouldn't have to call the Answer() application |
| 12:13.55 | Strom_M | you should NEVER send an Answer() until the handset is actually lifted (and then the phone sends answer supervision back automatically) |
| 12:14.26 | Strom_M | the only time you should ever call Answer() is if you actually need to go off-hook to play a recording or accept input |
| 12:15.31 | tinloaf | oh, right |
| 12:15.41 | tinloaf | the problem was a mistake in my dialplan |
| 12:16.06 | tinloaf | seems like asterisk doesnt recognize an extension if the first priority is not 1 |
| 12:16.46 | Strom_M | well, duh |
| 12:22.21 | mort_gib | tzafrir_laptop: Just finished setting up monit, that works very well! |
| 12:23.42 | *** join/#asterisk alphanet (i=ircuser@shakotay.alphanet.ch) |
| 12:24.57 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 12:40.22 | MatBoy | when I have a ringroup where I want to dial an external number from, what kind of dial pattern do I need from internal ? |
| 12:42.20 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:45.56 | MatBoy | in the "trunk" ? (which i may not call so, but every sstem calls it like that :) |
| 12:47.09 | *** join/#asterisk [gnubie] (n=gnubie@222.127.211.226) |
| 12:53.15 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 12:55.50 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 12:56.15 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 12:56.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:57.00 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:57.30 | mort_gib | i get chan_sip.c: Maximum retries exceeded on transmission XXXXXX -What does this mean?? |
| 13:02.12 | *** join/#asterisk emiller (n=ed@206.166.206.34) |
| 13:29.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:34.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:39.33 | riddlebox | ~dahdi |
| 13:39.34 | jbot | [DAhdi] Digium/Asterisk Hardware Device Interface. The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi, and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav, |
| 13:40.36 | *** join/#asterisk JackEStorm (n=no@ip70-171-83-146.no.no.cox.net) |
| 13:41.05 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:05.18 | mikkel | What is it with this Digium Analog card. If I call an hangs up, it takes sometime before I can call again. Asterisk don't detect incoming calls ? |
| 14:06.29 | pputman- | mikkel, what type of signalling? |
| 14:06.35 | Mike8861 | mikkel: hello, can uj provide more information on the card |
| 14:06.35 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 14:06.48 | Mike8861 | mikkel: is that a FXO or FXS ? |
| 14:07.42 | codefreeze-lap | jbot is wrong! if you try to fetch the dahdi.wav file, you (or at the very least, I) get: Not Found The requested URL /dahdi.wav, was not found on this server. |
| 14:07.47 | mikkel | Mike8861, My incoming line FXS |
| 14:08.32 | mikkel | pputman, TDM422P |
| 14:09.11 | Mike8861 | mikkel: never have a problem with an FXS |
| 14:09.15 | pputman- | mikkel, pastebin your zaptel.conf zapata.conf and turn up verbosity and the console log and we might be able to help you better. |
| 14:10.37 | Mike8861 | pputman-: should u suggest using analog card or linksys sipura as a pstn trunk ? |
| 14:10.55 | mikkel | pputman, http://pastebin.com/m777c3900 |
| 14:10.58 | pputman- | Mike8861, dunno, would depend on your setup I guess. |
| 14:10.58 | Mike8861 | pputman-: for a noob who wants easy setup |
| 14:11.25 | Kyoshi | n00b |
| 14:11.26 | Kyoshi | heh |
| 14:11.28 | pputman- | Mike8861, I know a lot of voip providers are significantly cheaper than buying pstn service if you want full us coverage and voicemail, etc... |
| 14:11.33 | *** join/#asterisk ice_croft (n=323@81.26.135.117) |
| 14:11.44 | ice_croft | hi ppl |
| 14:11.50 | Kyoshi | mike, why do you want to use analog trunking? |
| 14:11.59 | Kyoshi | im curious |
| 14:12.07 | ice_croft | need help with audiocodes and avaya station |
| 14:12.11 | Mike8861 | pputman-: thank you for ur advice, but i am not a US resident(i want to be) |
| 14:12.11 | mikkel | Mike8861, I guess it is a FXO. (All this upperside stuff still confuses me) |
| 14:12.15 | Mike8861 | pputman-: living in HKSAR |
| 14:12.16 | pputman- | Mike8861, yeah, I can only speak from personal experience though. |
| 14:12.40 | Kyoshi | mike, nay yew jo analog trunking la, dim gai yah? |
| 14:12.41 | pputman- | mikkel, what version of zaptel and asterisk? |
| 14:12.45 | Mike8861 | mikkel: for FXO, either setup something in the conf file, or buy a busy tone detector |
| 14:12.47 | ice_croft | when im dialin from audoicodes through avaya, and hang up the avaya line, fxo cant see hung line |
| 14:12.50 | ice_croft | what to do? |
| 14:13.04 | Kyoshi | heh |
| 14:13.08 | Kyoshi | tried |
| 14:13.11 | Mike8861 | Kyoshi: are u hk ? |
| 14:13.21 | Kyoshi | nopes but i know the system there well |
| 14:13.29 | mikkel | pputman, zaptel-1.4.10.1 and Asterisk-1.6.0b8 |
| 14:13.43 | Mike8861 | Kyoshi: any suggestion ? |
| 14:13.45 | ice_croft | any issues? |
| 14:13.58 | Mike8861 | Kyoshi: in hk, we are getting HKBN 2B sip trunk |
| 14:14.01 | pputman- | mikkel, I would compile zaptel-1.4.11 and asterisk 1.4.21.1 and see if you can reproduce the problem with those versions. |
| 14:16.33 | *** join/#asterisk Aquahallic (n=aquaholi@CPE0018f85ad169-CM001ac3152b2a.cpe.net.cable.rogers.com) |
| 14:16.52 | pputman- | mikkel, if that doesn't fix the problem and you can't figure it out, please contact support. |
| 14:17.01 | mikkel | pputman, I'm pretty sure that asterisk has nothing to do with it. I have heard someone talk about this before and it is a problem with analog lines. |
| 14:17.42 | pputman- | mikkel, well sometimes a telephone company doesn't have hangup supervision, but that doesn't mean this is necessarily the case, it's always good to do as a troubleshooting step. |
| 14:18.39 | mikkel | pputman, Okay, will do that. Thanks. |
| 14:18.54 | pputman- | usually best to try to isolate and rule out whatever you can first. |
| 14:19.22 | *** join/#asterisk academy (n=adam@unaffilated/academy) |
| 14:19.36 | academy | Besides using a vpn, is there any way of using SIP without opening ports? |
| 14:22.11 | Aquahallic | Mornin' folks |
| 14:31.02 | macros73 | Trying to get g729 pass-through working. I have an Aastra 480i. I set g729&ulaw under allow for that extension. I set allow g729&ulaw&gsm to the outgoing trunk. Yet it still defaults to ulaw. |
| 14:31.32 | Strom_C | what happens if you just set it to g729? |
| 14:31.49 | Mike8861 | macros73: please pastbin sip.conf |
| 14:32.07 | Strom_C | Mike8861: please learn to spell pastebin :) |
