| 00:00.28 | ChannelZ | psykon: it's probably crashing because you've sort of created a loop |
| 00:00.58 | brad_mssw | pabelanger: core show help pri: pri intense debug span <no description available> |
| 00:01.03 | pabelanger | brad_mssw: *CLI> module show like dahdi |
| 00:01.08 | brad_mssw | pabelanger: no pri show spans, etc ... |
| 00:01.23 | brad_mssw | pabelanger: module show like dahdi |
| 00:01.31 | brad_mssw | pabelanger: possibly didn't downgrade everything? |
| 00:01.42 | pabelanger | maybe |
| 00:01.57 | pabelanger | $ sudo apt-get install asterisk-dahdi |
| 00:02.39 | ChannelZ | psykon: oh maybe not these are two different numbers |
| 00:05.05 | brad_mssw | pabelanger: ok, identified a few other packages too, voicemail, etc |
| 00:05.29 | brad_mssw | pabelanger: ugh, why doesn't ubuntu provide an easy way to back off to packages actually in the repos currently subscribed |
| 00:05.44 | ChannelZ | psykon: hmm well I did it a slightly different way due to my setup but it worked. |
| 00:05.55 | psykon | ChannelZ, what did you do? |
| 00:06.00 | pabelanger | brad_mssw: not sure I understand |
| 00:06.16 | ChannelZ | well I mean I had to replace my phone numbers and extensions |
| 00:06.45 | brad_mssw | pabelanger: eh, not your fault ... just saying when I removed the proposed repo, I should have been able to do an apt-get update && apt-get upgrade and it should have downgraded me ... but it doesn't |
| 00:06.50 | brad_mssw | pabelanger: it's working now, thanks |
| 00:07.01 | brad_mssw | pabelanger: pri show spans, etc |
| 00:07.06 | pabelanger | okay, cool |
| 00:07.42 | ChannelZ | psykon: does your thing crash when it hits a specific part of the dialplan (the Set?) if you watch the console on verbose when you send the AMI command? |
| 00:07.43 | psykon | ChannelZ, Were both legs on one asterisk box? In my case they were. |
| 00:07.53 | ChannelZ | psykon: yes |
| 00:08.18 | ChannelZ | psykon: well yes-ish, DAHDI dialing out to my cell phone and SIP to my desk phone |
| 00:08.42 | psykon | I am doing the originate from a php script so basically once I click the submit button asterisk crashes |
| 00:09.23 | brad_mssw | pabelanger: thanks for your help, all appears good (well, except my voicemail duration stuff .... apparently it wasn't something newly introduced in 1.8.6.0-rc1) |
| 00:09.40 | ChannelZ | yes but asterisk -rvvv so you can watch the console, see what it's outputing and where specifically it's dying. It might be telling you something |
| 00:10.48 | psykon | ChannelZ, I watch it again. I have verbosity set to 21. THat is usually where I keep it. |
| 00:11.07 | pabelanger | brad_mssw: Ya, sorry I cannot help with vmail |
| 00:12.18 | ChannelZ | I don't think it does much more above 5 IIRC but that's fine |
| 00:13.02 | psykon | ChannelZ, I jus tried it from the CLI and it took a dump |
| 00:13.42 | ChannelZ | what do you mean, with 'channel originate'? |
| 00:13.54 | psykon | ChannelZ, http://pastie.org/2414186 |
| 00:15.33 | ChannelZ | sorry I have to dash off and do something. BBL |
| 00:17.49 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 00:18.48 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
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| 00:28.03 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 00:30.08 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 01:13.54 | Atriks | good night :) |
| 01:20.44 | *** join/#asterisk ssfsdf (~treborsux@75-148-67-49-Jacksonville.hfc.comcastbusiness.net) |
| 01:20.48 | ssfsdf | hey |
| 01:21.42 | ssfsdf | i can get into the web interface of my 550s but how do i get into the web interface of a 501 |
| 01:21.42 | ssfsdf | i typed in address of 501 and nothing came up |
| 01:22.06 | ssfsdf | do i need a certain port? |
| 01:22.20 | ssfsdf | anyone here? |
| 01:23.28 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 01:23.29 | Maliuta | ssfsdf: 501 whats? and have you tried nmaping the IP to see what ports are open? |
| 01:23.45 | *** join/#asterisk dwmw2_gone__ (~ctrlproxy@twosheds.infradead.org) |
| 01:23.57 | p3nguin | Cisco SPA-501? |
| 01:24.13 | treborsux | good point |
| 01:24.21 | treborsux | just though someone would now |
| 01:24.24 | treborsux | know |
| 01:24.48 | p3nguin | We'd have to know wtf you're talking about first? |
| 01:24.57 | Maliuta | treborsux: well to start with, fully describing the product would help |
| 01:25.13 | p3nguin | s/?/./ |
| 01:25.39 | treborsux | SoundPoint IP, SoundStation IP and Polycom VVX BootROM 4.1.4 |
| 01:25.40 | treborsux | is that the one i want to use for 501 |
| 01:35.51 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-140-42.chyn.qwest.net) |
| 01:54.32 | treborsux | <PROTECTED> |
| 01:56.05 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 01:57.23 | *** join/#asterisk gravin (~gravin@96.70.50.60.brf01-home.tm.net.my) |
| 02:10.10 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 02:24.15 | treborsux | where the heck do i put secret in this 560 |
| 02:24.36 | treborsux | cant get it to register |
| 02:24.43 | treborsux | it sees it as endpoint |
| 02:25.18 | treborsux | is secret the same as the pasword for the web interface |
| 02:26.45 | p3nguin | secret means password, if that's what you are asking. |
| 02:30.03 | *** join/#asterisk kl4m (~kl4m@2001:5c0:1100:7f00:a00:27ff:fed3:3561) |
| 02:33.13 | kl4m | I have a question regarding VoiceMailMain: since the old syntax is VoiceMailMain([[s|p]mailbox][@context]), how can I escape a leading "p" in a mailbox name? |
| 02:34.09 | p3nguin | You can't just use the new syntax? |
| 02:34.24 | kl4m | It still "accepts" the old one |
| 02:34.37 | p3nguin | even with the new syntax being used? |
| 02:35.27 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 02:36.07 | p3nguin | VoiceMailMain(paul@default,p) <-- is this seen as aul's mailbox with two p options? |
| 02:36.16 | kl4m | OK, so putting empty options VoicemailMain(psomething@context,) fixed it. Thanks |
| 02:57.17 | *** join/#asterisk BuenGenio (~BuenGenio@059148208218.ctinets.com) |
| 02:57.19 | BuenGenio | hello |
| 02:58.51 | BuenGenio | we're currently facing a dillema - our partners use the Polycom HDX7000 video converencing solution, and want us to get it also (~$12000 US) |
| 02:58.55 | BuenGenio | we think it's a waste of money |
| 02:59.23 | BuenGenio | is there a way to use off-the-shelf HD webcam and use Asterisk to connect with them? |
| 02:59.53 | p3nguin | I guess if the cam does h.323. I think Asterisk can do video with h.323. |
| 03:00.53 | BuenGenio | how do set it up? |
| 03:07.15 | BuenGenio | can this work out of the box, or do I need to write some custom DialPlan rules to connect with the other side? |
| 03:07.43 | p3nguin | Asterisk does nothing out of the box -- you have to set it up to do what you want done. |
| 03:08.04 | p3nguin | Asterisk is a toolkit. |
| 03:08.41 | p3nguin | But... |
| 03:09.35 | p3nguin | There are some packaged platforms with Asterisk pre-installed and already built to do certain things. Check out AsteriskNOW to see if it does what you want to do without having to do it starting from the ground up. |
| 03:11.22 | BuenGenio | already have it set up... |
| 03:11.27 | BuenGenio | has a nice web interface |
| 03:11.35 | p3nguin | AsteriskNOW? |
| 03:11.50 | p3nguin | Did you use the Asterisk GUI or FreePBX? |
| 03:12.07 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 03:12.19 | BuenGenio | Could be FreePBX |
| 03:12.45 | BuenGenio | all very new to me... |
| 03:14.24 | p3nguin | I'll set the stage for you: if you've installed FreePBX, we can't help you with very much stuff in this channel; you'll need to go to #freepbx for all questions pertaining to that interface. |
| 03:14.46 | p3nguin | ~freepbx |
| 03:14.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:17.33 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:17.37 | p3nguin | Basic questions regarding asterisk and parts of asterisk which are not affected by freepbx may still get adequate attention here. |
| 03:27.49 | *** join/#asterisk romich (~kvirc@91.229.188.2) |
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| 03:52.29 | *** join/#asterisk Ast001 (~uros@cable-89-216-168-192.dynamic.sbb.rs) |
| 03:54.13 | Ast001 | Hello, I have agents in n queues with wrapup in each queue of 5 s. When my agent gets call in queue 1 and hangup he gets call from queue2 emidietely if thereis someone waiting there. I don't want that. I notices wrapuptime in ms in agents.conf . Should i enable that (set to 5000) to avoid this situation ? |
| 03:54.54 | Ast001 | Is this some sort of bug in asterisk or my misconfiguration ? |
| 03:56.40 | Ast001 | that wrapuptime in agents.conf is at the moment comented out. I wonder does it have some default value ? |
| 03:56.59 | Ast001 | I meant it is commented. |
| 04:05.54 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
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| 04:53.51 | Ast001 | I am sure someone knows is there default value for wrapuptime in agents.conf. Is it active or not if it is commented ? |
| 04:55.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 05:00.55 | rizwank | Hi there. I'm looking to set up some sort of NAT transversal for my already-installed SIP Servers; STUN works fine unless we're using 3g [symmetric NAT] -- can I use Asterisk's NAT transversal as a SIP/RTP proxy? Or can someone point me in the right direction; I've found so many different products claiming to help that haven't' so far. |
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| 06:44.03 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 06:45.00 | f2Knight | Question: Has anyone used StarPY or FastAGI in general? I am using netcat to view output of a fastagi call. However I am unable to hangup the channel from the AGI script. |
| 06:45.00 | *** join/#asterisk gg0 (~gg0@unaffiliated/gg0) |
| 06:45.16 | f2Knight | or other wise return to allowing the Dialplan to run. |
| 06:46.10 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 06:47.23 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 06:47.26 | jacc0 | hi all |
| 06:47.35 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 06:47.36 | kaldemar | f2Knight: exit the script to return to dialplan or use the agi hangup command. |
| 06:48.08 | jacc0 | I have this memory leak in 1.8.6-rc1 |
| 06:48.09 | f2Knight | kaldemar, I was thinking the HANGUP SIP/2000-00000000 would do just that. |
| 06:48.43 | f2Knight | kaldemar, of course I am testing over netcat, maybe this is the issue? but its a way for interactive use. I would hope thats not the issue |
| 06:48.51 | *** join/#asterisk BugKhaM (~BugKhaM@101.108.119.193) |
| 06:49.10 | BugKhaM | Anyone has a link explaining how the parameters maxexpiry,defaultexpiry and minexpiry work? |
| 06:49.15 | kaldemar | f2Knight: enable agi debug in asterisk and see what the CLI says. |
| 06:49.18 | BugKhaM | and is defaultexpirey also valid for asterisk 1.2.x? |
| 06:49.40 | kaldemar | BugKhaM: http://svn.digium.com/svn/asterisk/tags/1.8.5.0/configs/sip.conf.sample |
| 06:49.58 | jacc0 | how can I investegate the source of a memory leak in asterisk? |
| 06:50.05 | kaldemar | BugKhaM: and yes, 1.2 has the defaultexpiry option. |
| 06:51.03 | jacc0 | someone told me I should do a "core show memory" |
| 06:51.07 | jacc0 | but it doesn't work |
| 06:51.35 | BugKhaM | kaldemar: with the spelling "defaultexpirey" ? |
| 06:51.41 | f2Knight | kaldemar, I get a 200 result=1 |
| 06:51.43 | f2Knight | *CLI> agi set debug on |
| 06:51.43 | f2Knight | AGI Debugging Enabled |
| 06:51.43 | f2Knight | *CLI> <SIP/2000-00000001>AGI Rx << HANGUP |
| 06:51.43 | f2Knight | <SIP/2000-00000001>AGI Tx >> 200 result=1 |
| 06:52.01 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 06:52.22 | kaldemar | BugKhaM: 1.2.X does not have the minexpiry option though. the default option may be spelled defaultexpiry or defaultexpirey in 1.2 |
| 06:52.23 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 06:52.24 | jacc0 | I've it compiled with\ MALLOC_DEBUG flag |
| 06:53.26 | BugKhaM | kaldemar: i see, thanks |
| 06:54.27 | f2Knight | kaldemar, other things work find like I can use GET VARIABLE <var_name> |
| 06:54.27 | kaldemar | BugKhaM: actually all three of them may be spelled expiry or expirey, i checked 1.8.5 and 1.2. |
| 06:55.19 | kaldemar | f2Knight: does the channel not get hung up on hangup? |
| 06:55.49 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 06:56.15 | f2Knight | kaldemar, no it does not. It still hangs open. But after a while (or after some commands) it says 511 something about dead channel but the softphone is still 'connected' and the CLI shows it is still active (core show channels) |
| 06:57.26 | BugKhaM | kaldemar: yeah, but what will aster do if I set "minexpiry=60" and the client registers every 30 secs |
| 06:58.10 | BugKhaM | kaldemar: reject the registration? |
| 07:02.21 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 07:02.49 | f2knight | kaldemar, sorry lost connection, |
| 07:05.09 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:05.10 | Polysics | hello |
| 07:05.19 | Polysics | is it still recommended to run 1.8 as not root? |
| 07:06.23 | jacc0 | yes |
| 07:07.08 | Polysics | is there a list of things to do somewhere, pelase? i have one for 1.6 |
| 07:08.02 | jacc0 | I always use .deb package to get users/groups and all right |
| 07:08.13 | Polysics | if it is a development only local machine, can i just leave it as root? |
| 07:08.20 | Polysics | it's not even started automatically |
| 07:08.34 | jacc0 | sure you can |
| 07:08.36 | Polysics | what do you mean? install deb first then source? |
| 07:08.42 | jacc0 | yes |
| 07:08.50 | ollii | what is the best way to create a .deb package from source? i tried checkinstall ... its working good, but is there a better tool? |
| 07:09.00 | ollii | asterisk 1.8 source |
| 07:09.11 | Polysics | jacc0, so you just apt-get install the package, then compile normally from latest? |
| 07:09.22 | jacc0 | that will compile 1.6 |
| 07:09.47 | kaldemar | BugKhaM: hmm. looks like registration is accepted but publishes and subscribes are responded to with 423 Interval too small. |
| 07:10.29 | jacc0 | do this first :http://pastebin.com/y5Zdrv0w |
| 07:10.43 | jacc0 | then apt-get install asterisk-1.8 |
| 07:10.48 | jacc0 | or just asterisk |
| 07:10.49 | ChannelZ | humm. I think it's res_jabber not parsing the buddy list properly that is causing my angst |
| 07:11.11 | Polysics | oh, there is an official repository? i never noticed :-) |
| 07:11.16 | jacc0 | ;) |
| 07:11.16 | Polysics | thanks a lot |
| 07:11.19 | jacc0 | yw |
| 07:11.34 | Polysics | will try that, i just installed anyway |
| 07:11.42 | Polysics | does it have mysql_config module? |
| 07:11.50 | jacc0 | not sure |
| 07:12.02 | Polysics | will poke around to find out, thanks |
| 07:12.50 | jacc0 | anyone here that can help me with the memory leak? asterisk 1.8.6-rc1 is taking up 940mb in idle state |
| 07:13.03 | jacc0 | after running for 2 days |
| 07:13.13 | jacc0 | it started out using only 134mb |
| 07:13.24 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 07:13.34 | *** join/#asterisk jkroon (~jkroon@dsl-241-246-168.telkomadsl.co.za) |
| 07:16.31 | BugKhaM | kaldemar: looks like asterisk replaces Re-Register Time with the value set in minexpiry |
| 07:17.36 | kaldemar | Polysics: the asterisk package has res_config_mysql. |
| 07:17.46 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 07:17.47 | Polysics | is there a command to have init.d scripts installed when installing fro msource? |
| 07:18.10 | Polysics | i am almost done on this machine with source, will do the next with package and play spot the difference |
| 07:18.26 | kaldemar | Polysics: make config |
| 07:19.49 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 07:27.29 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 07:36.31 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:36.39 | jacc0 | <PROTECTED> |
| 07:37.20 | jacc0 | lock.h is the source of the mem leak |
| 07:38.24 | jacc0 | 110116980 bytes in 71135 allocations in file '/usr/src/asterisk-1.8.6.0-rc1/include/asterisk/lock.h' |
| 07:38.39 | jacc0 | why is it pointing to my source folder? |
| 07:38.56 | jacc0 | is it becasue of the debuging flags ? |
| 07:39.07 | jacc0 | *because |
| 07:44.35 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:44.45 | Polysics | sorry, i lost the link to the debian packages for 1.8 |
| 07:46.10 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 07:46.13 | jacc0 | new issue : https://issues.asterisk.org/jira/browse/ASTERISK-18323 |