| 14:32.08 | macros73 | Strom_C: All calls from the phone fail. I don't have a g729 license on this * server. |
| 14:32.31 | Strom_C | macros73: outbound calls are SIP as well? |
| 14:32.40 | Strom_C | (the "trunk") |
| 14:32.51 | macros73 | Strom_C: Yes. |
| 14:33.01 | MCooper | pputman, Good day good sir... How are things going? |
| 14:33.13 | Strom_C | does the device your asterisk box connects to support g729? |
| 14:35.20 | macros73 | http://pastebin.com/d30e05724 |
| 14:35.23 | macros73 | Strom_C: Yes. |
| 14:35.44 | macros73 | The Aastra 480i natively supports g729 |
| 14:35.54 | Strom_C | and the ITSP? |
| 14:36.00 | Strom_C | do they support g729? |
| 14:36.13 | Strom_C | pastebin the relevant outbound segment of extensions.conf |
| 14:36.19 | macros73 | Strom_C: yes. |
| 14:36.25 | macros73 | Strom_C: http://pastebin.com/d30e05724 |
| 14:36.32 | Strom_C | EXTENSIONS.CONF |
| 14:36.44 | Strom_C | please read and think instead of just reacting |
| 14:36.46 | macros73 | I'll extend your conf. |
| 14:38.44 | macros73 | Strom_C: No parts seem relevant. All the extensions files appear generic between PiaF installs. |
| 14:39.01 | Strom_C | PiaF? |
| 14:39.23 | macros73 | PBX in a Flash, and I was wrong, I just saw parts labeled for vitel. Hang on, will paste. |
| 14:40.50 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:41.23 | jaytee | mornin [TK |
| 14:41.35 | jaytee | oops, mornin [TK]D-Fender |
| 14:42.19 | [TK]D-Fender | *yawn* |
| 14:42.32 | macros73 | Strom_C: http://pastebin.com/d22bc8d39 |
| 14:42.41 | jaytee | hands [TK]D-Fender a cup of fresh roasted Kona |
| 14:43.04 | Strom_C | macros73: none of that is relevant |
| 14:43.16 | Strom_C | pastebin the CLI output of a failed outbound call at verbosity 10 |
| 14:47.24 | macros73 | Strom_C: nm, got it working. Needed to add disallow=all to the extension. |
| 14:47.34 | Strom_C | ok |
| 14:47.43 | macros73 | sip show channels confirmed it was passing g729 |
| 14:48.09 | macros73 | Now I need to debug why there was no ring tone and why it took 10-15 seconds for the call to go through. :D |
| 14:49.04 | macros73 | ok, this time it's ringing..weird |
| 14:52.47 | *** join/#asterisk uTx (n=unix@modemcable074.229-82-70.mc.videotron.ca) |
| 15:01.40 | *** join/#asterisk jack_sparo (n=eddy@dxb-b15780.alshamil.net.ae) |
| 15:01.51 | jack_sparo | hi |
| 15:01.54 | jack_sparo | where are announcments saved on asterisk? |
| 15:02.20 | Strom_C | jack_sparo: look in asterisk.conf for the path to the sounds directory on your system |
| 15:03.05 | jaytee | jack_sparo, usually /var/lib/asterisk/sounds |
| 15:05.16 | uTx | hello, I'm using the Dial application with the |M() option to run a macro that plays back a sound file when the called is connected. |
| 15:05.25 | uTx | My problem is that I want the sound to be heard by the person placing the call and not the called person. |
| 15:05.48 | uTx | Right now the sound file gets played to the called person and then the call is bridged |
| 15:06.54 | uTx | how can I make the calling person hear the sound file? |
| 15:07.07 | Strom_C | uTx: lemme look |
| 15:07.08 | Strom_C | hang on |
| 15:07.34 | uTx | thx |
| 15:08.46 | Strom_C | doesn't look like that's possible |
| 15:09.00 | Strom_C | why do you need to play a sound file to the calling party upon supervision? |
| 15:09.29 | uTx | to make him know he is connected |
| 15:09.33 | uTx | and if there is FAS |
| 15:09.43 | Strom_C | ...FAS? |
| 15:09.58 | uTx | false answer supervision |
| 15:10.12 | [TK]D-Fender | uTx: Sorry.. this STILL isn't going to work without real coding... |
| 15:10.26 | uTx | damn |
| 15:10.27 | Strom_C | [TK]D-Fender: oh, he's been here before asking about this? |
| 15:10.55 | [TK]D-Fender | Strom_M: He's working on a setup that invloves many foreign telcos and hopes to determine real status where some progress is only inband, and worse... |
| 15:11.02 | Strom_C | ugh |
| 15:11.21 | uTx | <PROTECTED> |
| 15:11.43 | uTx | <PROTECTED> |
| 15:11.51 | [TK]D-Fender | uTx: For diffrent things liely... but it clearly states its only for the CALLEE |
| 15:11.59 | [TK]D-Fender | likely* |
| 15:13.06 | uTx | well it says when the called party answers |
| 15:13.16 | uTx | not who hears what |
| 15:13.45 | uTx | aahh if the callee answers |
| 15:13.48 | uTx | but still |
| 15:14.23 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:14.24 | uTx | ok so who is the coder that can help me |
| 15:19.15 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 15:19.24 | [TK]D-Fender | uTx: go to the WIKI, put up a bounty |
| 15:20.05 | uTx | bounty hey |
| 15:25.37 | *** join/#asterisk bmg505 (n=leon@196-209-79-83-tbnb-esr-2.dynamic.isadsl.co.za) |
| 15:25.59 | Mike8861 | [TK]D-Fender: hey hey hey, you just wake up eh |
| 15:27.18 | *** join/#asterisk ManxPower (n=manxpowe@201.sub-75-202-187.myvzw.com) |
| 15:27.34 | mandd | [TK]D-Fender |
| 15:27.35 | mandd | http://pastebin.com/m1bbf64d8 |
| 15:27.39 | mandd | thats a verbose 10 |
| 15:28.05 | mandd | asterisk shows different phone numbers from the same provider as the same |
| 15:28.17 | mandd | it's biased by the host/ip I guess |
| 15:29.07 | [TK]D-Fender | mandd: what "number" are you talking about? |
| 15:29.19 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:30.00 | mandd | 416 849 1170 |
| 15:30.03 | mandd | 416 849 1130 |
| 15:30.16 | mandd | same provider, different numbers (two accounts) |
| 15:30.53 | mandd | while asterisk thinks its a same number, and routes them both to fax |
| 15:31.02 | [TK]D-Fender | mandd: indeed it is hitting the "other" peer. |
| 15:31.30 | mandd | is that a dialplan problem, or sometihng esle? |
| 15:31.39 | [TK]D-Fender | mandd: See the bigger "oops" is that it is sending the call to exten "s" because you did not tell them a better # to dial instead of "s" in your "REGISTER =>" line |
| 15:31.51 | [TK]D-Fender | mandd: THAT would have helped you differentiate calls. |
| 15:32.19 | [TK]D-Fender | mandd: "register => user:pass@host/ifyoudontfillmeini'mgoingtotellthemS |
| 15:32.28 | *** join/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 15:32.54 | mandd | oh, I see |
| 15:33.09 | [TK]D-Fender | mandd: INVITE sip:s@76.66.164.250 SIP/2.0 <- clean evidence of something you can fix to better ID the call coming in. |
| 15:33.31 | *** join/#asterisk diegoviola (n=diego@adsl-132-64.click.com.py) |
| 15:34.11 | mandd | i am gonna go google to see how to send REGISTER => to a different extension |
| 15:34.20 | ManxPower | SOME providers send all calls to all numbers on an account to the same destination |
| 15:34.38 | ManxPower | mandd: paste JUST the register => line without password. |
| 15:34.55 | [TK]D-Fender | mandd: I just TOLD you the exact formatting of that line |
| 15:35.05 | mandd | register=>4168491170:*****@tor3.voipportal.ca |
| 15:35.05 | mandd | register=>4168491130:****@tor3.voipportal.ca |
| 15:35.17 | [TK]D-Fender | mandd: you are missing the "/416blahblah" at the END of your register |
| 15:35.19 | mandd | oh, hehe |
| 15:35.22 | Strom | today's #asterisk is brought to you by Maalox |
| 15:35.30 | ManxPower | [TK]D-Fender: most providers default to the DID |
| 15:35.52 | mandd | and then I just do username=ifyoudontfillmeini'mgoingtotellthemS |
| 15:36.08 | [TK]D-Fender | ManxPower: meh... untrustworthy. Anything worth doing is worth doing yourself |
| 15:36.17 | [TK]D-Fender | mandd: NO. |
| 15:36.26 | [TK]D-Fender | mandd: you only have to change your register line |
| 15:36.33 | mandd | register=>4168491130:****@tor3.voipportal.ca/office |
| 15:36.36 | mandd | register=>4168491170:*****@tor3.voipportal.ca/fax |
| 15:36.40 | [TK]D-Fender | mandd: register=>4168491170:*****@tor3.voipportal.ca/4168491170 |
| 15:36.44 | mandd | oooo |
| 15:36.45 | mandd | hehe |
| 15:36.57 | ManxPower | mandd: stick to the DID, it will make life easier. |
| 15:37.06 | [TK]D-Fender | mandd: Jesus ^%#$ Christ. Follow dammit! |
| 15:37.12 | [TK]D-Fender | :p |
| 15:37.14 | mandd | sorry [TK]D-Fender |
| 15:37.38 | diegoviola | lol |
| 15:38.06 | [TK]D-Fender | mandd: Sure, you probably could put "office. It would then try to hit : exten => office,1,NoOp(remind me why I didn't jsut use a nice sane number again... I forget...) |
| 15:39.24 | ManxPower | we dump all our incoming DID calls into a special context just to handle the DID routing |
| 15:39.33 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-6-189.revip2.asianet.co.th) |
| 15:42.03 | *** join/#asterisk truent (n=truent@c-67-181-8-115.hsd1.ca.comcast.net) |
| 15:42.35 | mandd | http://pastebin.com/m17809d0e |
| 15:42.48 | mandd | the fex worked fine, but the office number didnt |
| 15:42.55 | mandd | now the CLI is goign insane |
| 15:43.50 | [TK]D-Fender | mandd: You can now stick to a singe inbound context as well for those 2 |
| 15:44.14 | [TK]D-Fender | mandd: and it looked fine in that last PB |
| 15:46.20 | mandd | aha, in a same context is works |
| 15:46.21 | mandd | wow |
| 15:46.23 | mandd | awesome |
| 15:46.28 | mandd | thanks so much [TK]D-Fender |
| 15:46.41 | mandd | you too ManxPower |
| 15:48.05 | [TK]D-Fender | mandd: You're welcome |
| 15:52.48 | jaytee | yep, no ifs, ands or buts, you guys rock! |
| 15:53.13 | ManxPower | we should go on strike for a better benefits package. |
| 15:53.35 | ManxPower | and an increase in salary too! |
| 15:53.46 | jaytee | hehehe |
| 15:55.06 | jaytee | maybe start your own IRC channel that is by subscription only? |
| 15:55.28 | jaytee | call it #asteriskpro |
| 15:55.50 | jaytee | or #asktheasteriskgurus |
| 15:56.08 | diegoviola | ... |
| 15:57.00 | mandd | or #asterisk |
| 15:57.42 | jaytee | and then you'd have all kinds of people joining up and even though you made them click the check box on the user agreement that clearly states trixbox etc are not supported they'll still pay to come in and ask and you can just ignore them and still keep the money! |
| 16:00.28 | [TK]D-Fender | jaytee: This channels topic is due for a change in that direction shortly. |
| 16:00.38 | [TK]D-Fender | jaytee: I just haven't gotten around to it |
| 16:02.30 | jaytee | everyone needs a round TUIT but they are in such short supply these days :-) |
| 16:02.51 | [TK]D-Fender | jaytee: Yes, the round ones are the most special and powerful. |
| 16:03.20 | [TK]D-Fender | jaytee: Dear God the things I could acheive if I I could only get a round TUIT. |
| 16:03.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:04.22 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 16:05.13 | errr | My grandpa used to have a whole box of them |
| 16:05.34 | errr | he would give them out to people all the time |
| 16:05.37 | jaytee | I worked on a team doing code conversions for cobol apps and one of the guys on the team was the poster boy for procrastination. I'd ask if he finished his mods that I was waiting on and he's usually reply "Not yet, but I'll get around to it." I started keeping cardboard disks with TUIT written on them in big bold black letters. I'd hand him one and say, "There ya go! Now go finish those damn mods!!!" |
| 16:05.41 | *** join/#asterisk CVirus (n=Satan@196.218.41.31) |
| 16:06.07 | errr | lol |
| 16:06.25 | CVirus | asterisk CLI just spit this out -- Remote UNIX connection |
| 16:06.27 | CVirus | what is that ? |
| 16:06.37 | mandd | Snom 220 - anyone ever used? |
| 16:07.06 | mandd | they look awfully nice |
| 16:07.07 | mandd | http://www.linuxdevices.com/articles/AT6041627024.html |
| 16:07.14 | CVirus | ah realized that I connected to asterisk from another terminal |
| 16:08.50 | unpaidbill | fart smellow! |
| 16:10.02 | Mike8861 | hello unpaidbill |
| 16:10.24 | Mike8861 | any one can provide information about "voice computer entertainment systems ", like those provide games via phone lines ? |
| 16:11.18 | mikkel | Is there no way to get a Queue to dalay playing music for 5 sec. I don't like people calling getting music immedietly. They should hear a couple of rings first ? |
| 16:12.09 | Strom | mikkel: how about play ringing tone, wait five seconds, and then dump them into the queue? |
| 16:12.35 | Mike8861 | mikkel: try adding a line to your dialplan before u play your ringtone |
| 16:12.49 | mikkel | Storm, then the phones is not ringing. |
| 16:13.52 | *** join/#asterisk d-k-t (n=dt@vpn21.witopia.net) |
| 16:14.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:14.26 | mikkel | I now call my Queue two time, one with a timeout of 5 sec and supression of music and the I call it normally. But in the trasition the phones stop ringing and you can't answer the call. |
| 16:14.33 | [TK]D-Fender | mandd: Given your location, forget anything except Polycom or Linksys |
| 16:14.43 | Strom | mikkel: you have to choose either music or ringing, then. |
| 16:15.24 | Strom | mikkel: or, alternatively, stop thinking up arguably stupid progressions of progress indications. |
| 16:16.03 | mikkel | Strom, Yes, or accept that sometime you will pick up the phone and noone is there and the secend you hangup it rings again (that sucks) |
| 16:16.19 | Strom | mikkel: no, that's a brain-dead way of doing ti |
| 16:16.21 | Strom | s/ti/it/ |
| 16:18.46 | mikkel | Strom, I don't believe iøm the first one that don't want music to start the second you connect... |
| 16:19.17 | Strom | mikkel: what's your expected average hold time for queued callers? |
| 16:20.08 | mikkel | Strom, no more than 10 sec. |
| 16:20.19 | Strom | mikkel: so screw the music and just play ring tone then |