| 07:46.41 | jacc0 | http://pastebin.com/y5Zdrv0w |
| 07:47.04 | jacc0 | now adding debug info |
| 07:47.40 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-laojcwdhgubrabcz) |
| 07:50.50 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 07:51.27 | kaldemar | jacc0: do you have chan_iax2 loaded? |
| 07:52.01 | jacc0 | yes |
| 07:52.16 | kaldemar | see what happens to the allocations of lock.h when you unload it. |
| 07:52.41 | jacc0 | okay |
| 07:52.59 | jacc0 | let me first take some debug info |
| 07:53.22 | *** join/#asterisk xnfinite (~xnfinite@62.82.191.170.static.user.ono.com) |
| 07:53.49 | kaldemar | mine drops from "102979152 bytes in 66524 allocations" to "1456668 bytes in 941 allocations" on a freshly started asterisk. |
| 07:54.23 | jacc0 | hehehe |
| 08:00.09 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 08:01.01 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 08:05.07 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 08:06.37 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 08:08.45 | jacc0 | 110273328 bytes in 71236 allocations in file '/usr/src/asterisk-1.8.6.0-rc1/include/asterisk/lock.h' |
| 08:10.27 | jacc0 | 8770968 bytes in 5666 allocations in file '/usr/src/asterisk-1.8.6.0-rc1/include/asterisk/lock.h' |
| 08:11.01 | jacc0 | the last one is after unload of chan_iax2.so |
| 08:13.42 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:14.12 | Polysics | what do you use as realm on a local machine? |
| 08:14.43 | jacc0 | @kaldemar:any way to fix this leak? |
| 08:14.55 | Polysics | in sip.conf |
| 08:14.59 | jacc0 | I use "asterisk" as realm |
| 08:17.51 | Sakuranbo | hey guys, a simple idiotic question to ask regarding the vm |
| 08:17.58 | Polysics | i wonder why i am not seeing any SIP connection in the console |
| 08:18.14 | jacc0 | Sakuranbo: bring it on!! |
| 08:18.16 | Polysics | or anything at all, to be honest |
| 08:18.36 | Polysics | i configured Blink to connect to 10001@127.0.0.1 |
| 08:18.41 | Sakuranbo | my end users are trying to access their vm but the after the ivr the predefined pwd don't work |
| 08:18.47 | Sakuranbo | I got " app_voicemail.c: Unable to read password" |
| 08:18.52 | Polysics | shouldn't i at least get some sort of connection failed message in the asterisk console? |
| 08:19.04 | Polysics | asterisk 1.8 installed from deb |
| 08:19.26 | jacc0 | not sure |
| 08:19.34 | ChannelZ | Sakuranbo: you might have a dtmf problem - do your IVRs require other digit entries that work? |
| 08:19.45 | Sakuranbo | I ve checked up the Voip.info |
| 08:19.48 | jacc0 | see what is going on using:tshark -r "sip" |
| 08:19.52 | Sakuranbo | not much clues on it |
| 08:20.08 | Sakuranbo | no |
| 08:20.19 | Sakuranbo | straight into the vm box |
| 08:20.24 | Polysics | my sip.conf https://gist.github.com/1164627 |
| 08:20.53 | jacc0 | @polysics : apt-get install tshark |
| 08:21.01 | Polysics | i am installing it :-D |
| 08:21.05 | Sakuranbo | should I enable my GXP2020 endpoint to use RFC2833 as well? |
| 08:21.06 | jacc0 | or : nano /etc/asterisk/logger.conf |
| 08:21.12 | jacc0 | and enable more info |
| 08:21.42 | Polysics | i do not have the logger.conf, is it bad? |
| 08:21.50 | Polysics | i installed my own set of configuration |
| 08:21.56 | jacc0 | tshark -R sip |
| 08:21.59 | ChannelZ | Sakuranbo: yes you should generally be using rfc2833 in SIP almost universally |
| 08:21.59 | Polysics | which works on another server |
| 08:22.01 | jacc0 | with capital R |
| 08:22.05 | kaldemar | jacc0: is it really a leak? |
| 08:22.27 | jacc0 | it is growing over time |
| 08:22.29 | Polysics | there are no interfaces on which capture can be done? now what? |
| 08:22.37 | Sakuranbo | the current default is using "in-ausio" |
| 08:22.42 | Sakuranbo | audio |
| 08:23.18 | ChannelZ | Unless your * is setup for inband too then the DTMF is probably being thrown away |
| 08:23.35 | Polysics | 8.165338 81.23.228.150 -> 192.168.11.36 SIP Request: NOTIFY sip:95.228.142.210:54488 |
| 08:23.45 | Polysics | where do those IPs come from? lol |
| 08:23.50 | ChannelZ | But you pretty much have to use rfc if you're using g.729 or gsm or other compressed codecs |
| 08:23.56 | jacc0 | @polysics: make samples |
| 08:24.02 | jacc0 | to create logger.conf |
| 08:24.12 | Sakuranbo | as the internal network does not have QoS issue |
| 08:24.27 | Sakuranbo | all I use is PCMU |
| 08:24.43 | kaldemar | Polysics: asterisk and blink are on the same machine? are they trying to use the same port? |
| 08:25.09 | ChannelZ | That's fine but rfc is still better as it's just transported around as data rather than asterisk having to listen for inband DTMF and decode it |
| 08:25.39 | Sakuranbo | ok, then I need to do it tonight remotely as the market is still open |
| 08:26.06 | Polysics | kaldemar, could it be that? |
| 08:26.16 | Polysics | yes, it is the same machine |
| 08:26.24 | Polysics | i need to test some adhearsion stuff, though to use that |
| 08:26.28 | Sakuranbo | Thanks ChannelZ, |
| 08:26.49 | Sakuranbo | but I am not sure it could solve the issue as I check the voicemail.conf |
| 08:26.54 | ChannelZ | you can make a couple of test extensions just to verify that you're not reading dtmf properly. Like make an exten 5555 that does a Background(silence/10) and then a exten 6666 that does a Playback(tt-monkeys) or something. Then dial 5555, wait a sec, and dial 6666. |
| 08:26.55 | Sakuranbo | settings are clean |
| 08:26.59 | ChannelZ | No monkeys, no dtmf working |
| 08:27.45 | ChannelZ | (just to rule out something else odd happening specifically with your voicemail config) |
| 08:27.54 | Sakuranbo | I dont have a gummy hand, if need to do it, I have to go a few blocks away to the client site |
| 08:28.08 | Sakuranbo | sure! |
| 08:28.35 | Polysics | good, sorted out |
| 08:28.37 | ChannelZ | no ssh!? how do you live!? :) |
| 08:28.38 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-200-219.w83-203.abo.wanadoo.fr) |
| 08:28.43 | Polysics | kaldemar, can't thank you enough |
| 08:28.44 | *** join/#asterisk gravin (~gravin@113.210.252.17) |
| 08:28.54 | Sakuranbo | as it's already in production |
| 08:29.03 | Polysics | enabling debug in the console allowed me to see what was wrong |
| 08:29.11 | Sakuranbo | I will be hung to death if anything goes wrong |
| 08:29.37 | Polysics | can you get hung not to death? |
| 08:29.43 | Polysics | i guess it would be even worse |
| 08:30.05 | Sakuranbo | is there way to simulate a dial on a console without installing anything? |
| 08:30.26 | Sakuranbo | GXP default use *97 to access the VM |
| 08:31.12 | kaldemar | Sakuranbo: if you have chan_oss or chan_alsa, see "core show help console dial". |
| 08:31.30 | Sakuranbo | let me try now |
| 08:31.48 | kaldemar | or chan_console, forgot about that one. |
| 08:32.11 | Sakuranbo | try ""core show help console dial"" right? |
| 08:32.43 | ChannelZ | you're not physically at the box right? |
| 08:32.47 | Sakuranbo | my asterisk is pretty old |
| 08:32.55 | Sakuranbo | no such command |
| 08:33.07 | Sakuranbo | yup, I am VPN-ing to the site |
| 08:33.16 | ChannelZ | none of that will help you then |
| 08:33.28 | Sakuranbo | Asterisk 1.4.24 |
| 08:33.59 | ChannelZ | you need a device (softphone, phone) connected to the system to test |
| 08:34.14 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 08:34.17 | Sakuranbo | Oh I have X-lite! |
| 08:34.47 | Sakuranbo | try it out now |
| 08:35.35 | ChannelZ | do "sip show settings" on the console - down near the bottom it should say something like 'default settings' and then show you "DTMF: something" |
| 08:35.50 | *** join/#asterisk sourcode (~code@ppp-58-8-48-121.revip2.asianet.co.th) |
| 08:36.02 | *** join/#asterisk gravin (~gravin@113.210.252.40) |
| 08:36.54 | kaldemar | Sakuranbo: Usage: console dial [extension[@context]] |
| 08:37.10 | ChannelZ | noooo don't |
| 08:37.17 | Sakuranbo | <PROTECTED> |
| 08:37.18 | Sakuranbo | <PROTECTED> |
| 08:37.18 | Sakuranbo | <PROTECTED> |
| 08:37.54 | ChannelZ | Ok so if all your phones are setup for inband ("in-audio" I assume is what they are calling it) then it's no wonder it doesn't work :) |
| 08:38.07 | ChannelZ | Your x-lite is probably setup for rfc by default and will work |
| 08:39.22 | Sakuranbo | yes that ext works !! |
| 08:39.52 | ChannelZ | Do the phones in the office dial out to the world via SIP or PRI or something? |
| 08:40.05 | Sakuranbo | IDAP for external |
| 08:40.05 | merlin8282 | Hi ! Any idea why parking does not work ? I have "parkcall => *7" in features.conf, and "include => parkedcalls" in my dialplan, in the context [intern], which is included by [default]. See here for a test call: http://pastebin.archlinux.fr/433661 |
| 08:40.13 | Sakuranbo | T1 |
| 08:40.44 | Sakuranbo | (^33^) |
| 08:40.52 | Sakuranbo | thanks guys! |
| 08:41.13 | ChannelZ | ok. Well you should be pretty safe changing all of your phones' configs to rfc. Do one, test it with your VM, and then make a call to some other place and make sure DTMF works there too (call your cable company or something, they have lots of phone mazes to test on) |
| 08:41.28 | f2knight | Question: Does anyone know of a FastAGI python script that does not require twisted? |
| 08:45.43 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 08:45.55 | ChannelZ | merlin8282: looks like it parked to me? "== Parked SIP/10-00000004 on 81 (lot default). Will timeout back to extension [default] 24, 4 in 120 seconds" |
| 08:47.39 | Sakuranbo | One more thing is configuring the Digium TDM410 card for the fax |
| 08:47.51 | Sakuranbo | I changed the dahdi-channels.conf |
| 08:48.01 | Sakuranbo | and do a dahdi_scan |
| 08:48.32 | Sakuranbo | the 3 FXS ports are not all open as I configured they should be |
| 08:48.42 | Sakuranbo | any clues guys? |
| 08:48.48 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:49.18 | ChannelZ | what do you mean not all open |
| 08:49.55 | Sakuranbo | wait I give you the dump |
| 08:49.56 | ChannelZ | as in they show 'port=1,none'? |
| 08:50.04 | *** join/#asterisk gravin (~gravin@113.210.253.205) |
| 08:51.12 | ChannelZ | ~pb |
| 08:51.12 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 08:53.05 | ChannelZ | merlin8282: oh now that I read the rest of your paste, the pickup isn't working. Does your dialplan have a pattern or exten that is picking up the 81 specifically? It looks like it's doing a Goto(i,1) on purpose for some reason |
| 08:53.35 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:53.44 | Sakuranbo | ChannelZ: [1] |
| 08:53.45 | Sakuranbo | active=yes |
| 08:53.45 | Sakuranbo | alarms=RED |
| 08:53.45 | Sakuranbo | description=Wildcard TE122 Card 0 |
| 08:53.45 | Sakuranbo | name=WCT1/0 |
| 08:53.45 | Sakuranbo | manufacturer=Digium |
| 08:53.46 | Sakuranbo | devicetype=Wildcard TE122 |
| 08:53.46 | Sakuranbo | location=PCI Bus 01 Slot 13 |
| 08:53.47 | Sakuranbo | basechan=1 |
| 08:53.47 | Sakuranbo | totchans=24 |
| 08:53.50 | ChannelZ | damnit |
| 08:53.51 | ChannelZ | ~pb |
| 08:53.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 08:54.08 | Sakuranbo | type=analog |
| 08:54.08 | Sakuranbo | port=25,FXS |
| 08:54.09 | Sakuranbo | port=26,none |
| 08:54.09 | Sakuranbo | port=27,none |
| 08:54.09 | Sakuranbo | port=28,FXO |
| 08:54.09 | Sakuranbo | [3] |
| 08:54.15 | ChannelZ | STOP PASTING THAT HERE! |
| 08:54.22 | ChannelZ | please read above |
| 08:54.37 | f2knight | kaldemar, I figured out what it was... The AGI command HANGUP does not actually hangup the channel but rather "MARKS" it to be hungup. |
| 08:54.42 | Sakuranbo | Sorry m(__)m |
| 08:54.55 | Sakuranbo | my 1st time in the channel |
| 08:55.15 | ChannelZ | also it doesn't seem great that your TE122 has red alarms... |
| 08:55.35 | Sakuranbo | as there is no T1 plugged yet |
| 08:55.49 | Sakuranbo | a lab without PRI |
| 08:55.50 | ChannelZ | Oh. I thought you said this system was "live" |
| 08:56.16 | Sakuranbo | this is another one I need to swap the live one (highly likely) |
| 08:56.29 | ChannelZ | oh. |
| 08:56.47 | Sakuranbo | as the live one is a time bomb |
| 08:56.48 | ChannelZ | Well anyway it looks like your FXS is channel 25. What is the problem? |
| 08:57.05 | *** join/#asterisk maxagaz (~maxagaz@220.231.37.106) |
| 08:57.15 | Sakuranbo | for 26,27 that it's not open for usage |
| 08:58.00 | ChannelZ | what modules do you have on that TDM card? |
| 08:58.15 | Sakuranbo | yes, there is no problem for the FXS but how to open 26 and 27 which I will connect 2 fax to modem to test the fax function |
| 08:58.47 | Sakuranbo | 1FXO 3FXS |
| 08:58.58 | Sakuranbo | 25=FXS |
| 08:59.16 | ChannelZ | IE you actually do have 3 green cards and one red? |
| 08:59.50 | Sakuranbo | 2 green, 25 and 28 are green |
| 09:00.10 | Sakuranbo | 26 and 27 are out |
| 09:00.15 | ChannelZ | well that doesn't make sense |
| 09:00.45 | ChannelZ | but if you have only 2 modules plugged into the thing, I don't know how you expect there to be 3 FXS channels!? |
| 09:00.58 | Sakuranbo | as port=26,none and port=27,none from dahdi scan |
| 09:04.19 | Sakuranbo | I didnt plug anything on port 28 but it's green |
| 09:04.20 | ChannelZ | You have a problem if you have 2 green cards plugged in but have one FXS and one FXO showing |
| 09:04.20 | ChannelZ | did you configure /etc/dahdi/system.conf correctly? |
| 09:04.20 | Sakuranbo | yes |
| 09:04.21 | ChannelZ | Really? Cause what you're saying and what you've shown don't agree |
| 09:04.21 | Sakuranbo | do u mean the framing, clocking, line code right? |
| 09:04.21 | ChannelZ | no that has to do with your T1 |
| 09:04.21 | ChannelZ | You should have something like "fxoks=25" for an FXS port |
| 09:05.35 | ChannelZ | in any case dahdi_scan should be showing you really what the hardware sees |
| 09:06.15 | Sakuranbo | that's why I am puzzling, let me check first. Thanks Channelz |
| 09:06.25 | ChannelZ | The reason channels 26 and 27 are 'none' is because you just said all the modules aren't even plugged into the card. |
| 09:07.07 | ChannelZ | also make sure you've got the molex power connector connected or it won't work even if you sort the rest out |
| 09:10.09 | ChannelZ | way past my bedtime. good luck |
| 09:13.10 | Sakuranbo | sure, goodnite |
| 09:13.59 | Sakuranbo | the fax-to-modems are plugged but the LEDs are still out |
| 09:14.09 | Sakuranbo | to 26,27 port |
| 09:19.57 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:25.58 | Polysics | caller calls, i start MoH, then originate a call to the destination, if destination accepts i need to bridge them |
| 09:26.10 | jacc0 | :) |
| 09:26.15 | Polysics | is there an AGI command to bridge two cchannels? AGI, not AMI |
| 09:26.16 | jacc0 | that is about what I made |
| 09:26.26 | jacc0 | not an agi |
| 09:26.33 | Polysics | or i can use an UserEvent |
| 09:26.34 | jacc0 | just the dialplan bridge() command |
| 09:26.50 | Polysics | yes, sorry, i mis-expressed myself |
| 09:26.58 | Polysics | a dialplan application |
| 09:27.21 | jacc0 | bridge() is what you are looking for |
| 09:28.30 | jacc0 | also; the app_originate patch I've made might come in handy |
| 09:28.42 | Polysics | where can i see that, please? |
| 09:28.52 | jacc0 | it enables you to set the timeout of the dialplan command originate() |
| 09:29.52 | jacc0 | https://reviewboard.asterisk.org/r/1310/ |
| 09:34.40 | Polysics | which variable holds the current channel full name? eg. SIP/10234-00000000f |
| 09:34.50 | jacc0 | ${CHANNEL} |
| 09:50.28 | irroot | having fun with 2 digium b410p cards |
| 09:50.50 | irroot | 4 in nt 4 in te seems to work |
| 09:51.23 | jacc0 | :0 |
| 09:51.51 | irroot | make all cross connected make one call it loops till all lights go green |
| 09:52.22 | WIMPy | waits for the non working part. |
| 09:52.33 | irroot | still dont like going inline with telco and legacy pbx with BRI |
| 09:57.52 | jacc0 | lol |
| 09:59.57 | *** join/#asterisk nunne (~nunne@217-211-182-66-o871.telia.com) |
| 10:01.00 | nunne | i have configured isdn as te_ptp in misdn.conf.. but when doing "misdn show port 1" for example it comes up as TE PMP... and that L1 is UP but L2 is DOWN.. what gives? :( I have tried both a regular and crossover cable for it. |
| 10:01.28 | nunne | and yeah, i'm trying to connect it to a ISDN provider. not to another PBX etc. |
| 10:02.03 | WIMPy | Straight cable (e.g. network patch cable), but the L1 status schould already tell you. |
| 10:02.45 | nunne | yeah. i have tried that. but still L2 is down and it shows as PMP in asterisk. even though it's PTP in misdn.conf |