| 16:20.28 | Strom | s/ring/alerting. |
| 16:20.31 | Strom | ah, fuck it. |
| 16:20.39 | Strom | mikkel: so screw the music and just play alerting tone :) |
| 16:21.11 | Strom | 10 seconds is less than two full north american ring cycles |
| 16:21.17 | mikkel | Strom, And I would like the caller not to hear music must of the time. But if a second caller is calling he should here music after 10 sec. |
| 16:22.37 | Strom | mikkel: you've got unrealistic expectations of how this sort of thing should work. If you're that concerned that a caller is going to hang up after more than ten seconds of alerting tone, then answer the call, play a recording that says "Please hold while I connect you to someone" (or whatever), and then dump them into the queue and play music. |
| 16:23.17 | Strom | having the audio abruptly shift from ringing to music is confusing to the caller. |
| 16:23.18 | ManxPower | mikkel: you are trying to have too fine grained control. Why don't you just make it work OK, then go back when you understand more of Asterisk to try making it work better? |
| 16:23.45 | jeev | Strom |
| 16:23.50 | Strom | cocks |
| 16:23.51 | jeev | i've had bigger chunks of corn in my crap! |
| 16:23.52 | ManxPower | heck you can even run Ringing() for a few seconds before sending the call to the queue, that's what we do |
| 16:23.55 | jeev | what is there to do in palm springs |
| 16:23.58 | jeev | i hate that place |
| 16:24.02 | jeev | but my friend wants to go gamble |
| 16:24.09 | Strom | jeev: i don't know; I've never been. |
| 16:24.13 | mikkel | Strom, I don't se how I can answer the call if the caller is no transfered into the Queue. |
| 16:24.22 | jeev | it's a lame place bah |
| 16:24.28 | ManxPower | mikkel: Answer() |
| 16:24.35 | mikkel | Strom, The Queue is what makes the phone ring. |
| 16:24.47 | ManxPower | jeev: Las Vegas or at least Nevada is not far away from Palm Speings |
| 16:25.03 | ManxPower | mikkel: but Queue is not the only thing that can make the caller HEAR a ring. |
| 16:25.03 | jeev | yea i know manx, i own a condo in vegas |
| 16:25.11 | jeev | but i dont wanna go cause my friend is here from vegas.. at least he's comin tomorrow |
| 16:25.12 | jeev | so it's poitnless |
| 16:25.29 | Strom | jeev: your friend lives in vegas and wants to go gambling? |
| 16:25.36 | Strom | jeev: he must be new to vegas |
| 16:25.39 | jeev | no, other friend wants to go to palm springs |
| 16:25.46 | jeev | my friend from vegas is coming here tomorrow, no reason to go to vegas |
| 16:25.51 | mikkel | ManxPower, Not but make the caller hear a ringtone, without me being able to answer is stupid. |
| 16:26.02 | Strom | mikkel: not necessarily |
| 16:26.26 | Strom | mikkel: your idea of abruptly shifting from alerting tone to music is far stupider. |
| 16:26.53 | ManxPower | not in the least. If the caller hears a couple of rings before anything else it makes them feel comfortable because they know the call went thru |
| 16:28.00 | ManxPower | mikkel: what you want to do is very difficult -- maybe impossible to do with Asterisk. |
| 16:28.27 | ManxPower | And personally, I think time is better spent learning useful things in Asterisk, designing your dialplan, testing stuff, etc |
| 16:28.27 | Strom | playing music without some indication that you've reached a queue is confusing. |
| 16:28.40 | mikkel | ManxPower, exactly my point. But I would like to be able to answer the call before the caller hears music. Since my phones is in the Queue it will not be able to answer before the user hears music. |
| 16:29.04 | ManxPower | mikkel: so give up or use a different product if that feature is SO terribly important. |
| 16:29.25 | [TK]D-Fender | mikkel: I have a solution for you. |
| 16:29.35 | [TK]D-Fender | mikkel: a nice flat & simple one too. |
| 16:29.57 | mikkel | [TK]D-Fender, That is ? |
| 16:30.37 | [TK]D-Fender | mikkel: Se a MoH class to a folder with ONE file in it. that file will have your few seconds of ringing in it followed by a long mashup of "whatever". Set it so that it always starts from the beginning. |
| 16:31.36 | mikkel | Strom, I see your point. And I should alert callers that they are entering a queue. But the problem that I'm try to avoid is that while transfering them to the Queue (and alerting) I loose the abillity to answer. |
| 16:31.56 | Strom | mikkel: well, how about this then |
| 16:32.02 | ManxPower | mikkel: there are at least 5 different ways to Answer, be more specific |
| 16:32.12 | Strom | (a) learn the difference between "lose" and "loose" so you don't sound like a total idiot |
| 16:32.44 | Strom | (b) consider ringing your telephone sets for 12 seconds, then transferring to a recording saying "please hold while we queue you up" and then do the queue with music |
| 16:32.54 | Strom | (c) use [TK]D-Fender's suggestion |
| 16:33.07 | Strom | (d) learn the psychology of setting up telephone systems |
| 16:34.13 | ManxPower | 20 - 30 seconds is also a good number, 20 is a good one since most people expect SOMETHING to happen within 20 seconds |
| 16:34.43 | [TK]D-Fender | mikkel: you ring. Then you play the announcement that they are entering a queue, and then continue music. to THEM they can't tell the difference, but they were ALWAYS in the queue |
| 16:35.03 | RoyK | ~asterisk |
| 16:35.03 | jbot | hmm... asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 16:35.06 | *** part/#asterisk mond0 (n=lvsyijnc@galactic.umflint.edu) |
| 16:35.07 | [TK]D-Fender | mikkel: and you never lose the ability to answer. its rings jsut like normal, because its just a queue. |
| 16:35.08 | RoyK | ~asstrix |
| 16:35.09 | jbot | i guess asstrix is a nickname given to the Aterisk PBX, hinting on various design flaws and "features" |
| 16:35.51 | mikkel | (a) English is not my first language. (b) That is what I do now, but would could pick up the phone when it is transfered and no one is there. (c) That is maybe a solution (Don't have many callers so CPU is no problem) (d) ? |
| 16:36.07 | mikkel | [TK]D-Fender, I think that is a good idea. |
| 16:36.16 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 16:37.02 | [TK]D-Fender | mikkel: in my way there is no such thing as "transferred". Each agent callout is just an agent callout. They get the same normal amount of time as each other. |
| 16:37.10 | [TK]D-Fender | mikkel: there is no "2nd" ring to them, etc. |
| 16:37.35 | [TK]D-Fender | mikkel: If your users want to live an illusion, there you have it. It is just that, an illusion. |
| 16:41.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:43.13 | *** join/#asterisk CVirus (n=Satan@196.218.41.31) |
| 16:46.09 | ManxPower | mikkel: WHY would nobody be there when you pick up the phone? |
| 16:46.47 | ManxPower | ~royk |
| 16:46.48 | jbot | methinks royk is that viking asterisk guru, or your friend |
| 16:47.14 | mikkel | ManxPower, Because the caller is taken out of the Queue and inserted into a new Queue. |
| 16:56.40 | *** join/#asterisk phlax (n=ryan@62.24.178.184) |
| 17:06.33 | [TK]D-Fender | mikkel: which is why that plan is unadvisable, plus you're creating extra calls for nothing. All things that my method does not fall victim to. |
| 17:06.43 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 17:07.11 | ThatKidKel | is it possible from the console to take two different legs of a call and transfer them into a meetme bridge |
| 17:09.35 | mikkel | [TK]D-Fender, Yes, I will go that way. Just can't get it to play "periodic-announce" |
| 17:09.56 | *** join/#asterisk zeirthz (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 17:10.02 | zeirthz | hi |
| 17:10.20 | zeirthz | i got grandstream 486 adaptor |
| 17:10.31 | zeirthz | with 1 FXS port and 1 FXO |
| 17:11.14 | [TK]D-Fender | mikkel: INVENT it yourself, or set it so that it starts with a far enough dealy so as to cooperate with your "fake-out" |
| 17:11.21 | zeirthz | i connect my Xtelecom line to fxs port |
| 17:11.31 | zeirthz | i connect my Xtelecom line to fxO port |
| 17:11.54 | zeirthz | and i connect my asterisk server line to the FXS port |
| 17:12.48 | zeirthz | i also create 5 sip line |
| 17:13.06 | zeirthz | one sip line i use it from my home |
| 17:13.11 | zeirthz | one sip line i use it from my work sorry |
| 17:13.35 | zeirthz | my question is how to use my home pstn line from my work ? |
| 17:16.36 | [TK]D-Fender | zeirthz: that devices "FXO" port is NOT an FXO port. It is a dumb pass-through port only. It is not usable by Asterisk |
| 17:18.31 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 17:19.23 | zeirthz | ok |
| 17:19.32 | zeirthz | thx tk |
| 17:19.40 | tzafrir_laptop | so the 486 isn't something like the sipura 3000? |
| 17:19.42 | zeirthz | i also got sipura 3000 |
| 17:20.01 | zeirthz | is it possible to do the thing |
| 17:20.04 | zeirthz | i want |
| 17:20.05 | zeirthz | ??? |
| 17:20.26 | zeirthz | from sipura 3000 ? |
| 17:20.29 | zeirthz | help |
| 17:20.31 | zeirthz | plz |
| 17:20.42 | zeirthz | i past one month on this problem |
| 17:21.00 | zeirthz | i can't get all correct information |
| 17:22.18 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 17:24.51 | *** join/#asterisk jbeez (i=jbeez@jbeez.net) |
| 17:25.08 | tzafrir_laptop | <zeirthz> and i connect my asterisk server line to the FXS port |
| 17:25.18 | jbeez | Anyone know how to setup dhcp/tftp for a polycom phone to download a SIP configuration? |
| 17:25.25 | tzafrir_laptop | that makes no sense - connect to your asterisk server through SIP |
| 17:25.42 | jbeez | I have dhcp and tftp working to pxe boot some OSes, so most of the basic part is setup, just not sure on the phone part, and options, etc |
| 17:27.19 | mikkel | [TK]D-Fender, Hmm, doing the annoucment, the phonse stop ringing...... |
| 17:28.02 | mikkel | [TK]D-Fender, Will continue tomorrow. Thanks for all help. |
| 17:33.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:33.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:35.35 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 17:47.23 | *** join/#asterisk T_Boone_Pickens (n=tboone@user-24-214-126-81.knology.net) |
| 17:51.06 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 17:56.43 | *** join/#asterisk KurtKraut (n=ktk@unaffiliated/kurtkraut) |
| 17:57.20 | KurtKraut | I need to run a SIP server to do conference calls using softphones only. Can anyone recommend me a HowTo or tutorial? |
| 17:57.29 | Strom | ~book |
| 17:57.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:57.39 | Strom | KurtKraut: ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:58.32 | KurtKraut | Strom, are you sure that book cover this topic ? |
| 17:58.35 | *** join/#asterisk trifon (n=chatzill@87-126-48-189.btc-net.bg) |
| 17:58.49 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:58.50 | Strom | KurtKraut: the book contains everything you'll need to know (and more) |
| 17:59.17 | *** join/#asterisk bl4q (i=Bl@dslb-088-064-158-017.pools.arcor-ip.net) |
| 17:59.24 | KurtKraut | Strom, I'm glad to hear that. Thanks |
| 17:59.48 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:06.53 | Kyoshi | Dialplan help with realtime please... http://pastebin.com/m1c339312 |
| 18:07.43 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 18:09.17 | T_Boone_Pickens | qwell |
| 18:09.33 | hsv-al | i was looking at this openmoko, seems tight |
| 18:09.36 | hsv-al | but its only gprs |
| 18:09.38 | hsv-al | ftl |
| 18:11.33 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 18:24.33 | tzafrir_laptop | Someone should update the asterisk package for it |
| 18:24.36 | tzafrir_laptop | ~someone |
| 18:26.39 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:26.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:28.37 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 18:30.35 | academy | Is it possible to use SIP without opening any ports? |
| 18:33.39 | j0 | academy: you can have a client behind nat, but the "server" would have to have open ports |
| 18:38.15 | mandd | which level logging do I set in CLI so that I can see incoming call number |
| 18:39.02 | academy | j0: I thought the client hat to open and forward ports on their end too. |
| 18:41.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:53.12 | academy | If the client is behind NAT, do you have to open any ports to allow the client to use SIP? |
| 18:57.12 | hsv-al | tzafrir |
| 18:57.13 | hsv-al | openM? |
| 18:57.32 | jaytee | ~sipnat |
| 18:57.33 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:59.05 | academy | jaytee: thank you |
| 18:59.36 | jaytee | academy, yw |
| 19:03.42 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:03.52 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 19:06.01 | *** join/#asterisk teknologika (n=ariel@172puntacana02.codetel.net.do) |
| 19:06.15 | teknologika | hello world |
| 19:06.40 | teknologika | is somebody at home? |
| 19:06.57 | jbeez | no |
| 19:07.09 | teknologika | mmmm |
| 19:07.49 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 19:08.00 | teknologika | great |
| 19:08.16 | jbeez | :D |
| 19:08.29 | teknologika | how u doing |
| 19:09.06 | jbeez | ok, was just messing with a sip phone |
| 19:10.00 | teknologika | kool |
| 19:10.12 | teknologika | im new on asterisk systems |
| 19:10.20 | jbeez | im pretty new myself |
| 19:10.57 | teknologika | well is funny for me |