| 10:02.57 | WIMPy | Don;t you already specify ptp/ptmp on module load with the old misdn? I can't rally remember. |
| 10:02.59 | nunne | (not using latest misdn.. and asterisk 1.4.. embedded platform) |
| 10:03.23 | nunne | WIMPy, yes. you specify when you load the module.. which i do. but why does it come up as |
| 10:03.27 | nunne | PMP |
| 10:03.51 | nunne | but shouldnt it come up as PP? |
| 10:04.46 | WIMPy | What does misdn_info say? |
| 10:06.01 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 10:07.38 | nunne | the thing is this.. the ports only react with a straight cable when i have it in PTMP (misdn-init.conf) ... saying L1 activated, deactivated etc |
| 10:07.42 | nunne | misdnportinfo shows |
| 10:07.51 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:08.35 | nunne | http://pastebin.com/WA4DcrKY |
| 10:08.43 | nunne | te_ptp=1,2 |
| 10:08.47 | nunne | te_ptmp=3,4 |
| 10:09.17 | WIMPy | That means you only get L1 up on ports 3 and 4? |
| 10:10.17 | WIMPy | Well, it moans about a bad layermask. |
| 10:11.21 | nunne | layermask is 0xf, which should be the correct one :/ but i shouldnt really use PTMP either way? why cant i get L1 up on my PTP?? or does this mean the NT is in PTMP-mode?? |
| 10:12.31 | WIMPy | The NT is nothing. And you are the only one her who could know how your line(s) are configured. |
| 10:13.48 | WIMPy | wonders if TEI management falls into L1. |
| 10:14.50 | nunne | well, i can see and touch the NT-modules. it's 2 little boxes on the wall :) and usually just plug them in and set to go.. but since it's an old installation i dont know if they used to have PTP or PTMP.. but seing as they had an old ISDN PBX her i would think it would be PTP |
| 10:15.19 | WIMPy | Not neccessarily. |
| 10:16.59 | Polysics | ok, i managed to get a caller in MoH, originate to the destination, if he accepts, they are bridged |
| 10:17.31 | Polysics | how do i know if he accepts from the original call control in AGI? if he does not accept, i need to go over to the next in the list |
| 10:18.55 | *** join/#asterisk Sakuranbo (~Sakuranbo@59.152.236.158) |
| 10:19.15 | nunne | WIMPy, but i can forget about using a crossover cable? correct? just so i dont dwell into that.. i should only use that when trying to emulate a NT-device i would guess? |
| 10:19.35 | WIMPy | correct |
| 10:19.40 | Polysics | can i return a response over AMI from an originate call by setting something in the dialed extension? |
| 10:20.40 | WIMPy | nunne: I found a reference to layermask also being affected by ptp/ptmp, but I can't find values for that. |
| 10:21.38 | WIMPy | Polysics: Depends when/how you want to know. You can access variables, But you could also use a NoOp and parse it's data. |
| 10:22.41 | Polysics | WIMPy, i am doing the following: caller enters AGI, an UserEvent is generated, that is received and starts dialing using Originate |
| 10:22.44 | nunne | WIMPy, it's from switchfin firmware. should be okay. atleast it has worked for me in NT-mode.. but i can get the L1 up.. what could i possible do to tell what could be wrong? or maybe it's even wrong with the lines? |
| 10:22.46 | *** join/#asterisk dwmw2_gone__ (~ctrlproxy@twosheds.infradead.org) |
| 10:22.49 | Polysics | every receiver can press 1 to accept or not |
| 10:23.11 | Polysics | i basically need to know if the call went up in the Originate or not |
| 10:23.36 | WIMPy | nunne: Hmm. Are you sure it doesn't work? Have you tried? |
| 10:23.52 | *** join/#asterisk Ulrar (~quassel@2a01:e0b:1:136:62eb:69ff:fe8f:18a0) |
| 10:24.02 | nunne | WIMPy, you mean that the lines work even though L2 is down? trying by calling etc? |
| 10:24.19 | Ulrar | Hi. What happen to an AGI script when the channel is hang up ? |
| 10:24.41 | WIMPy | Polysics: If 1 was pressed you could just goto some special extension that you wait for on AMI. |
| 10:25.00 | WIMPy | nunne: Yes. |
| 10:25.36 | Polysics | i am finding references to an OriginateResponse on google but i can't tell where it is defined or what it is |
| 10:25.43 | nunne | WIMPy, i get this when trying to dial |
| 10:25.44 | nunne | http://pastebin.com/wyTcVpF7 |
| 10:26.53 | WIMPy | nunne: Better try the other way. |
| 10:26.58 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 10:27.53 | WIMPy | There's a dialplan application to activate the line. Cant' remember the name checkl1l2 or something. |
| 10:28.37 | nunne | WIMPy, i dont see anything in the console and get a congestion-tone on my cell |
| 10:31.14 | nunne | WIMPy, no success with that either :( |
| 10:31.34 | WIMPy | If you dial out on a group, I think it will never try a line that isn't up/up. |
| 10:33.24 | WIMPy | Might be a good idea to look in to misdn_log. Or did you have to do that via chan_misdn? |
| 10:35.45 | nunne | what do you mean? my loggin capabilites are a bit limited on this embedded system |
| 10:36.14 | WIMPy | Try to see if there is any communication going one. |
| 10:36.29 | WIMPy | -e |
| 10:38.48 | nunne | how do i see that the best way? turning misdn debug on just shows P[ 0] Got empty Msg.. |
| 10:39.44 | nunne | gotta break for lunch! brb |
| 10:40.57 | Ulrar | No one can tell me what happen to an AGI script when the channel is hunp up ? Is it killed ? |
| 10:45.33 | dwmw2_gone__ | hm, is it possible to register a 'filler frame' that will always be sent rather than letting the line go idle? |
| 10:51.43 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 10:52.14 | merlin8282 | ChannelZ: I have this http://pastebin.archlinux.fr/433663 , but "include => parkedcalls" should provide the ability to pick up 81, no ? |
| 10:54.24 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 10:55.16 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 10:56.27 | mandla | Hello, can anyone hook me up with the simpliest pattern to route ANY incoming call to my pbx |
| 10:57.16 | mandla | A pattern that can allow ANY call. |
| 10:58.18 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 11:00.14 | jacc0 | exten = _X.,1,............................ |
| 11:00.26 | singler | exten => |
| 11:00.45 | jacc0 | allows al nummeric |
| 11:00.49 | kaldemar | X. only matches something starting with a digit and with length of at least two. |
| 11:00.50 | singler | and in this case exten length should be at least 2 |
| 11:01.01 | jacc0 | true |
| 11:01.12 | jacc0 | s,1,.......... |
| 11:01.24 | jacc0 | maybe |
| 11:01.34 | singler | "." could be used, but then special extens should be defined |
| 11:01.39 | kaldemar | s matches when there is no extension. it is not a wildcard. |
| 11:01.45 | mandla | cant it be _X! |
| 11:01.46 | singler | because it will match "i" and "t" |
| 11:01.56 | singler | mandla: _X will be one digit |
| 11:02.27 | mandla | oh ok so what should i use?? |
| 11:02.37 | jacc0 | s,1,.......... |
| 11:02.45 | jacc0 | if you have no other extensions |
| 11:02.47 | singler | exten => _.,1,<..> |
| 11:02.59 | kaldemar | merlin8282: the XX extension wins any matching included extension. |
| 11:03.22 | mandla | ok thanx guys. |
| 11:03.33 | kaldemar | s will only match s. it does not match 123 for example. |
| 11:05.38 | mandla | exten => s,1,Goto(default,6000,1) |
| 11:05.43 | mandla | is that fine?? |
| 11:05.55 | merlin8282 | kaldemar: ok, so I should write into my dialplan something like "exten => _8Z,1,ParkedCall(${EXTEN})" for the parked calls to be able to be picked up ? |
| 11:06.02 | singler | mandla: read http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns and http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 11:07.12 | merlin8282 | kaldemar: ok, with this it works. |
| 11:07.15 | mandla | singler, is exten => s,1,Goto(default,6000,1) fine?? |
| 11:07.15 | merlin8282 | thanks |
| 11:08.41 | singler | it is not fine if you want to match any extension, you should use "exten => _." but then special extensions will be matched too, but not sure if it will negative effect |
| 11:12.18 | merlin8282 | kaldemar: I get the logic behind it : http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting . So if I understand correctly, it's because the parking lots are not in the same context as "exten => _X.", and so the extensions for the parkinglots must be explicitly defined ? |
| 11:14.27 | merlin8282 | mmm, it seems that then I don't need to include the parkedcalls at all... |
| 11:15.00 | kaldemar | merlin8282: the order is extensions first, then includes. either you define an extension for the extensions or change the context structure of your dialplan. i'd do the latter. |
| 11:16.41 | merlin8282 | kaldemar: ok. Would it be sufficient if I move only the _X extension to a context that I include ? In this context there would then be only this _X extension. |
| 11:17.22 | merlin8282 | (this context would of course be included in the "default" context) |
| 11:18.35 | kaldemar | merlin8282: yes. |
| 11:19.08 | kaldemar | merlin8282: remember that included contexts get matched in the order they appear in extensions.conf. |
| 11:19.23 | merlin8282 | ah ! ok. |
| 11:19.43 | merlin8282 | So I put it after the context [intern] |
| 11:20.38 | merlin8282 | Good ! It works now as expected ! Thank you for your help kaldemar :) |
| 11:21.32 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 11:34.31 | jacc0 | is there a simple cli command that shows if asterisk is in idle state? |
| 11:35.09 | jacc0 | in the same way that "core stop gracefully" checks if its idle? |
| 11:35.10 | WIMPy | Define idle |
| 11:35.46 | WIMPy | 'core show calls' should be about it. |
| 11:36.59 | *** join/#asterisk delki8 (~delki8@189.5.136.31) |
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| 12:00.11 | jacc0 | ty |
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| 12:26.08 | azv4 | Panasonic Digital Hybrid, are the cards hotswapable? |
| 12:35.32 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 12:35.43 | AdvoWork | is there a way I can check if a trunk is working or how many channels are working etc? |
| 12:39.22 | kaldemar | AdvoWork: make calls. what kind of a trunk you talking about? |
| 12:46.02 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-orcnhxeodbezzaai) |
| 12:47.06 | AdvoWork | kaldemar, an iax2 trunk, it should use the trunk unless congested or full, then use another service, but its using this other service more than it should, by a lot. now im trying to work out if theres a problem on the trunk, or the channels or? |
| 12:50.40 | *** join/#asterisk gravin (~gravin@255.189.159.110.tm-hsbb.tm.net.my) |
| 12:59.24 | kaldemar | AdvoWork: you'd need to define full yourself, there are really no number of channels in VoIP connections. you could count the numbers of calls with GROUP and GROUP_COUNT for example. |
| 13:00.03 | Katty | drags in |
| 13:00.10 | Katty | plops |
| 13:01.47 | beek | hands Katty a cup of coffee and a donut |
| 13:02.32 | Katty | hugs beek |
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| 14:46.57 | Katty | oakwneofqiwjeflkJALKJFOEINFLKSJldfoan |
| 14:48.45 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 14:50.35 | ruben23 | hi guys troubleshooting for couple of days already still i can figure out this erro code flooding my asterisl console and also affecting my recordings---------> http://pastebin.com/4VmeM0g3 <----------------------------any help guys..thanks you in advance |
| 14:51.43 | *** join/#asterisk albertoandrade (~albertoan@200.195.150.202) |
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| 15:08.48 | TobyRulez | ruben, looks like it might be related to large file size somewhere |
| 15:09.11 | TobyRulez | http://forums.digium.com/viewtopic.php?p=125514&sid=3e94d0b43dcbdf02dbcba5d85c598f2b |
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| 15:11.54 | ChannelZ | ruben23: asked and unanswered before... what is the wav file(s) you're trying to play? are they valid? |
| 15:13.01 | ruben23 | ChannelZ: i cant track what file is generating this its hard to trace, where should i start..? |
| 15:13.53 | *** join/#asterisk navaismo (~navaismo@187.170.13.36) |
| 15:13.57 | ChannelZ | turn on verbose, *something* is playing it |
| 15:14.20 | TobyRulez | have you checked the full log right before the errors? anything suspicious? |
| 15:14.40 | ChannelZ | Do you have your own system sounds or MOH you've made as wav? |
| 15:15.20 | ruben23 | ChannelZ:i did not do MOH and also im running on verbose already the warning is really flooding the screen, can see a thing |
| 15:15.50 | TobyRulez | look at the actual log file, see if there is anything right before the errors start |
| 15:16.09 | navaismo | hi everuone |
| 15:16.11 | TobyRulez | not sure what os, in CentOS it should be /var/log/asterisk/full |
| 15:16.20 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 15:16.22 | Katty | hi |
| 15:16.49 | ruben23 | TobyRulez: im suing ubuntu-server |
| 15:17.15 | ChannelZ | you might have to edit your logger.conf and make it log verbose, it might not be doing it |
| 15:17.27 | p3nguin | Ubuntu? There's yer problem. |
| 15:17.50 | ChannelZ | No, that seems to be YOUR problem. |
| 15:18.37 | Kobaz | nothing wrong with ubuntu |
| 15:18.48 | Kobaz | it's debian-based, so it's all good |
| 15:19.32 | *** join/#asterisk AlecTaylor (AlecTaylor@unaffiliated/alectaylor) |
| 15:19.33 | AlecTaylor | hi |
| 15:19.34 | AlecTaylor | Do you know of a locally-hostable project which allows for call-in radio-shows to be hosted (and interacted with) through a web-interface? |
| 15:20.01 | Katty | http://thepioneerwoman.com/cooking/2010/07/make-ahead-muffin-melts/ <- |
| 15:20.46 | jaytee | I like Ubuntu as a desktop, much prefer a Red Hat or derivative for servers but that's just my preference for managing a system. Ubuntu works fine as a server for a lot of people. |
| 15:21.04 | *** join/#asterisk gnuday (~gnuday@79.135.102.11) |
| 15:21.18 | ruben23 | guys i see this setting up on verbose--------> http://pastebin.com/zvQGkDb9 |
| 15:21.52 | ChannelZ | ruben23: sip-silence isn't a standard file that I know of. Find it and see what it is |
| 15:21.58 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 15:22.11 | ChannelZ | (generally somewhere in /var/lib/asterisk/sounds/) |
| 15:22.52 | gnuday | Hi I'm looking for an open source load testing solution for asterisk to generate high call volumes between two asterisk servers. Any suggestions? Many thanks |
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| 15:23.14 | TobyRulez | ruben23: looks like it's trying to write a monitor file, have you checked write permissions for the directory? |
| 15:23.52 | TobyRulez | or if the directory exists |
| 15:24.17 | ChannelZ | ah yeah I didn't see that up top |
| 15:26.03 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:27.35 | ruben23 | yes i have the directory exist ---> /var/spool/asterisk/monitor/ |
| 15:28.00 | Qwell | gnuday: sipp |
| 15:28.18 | *** join/#asterisk linusXtorvalds (~e_dot_zil@pool-98-118-168-221.bflony.fios.verizon.net) |
| 15:28.26 | linusXtorvalds | im back bitches |
| 15:28.26 | TobyRulez | ruben23: does /var/spool/asterisk/monitor/MIX exist? |
| 15:28.34 | TobyRulez | ruben23: owner/group? |
| 15:28.43 | *** mode/#asterisk [+b *!*@pool-98-118-168-221.bflony.fios.verizon.net] by Qwell |
| 15:28.44 | TobyRulez | ruben23: folder permissions? |
| 15:28.48 | *** kick/#asterisk [linusXtorvalds!~north@pdpc/sponsor/digium/Qwell] by Qwell (No you aren't. Go away.) |
| 15:29.12 | ruben23 | drwxr-xr-x 2 root root 196240 2011-08-23 08:28 MIX |
| 15:29.32 | anonymouz666 | gnuday: be careful, sipp makes you think when you are about to upgrade the version ;) |
| 15:30.46 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 15:31.12 | anonymouz666 | Qwell: you kicked Linus ! |
| 15:31.13 | anonymouz666 | :P |
| 15:32.40 | *** join/#asterisk jayson_r (~jayson@rrcs-70-61-219-69.midsouth.biz.rr.com) |
| 15:34.19 | TobyRulez | ruben23: hmmm, does asterisk run as the root user? |
| 15:35.22 | ruben23 | yes |
| 15:37.52 | gnuday | Sipp looks perfect for the task but involves a very severe learning curve. I'm really struggling against the clock at the moment. Is there anything quicker/easier or can somebody point me towards a good source of documentation for sipp with many examples? Thanks again |
| 15:38.46 | AlecTaylor | FreePBX maybe? |
| 15:39.57 | navaismo | The warning in the pastebin is about the wav file maybe its corrupted the wav file |
| 15:40.27 | navaismo | use audacity to verify, resample or other thing |
| 15:40.32 | navaismo | or sox |
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| 15:52.17 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 15:52.57 | f2Knight | Q: Anyone using FastAGI? |