| 19:11.17 | teknologika | i just installed and is working pretty good for me |
| 19:11.26 | teknologika | with grandstream 101 |
| 19:11.33 | teknologika | voip phone |
| 19:11.50 | teknologika | how are you doing yours |
| 19:12.58 | jbeez | ok, just got it working last week, and I just tested bringing a phone home friday |
| 19:13.12 | jbeez | I can dial extensions on our traditional pbx from this at my house |
| 19:13.25 | teknologika | its great! |
| 19:13.31 | jbeez | i can't make outgoing calls though, I think the T1 to the asterisk box is down since we weren't using it, so I gotta call the telco monday |
| 19:14.12 | teknologika | well thats can be, or something with the zap |
| 19:14.48 | teknologika | check if the hardware is up |
| 19:14.59 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 19:15.08 | hsv-al | http://blog.tmcnet.com/blog/tom-keating/apple/sipgate-adds-sip-voip-support-to-iphone.asp |
| 19:15.18 | hsv-al | (7/12/2008) sipgate today announced the a free VoIP application for the iPhone that they claim will allow consumers to use virtually every any VoIP service on the iPhone. |
| 19:15.43 | teknologika | yeah thats old |
| 19:16.01 | hsv-al | wtf, "Of course, as I have written before, this service will only work over a Wi-Fi connection and not the iPhone's cellular data connection." |
| 19:16.03 | teknologika | but the good new can be if is blackberry doing the same |
| 19:16.12 | hsv-al | are there any coming out that will ride on the 3G? |
| 19:16.26 | teknologika | no no no |
| 19:16.32 | teknologika | is the best |
| 19:16.36 | teknologika | for a while |
| 19:16.42 | jbeez | iphones cellular data connection is pretty weaksauce |
| 19:16.51 | jbeez | I hate the data on my iphone if im not using wifi |
| 19:16.56 | teknologika | i said 3g speeds |
| 19:17.26 | teknologika | hey |
| 19:17.36 | teknologika | i just waiting for make some test with palm |
| 19:18.03 | teknologika | and is not bad idea have an account on your mobile |
| 19:18.06 | teknologika | ;-) |
| 19:18.43 | teknologika | maybe can you explain to me |
| 19:19.08 | teknologika | how can I play background music on my speakers phone |
| 19:19.32 | teknologika | any idea? |
| 19:20.08 | jbeez | noidea |
| 19:21.11 | *** part/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 19:21.17 | teknologika | I dont have either |
| 19:21.48 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 19:22.04 | teknologika | and did you check your zap status |
| 19:22.44 | jbeez | ? |
| 19:22.48 | jbeez | who are you talking to? |
| 19:22.55 | teknologika | to you |
| 19:23.00 | teknologika | about your t1 |
| 19:23.05 | jbeez | no |
| 19:23.12 | jbeez | I didn't have time to mess with it last week, very busy |
| 19:23.27 | teknologika | i know that |
| 19:25.09 | teknologika | well i got to go |
| 19:25.19 | teknologika | good look |
| 19:25.27 | teknologika | with the t1 |
| 19:25.30 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 19:26.06 | *** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-efacbac7c76f32ef) |
| 19:27.19 | TJNII | Oh what in the hell..... No I'm getting one way audio and no DTMF from my ITSP...... |
| 19:27.28 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000517.dsl.bell.ca) |
| 19:27.55 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 19:28.09 | ice_croft | Fender, hi |
| 19:28.16 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 19:31.40 | hardwire | why does chan_local reset cdr? |
| 19:36.14 | TJNII | Problem on their end.... changes hosts and it went away. |
| 19:40.14 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 19:52.56 | ice_croft | hey, is any gurus here now? |
| 19:53.17 | l2cache | yeah |
| 19:54.01 | ice_croft | man, my fxo-gtw can see that avaya line hung |
| 19:54.06 | ice_croft | where to dig? |
| 19:54.26 | ice_croft | when i put down avaya phone, fxo still hold the channel |
| 19:54.31 | ice_croft | no hangup ( |
| 19:54.41 | Kyoshi | Dialplan help with realtime please... http://pastebin.com/m1c339312 |
| 19:55.33 | ice_croft | this doesnt happen when i hang pstn line |
| 19:56.06 | ice_croft | l2cache> any issues? |
| 20:10.13 | *** join/#asterisk th1 (n=th@pdpc/supporter/sustaining/th1) |
| 20:11.08 | *** join/#asterisk ftugrul (n=tugrul@unaffiliated/ftugrul) |
| 20:11.46 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 20:21.39 | lesouvage | I'm trying to set up a sip conenction between two asterisk server within my network following the instructions on http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ . When trying to call from Paris to Lyon I got the following message on Lyon server . Is somethng missing in the instruction or am I doing something really wrong? |
| 20:22.17 | *** part/#asterisk ftugrul (n=tugrul@unaffiliated/ftugrul) |
| 20:24.58 | *** part/#asterisk phlax (n=ryan@62.24.178.184) |
| 20:26.55 | *** join/#asterisk Imagineer66 (n=sostrate@c-67-177-134-11.hsd1.mi.comcast.net) |
| 20:28.00 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-186-173.telkomadsl.co.za) |
| 20:28.40 | l2cache | If anyone is looking for a failover solution, I just released a script at www.thiscoolsite.com |
| 20:31.53 | TJNII | lesouvage: hard to say seeing as how you geve no specifics at all. |
| 20:32.49 | *** part/#asterisk Imagineer66 (n=sostrate@c-67-177-134-11.hsd1.mi.comcast.net) |
| 20:33.43 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 20:34.13 | lesouvage | TJNII: I gave a link to an instruction I'm following. I'm just trying to replicate the instruction but with mixed success. The actual question is: should it work if I follow this instructions. Paris and Lyn are the names of the two servers used in the instruction |
| 20:34.20 | xacatecas | hi guys, when making a call between a SNOM M3 and a Grandstream BT-200 I get some stutter on the BT-200 side, codec is ulaw with rtp passing through an asterisk server. based on a tcpdump it doesn't look like VAD packets is being sent so I'm rather baffled. |
| 20:35.02 | lesouvage | xacatecas: any change that silence surpression is enabled? |
| 20:35.29 | lesouvage | On one of the phones |
| 20:36.01 | TJNII | lesouvage: That's spectacular. If you want support you need to state the problem you are having including the error message. Saying "I'm following a tutorial and its not working" is not a answerable question. |
| 20:36.26 | xacatecas | lesouvage, i've checked and rechecked, so i don't think so. |
| 20:36.39 | *** join/#asterisk th1 (n=th@pdpc/supporter/sustaining/th1) |
| 20:37.31 | xacatecas | in fact, i've switched it on on the bt200 to see if it makes any difference, but it doesn't. I've switched it off again. |
| 20:37.41 | xacatecas | struggling to find the option on the M3 side. |
| 20:37.47 | *** join/#asterisk Absnthe (n=Christop@ool-43561407.dyn.optonline.net) |
| 20:37.54 | *** part/#asterisk Absnthe (n=Christop@ool-43561407.dyn.optonline.net) |
| 20:39.34 | lesouvage | xacatecas: silence surprssion is a setting that goes under difference names and saves bandwidth because is silence is't sed anymore. Asterisk can't handle that very well and the result is a stuttering sound. |
| 20:40.07 | xacatecas | ok, so the incoming from the M3 may have it on? but the outbound won't? |
| 20:40.22 | xacatecas | any way to ask wireshark to quickly search for rtp packets indicating VAD? |
| 20:41.48 | xacatecas | ok, the packets seems to be alternating nicely between the asterisk server and the m3 so i'm guessing there's no VAD ... not wanting to read through all the packets again like for the bt200. |
| 20:42.31 | lesouvage | TJMII: The idea of my question was to get conformation about the acuracy of the instruction to set up a sip trunk because I run into trouble following them. |
| 20:47.31 | xacatecas | i cannot reproduce between two bt200 phones. |
| 20:47.38 | xacatecas | i don't have another snom to test with :( |
| 21:02.06 | *** join/#asterisk implicit (n=bayan@ip68-4-97-211.oc.oc.cox.net) |
| 21:03.03 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 21:03.49 | lesouvage | TJMII: I'm half way. Making phonecalls frm server 1 (Lyon) to server 2 (Paris) works ok but the other way around I got a message in the cli of server 1. Failed to authenticate user "501" <sip:501@192.168.1.113>;tag=as3844775e . This is kid of strange because 501 is the number of the phone registered on server 2. Server 1 shouldn't have anything to do with it. Any suggestions? |
| 21:04.46 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 21:04.47 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 21:11.26 | xacatecas | lesouvage, complete setup? |
| 21:12.12 | xacatecas | hehe, here's the funny. the snom m3 doesn't (according to the user manual at least) support vad. |
| 21:12.47 | *** join/#asterisk cabbiepete (n=cabbiepe@host86-143-32-91.range86-143.btcentralplus.com) |
| 21:13.02 | *** join/#asterisk th1 (n=th@pdpc/supporter/sustaining/th1) |
| 21:16.14 | xacatecas | hmm, interesting, the latency as shown by sip show peers is 6ms for the snom as compared to 20 for the bt200s. |
| 21:17.18 | lesouvage | xacatecas: On server 2 I am using sip phone 501 and on server 1 I also had a sip phone entry 501. Now tht I removed the si entry 501 on server 1 it is is working. I do not completely understand it bt I seem to be out of trouble and have a proper setting to test my early media challanges. |
| 21:18.26 | lesouvage | I also had to add insecure=very. |
| 21:18.54 | xacatecas | why?!? i've only ever needed when i connected to a rather braindead voip provider. |
| 21:21.18 | lesouvage | xacatecas: Wel I will try without and looks if it is still working. It just for a test setup so it doesn't actually matter. |
| 21:21.38 | xacatecas | i still like understanding. |
| 21:21.52 | xacatecas | at the moment I've got to say SIP is irritating the daylights out of me. |
| 21:24.43 | *** join/#asterisk Freeaqingme (i=5064b031@gateway/web/ajax/mibbit.com/x-aa819d4a56884313) |
| 21:25.33 | Freeaqingme | hey guys, I'm trying to get started with asterisk. What (n00b-)tutorial do you advise? |
| 21:25.35 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 21:26.55 | lesouvage | xacatecas: you are right, don't need the insecure=very line to have it working. |
| 21:27.16 | xacatecas | ~book |
| 21:27.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:27.54 | Freeaqingme | ty xacatecas |
| 21:29.19 | lesouvage | xacatecas: ngrep -p -q -W byline port 5060 > /home/erik/sip_messages.txt is a handy line to trace all sip packages. |
| 21:30.02 | lesouvage | xacatecas: the result might be that sip will starts to irritate you even more ;-) |
| 21:31.06 | xacatecas | hehe, yea well. it can't get much worse. on the upside, windows still irritates me even more. i'll admit that sip works pretty well on a lan between my phones and asterisk server. |
| 21:44.20 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 21:46.17 | lesouvage | Can somebody explain to me or point me to info why it is a problem to have the same sip entry on two asterisk servers that are connected with a so called sip trunk ([TK]D-Fender: I really would like to stop irritating you and stpping with the use of the word sip trunk but how would you like me to name it?) |
| 21:48.24 | lesouvage | This is the cli output: WARNING[2295]: chan_sip.c:8385 check_auth: username mismatch, have <501>, digest has <trunk-lyon-paris> |
| 21:48.44 | lesouvage | chan_sip.c:13821 handle_request_invite: Failed to authenticate user "501" <sip:501@192.168.1.113>;tag=as58c05155 |
| 21:49.05 | xacatecas | lesouvage, is the call coming from 501 through to 501 on the other side? |
| 21:51.37 | lesouvage | No it is a call from server 2 using sip phone number 501 to an extension on server 1. on 501 there is a (not in use) sip entry in sip.conf. If I remove this sip entry and reload the problem is gone and the sip trunk is up and running . |
| 21:58.49 | *** join/#asterisk lucky|aba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 21:59.07 | lucky|aba | can someone explain what the context in the conf files is? |
| 21:59.11 | lucky|aba | is it just a label? |
| 22:00.48 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 22:05.38 | *** join/#asterisk d-k-t (n=dt@vpn21.witopia.net) |
| 22:08.46 | xacatecas | lucky|aba, in which context :)? - which file? |
| 22:09.13 | lucky|aba | sip, extensions, .. .conf |
| 22:09.20 | lucky|aba | I am wondering if they have to match up or are they just there to help you remember what is what |
| 22:10.27 | lucky|aba | specifically, i am trying to setup voicemail and the [mb_tutorial] is in there |
| 22:16.55 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 22:18.36 | *** join/#asterisk thansen|laptop (n=thansen@7.247.sfcn.org) |
| 22:19.29 | thansen|laptop | I'm trying out 1.6.0 beta and getting this error...channel.c:752 ast_best_codec: Don't know any of 0x0 formats...can someone guide me to get this resolved? |
| 22:22.13 | xacatecas | lucky|aba, it's the name of the context in which a call originating from the sip channel will start in, and should be defined somewhere in extensions.conf |
| 22:22.40 | lucky|aba | uhh, I think i get what your saying |
| 22:23.03 | lucky|aba | IS there some faster way in creating extensions for folks? like a gui or something? |
| 22:26.33 | xacatecas | freepbx |
| 22:26.41 | xacatecas | and a GUI is usually slower. write a script. |
| 22:27.07 | lucky|aba | hahah, oh yeah |