| 15:54.13 | AlecTaylor | Does it make Algae grow really fast? |
| 15:56.22 | *** mode/#asterisk [-b *!*@*.dsl.chcgil.ameritech.net] by Qwell |
| 15:57.07 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:57.51 | f2Knight | AlecTaylor, no its for running processes on a different server then asterisk |
| 15:58.40 | p3nguin | than |
| 15:58.41 | AlecTaylor | aww |
| 15:58.45 | AlecTaylor | I was excited |
| 15:59.27 | *** join/#asterisk voipguynumber1 (6276a8dd@gateway/web/freenode/ip.98.118.168.221) |
| 15:59.48 | f2Knight | AlecTaylor, sorry to burst your bubble. |
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| 16:11.06 | AlecTaylor | WAHH |
| 16:11.28 | ChannelZ | the man loves his algae |
| 16:12.56 | Qwell | AlecTaylor: but, you could use FastAGI to talk to a remove algae feeder on another box |
| 16:12.59 | Qwell | remote* |
| 16:13.18 | Katty | eyes Qwell |
| 16:13.25 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 16:13.41 | Qwell | Katty: what? if someone can use AGI to feed a plant - why not algae? |
| 16:17.23 | Katty | i uhmm |
| 16:17.27 | TobyRulez | i can see it now...oh man, i forgot to feed my fish. let me make a quick phone call... |
| 16:17.29 | Katty | yeah. |
| 16:17.35 | Qwell | TobyRulez: it's been done |
| 16:17.54 | TobyRulez | oh i'm sure, can probably do a lot of home automation |
| 16:18.08 | Katty | and more, with open ssh keys |
| 16:19.54 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-217.mobile.uci.edu) |
| 16:20.05 | TobyRulez | and here we've just been using asterisk for phone calls :P |
| 16:20.31 | Qwell | TobyRulez: Speak for yourself. :) Don't tell anyone, but Asterisk has been doing my job for me for the last 4 years. |
| 16:22.32 | TobyRulez | reminds me of "The IT Crowd" (british show about IT dept)...guy had a tape recording he would play for all incoming calls..."have you tried rebooting the computer? ..." |
| 16:25.44 | p3nguin | It really pisses me off when I have to call an ISP over some sort of outage and that's what they ask me. |
| 16:26.18 | p3nguin | As if rebooting a client on a LAN will fix the internet connection on the outside of the edge router. |
| 16:26.32 | citywok | p3nguin: it won't? |
| 16:26.52 | citywok | why, i did that yesterday and it worked just fine after the reboot! |
| 16:26.53 | p3nguin | Do you work for Verizon, AT&T, or Charter? |
| 16:27.26 | citywok | haha, no :P |
| 16:28.10 | aberrios | heh, our carrier once said that I hadn't rebooted our NTE equipment, he said "our panel is showing its still on", to which I replied "Well there's zero power to it, I've unplugged it to the mains, so its definately a problem your end"... Turned out to be a major network outage, of which we were the first to report. |
| 16:29.18 | citywok | hah |
| 16:29.30 | f2Knight | Q: Anyone using FastAGI? |
| 16:29.38 | citywok | my worst was spending 3 hours trying to get to the right department to report that my private ds3 was down, b/c i didn't have my circuit id stamped on my forehead. |
| 16:29.59 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 16:29.59 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 16:30.00 | f2Knight | Esp w/ Python or StarPY or some other python setup? |
| 16:30.00 | citywok | what's your circuitid? no idea, but i have my business name and address! |
| 16:30.14 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:30.23 | Qwell | stupid bip |
| 16:31.18 | p3nguin | I guess they can't do database queries to find it for you. |
| 16:32.11 | f2Knight | p3nguin, we spoke a while ago, was it you that was using starpy? |
| 16:32.18 | p3nguin | negative |
| 16:32.22 | anonymouz666 | anyone in here using 1.8.X under heavy load? I would like to know how realiable things are at this moment |
| 16:32.37 | aberrios | define heavy load? |
| 16:32.38 | citywok | anonymouz666: nope, only under low load (25 people or less) |
| 16:32.40 | Qwell | f2Knight: The Asterisk testsuite uses starpy |
| 16:33.07 | f2Knight | Qwell, Ohh..?? is it in the source tree? |
| 16:33.14 | Qwell | no |
| 16:33.15 | anonymouz666 | aberrios: heavy load it's anything above 150 active calls |
| 16:33.41 | citywok | anonymouz666: why don't you test it using your usage patterns? |
| 16:34.03 | anonymouz666 | just did. few deadlocks and 1 core dumped. |
| 16:34.16 | aberrios | anonymouz666, nope sorry, highest we get is stable at 46 concurrent calls. |
| 16:34.33 | anonymouz666 | aberrios: 1.8? |
| 16:34.44 | f2Knight | Qwell, I mostly am just having an issue retrieving channel variables. (it uses some odd coding) |
| 16:34.56 | aberrios | anonymouz666, 1.8.5.0 |
| 16:35.08 | citywok | aberrios: what kind of hardware do you do that on, and how many calls can you get in 1.6? |
| 16:35.09 | anonymouz666 | hey, it sounds good ! |
| 16:35.59 | aberrios | citywok, Overkill, Quad Core Xeon 2.4ghz, 8GB RAM |
| 16:36.09 | aberrios | citywok, callrecording started on a seperate machine |
| 16:36.41 | aberrios | citywok, 1.6 same, updated to 1.8 for some issue regarding freepbx and device state |
| 16:36.57 | anonymouz666 | aberrios: do you use distributed device state? |
| 16:38.44 | aberrios | anonymouz666, yes |
| 16:38.50 | aberrios | anonymouz666, for some extensions |
| 16:39.07 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-137-186.chyn.qwest.net) |
| 16:39.10 | aberrios | anonymouz666, extensions = devices |
| 16:40.36 | aberrios | anonymouz666, sorry, no, ignore what i just said |
| 16:40.51 | aberrios | anonymouz666, Just device state within one server. |
| 16:40.54 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 16:41.10 | citywok | aberrios: you can only get 46 calls out of that? yikes |
| 16:41.25 | aberrios | citywok, we're only inbound,, limited by PSTN channels |
| 16:41.52 | aberrios | citywok, I used to work at a bigger call center, they had 200 PSTN channels and mainly outbound predictive dialling.. now that was fun! |
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| 16:50.09 | Katty | poke |
| 16:50.32 | chuckf | ouch |
| 16:51.04 | *** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl) |
| 16:52.09 | Katty | hi chuckf |
| 16:53.47 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 16:53.50 | sequencer | hi all :) |
| 16:54.12 | sequencer | how can i change the default behaviour of follow me ? |
| 16:54.21 | *** join/#asterisk catphish_ (~catphish@gateway.office.atechmedia.net) |
| 16:54.40 | sequencer | i want it to grab the call once a peer picks up the phone handle instead of dialing a number |
| 16:55.38 | chuckf | how are ya today katty? |
| 16:55.56 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 16:58.28 | *** join/#asterisk jkroon (~jkroon@197.173.211.42) |
| 16:58.40 | TobyRulez | sequencer: could you elaborate a little more, not sure i follow you |
| 17:01.25 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 17:02.53 | sequencer | TobyRulez yeah |
| 17:03.19 | sequencer | with follow me, it requires tha the user picks up the phone and dials 1 to get a call from the hunt group |
| 17:03.39 | sequencer | i used to had it where the user gets the call immdiately once picking up the phone |
| 17:03.46 | sequencer | but not sure ow to do it |
| 17:04.01 | *** join/#asterisk Godfather_ (~estanteri@90.170.35.213) |
| 17:04.14 | TobyRulez | you using freepbx or anything? |
| 17:04.41 | f2knight | Qwell, Thanks for the reference to testsuite, but none of the tests seem to be reading variables from the channels on the fastagi |
| 17:07.16 | sequencer | am using asterisk Now with diguim gui |
| 17:07.24 | sequencer | but i won use that for configuration |
| 17:07.32 | sequencer | wont* |
| 17:07.57 | sequencer | becuase i made enormous chnges to the files and they will be overwritten if i did |
| 17:14.44 | TobyRulez | hmmm, i'm using FreePBX and you can turn it off in there, but it looks like they implement their own diaplan version of followme. not sure if you can turn it off using asterisks followme app |
| 17:15.04 | sequencer | oh.. |
| 17:15.10 | sequencer | so it seems i have to do my own |
| 17:16.13 | TobyRulez | looks that way. i did see a couple of simple examples on how to do it on voip-info.org |
| 17:16.38 | TobyRulez | example #3 at the bottom is one http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe |
| 17:17.44 | TobyRulez | a few more http://www.voip-info.org/wiki/view/Asterisk+tips+findme |
| 17:17.56 | TobyRulez | probably not exactly what your looking for but maybe will get you there |
| 17:22.12 | sequencer | ok great |
| 17:22.13 | sequencer | thanks!! |
| 17:22.14 | p3nguin | sequencer: If you just want the call to go to another phone after it tried a primary phone first, don't use FollowMe(). |
| 17:22.46 | sequencer | p3nguin actually i want it to ring 3-4 different phones simultaneously |
| 17:22.57 | p3nguin | Dial() does that. |
| 17:23.06 | sequencer | and once one of them picks the phone, it gets there. |
| 17:23.12 | sequencer | ok great let me try that |
| 17:23.17 | p3nguin | In my opinion, the main feature of FollowMe() is to seemlessly check if a person is willing to take a call at another number. |
| 17:23.26 | sequencer | oh i see |
| 17:23.52 | p3nguin | If the call is rejected, FollowMe exits and the next line of dial plan is executed. |
| 17:23.53 | sequencer | can i dial multiple phones at the same time ? |
| 17:23.56 | p3nguin | Yes. |
| 17:24.18 | prgmrchris | p3nguin: the problem i always have with followme and cellphones is once the voicemail picks up the call goes to that |
| 17:24.33 | sequencer | just by Dial(SIP/1234,SIP/2345,SIP/3456) ? |
| 17:24.39 | KavanS | prgmrchris, I found a "key sequence" to send to each provider to exit out of each voicemail by force |
| 17:24.40 | p3nguin | & not , |
| 17:24.46 | sequencer | oh ok |
| 17:24.48 | KavanS | prgmrchris, it's a "hack" but it's reliable |
| 17:24.51 | prgmrchris | KavanS: do tell, thats really good to know |
| 17:24.58 | KavanS | lemme find it... |
| 17:25.01 | p3nguin | prgmrchris: Make sure your outgoing mobile voicemail is longer than your FollowMe timeout value. |
| 17:25.36 | p3nguin | err... outgoing mobile voicemail message |
| 17:25.53 | p3nguin | or mobile voicemail outgoing message |
| 17:26.03 | sequencer | can i set the ringing timeout on Dial ? |
| 17:26.05 | TobyRulez | while we're on the subject of followme, maybe some can help me...we have a problem of lines being left open when ringing out to cell phones. seems like it happens if you are trying to pickup a call the same time someone else is (freepbx implementation of followme) |
| 17:26.15 | p3nguin | My outgoing message is a full minute. My followme timeout is 30 seconds. |
| 17:26.31 | p3nguin | sequencer: Yes. |
| 17:26.36 | KavanS | prgmrchris, http://pastebin.com/E9CTfKgQ |
| 17:26.36 | prgmrchris | imagine the poor souls who have to listen to a minute long message |
| 17:26.44 | p3nguin | sequencer: core show application Dial |
| 17:26.56 | TobyRulez | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 17:27.01 | prgmrchris | KavanS: thanks ill try it out |
| 17:27.02 | p3nguin | prgmrchris: That's the point. I don't want people to leave voicemail on my mobile. |
| 17:27.11 | prgmrchris | KavanS: how did you stumble on that |
| 17:27.16 | KavanS | prgmrchris, for at&t - exten => s,n,SendDTMF(#3331w#3331) |
| 17:27.19 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 17:27.21 | KavanS | I just came up with it... |
| 17:27.27 | KavanS | I figured it'd be a hack, but fuck it |
| 17:27.41 | prgmrchris | if it works, it works |
| 17:27.41 | KavanS | just found what keypresses for each provider would exit the voicemail |
| 17:27.50 | KavanS | our peeps got tired of "press 1 to accept call" on their voicemail |
| 17:27.56 | KavanS | causes a bunch of extra notifications/hassles |
| 17:28.00 | prgmrchris | yea |
| 17:28.02 | p3nguin | I think my logic makes enough sense. |
| 17:28.05 | KavanS | in IT, the less notifications/random shit/beeping, the better |
| 17:28.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:28.26 | sequencer | can i have s then a pattern then another s in a dialplan ? |
| 17:28.41 | prgmrchris | KavanS: agreed |
| 17:28.42 | p3nguin | If your outgoing vm message is 30 seconds long, make the followme timeout less than 30 seconds. Done. |
| 17:29.02 | p3nguin | sequencer: Yes, but why would you need to? |
| 17:29.12 | sequencer | like, exten = s,3 something() then exten = _XXXX, something() then exten = s,6,something() ? |
| 17:29.16 | KavanS | yeah, but how do you determine each "length" of different providers/users voicemail length? |
| 17:29.32 | p3nguin | Know your users. |
| 17:29.35 | sequencer | i need to define what phones to dial based on the dialed |
| 17:29.45 | KavanS | I just plugin the macro for each provider, and we've not had any voicemails since...as the caller you do get to "hear" the dtmf when you pick up |
| 17:29.49 | KavanS | yeah yeah....know your users |
| 17:30.11 | prgmrchris | i like KavanS it looks more badass, when you just tell people to have long voicemails you dont look like a matrix hacker |
| 17:30.15 | prgmrchris | :) |
| 17:30.22 | p3nguin | haha |
| 17:30.36 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 17:30.52 | p3nguin | sequencer: I don't understand what you're saying, but I'm sure it is possible. |
| 17:31.22 | KavanS | lol right on, well let me know how it works out |
| 17:31.37 | sequencer | i want, if the first(dialed) phone didnt anser, dial another 3 phones ( which are located next to missed one ) |
| 17:31.41 | KavanS | we liked not getting additional voicemails on our mobiles...was definitely a nice 'feature' |
| 17:31.47 | sequencer | but having 300 phones.. |
| 17:31.53 | sequencer | we can do the math |
| 17:32.51 | p3nguin | Dial() one phone and then Dial() three phones... that's the easy part. How will you know what devices to dial? |
| 17:33.02 | sequencer | lets say for instance |
| 17:33.25 | sequencer | 1111 rings, no body answers, i want 1112 and 1113 and 1114 to ring |
| 17:33.39 | sequencer | if 1211 rings, i want 1212 , 1213 , 1214 to ring |
| 17:33.42 | sequencer | and so on |
| 17:33.52 | sequencer | i have list of what should ring |
| 17:34.00 | p3nguin | You have phones with these numbers as their names? |
| 17:34.12 | sequencer | yes, these are the phone extensions |
| 17:35.38 | sequencer | can i have it where exten = _3333331111,4,Dial(SIP/1112&SIP/1113&SIP/1114) ? |
| 17:37.01 | ketas-av | aberrios: did you actually meant "big spam center"? |
| 17:41.01 | voipguynumber1 | hi |
| 17:41.39 | navaismo | hell-o |
| 17:41.42 | p3nguin | sequencer: Well, you've mixed terms again. They are either the phone names or the extensions used to dial the phones. If you've named your phones with those numbers, you've made a mistake. Phones should be named with something specific to the device. |
| 17:42.17 | p3nguin | If the extension is 3333331111, you don't need _ on the front of it. _ is for patterns. |
| 17:42.46 | p3nguin | But extension 3333331111 certainly can Dial() three phones. |
| 17:43.02 | p3nguin | It can even Dial() one first, and then three next. |
| 17:43.06 | sequencer | ok thanks |
| 17:43.17 | p3nguin | Just so we're clear, phones are not extensions. |
| 17:43.22 | sequencer | right.. |
| 17:43.25 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:43.30 | p3nguin | And extensions are not phones. |
| 17:43.36 | p3nguin | Extensions can Dial() phones, though. |
| 17:43.55 | sequencer | my question is , 33333331111 is the called number from an incoming call |
| 17:44.02 | sequencer | so i suppose this shoudl work ? |
| 17:44.03 | p3nguin | That's the extension. |
| 17:44.10 | p3nguin | extention 3333331111. |
| 17:44.13 | p3nguin | extension, rather |
| 17:44.16 | sequencer | yes |
| 17:44.29 | sequencer | extension.. |
| 17:44.41 | p3nguin | exten => 3333331111,1,DoWhateverYouWantHere(). |
| 17:44.44 | sequencer | like 4-digit extension? or should i define the full number ? |
| 17:45.00 | p3nguin | You need to define whatever number is being called. |
| 17:45.06 | sequencer | great |
| 17:45.13 | sequencer | then this should be correct |
| 17:45.13 | p3nguin | If the number is 3333331111, extension 1111 won't match. |
| 17:45.23 | TobyRulez | how about something like... exten => _XXX1,n,Dial(SIP/${EXTEN:0:3}2&SIP${EXTEN:0:3}3&${EXTEN:0:3}4 ...) |
| 17:45.30 | sequencer | yeah , usually someone will call the DID |
| 17:45.35 | TobyRulez | so if they dial 1111...it will dial 1112, 1113, etc |