| 22:27.21 | lucky|aba | That is a good idea |
| 22:32.14 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-220-rndf-esr-5.dynamic.isadsl.co.za) |
| 22:34.50 | *** join/#asterisk bsaxon (n=bsaxon@61.sub-70-222-90.myvzw.com) |
| 22:35.19 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 22:40.43 | th1 | any who are debian package maintainers for asterisk in here.. |
| 22:40.53 | th1 | because the debian init scripts are totally broken |
| 22:45.16 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:50.18 | lesouvage | luckylaba: the extensions.conf might be look a little bit confusing but as soon as you understand what a context, an extension and a priority is editing the files is the fastest way to create a dialplan and certainly the most maintainable way. Dial plans generated with a gui tend to be a total nightmare as soon as something goes wrong. |
| 22:54.23 | lucky|aba | fair enough |
| 22:54.38 | lucky|aba | So are the contexts like classifications for users? |
| 22:54.49 | lucky|aba | like [home] could only allow internal calls |
| 22:55.08 | lucky|aba | [away] could allow users with that context to make outbound calls? |
| 22:58.43 | lesouvage | Somethng like that, but is you who defines what is happening in a context. One basic principle for incoming calls is that there has always to be a match in the dial plan (just google for number matching asterisk and you will see lots of examples) |
| 22:59.40 | lucky|aba | I am just working with extensions on my LAN right now. Once i get the idea there i want to setup incoming and outgoing services |
| 23:01.56 | lesouvage | luckylaba: You really should spend some time reading, googling and trying. The book is a good start. some google tips are "asterisk guru" "asterisk russell" "asterisk cmd <asterisk command>" |
| 23:04.36 | Freeaqingme | I'\m trying to get started with asterisk. Asterisk is listening on port 5060, I\m trying to connect to that server through ekiga, but nothing happens |
| 23:04.56 | Freeaqingme | even when I do a strace on the asterisk process nothing happens, what could I be doing wrong? |
| 23:05.42 | lucky|aba | lesouvage, yeah that is what i am doing now. Going through the tutorials on voip-info and asterisk guru |
| 23:06.56 | lucky|aba | Did you register the extensions? That threw me for a while when i first tried setting it up |
| 23:07.13 | Freeaqingme | you're asking me? |
| 23:07.43 | Freeaqingme | I entered the info for both sip.conf and extensions.conf as described here.. http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client |
| 23:07.50 | Freeaqingme | so I suppose that should be correctly |
| 23:08.06 | lesouvage | Freeaqingme: are you sure Asterisk is running, have you add a sip entry for ekiga and is it registrered, have you reloaded after changing config files. |
| 23:08.48 | Freeaqingme | I have restarted the asterisk process several times, no errors of loading in the logfiles |
| 23:09.36 | Freeaqingme | I\m busy with it only 2 hours, guess that's not enough :P |
| 23:10.10 | lesouvage | are you at the cli (character line interface) of asterisk |
| 23:10.30 | Freeaqingme | yup |
| 23:10.36 | Freeaqingme | it doesn't show any channels |
| 23:10.57 | lesouvage | freeaqingme: have you tried "sip show peers" |
| 23:11.57 | Freeaqingme | lesouvage: that gives me one peer with status UNKNOWN, and host unspecified |
| 23:12.31 | lesouvage | what happens when you enter "reload" |
| 23:12.54 | lesouvage | on the cli |
| 23:14.25 | Freeaqingme | lesouvage: then it starts to output a whole lot of lines |
| 23:14.40 | Freeaqingme | so many that I'm not able to see them all in my output buffer... |
| 23:14.54 | lesouvage | And if you do a sip show peers again |
| 23:15.33 | Freeaqingme | no change |
| 23:15.55 | Freeaqingme | but when I do a strace on the asterisk process, nothing is shown, so it appears there\s nothing sent to the port/socket anyways |
| 23:16.21 | lesouvage | Are you sure the server and the pc running ekiga are on the same network. |
| 23:16.32 | Freeaqingme | :P |
| 23:16.36 | Freeaqingme | yeah, they are |
| 23:16.50 | lesouvage | and you didn't pick up a IP number from your neigbours ap |
| 23:17.05 | Freeaqingme | damn it, nvm :X |
| 23:17.14 | lucky|aba | lesouvage, "sip show peers" I was looking for exactly that earlier |
| 23:17.15 | lucky|aba | thanks |
| 23:17.37 | Freeaqingme | once a year, I internet over a wireless connection, which I do now too.... and guess what apparently my AP is set to NAT instead of Bridgemode :'( |
| 23:18.04 | Freeaqingme | thanks anyways lesouvage :P |
| 23:18.42 | lesouvage | can you ping from your laptop to your asterisk server? |
| 23:19.39 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 23:19.50 | lesouvage | luckylaba: there is also sip show registry and sip show users |
| 23:20.11 | lesouvage | and sip set debug and sip set debug off |
| 23:20.30 | Freeaqingme | lesouvage: yes I can |
| 23:20.38 | Freeaqingme | but it\s 1:20 am here, so I'm of |
| 23:20.39 | Freeaqingme | f |
| 23:21.03 | lesouvage | You are in the same time zone as me. Goodnight. |
| 23:21.40 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 23:22.54 | mandd | hey, with a regular pbx, say from Bell, office can have sereval lines, so that a line is never busy, even if it is dialed by different people at the same time. |
| 23:23.07 | mandd | how does it work with VOIP & asterisk ? |
| 23:28.38 | Strom | mandd: depends on the entrance facilities |
| 23:28.49 | Strom | analog lines with rotating hunt; ISDN PRI; SIP; IAX... |
| 23:29.00 | Strom | they all offer that functionality, but in different ways |
| 23:31.23 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:37.54 | lesouvage | mandd: but if you go voip yu basically can forget about the whole concepts of lines because there are no lines, just a cable for the data traffic. The number of calls on the same time depends on the bandwidth available and the codec used. |
| 23:54.08 | mandd | oh |
| 23:54.22 | mandd | well, some calls form analog to my voip line |
| 23:54.23 | mandd | tice |
| 23:54.28 | mandd | at thesame time |
| 23:55.47 | mandd | oh |
| 23:55.52 | mandd | it just called twice |
| 23:55.53 | mandd | haha |
| 23:55.55 | mandd | thats nice |
| 23:56.03 | mandd | how do I figure out the limit |
| 23:56.09 | thansen|laptop | does anyone know what the format should be for Read() in 1.6? |
| 23:56.16 | thansen|laptop | it's acting really weird on me |
| 23:57.09 | thansen|laptop | using | to separate stuff doesn't work |
| 23:57.19 | thansen|laptop | using , to separte args doesn't work |
| 23:57.36 | thansen|laptop | I have to use some crazy combination of the two |