| 17:45.53 | TobyRulez | same if 1231, it dials 1232, 1233, etc |
| 17:46.00 | sequencer | thats a great plan |
| 17:46.08 | sequencer | but o do have alot of outliers as well |
| 17:46.42 | p3nguin | Using those 4-digit numbers as the device names is probably a terrible idea. If the phones have those 4-digit numbers engraved into the plastic, then I'd say go for it. |
| 17:46.59 | p3nguin | Peer names should be based on some unique information from the hardware. |
| 17:47.12 | p3nguin | Most sane people use the MAC address. |
| 17:47.38 | p3nguin | Weird people and people without much experience tend to use arbitrary or even random numbers. |
| 17:48.03 | sequencer | am not using that though |
| 17:48.08 | sequencer | becuase i have matching DIDs |
| 17:48.20 | sequencer | on the last 4 digits of all my phones |
| 17:48.25 | p3nguin | You're not making sense again. |
| 17:48.36 | sequencer | phones/extensions |
| 17:48.41 | sequencer | am lost |
| 17:48.56 | p3nguin | Phones' sip names should be based on unique information FROM THE PHONES. |
| 17:49.16 | p3nguin | The MAC addresses on the phones are unique pieces of information from the phones. |
| 17:49.20 | p3nguin | 1111 is not. |
| 17:49.47 | p3nguin | People around here have a tendency to encourage bad practices. I'm not one of those people. |
| 17:50.40 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 17:51.24 | sequencer | thanks :) |
| 17:51.40 | sequencer | i got an auto fallthrough :s |
| 17:53.03 | p3nguin | <@leifmadsen> devices, extensions, and people should be entirely abstracted |
| 17:53.12 | p3nguin | <@leifmadsen> extension numbers are applied to people, and people are applied to a device |
| 17:53.22 | Qwell | Help! I'm stuck in a device! |
| 17:53.24 | p3nguin | <@leifmadsen> (which means you should name your devices something unique to the device, such as an ID tag, or a MAC address) |
| 17:53.40 | p3nguin | applied to, not forced inside of |
| 17:53.51 | Qwell | oh, phew |
| 17:53.54 | p3nguin | GET BACK IN YOUR BOX |
| 17:54.12 | Qwell | But yes, I support his comments fully. |
| 17:55.14 | p3nguin | Grabbing a phone off the shelf and deciding to call it 1115 arbitrarily doesn't make a lot of sense. |
| 17:55.53 | navaismo | why not? |
| 17:55.56 | p3nguin | But looking at the tag that is permanently affixed to that phone and using the ID number from it as the name for the phone... well, that's using your brain. |
| 17:56.22 | p3nguin | <@leifmadsen> it's hard enough to understand the logic in your head about an extension number and a device, than to call them the same thing |
| 17:57.03 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 17:57.05 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 17:58.33 | p3nguin | If you put asset tags on phones, you're familiar with recording the MAC address anyway. |
| 18:00.59 | p3nguin | qwell: What's your opinion of using the asset tag number as the phone's peer name? Too temporary? |
| 18:01.09 | Qwell | if it's short/unique, sure |
| 18:01.14 | *** join/#asterisk delki8 (~delki8@189.5.136.31) |
| 18:01.32 | Qwell | presumably they won't change if the sticker falls off (though, the sticker can fall off) |
| 18:01.56 | p3nguin | My concern was people peeling off stickers and switching them around. |
| 18:02.12 | Qwell | You've got a whole other problem on your hands in that case. |
| 18:02.27 | _Corey_ | Anyone else in the northeast? We just felt an earthquake in Philly |
| 18:02.33 | p3nguin | While there would still be a permanent record somewhere else, it would cause confusion for an admin. |
| 18:02.37 | beek | Have it in South Central PA |
| 18:02.51 | beek | _Corey_ ^^^^^ (State College) |
| 18:02.53 | _Corey_ | Looks like the epicenter was in Virginia |
| 18:02.56 | Kobaz | felt shakes in upper new york |
| 18:03.00 | beek | http://earthquake.usgs.gov/earthquakes/recenteqsww/Quakes/at00lqe6x3.php |
| 18:03.05 | Qwell | 5.8 outside of CA? Crazy. |
| 18:03.05 | TobyRulez | _Corey_ we just felt it in south carolina |
| 18:03.15 | _Corey_ | Seriously, wild stuff |
| 18:03.45 | Qwell | huh. there've been a bunch of note all over the US today. |
| 18:03.51 | thehar | wow |
| 18:04.00 | p3nguin | I've got the maintenance man running a power washer on the other side of the wall, so I haven't noticed anything like that. |
| 18:04.08 | _Corey_ | Yeah, I guess there was a pretty big one in CO yesterday |
| 18:04.12 | Kobaz | beek: ah. you're in state college? |
| 18:04.15 | Qwell | _Corey_: repeated today |
| 18:04.19 | beek | Kobaz: Just SW of it. |
| 18:04.25 | _Corey_ | hmm, wild |
| 18:04.28 | Qwell | _Corey_: todays was bigger, even. same place. |
| 18:04.33 | Kobaz | I used to live in bellwood (near tyrone) |
| 18:04.41 | beek | Wow... I'm in Huntingdon. |
| 18:04.50 | Kobaz | yeah, i used to paddle in huntingdon |
| 18:04.58 | beek | Small world. |
| 18:05.10 | beek | Kobaz: How long ago? |
| 18:05.12 | _Corey_ | Yeah, I'm kind-of wondering if the VA thing is going to repeat... |
| 18:05.16 | Kobaz | moved in may |
| 18:06.36 | Qwell | _Corey_: if so, it'll be ~6.5 |
| 18:07.09 | voipguynumber1 | _Corey_: just felt it in Buffalo |
| 18:07.16 | voipguynumber1 | around 10 minutes ago |
| 18:07.39 | Qwell | Anyways, offtopic. Carry on! |
| 18:08.23 | _Corey_ | It certainly traveled.. :) Looks like we have some PRI problems calling in |
| 18:08.43 | _Corey_ | goes back to work |
| 18:12.36 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 18:14.56 | TobyRulez | p3nguin: still unclear of why the need for abstraction of device and extension number like you were talking about. example? |
| 18:15.43 | p3nguin | I don't know how else to say it. |
| 18:16.42 | TobyRulez | i guess in our case (application), extensions aren't mapped to people, they remain in the same spot |
| 18:16.49 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 18:16.51 | TobyRulez | pharmacy enviroment |
| 18:16.56 | TobyRulez | not a typical office |
| 18:17.15 | TobyRulez | i can see how that applies in that case |
| 18:17.39 | p3nguin | The main focus of the topic was that the phones should not have arbitrary numbers given to them as their peer names, and then using that same number as the extension used to dial that phone. |
| 18:19.09 | leroybuckingham | I have a packet capture where I'm seeing DTMF being sent over rfc2833 both from the endpoint to asterisk, and from asterisk to the SIP provider, but the capture is showing pretty clearly that about a third of the DTMF signals are not being sent to the provider. Could this be a configuration issue or is it a problem with my asterisk version 1.6.2.18? |
| 18:21.18 | leroybuckingham | The same problem occurs when the endpoint transmits DTMF with sip info. The provider only accepts rfc2833. |
| 18:21.51 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 18:22.54 | jkroon | hi guys, is anybody aware of issues with "sip reload" in 1.8.5.0 ?? |
| 18:23.08 | Qwell | jkroon: issues such as? |
| 18:23.19 | jkroon | lockup of chan_sip |
| 18:23.44 | jkroon | and bunch of lines being output re register lines not containing = signs on lines in files that doesn't exist. |
| 18:23.54 | p3nguin | Does same => not work in 1.4.39? The README-bestpractices file suggests that it does, but I'm not having luck with it. |
| 18:24.00 | voipguynumber1 | leroybuckingham: are the endpoints are sending rtpmap:101 |
| 18:24.07 | Qwell | p3nguin: I want to say that's 1.6.0+ |
| 18:24.11 | Qwell | maybe even higher |
| 18:24.51 | p3nguin | I was reading in src/asterisk-1.4.39.2/README-SERIOUSLY.bestpractices.txt |
| 18:24.56 | p3nguin | and saw a reference to it. |
| 18:25.14 | Qwell | How are you using it? |
| 18:25.39 | p3nguin | exten => 3149691077,1,NoOp() |
| 18:25.40 | p3nguin | same => n,DoStuff() |
| 18:26.46 | p3nguin | I've used it in 1.8.something before, and I was pretty sure that was the syntax I used. |
| 18:27.25 | p3nguin | core verbosity indicated to me that there was no second priority to run when I used it in this 1.4 version. |
| 18:27.38 | p3nguin | The call eventually timed out and ran h. |
| 18:28.44 | p3nguin | I always thought that it did not work in 1.4, but when I saw it in that text file, I had to try it. |
| 18:31.55 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 18:32.05 | leroybuckingham | voipguynumber1: rtpmap:101 telephone-event/8000 is in the INVITE message from the endpoint |
| 18:32.43 | Kobaz | http://earthquake.usgs.gov/earthquakes/dyfi/events/us/c0005ild/us/form.en.disabled.html |
| 18:32.52 | eXcAliBuR | i'm trying to get my asterisk box to accept calls from pstn, I got the card in and did the configs on pages 140- of the 3rd edition of asterisk book |
| 18:33.01 | eXcAliBuR | when i call it says that my extension is busy... |
| 18:33.43 | eXcAliBuR | the cable i plugged into the digium card comes from a toshiba pbx, so i'm not sure if it needs something fancy to realize asterisk is there |
| 18:33.45 | eXcAliBuR | :/ |
| 18:34.08 | eXcAliBuR | how can i test to see if my card is properly configured and recongized? |
| 18:34.33 | jkroon | dahdi show channels? |
| 18:35.18 | eXcAliBuR | no such command |
| 18:35.40 | jkroon | do you have dahdi installed? |
| 18:35.44 | eXcAliBuR | yes |
| 18:35.45 | jkroon | loaded? |
| 18:35.48 | eXcAliBuR | don't know |
| 18:35.57 | jkroon | lsmod ? |
| 18:36.03 | eXcAliBuR | i can do /etc/init.d/dahdi restart |
| 18:36.45 | eXcAliBuR | looks like it's loaded |
| 18:36.51 | eXcAliBuR | it shows when i type lsmod |
| 18:38.04 | leroybuckingham | eXcAliBuR: what about "module load chan_dahdi.so" from the asterisk console? |
| 18:39.01 | eXcAliBuR | complains that it can't load channel 1 not found or no device |
| 18:39.51 | leroybuckingham | freepbx? |
| 18:40.00 | navaismo | lsdahdi from root console? |
| 18:40.56 | eXcAliBuR | i think i don't have the mod for my card being loaded |
| 18:41.08 | eXcAliBuR | i'll try adding something in /etc/dahdi/modules |
| 18:41.20 | navaismo | dahdi_hardware? |
| 18:41.31 | navaismo | what card is? |
| 18:41.44 | leroybuckingham | ls -l /etc/asterisk/chan_dahdi.conf |
| 18:42.08 | eXcAliBuR | i have a digitum 1x100MF FXO single module and a digium 1TDM410PLF 4 port PCI card |
| 18:43.56 | eXcAliBuR | i added wctdm24xxp |
| 18:44.52 | eXcAliBuR | [Aug 23 14:44:29] ERROR[7319]: chan_dahdi.c:16522 build_channels: Unable to register channel '1-4' |
| 18:45.11 | eXcAliBuR | i only have 1 channel so should I just put 1 and not 1-4 |
| 18:45.12 | eXcAliBuR | ? |
| 18:46.07 | malcolmd | eXcAliBuR: contact our Support department (http://www.digium.com/en/supportcenter/) they provide complimentary installation assistance |
| 18:46.14 | jkroon | Qwell, [Aug 13 13:08:26] WARNING[7590] config.c: No '=' (equal sign) in line 3598 of /etc/asterisk/sip-register.conf |
| 18:46.49 | jkroon | however, sip-register.conf in this case contains a single line of the format: register => user:pass@host/cli |
| 18:47.44 | ChannelZ | does it exist under [general]? |
| 18:48.03 | eXcAliBuR | okies |
| 18:51.07 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 18:51.44 | *** join/#asterisk dre- (~dre@69.38.200.246) |
| 18:57.27 | f2knight | Q: Anyone have experience with fastagi? |
| 18:59.17 | pabelanger | A: yes. It is a common way to write asterisk applications |
| 19:00.21 | eXcAliBuR | i wish i was smart... then i could be intelligent too :) |
| 19:02.37 | navaismo | i wish i have money a lot money... |
| 19:11.30 | p3nguin | Not me. |
| 19:11.39 | p3nguin | I wish stuff didn't cost money. |
| 19:12.06 | *** join/#asterisk sdh (~foo@steve.st) |
| 19:12.12 | p3nguin | If everything was free, I wouldn't need money. |
| 19:12.49 | navaismo | but we are in a capitalist world so i need money :( |
| 19:13.07 | atheos | I wish I was a little bit taller |
| 19:16.29 | eXcAliBuR | :P |
| 19:23.32 | *** join/#asterisk voipguynumber1 (6276a8dd@gateway/web/freenode/ip.98.118.168.221) |
| 19:24.57 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:30.38 | voipguynumber1 | did everyone survive the earthquake? |
| 19:32.49 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 19:44.38 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 19:45.03 | f2knight | FastAGI with starpy anyone? |
| 19:45.47 | p3nguin | What causes a call to land on the 'fax' or 'failed' extension? |
| 19:46.58 | TobyRulez | fax - if fax tones are heard...failed - if extension doesnt exist? |
| 19:47.05 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-lyorxqfhzamrjnnb) |
| 19:48.40 | JonathanRose | Anyone familiar with Openfire? More specifically, I was hoping someone could get me pointed in the direction of how to get started with pubsub in OpenFire. |
| 19:48.58 | p3nguin | Assume a fax call comes to me... if my ITSP sends my calls to 4154499909, the call will end up on my extension 4154499909. From there, how does it get to extension 'fax'? |
| 19:49.07 | voipguynumber1 | JonathanRose: i prefer red5, haven't used openfire |
| 19:49.45 | pabelanger | f2knight: yes, we use it alot in the Asterisk testsuite |
| 19:49.53 | p3nguin | I use openfire for a simple messaging platform. I'm not sure about pubsub, though. |
| 19:50.04 | *** join/#asterisk nunne (~nunne@c-56f0e355.021-109-73746f46.cust.bredbandsbolaget.se) |
| 19:50.08 | TobyRulez | from what i gather, as long as you have a fax extension in your context, asterisk does the rest |
| 19:50.17 | f2knight | pabelanger, Could you spare a moment to walk me through something? |
| 19:50.42 | p3nguin | It will magically change extensions after the call has gone to 4154499909? |
| 19:50.48 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:50.53 | p3nguin | That doesn't sound right. |
| 19:51.14 | TobyRulez | try it and see |
| 19:51.35 | p3nguin | Some fax machines don't send their tones before I answer, so how will it work? |
| 19:51.44 | anonymouz666 | JonathanRose: I talked to you in another channel, but it is EXACTLY what I am testing right now |
| 19:51.57 | p3nguin | There's nothing for me to try. I accept fax on the numbered extension just fine. |
| 19:51.59 | anonymouz666 | OPENFIRE + PUBSUB |
| 19:52.05 | TobyRulez | sure they do...if faxed properly |
| 19:52.18 | TobyRulez | sending machine should send ced and terminating machine send cng |
| 19:52.20 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-217.mobile.uci.edu) |
| 19:52.21 | pabelanger | f2knight: infact, we now maintain it on github: https://github.com/asterisk-org/starpy |
| 19:52.23 | TobyRulez | or vice versa |
| 19:52.28 | pabelanger | f2knight: sure, whats up |
| 19:52.34 | p3nguin | I might answer before the person sending the fax pressed SEND, so no, they don't. |
| 19:52.47 | f2knight | pabelanger, Some of the coding styles that are used in the examples have me confused, but the biggest thing is I am trying to do is read and write channel variables. |
| 19:53.07 | f2knight | pabelanger, but for the life of me I can not seem to get it working. |
| 19:53.08 | TobyRulez | yeah, we have that problem. basically we tell them to put the paper in and hit send |
| 19:53.16 | TobyRulez | don't pick up the handset |
| 19:53.30 | pabelanger | f2knight: ya, the examples need some work. |
| 19:53.43 | f2knight | pabelanger, It also kinda seems that I need to build my own class objects to even make use of it. I hope thats not so. |
| 19:53.44 | p3nguin | Not everyone has a dedicated fax number, so using the handset is reasonable. |
| 19:53.45 | pabelanger | let me see if we have something in the testsuite |
| 19:54.12 | f2knight | pabelanger, What I was hopping for was something like var1 = agi.getVariable('MYCHANVAR') |
| 19:54.12 | p3nguin | So how does a call get from extension 4154499909 to extension fax all by itself? |
| 19:54.13 | TobyRulez | right, but when you actually speak to them, you can let them know |
| 19:54.29 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 19:54.29 | JonathanRose | anonymouz666: Well that's exciting. I personally am just getting started with both for an issue I'm working on, but I really have no idea how to start setting up pubsub. |
| 19:54.53 | p3nguin | I would have thought some application would have to send a call to extension fax. |
| 19:55.03 | f2knight | pabelanger, the getVariable example doesnt even read the Channel variables it parses the doc string. |
| 19:55.58 | TobyRulez | app_rxfax and app_txfax...unless you go with a third party |
| 19:56.50 | f2knight | pabelanger, I have a working pyst local AGI that I am needing to convert to FastAGI, and well .. starpy looks like the only one really. |
| 19:56.52 | TobyRulez | also need to be sure faxdetect is setup properly in dahdi/zaptel conf |
| 19:56.54 | p3nguin | So exten 4154499909 need to run rxfax() to branch it over to exten 'fax'? |
| 19:57.20 | p3nguin | What if it isn't a fax call? Will rxfax() exit cleanly, allowing dialplan to progress? |
| 19:57.26 | TobyRulez | no, use "fax" exactly like you would use i or T or t, etc |
| 19:57.30 | TobyRulez | its automagic |
| 19:57.33 | *** part/#asterisk JonathanRose (~jonathan@nat/digium/x-lyorxqfhzamrjnnb) |
| 19:57.39 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-lyorxqfhzamrjnnb) |
| 19:57.42 | *** join/#asterisk Anthony- (~foo@ip68-104-173-24.ph.ph.cox.net) |
| 19:57.42 | *** join/#asterisk iamaham (~iamaham1@web2.supergreenhosting.com) |
| 19:57.44 | iamaham | greetings |
| 19:57.46 | TobyRulez | http://www.voip-info.org/wiki/view/Asterisk+fax |
| 19:57.48 | *** part/#asterisk Anthony- (~foo@ip68-104-173-24.ph.ph.cox.net) |
| 19:58.02 | iamaham | I keep getting dropped calls so I started monitoring and this is what I'm getting, any ideas? |
| 19:58.03 | iamaham | WARNING[14794]: chan_iax2.c:2289 __attempt_transmit: Max retries exceeded to host ipaddress on IAX2/hostname2-7973 (type = 6, subclass = 2, ts=105021, seqno=37) |
| 19:58.04 | pabelanger | f2knight: checkout the fastagi/execute test in the testsuite, it uses the getVariable() function in starpy |
| 19:58.05 | pabelanger | http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/fastagi/execute/ |
| 19:58.32 | f2knight | pabelanger, if I call AGI(agi://127.0.0.1:8888/myscript) I would think that the 'myscript' part should be the default function to be called. |
| 19:59.08 | f2knight | pabelanger, I did look in there, I see calls to it, but its all in a custom class object. Do I have to build my own class object just to use Staypy? |
| 19:59.26 | TobyRulez | p3nguin: set faxdetect=incoming (or outgoing, both, no). add fax => 1,rxfax(/path/to/fax) to context |
| 19:59.32 | iamaham | any tips is appreciated |
| 19:59.42 | iamaham | are appreciated rather :) |
| 19:59.44 | TobyRulez | when asterisk hears the cng, it will automatically route to fax extension |
| 19:59.55 | jaytee | iamaham, what do you get when you type "iax2 show peers" |
| 19:59.57 | p3nguin | So exten 'fax' is only relevant when using analog? |
| 20:00.25 | pabelanger | f2knight: no, you don't need a class. You can simply add the logic to your main() function |
| 20:00.28 | p3nguin | Assume sip. Assume the phone number is not dedicated to fax. How how's it going to work? |
| 20:00.29 | TobyRulez | we've got it on PRI and T1's setup exactly the same as POTS if thats what you mean |
| 20:00.42 | anonymouz666 | JonathanRose: you need to download the smack.zip, and then extract the jar file that is inside the smack. after that you need to upload it through the web interface - basically that is. |
| 20:00.54 | *** join/#asterisk iamaham (~iamaham1@web2.supergreenhosting.com) |
| 20:00.59 | iamaham | sorry client crashed what was the command? |
| 20:01.02 | pabelanger | however, you'll need to create a fastagi object and define callbacks for it |
| 20:01.05 | TobyRulez | i wouldn't do sip |
| 20:01.06 | jaytee | iax2 show peers |
| 20:01.11 | p3nguin | But I do. |
| 20:01.13 | iamaham | ok one sec checking |
| 20:01.14 | iamaham | tyvm |
| 20:01.23 | TobyRulez | you will have quality/reliability problems |
| 20:01.29 | p3nguin | I rarely do. |
| 20:01.32 | TobyRulez | for fax? |
| 20:01.34 | anonymouz666 | JonathanRose: I am trying the OPENFIRE just because the res_ais is not working here (deadlock suspect), but is million times easier |
| 20:01.56 | TobyRulez | sip is great for voice, but not fax |
| 20:01.57 | p3nguin | I'm not asking for commentary on the ability to fax over sip successfully. |
| 20:01.58 | navaismo | imaham the calls droped are natted or with external(outside lan) extension? |
| 20:01.58 | iamaham | it lists my 2 other asterisk servers |
| 20:02.11 | JonathanRose | anoymouz666 thanks |
| 20:02.27 | jaytee | iamaham, can you pastebin the output of the command? |
| 20:02.29 | iamaham | yeah have 3 asterisk servers all connected via iaxy2 protocol |
| 20:02.31 | jaytee | ~pb |
| 20:02.31 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:02.32 | p3nguin | I want to know how will a call to my DID which ends up on a specific extension arrive at either a phone or to the fax extension. |
| 20:02.44 | iamaham | one sec |
| 20:03.13 | TobyRulez | exten => fax,1,rxfax(/path/to/faxfile) |
| 20:03.19 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 20:03.26 | p3nguin | nope, not going to happen. |
| 20:03.35 | TobyRulez | ok, then you're on your own |
| 20:03.44 | p3nguin | The call is sent to extension 4154499909, not fax. |
| 20:03.46 | iamaham | http://pastebin.com/3nYp9nRs |
| 20:04.01 | jaytee | iamaham, good, gimme a sec to look at it |
| 20:04.04 | TobyRulez | http://www.voip-info.org/wiki/view/Asterisk+fax |
| 20:04.19 | p3nguin | You already suggested that page, and my answer is not on it. |
| 20:04.25 | navaismo | imaham does the connection with the other servers are alive |
| 20:04.26 | f2knight | pabelanger, no chance you have a working example would you.? Because I don't really understand twisted. ANd what totally gets me confused with starpy is this assignment of the Factory and that just confuses me more because the examples majicly have an 'agi' reference and I can not find anywhere it gets assigned at. |
| 20:05.02 | TobyRulez | sure it is |
| 20:05.06 | iamaham | yeah they all have static IP's. It's been working for years, so this is just a recent event (no system changes) |
| 20:05.08 | p3nguin | No, it isn't. |
| 20:05.25 | jaytee | iamaham, did you mask the actual IP addresses or is that the actual output? |
| 20:05.26 | iamaham | it's sip->asterisk ->iaxy2 over the internet ->asterisk2->sip phone |
| 20:05.34 | iamaham | I changed the name and IP addresses |
| 20:05.34 | pabelanger | f2knight: nothing more then the testsuite. |
| 20:05.39 | TobyRulez | about halfway down, "Emailing a faxe based on DID", not exactly what you are looking for, but example of how its done |
| 20:05.51 | iamaham | didn't want to give production IP addys out :) |
| 20:06.08 | jaytee | ok, but can you ping the WAN IP address for the server the call to failed on? |
| 20:06.15 | pabelanger | f2knight: you might want to start with using starpy to log into the AMI. Then built it up from there. Once you understand twisted, things will become cleared. |
| 20:06.30 | pabelanger | the manager/login test is pretty basic |
| 20:06.36 | iamaham | yup, line is up... I can make calls it just auto hangs up after a minute or so |
| 20:06.36 | JonathanRose | wonders how long a "short time for the plugin to appear in the list of installed plugins" is. |
| 20:06.49 | TobyRulez | If you are trying to detect faxes over IAX, SIP, or for that matter any type of channels, Newman has created NVFaxDetect and updated BackgroundDetect as NVBackgroundDetect for that purpose. We have had near perfect results on decent IAX connections using ULAW/ALAW. Fax detection utilizes Asterisk DSP and works in the same way. Once detected, faxes are sent to the fax extension — look at |
| 20:06.50 | TobyRulez | Zap fax detection abov |
| 20:07.14 | anonymouz666 | JonathanRose: it doesn't show |
| 20:07.32 | navaismo | asterisk hangup because dont fnd the oath to your other asterisk |
| 20:07.37 | navaismo | find* |
| 20:07.39 | JonathanRose | Oh blimey. |
| 20:07.49 | anonymouz666 | at least in here. |
| 20:07.55 | leroybuckingham | what could cause asterisk not to queue DTMF? |
| 20:08.00 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:08.13 | anonymouz666 | leroybuckingham: native bridge |
| 20:08.42 | f2knight | pabelanger, http://pastebin.com/F2W79CQS |
| 20:08.43 | iamaham | when it hangs up I get this error |
| 20:08.47 | iamaham | chan_iax2.c:2289 __attempt_transmit: Max retries exceeded to host |
| 20:09.02 | navaismo | yep asterisk cant find the host |
| 20:09.03 | iamaham | maybe just instability in the net in general? routing issue, etc |
| 20:09.18 | jaytee | iamaham, latency most likely |
| 20:09.37 | iamaham | it's a DSL line (feel free to laugh, 1meg down 1 meg up) |
| 20:09.44 | iamaham | but only service offered to that site |
| 20:10.13 | leroybuckingham | anonymouz666: I'm seeing this issue when I use 1.6.2.18 but not 1.6.2.13 |
| 20:10.19 | leroybuckingham | does that make sense? |
| 20:10.23 | leroybuckingham | all configs are being left alone |
| 20:10.32 | iamaham | hrm ok so pretty much what I was thinking network issue. those dsl lines are flaky as hell |
| 20:10.36 | pabelanger | f2knight: what is that suppose to do? |
| 20:10.38 | p3nguin | I've never heard of anyone using this NVFaxDetect app to get calls to the fax extension before. |
| 20:10.51 | iamaham | appreciate your help jaytee and navaismo |
| 20:10.54 | jaytee | iamaham, I understand. Alot of my clients are in Bugtussell and Hooterville and they only get dialup |
| 20:10.56 | p3nguin | iamaham: Gotta watch out for those digital subscriber line lines. |
| 20:11.14 | p3nguin | Also have to watch out for the alots. |
| 20:11.19 | p3nguin | They might eat you! |
| 20:11.27 | iamaham | heh |
| 20:11.41 | iamaham | well have a good one |
| 20:11.49 | jaytee | iamaham, you might want to add qualify=yes for your iax peers so you can observe the latency and get notices if a peer goes "unreachable" |
| 20:12.22 | f2knight | pabelanger, well it DOES nothing, but what it is wanted to do was read the Channel Variable LICENSE and assign it to a variable, print that variable, and disconnect the connection. |
| 20:12.51 | f2knight | pabelanger, but what it does is prints out <Deferred at 0x2342440> |
| 20:12.51 | f2knight | <PROTECTED> |
| 20:14.15 | pabelanger | f2knight: right, because you need to first create a FastAGIFactory() |
| 20:15.00 | TobyRulez | p3nguin: which version of asterisk are you running? |
| 20:15.08 | p3nguin | 1.4.39.2 |
| 20:15.32 | p3nguin | I'm currently using ffa successfully. |
| 20:15.59 | TobyRulez | hmmm, nvfaxdetect says may not work on versions > 1.4 ...doesn't say exactly where the line is drawn |
| 20:16.01 | p3nguin | A call comes in to my fax number, I answer it, and run ReceiveFAX() to accept a fax. |
| 20:16.21 | p3nguin | And my setup has absolutely nothing to do with my original question. |
| 20:16.36 | pabelanger | f2knight: the best example to follow is fastagi/execute in the testsuite. It will create a fastagi.FastAGIFactory on port 4574, then launch the do_test() function when your dialplan connects to it. You can then follow along who the callback events are processed as the application moves forward |
| 20:17.27 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 20:17.31 | TobyRulez | you asked how to route a fax to the fax extension in a sip environment. my original response was wrong. you would need something like nvfaxdetect for that. nvfaxdetect doesn't work on versions > 1.4, however 1.8 has built in fax detection. seems relevant |
| 20:18.29 | p3nguin | I just wanted to know what makes a call to my DID number, which arrives in asterisk at the extension that is my DID number, somehow branch off to either a phone or to exten 'fax'. |
| 20:18.34 | f2knight | pabelanger, I keep looking at it. I see that do_test, but where is it getting the agi from? (def do_test(self, agi): because the calling line is passing no arguments to it. |
| 20:19.38 | TobyRulez | and my response still stands. you need fax detection (in this case nvfaxdetect might work for you). and then you include a fax extension in the same context as your regular extension (same as i, t, T, fail, etc) |
| 20:19.54 | *** join/#asterisk MiserySoft (~lnd@host81-148-14-51.in-addr.btopenworld.com) |
| 20:19.55 | f2knight | pabelanger, self.agi_factory = fastagi.FastAGIFactory(self.do_test) what is more is following from there half the functions never have a call from the d0_test. |
| 20:20.32 | f2knight | pabelanger, and that is making me confused. |
| 20:21.16 | p3nguin | If a fax detection application is required for it to happen, that takes out the magic that had my puzzled. Having an app rather than magic making the determination makes more sense. |
| 20:21.56 | TobyRulez | yes, fax detection occurs in either dahdi/zaptel drivers OR in your case, a third party app such as nvfaxdetect |
| 20:22.04 | p3nguin | But with a fax detection app standing in the way, a late fax tone would cause phones to ring rather than going to exten fax. |
| 20:22.38 | TobyRulez | yes, you would probably want to add an extra ring or two before you actually ring the extension to give it a few seconds to detect |
| 20:22.57 | TobyRulez | in our case, we have an ivr that adds that extra time |
| 20:23.11 | pabelanger | f2knight: you need to go read up on how the twisted protocol works. By registering do_test() as the callback function in fastagi.FastAGIFactory(), starpy will then pass the 'agi' arguments into do_test(self, agi). |
| 20:23.47 | pabelanger | do_test() is the callback function you want starpy to use |
| 20:28.17 | pabelanger | f2knight: I recommend reading: http://krondo.com/blog/?p=1209 |
| 20:28.51 | pabelanger | One of the best, and easiest, tutorials to follow on twised |
| 20:29.03 | pabelanger | s/twised/twisted/ |
| 20:30.31 | f2knight | pabelanger, guess I will get started reading. Quick Quesiton about starpy ... if its using twisted for all the service interfacing, how many connections can be dumped in to it at once? |
| 20:32.12 | pabelanger | f2knight: not sure, you'd have to test and see. How many are you looking to do? |
| 20:32.55 | f2knight | pabelanger, about 80 lookups a second. |
| 20:33.54 | pabelanger | give it a try and see. Like you said, starpy is built on twisted, and it has been around for a while |
| 20:36.58 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-nznyddujgtfnjgub) |
| 20:44.38 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 20:56.21 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:56.22 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 20:57.51 | JasonL | I'm seeing high send-q/recv-q on UDP 5060 when i do a netstat -an at peak times and people report system problems like dropped calls... any ideas? |
| 21:04.15 | navaismo | and what show the cli when the call is dropped |
| 21:06.34 | *** join/#asterisk nmjnb (~nmjnb@c-567e72d5.026-18-73746f23.cust.bredbandsbolaget.se) |
| 21:06.58 | nmjnb | how can I see what the different trunk statuses mean? |
| 21:08.03 | navaismo | nmjnb excuse me? |
| 21:08.39 | navaismo | don't understand |
| 21:08.44 | nmjnb | navaismo: I'm logged on to my asterisk, and I added 2 trunks, and I'd like to understand their statuses as they're not working. |
| 21:09.07 | navaismo | its a sip trunk? |
| 21:09.11 | WIMPy | What statuses? |
| 21:09.14 | nmjnb | meaning I can call within the asterisk users but not reach the "normal" phones |
| 21:09.16 | WIMPy | ~siptrunk |
| 21:09.16 | infobot | rumour has it, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 21:09.55 | navaismo | unreachable, ok, or Registered, failed, request sent? |
| 21:10.09 | nmjnb | the trunk status is a number |
| 21:10.19 | navaismo | 0_o |
| 21:10.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:10.40 | WIMPy | Are you talking about Asterisk? |
| 21:10.58 | nmjnb | WIMPy: I could choose between SIP or IAX when setting the trunks |
| 21:11.01 | nmjnb | yes |
| 21:11.18 | WIMPy | And where are you seeing those numbers? |
| 21:11.29 | navaismo | mm in asterisk sip show peers or sip show registry the state column is a word |
| 21:11.53 | nmjnb | WIMPy: under System Status |
| 21:11.54 | navaismo | for sip show peers states: Unreachable, unknow, ok, unmonitored |
| 21:12.28 | WIMPy | What System status? Are you talking about some GUI thing _for_ Asterisk? |
| 21:12.30 | navaismo | for sip show registry: Registered, Failed and REquest sent |
| 21:13.04 | nmjnb | WIMPy: yes |
| 21:13.23 | WIMPy | Then you should ask in the appropriate channel. |
| 21:13.31 | p3nguin | We probably don't know what your GUI crap does and what the terms mean. |
| 21:13.44 | p3nguin | We might, but do not expect that we do. |
| 21:13.45 | navaismo | what gui? |
| 21:13.51 | nmjnb | digium |
| 21:13.57 | nmjnb | came with Asterisknow |
| 21:14.27 | p3nguin | I suppose #asterisk-gui is still just as dead as always. |
| 21:15.00 | nmjnb | I don't mind doing it the CLI way, but I'd like some guides to read for that |
| 21:15.08 | nmjnb | Asterisknow 1.7.1 |
| 21:15.08 | serafie | p3nguin: it's dead because it's better than it used to be. :P |
| 21:15.21 | nmjnb | or perhaps I'm better off installing 1.8 or 10? |
| 21:15.35 | navaismo | i can help you i never used |
| 21:15.46 | navaismo | can't* |
| 21:16.37 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 21:17.29 | nmjnb | anyone know of any good guides to learning asterisk the CLI way? |
| 21:18.21 | navaismo | the book of asterisk the future of telephony |
| 21:18.35 | WIMPy | ~book |
| 21:18.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 21:19.16 | WIMPy | There are others, but that's probably the most up-to-date one ATM. |
| 21:19.32 | nmjnb | great, the pdf version of that book that I have is for 1.4 |
| 21:21.19 | TobyRulez | still plenty of good reading in there |
| 21:24.28 | nmjnb | perhaps, but I've had some bad experiences of guides from another version being wrong so.. |
| 21:24.57 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 21:25.01 | sequencer | hi again :) |
| 21:27.25 | TobyRulez | nmjnb: no doubt. looks like the link above is for 1.8 so you should be good |
| 21:27.41 | sequencer | if a call comes in , how can i enable someone to pickup the call by dialing *8 ? |
| 21:27.53 | navaismo | edit features.conf |
| 21:28.03 | navaismo | but that its the default |
| 21:28.09 | sequencer | it is ? :s |
| 21:28.13 | navaismo | so you may check the callgroup |
| 21:28.53 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:29.00 | *** part/#asterisk TobyRulez (~TobyRulez@66-191-161-122.dhcp.gnvl.sc.charter.com) |
| 21:29.52 | sequencer | all are callgroup=1 |
| 21:29.55 | navaismo | type in the cli features show to see the actual dtmf combination |
| 21:30.05 | sequencer | but i want to change it to 8 |
| 21:30.10 | sequencer | #8 |
| 21:30.20 | sequencer | alright |
| 21:30.25 | navaismo | ok, edit the features.conf |
| 21:30.31 | navaismo | and set it |
| 21:30.33 | leroybuckingham | The change in SVN Revision 301505 breaks DTMF Queuing in asterisk 1.6 |
| 21:30.50 | navaismo | thenin the cli type features reload |
| 21:30.57 | leroybuckingham | It's still broken in 1.6.2.20, 1.6.2.16.2 is the last good release. |
| 21:31.28 | f2knight | pabelanger, thank you for your guidance.. but I am still lost. All I need to do is know how to access the agi model. |
| 21:33.55 | sequencer | woohoo! |
| 21:33.58 | sequencer | it worked! |
| 21:34.08 | *** join/#asterisk xnfinite (~xnfinite@225.139.22.95.dynamic.jazztel.es) |
| 21:34.15 | sequencer | now to the fun part ;) |
| 21:35.18 | sequencer | how can i set up an automated call recording ? |
| 21:37.02 | navaismo | maybe using mixmonitor() |
| 21:37.41 | navaismo | before the dial |
| 21:38.04 | navaismo | command, and use the option b, for record only bridged calls |
| 21:38.04 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:39.56 | *** join/#asterisk xnfinite (~xnfinite@225.139.22.95.dynamic.jazztel.es) |
| 21:40.09 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 21:40.12 | p3nguin | If you want all calls to a specific extension to be recorded, MixMonitor() is certainly the way to go. If you want to choose which calls are recorded, you can use automon, which is configurable in features.conf. |
| 21:40.45 | sequencer | i would like all of incoming & outoging calls to be recorded, regardless of the extension |
| 21:41.06 | sequencer | of course i will setup an announcement before the recording |
| 21:42.12 | p3nguin | I can't see how you'll record the calls without MixMonitor (or another recording app) being executing it in an extension. |
| 21:42.54 | p3nguin | Typically, decide what extension(s) will need to be recorded, and then put MixMonitor() on the extension. |
| 21:43.28 | sequencer | cant i just put MixMonitor() in incoming context ? |
| 21:43.36 | sequencer | and outgoing context |
| 21:43.42 | p3nguin | It has to be run IN AN EXTENSION. |
| 21:43.50 | p3nguin | Extensions go in contexts. |
| 21:43.53 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 21:44.00 | p3nguin | So the choice of context and extension will be yours. |
| 21:44.11 | sequencer | right |
| 21:44.41 | p3nguin | In the case of extension s: |
| 21:44.44 | p3nguin | exten => s,n,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.WAV,a); |
| 21:44.47 | p3nguin | exten => s,n,Playback(silence/1&this-call-may-be-monitored-or-recorded); |
| 21:45.27 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 21:46.00 | p3nguin | That's how I do it. |
| 21:46.31 | p3nguin | But you can't just throw it into a context; it has to be run in an extension. |
| 21:47.12 | WIMPy | It would solve endless problems if you could, however. |
| 21:47.36 | p3nguin | It might save some redundancy. |
| 21:48.40 | WIMPy | A pseudo extension that is alwaus executed before any real extension would make things possible that are currently hard to impossible. |
| 21:48.43 | sequencer | yeah, i agree |
| 21:49.04 | p3nguin | That's a great idea. You should file a feature request. |
| 21:49.38 | WIMPy | It's not really new. |
| 21:50.17 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 21:50.38 | treborsux | why does the soundpoint 560 and the 501 take 20 minutes to reboot and run sip app |
| 21:50.43 | WIMPy | I thought about requesting a BOB feature for Dial. |
| 21:51.07 | p3nguin | treborsux: Because you don't have the appropriate server supplying the files the phones want, I presume. |
| 21:51.19 | treborsux | gotchya |
| 21:51.32 | treborsux | i have no server supplying any files |
| 21:51.36 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 21:51.46 | p3nguin | When you have it, the phones will load the files rather quickly and be ready to go. |
| 21:51.49 | treborsux | just asterisk server looking for sip |
| 21:52.19 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:52.20 | treborsux | so just by themselves they are very slow i can still register sip though right just for little test for now |
| 21:52.45 | p3nguin | My Cisco phones, for example, when using SIP take about 15 seconds with a tftp server supplying the files or 5 minutes without the server giving files. |
| 21:53.03 | treborsux | gotchya |
| 21:53.36 | p3nguin | They are built to try and try and try before finally falling back to what they already know internally. |
| 21:53.53 | treborsux | I know how to make a ftp I just need to read and find out what files need to be in in for my 501s and my 560 |
| 21:54.15 | p3nguin | The Polycom admin guide can surely help with that. |
| 21:54.16 | treborsux | well that makes sense |
| 21:55.14 | p3nguin | I don't use Polycom, so I can't saw if it is possible or not... but maybe you can change a setting in the phone to not look for a server for so long. |
| 21:55.22 | p3nguin | can't say, rather. |
| 21:55.37 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 21:56.00 | IsUp | maybe a DNS resolution problem? |
| 21:56.16 | p3nguin | What would it be trying to resolve? |
| 21:56.17 | IsUp | it happens on Grandstream phones |
| 21:56.59 | p3nguin | asterisk's hostname, or something else? |
| 21:57.21 | IsUp | maybe, i dont know the case i am just telling my experience :p |
| 21:58.11 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:00.57 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 22:01.43 | treborsux | i cant get the 560 to register nor the 501 |
| 22:02.07 | treborsux | i guess i need to make ftp server |
| 22:02.07 | p3nguin | Are you configuring them from their web interfaces? |
| 22:02.15 | p3nguin | You don't really need to. |
| 22:02.15 | treborsux | yes from web |
| 22:02.27 | treborsux | actually i never got into the 501 |
| 22:02.45 | treborsux | i must not know the port because its address brings up nothing |
| 22:02.46 | p3nguin | There are a couple settings that you should do on the phone display and a few more on the web interface. |
| 22:02.52 | treborsux | the 560 does though |
| 22:04.24 | treborsux | what i dont get is what is the default secret to out in the asterisk setting |
| 22:04.27 | treborsux | 456? |
| 22:04.27 | p3nguin | Go into the phone menu on the phone display. |
| 22:04.47 | treborsux | i did on the 501 pointed it at the asterisk server |
| 22:05.09 | treborsux | line and sip |
| 22:05.34 | treborsux | on pbx the endpoint configurator sees them and i can say what extension |
| 22:05.35 | p3nguin | Go to admin settings, and reset to default and reset local config. |
| 22:05.39 | treborsux | they never register |
| 22:07.34 | p3nguin | After it reboots, then go back into Menu, Settings, Advanced. |
| 22:07.45 | treborsux | resetting local config now then default |
| 22:08.30 | p3nguin | I don't know if the order is important, but if it is, you're doing it backward. |
| 22:09.08 | treborsux | then ill do local again after |
| 22:11.04 | treborsux | asterisk specific firware for polycom phones would be kewl |
| 22:11.08 | treborsux | :> |
| 22:12.17 | p3nguin | After the reboot and you've gone back to the advanced menu, you'll need to go to the admin settings, and set the appropriate values in network configuration and sip configuration. |
| 22:12.39 | treborsux | i gues i dont know where |
| 22:13.01 | treborsux | i know where to put in address from this http://pbxinaflash.com/forum/showthread.php?t=3168 |
| 22:13.03 | p3nguin | I don't have a phone on me, or I'd poke through it with you. |
| 22:13.17 | treborsux | but what value in the phone is the secret value in asterisk??? |
| 22:13.25 | treborsux | the password 456? |
| 22:13.31 | treborsux | of the username Polycom |
| 22:13.36 | treborsux | I dont get it |
| 22:14.08 | p3nguin | When you press Menu, then 3, then 2... the admin password should be 456. |
| 22:14.25 | p3nguin | Then choose Admin Settings. |
| 22:14.29 | treborsux | right but in asterisk is that the value for secret |
| 22:14.45 | p3nguin | No, that has nothing to do with asterisk. |
| 22:14.57 | p3nguin | Just follow with me, doing the steps. |
| 22:15.13 | treborsux | ok |
| 22:15.34 | p3nguin | So you've made it to admin settings? |
| 22:17.00 | p3nguin | First, go into network configuration and fill any any pertinent values. Then go into sip configuration and put in asterisk's IP address in server address, and 5060 in server port. |
| 22:17.54 | treborsux | still waiting for either one to reboot |
| 22:18.11 | treborsux | processing configuration....... |
| 22:18.18 | p3nguin | Neither came back up yet? |
| 22:18.30 | treborsux | ok 560 is up |
| 22:19.00 | treborsux | net config is dhcp i have the mac assigned |
| 22:19.26 | treborsux | there is not a sip configuration listed |
| 22:19.29 | p3nguin | If there are any other values in there that you need to set, go ahead and do that before moving on to sip configuration. |
| 22:19.41 | p3nguin | uh, that's not a good sign. |
| 22:19.41 | treborsux | line coniguration |
| 22:19.52 | treborsux | call server configuration |
| 22:20.21 | p3nguin | Oh, the admin settings contains network as well as sip, register, line, display, etc, right? |
| 22:21.13 | treborsux | no |
| 22:21.17 | treborsux | on 560 |
| 22:22.06 | treborsux | admin is 1 network config 2 line configuration 3 call server configuration |
| 22:22.08 | p3nguin | I guess call server configuration is the same thing. |
| 22:23.58 | treborsux | transport naptr? |
| 22:24.24 | p3nguin | You'll need to fill in the line configuration with what each line key on the side of the phone is supposed to do. |
| 22:24.27 | treborsux | i did call sever |
| 22:24.50 | p3nguin | If you find one that allows you to set Register to yes or no, set it to no. |
| 22:25.30 | sequencer | what would be the default directory to place a recording to be played by Play() ? |
| 22:26.13 | p3nguin | Playback()? /var/lib/asterisk/sounds or /var/lib/asterisk/sounds/en probably. |
| 22:26.22 | sequencer | Great! |
| 22:26.52 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 22:27.08 | p3nguin | Just look for the directory that has all the sound files in it. ;) |
| 22:27.16 | sequencer | ;) |
| 22:27.34 | treborsux | why no to register? |
| 22:27.57 | treborsux | i dared change something so restarting 10 minutes |
| 22:28.20 | treborsux | maybe i should statrt witth ftp so this is quicker |
| 22:28.45 | p3nguin | After you set the other stuff in the phone menu, you'll go to the web interface and turn on which line should register to asterisk. |
| 22:29.50 | p3nguin | You'll also configure the voice mail stuff in the web interface. |
| 22:30.03 | treborsux | cant keep making a setting and waiting 10 minuteses |
| 22:30.14 | treborsux | when i make ftp is there separate folder for each phone |
| 22:30.19 | treborsux | or one for each kind |
| 22:30.21 | p3nguin | Why does it restart after you fill in settings? |
| 22:30.33 | treborsux | anytime i save a setting |
| 22:30.49 | p3nguin | You can't fill in the settings before hitting save? |
| 22:31.04 | p3nguin | One setting, move to next setting, etc, then save? |
| 22:31.07 | *** join/#asterisk trelane` (trelane@funtoo/staff/trelane) |
| 22:31.28 | treborsux | when i go back if i dont save it wipes out |
| 22:31.32 | p3nguin | ick |
| 22:31.43 | p3nguin | I'm glad I don't deal with that stuff. |
| 22:31.53 | treborsux | I think I need to set up ftp first to boot these things right |
| 22:31.54 | trelane` | I'm attempting to send data to a peer which does not allow registering. |
| 22:31.54 | trelane` | From: "Unknown" <sip:Unknown@208.72.22.3>;tag=as56c07ed9 |
| 22:31.54 | trelane` | <PROTECTED> |
| 22:32.25 | p3nguin | If you don't want to continue doing it in the phone, ftp is probably the best option. |
| 22:32.26 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 22:33.01 | p3nguin | For just one or two phones, I'd continue in the phones then do the two settings in the web interface, then use my phones. |
| 22:33.09 | sequencer | is this correct: exten => s,1,Playback("cal_mon.wav") |
| 22:33.11 | sequencer | ? |
| 22:33.15 | p3nguin | no |
| 22:33.23 | p3nguin | Playback(cal_mon) |
| 22:33.29 | p3nguin | If the file name is cal_mon.wav |
| 22:33.32 | sequencer | without .wav |
| 22:33.33 | sequencer | oh |
| 22:33.37 | trelane` | no quotes either |
| 22:33.54 | treborsux | what do i set transport for sip setting to |
| 22:33.58 | p3nguin | I'd let Allison do her thing. |
| 22:34.20 | p3nguin | this-call-may-be-monitored-or-recorded is an Allison file. |
| 22:34.37 | treborsux | naptr in sip setting? |
| 22:34.45 | trelane` | because-we-are-paranoid is good too! :) |
| 22:34.57 | treborsux | on 501 it finally reset after and it has sip setting |
| 22:35.04 | p3nguin | Do you have other choices besides naptr? |
| 22:35.06 | treborsux | what transport in the sip setting |
| 22:35.09 | treborsux | yes |
| 22:35.24 | treborsux | tcp udp tls |
| 22:35.30 | treborsux | tcp preferred |
| 22:35.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:35.51 | p3nguin | I'd use UDP, since that's what asterisk uses. |
| 22:36.08 | p3nguin | If those were the only choices, that is. |
| 22:36.17 | treborsux | outbound proxy? |
| 22:36.40 | treborsux | is that the asterisk server? |
| 22:37.24 | p3nguin | If you already put in the asterisk IP address in another field, I'd leave outbound proxy blank. |
| 22:38.39 | treborsux | line 1 |
| 22:38.59 | treborsux | name got it what i ma i made ext in asterisk |
| 22:39.10 | treborsux | is address under line one the ext number? |
| 22:39.35 | p3nguin | I believe it is the phone's name as you configured it in asterisk sip.conf. |
| 22:39.59 | p3nguin | probably the MAC address of the phone. |
| 22:40.40 | sequencer | whats the difference between exten= and exten => ? |
| 22:40.58 | p3nguin | The former is wrong, the latter is right. |
| 22:41.28 | treborsux | using freepbx |
| 22:41.31 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 22:41.39 | treborsux | so ididnt make any thing in a file |
| 22:41.44 | p3nguin | I don't support FreePBX. |
| 22:41.55 | p3nguin | Decide what the phone's name is. |
| 22:41.58 | treborsux | so i guess that is the file freepbx edits when i add extension |
| 22:42.04 | p3nguin | maybe |
| 22:43.34 | carrar | guessing is good |
| 22:43.41 | carrar | that will surely fix it |
| 22:43.45 | treborsux | lol |
| 22:43.46 | sequencer | exten => s,1,Playback(cal_mon) Still cant get it work.. loosing my mind here |
| 22:43.47 | p3nguin | That's what I'd be doing if I said yes. |
| 22:43.54 | trelane` | I'm attempting to send data to a peer which does not allow registering. I need to replace From: "Unknown" <sip:Unknown@ with a real phone number. I've tried callerid and FromUser. What am I missing? |
| 22:43.56 | treborsux | not broke just new and setting |
| 22:43.58 | treborsux | up |
| 22:44.30 | treborsux | the 2 soft phones work fine one sip and one iax |
| 22:44.42 | treborsux | trying to get a 501 to register |
| 22:45.24 | p3nguin | Whatever auth name you used on the softphone is the auth name you'd use for the hardphone. If you're using the same account, that is. |
| 22:45.43 | p3nguin | sequencer: Do calls go to extension s? |
| 22:46.00 | sequencer | p3nguin good catch ! :s |
| 22:46.01 | p3nguin | sequencer: If calls are going to a different extension, don't expect extension s to do you any good. |
| 22:46.07 | sequencer | i thought is is start |
| 22:46.22 | p3nguin | it's literally 's' |
| 22:46.42 | p3nguin | Sometimes calls go to extension s. Most of the time they don't. |
| 22:46.58 | treborsux | 501 reboot |
| 22:47.00 | sequencer | in my case they always dont |
| 22:47.06 | treborsux | all settings are in |
| 22:47.16 | treborsux | how do i get it to register? |
| 22:47.23 | p3nguin | For SIP, they rarely go to s; if they do, you've misconfigured something. |
| 22:47.53 | sequencer | here's a new one |
| 22:47.54 | sequencer | format_wav.c:153 check_header: Can only handle 16bits per sample: 1 |
| 22:48.17 | treborsux | endpoint configurator sees the phone |
| 22:48.25 | treborsux | says configured without incident |
| 22:48.34 | treborsux | but phone never registers |
| 22:48.35 | p3nguin | Once you have the settings in the phone, including auth user id and auth password, and it has restarted, then go to the web interface and turn on which lines you want to register to asterisk. |
| 22:48.55 | navaismo | sequencer the wav needs to be mono 16pcm 8Khz |
| 22:49.05 | sequencer | lets do it.. |
| 22:49.23 | p3nguin | Oh, trying to use the wrong format of wav. |
| 22:49.31 | sequencer | treborsux that would be in the "extensions" menu |
| 22:49.40 | treborsux | dont know how to get into web interface of 501 |
| 22:50.11 | sequencer | treborsux i have 560's and they go by http://ipaddress |
| 22:50.22 | sequencer | ipaddress is the ip for you polycom |
| 22:50.25 | treborsux | 560s do but 501s dont |
| 22:50.39 | sequencer | oh, i have 560s and 610s |
| 22:51.22 | p3nguin | The 501 is supposed to. Did you reset to defaults and reset local config? |
| 22:51.57 | p3nguin | If that fails, I'd probably go for the big daddy reset. |
| 22:52.29 | p3nguin | (the "format file system" option) |
| 22:52.30 | treborsux | yes i did |
| 22:52.43 | treborsux | but there is no web interface |
| 22:52.54 | p3nguin | Sounds broken to me. |
| 22:53.05 | treborsux | ok ill hook up diffent one |
| 22:54.06 | treborsux | I have a 560 |
| 22:54.09 | treborsux | too |
| 22:54.15 | treborsux | cant get that to register either |
| 22:54.49 | p3nguin | Once you filled in the values appropriately and used the web interface to register the line(s), it should register. |
| 22:54.57 | treborsux | under line configuator it says siplay name |
| 22:55.13 | treborsux | then it says address is that the extension it is? |
| 22:55.14 | p3nguin | That's the name to put on the phone, such as your name. |
| 22:55.39 | p3nguin | Address should be the name of the phone, probably the MAC address, as configured in asterisk. |
| 22:55.43 | treborsux | so display name and address are the same? |
| 22:55.47 | p3nguin | no |
| 22:55.52 | p3nguin | Display name is YOUR NAME |
| 22:56.04 | p3nguin | Address is the phone's name as configured in asterisk. |
| 22:56.30 | p3nguin | Auth user ID is also the phone's name as configured in asterisk. |
| 22:56.42 | treborsux | so its the phones name in extensions? |
| 22:56.51 | p3nguin | Not really, no. |
| 22:57.00 | p3nguin | Auth password is the phone's password as configured in asterisk. |
| 22:57.13 | treborsux | secret? |
| 22:57.33 | p3nguin | The Auth password in the phone is the secret as configured in asterisk. |
| 22:57.48 | treborsux | So where in the phone do i tell it what extension it is? |
| 22:57.57 | p3nguin | The phone doesn't care what extension is used to reach it. |
| 22:58.01 | p3nguin | That's a human concept. |
| 22:58.06 | *** join/#asterisk Crowb4r (u1012@gateway/web/irccloud.com/x-zaedmysphuqilgsk) |
| 22:58.16 | p3nguin | The phone does care what its name is, though. |
| 22:58.50 | treborsux | Where in freepbx am i putting the phones name |
| 22:59.15 | p3nguin | Let me give you an example. My phone's name is 0000AAAABBBB, which is configured in sip.conf. |
| 22:59.27 | p3nguin | The extension used to call my phone from other phones is 762. |
| 22:59.42 | p3nguin | The phone doesn't care what extension 762 does. It doesn't need to know it. |
| 22:59.51 | p3nguin | I don't support FreePBX. |
| 23:00.03 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 23:00.11 | sequencer | treborsux that would be in the extensions menu |
| 23:00.17 | sequencer | to your left |
| 23:00.45 | sequencer | click on the extension an type the Display Name |
| 23:01.05 | sequencer | and in the SIP Alias |
| 23:02.12 | p3nguin | Just out of curiosity, what is the purpose of SIP alias? |
| 23:02.18 | p3nguin | What's it do? |
| 23:02.30 | p3nguin | I'll try to translate it to what it does in asterisk. |
| 23:02.43 | sequencer | it would provide a different CID number/name |
| 23:02.53 | sequencer | when making local calls ( calls to local extensions ) |
| 23:03.04 | sequencer | that are not using DID trunks |
| 23:03.16 | p3nguin | Weird. I would have called that something completely different. Maybe something like CallerID or something. |
| 23:03.45 | sequencer | caller ID is what appears on the other party's phone when making an outside call |
| 23:03.46 | p3nguin | I guess that's why we don't support FreePBX or its configuration here. |
| 23:04.08 | treborsux | waiting for reboot |
| 23:04.21 | sequencer | but for local extensions, you can use SIP alias to show your actual extension number |
| 23:04.31 | sequencer | instead of your corporate DID |
| 23:04.37 | sequencer | for instance.. |
| 23:04.43 | p3nguin | I use callerid for that. |
| 23:04.52 | treborsux | so use ext number for sip alias? |
| 23:04.54 | p3nguin | callerid=Rob <762> |
| 23:05.08 | treborsux | so under address on phone under line use ext number? |
| 23:05.11 | sequencer | but wehn you call to outside the box ? |
| 23:05.26 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:05.31 | treborsux | I am so lost |
| 23:05.38 | p3nguin | For external calling, I use a variable. setvar=externalCID=3149691077 |
| 23:05.47 | sequencer | oh |
| 23:06.03 | sequencer | so thats how you do it, its almost same concept with fpbx |
| 23:06.05 | p3nguin | Then the extension that calls out sets the callerid number to the value of that variable. |
| 23:06.21 | sequencer | treborsux whats that you need ? |
| 23:07.19 | treborsux | sequencer are you in #freepbx also |
| 23:07.31 | p3nguin | By using the callerid setting in my peer, I don't have to fiddle around with the caller id for calls to other phones on the system. |
| 23:07.41 | sequencer | treborsuxnope am just here |
| 23:08.00 | p3nguin | But it does require that every phone has an externalCID value set. |
| 23:08.03 | treborsux | i need to start with freepbx first |
| 23:08.12 | carrar | use a db |
| 23:08.23 | sequencer | sure |
| 23:08.26 | treborsux | i think i dont have that right in the first place to see this pohone |
| 23:08.28 | sequencer | what you need to start with ? |
| 23:08.50 | sequencer | did add an extension in the extensions ? |
| 23:08.51 | p3nguin | What do you want to put in the DB? |
| 23:08.52 | carrar | simple agi to to see if the extension has a external DID mapped to it |
| 23:09.01 | sequencer | you need to define the extension number and secret |
| 23:09.04 | carrar | if not, use a 'default' |
| 23:09.10 | p3nguin | I just use basic dialplan. ExecIf() works just fine. |
| 23:09.11 | sequencer | p3nguin alot of stuff |
| 23:09.24 | p3nguin | Don't feed the alot. |
| 23:09.38 | treborsux | ok i put Dorothy in as display name ext is 356 and sip alias is 356 |
| 23:09.49 | treborsux | then secret is 456 |
| 23:09.57 | sequencer | p3nguin mixMonitor isnt working for me :s |
| 23:09.57 | sequencer | file.c:750 ast_readaudio_callback: Failed to write frame |
| 23:10.08 | sequencer | treborsux ok good |
| 23:10.10 | p3nguin | Don't use the admin password for your secret. |
| 23:10.12 | sequencer | now click save |
| 23:10.42 | p3nguin | Hang on a minute, I have to go make some calls on treborsux's box. |
| 23:10.45 | sequencer | and on the top you'll see reload orange button |
| 23:11.19 | sequencer | p3nguin well.. |
| 23:11.29 | treborsux | lol this is just a test |
| 23:11.43 | sequencer | treborsux you can allow/deny certain IP's or subnets also in the same page ;) |
| 23:11.44 | treborsux | i wont leave it this way it is on closed network right now |
| 23:12.08 | treborsux | i saved the ext |
| 23:12.25 | sequencer | ok good, noe reload |
| 23:12.35 | sequencer | p3nguin have you seen this before :s file.c:750 ast_readaudio_callback: Failed to write frame |
| 23:12.43 | p3nguin | probably |
| 23:12.53 | treborsux | i did reload |
| 23:13.07 | sequencer | did it register treborsux ? |
| 23:13.19 | treborsux | what? |
| 23:13.28 | treborsux | no idea what to set the phone to |
| 23:13.29 | sequencer | on top go to status |
| 23:13.35 | sequencer | oh |
| 23:13.41 | p3nguin | I'd guess that you don't have permission to write to the monitor directory. Make sure asterisk is running as user asterisk group asterisk, and that /var/spool/asterisk/monitor is owned appropriately. |
| 23:13.56 | treborsux | its a 560 |
| 23:14.25 | sequencer | why would monitor be owned to root :s |
| 23:14.54 | treborsux | line configuratipon |
| 23:14.59 | treborsux | line1 |
| 23:15.10 | treborsux | display name Dorothy |
| 23:15.11 | sequencer | server is you fpbx box |
| 23:15.17 | treborsux | is the address the sip alias? |
| 23:15.21 | sequencer | username is your extension |
| 23:15.23 | sequencer | number |
| 23:15.30 | sequencer | password is your secret |
| 23:15.46 | p3nguin | I often have to chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk |
| 23:15.57 | sequencer | p3nguin still same issue even after chowning |
| 23:16.17 | p3nguin | Without more details, I don't know what's wrong. |
| 23:16.32 | sequencer | one sec.. |
| 23:16.45 | treborsux | what is address? |
| 23:16.49 | treborsux | is that the sip alias? |
| 23:16.57 | sequencer | address is the server ip address |
| 23:17.02 | sequencer | of your fpbx |
| 23:17.12 | treborsux | but down a bit it says server |
| 23:17.18 | p3nguin | Address should be the phone's name. |
| 23:17.27 | p3nguin | (as I said three times before) |
| 23:17.32 | treborsux | phones display name or sip alias?? |
| 23:17.46 | p3nguin | The peer name that you use to Dial() the damn phone, of course. |
| 23:18.14 | sequencer | p3nguin i messed sth up in the dialplan :s |
| 23:18.26 | p3nguin | What was smith doing in the dial plan to begin with? |
| 23:18.43 | treborsux | peer name??? |
| 23:18.44 | treborsux | what |
| 23:18.49 | treborsux | I dont understand |
| 23:18.57 | sequencer | peer name is extension number |
| 23:18.58 | p3nguin | You're in #asterisk and don't know what a peer name is? |
| 23:19.07 | p3nguin | Have you read the book yet? |
| 23:19.08 | p3nguin | ~book |
| 23:19.09 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 23:19.19 | sequencer | p3nguin |
| 23:19.26 | sequencer | many people wont read the book |
| 23:19.31 | sequencer | thats whywe use GUi |
| 23:19.37 | p3nguin | THIS IS WHY WE DON'T SUPPORT THAT FUCKING FREEPBX SHIT. |
| 23:19.47 | sequencer | i understand that |
| 23:19.51 | p3nguin | Learn some stuff, then we can help you. |
| 23:19.54 | treborsux | I get that |
| 23:20.03 | sequencer | but also you need to look at other people's POV |
| 23:20.05 | navaismo | treborsux http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_sip_3_0_2_english_rev_A3.pdf |
| 23:20.14 | treborsux | ill ask in Freepbx if it is a problem I appriaciate the help and thank you either way |
| 23:20.32 | p3nguin | I do? I don't get paid enough to look at it from other people's points of view. |
| 23:20.53 | sequencer | nobody's getting paid enough ;) |
| 23:21.04 | sequencer | but everyone appreciates the help of others :) |
| 23:21.19 | p3nguin | I help where I can. I help within the scope of this channel. |
| 23:21.39 | sequencer | well.. i dont know much of bare-asterisk |
| 23:21.45 | p3nguin | I even deviate outside the scope where I am capable. |
| 23:21.48 | sequencer | but i operated fpbx for 8 months |
| 23:22.02 | sequencer | so i know some terms.. thats all |
| 23:22.28 | treborsux | i think i am close |
| 23:22.34 | treborsux | phone is rebooting |
| 23:22.34 | sequencer | and i know how it feels , just like how i do now with a non-working diguim gui and ( do it yourself asterisk ) |
| 23:22.41 | sequencer | treborsux good :) |
| 23:22.49 | sequencer | you should see it in system status |
| 23:23.12 | sequencer | in IP Phones online it shuold say 1 |
| 23:23.18 | navaismo | treborsux you need to read the Admin guide of polycom |
| 23:23.43 | treborsux | Will it relate terms to asterisk or freepbx? |
| 23:24.00 | p3nguin | My guess would be neither. |
| 23:24.28 | sequencer | yep.. just generic SIP - IP Phone |
| 23:24.48 | sequencer | treborsux |
| 23:25.04 | sequencer | i usually use x-lite and eyebeam to test pbx |
| 23:25.18 | sequencer | saes you alot of headache |
| 23:25.21 | sequencer | saves* |
| 23:25.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fuyzhgskxonophso) |
| 23:25.41 | treborsux | i ahve zoiper working i can recieve and make calls with the fxo card i setup with 8 lines |
| 23:25.57 | treborsux | ill really send you through the roof when i tell you it is an elastix system |
| 23:26.09 | p3nguin | I don't get it. You'll make a correction to "save" but not to "alot." |
| 23:26.26 | p3nguin | err, "saes" |
| 23:26.49 | p3nguin | People are so strange sometimes. |
| 23:26.49 | treborsux | what is x-lite and eybeam |
| 23:26.55 | sequencer | alot is understandable |
| 23:27.01 | sequencer | saes could mean says |
| 23:27.02 | p3nguin | hmm? |
| 23:27.07 | p3nguin | allot? |
| 23:27.11 | sequencer | these are soft phones |
| 23:27.13 | p3nguin | To make an allocation? |
| 23:27.25 | navaismo | jeje i think some is very sensible today |
| 23:27.31 | sequencer | slot = a lot |
| 23:27.31 | navaismo | some one* |
| 23:27.35 | sequencer | err, alot |
| 23:28.10 | sequencer | treborsux you can downalod x-lite from counterpath |
| 23:28.20 | p3nguin | Now why would I have used IF() inside of ExecIf()? I must have been on the crack when I wrote that. |
| 23:28.50 | sequencer | is on crack always :D |
| 23:28.56 | treborsux | oso it is like zoiper |
| 23:29.04 | sequencer | i just messed up my dial plan lol |
| 23:29.33 | p3nguin | I have no idea how I arrived at this: ExecIf($[${IF($["${externalCID}" != ""]?1)}]...) |
| 23:29.43 | p3nguin | I guess it was a change to something else. |
| 23:29.56 | sequencer | p3nguin cant you ,... undo ? |
| 23:30.07 | navaismo | im so bored |
| 23:30.08 | p3nguin | It should have been ExecIf($["${externalCID}" != ""]...) |
| 23:30.26 | p3nguin | I'm going to fix it. I just found it in dial plan. |
| 23:30.35 | p3nguin | Trying to understand how I arrived at it. |
| 23:30.43 | p3nguin | I blame the crack. |
| 23:30.43 | sequencer | navaismo |
| 23:30.57 | navaismo | ?? |
| 23:31.03 | sequencer | i dont know how i arrived to this also .. :s |
| 23:31.24 | sequencer | <PROTECTED> |
| 23:31.30 | sequencer | Auto fallthrough, channel 'SIP/trunk_1-00000050' status is 'UNKNOWN' |
| 23:31.33 | sequencer | :s |
| 23:31.55 | navaismo | i need to go, the way to my home take at least 2:30hrs |
| 23:31.55 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 23:31.56 | p3nguin | I guess you don't have another priority after that one. |
| 23:31.58 | treborsux | yay it registered!!!!!! |
| 23:32.05 | sequencer | treborsux congrats! |
| 23:32.18 | navaismo | quit |
| 23:32.58 | sequencer | am still having this file.c:750 ast_readaudio_callback: Failed to write frame |
| 23:33.07 | sequencer | after all of the chowns i did :s |
| 23:33.20 | p3nguin | Are the permissions set correctly? |
| 23:33.32 | sequencer | they should.. ill check one more time |
| 23:34.24 | treborsux | called wife and it works |
| 23:34.54 | sequencer | treborsux if its wife it has to work :| |
| 23:35.02 | p3nguin | http://pastebin.com/U7uP1LZ1 |
| 23:37.37 | sequencer | my namei doesnt have -l :s |
| 23:37.44 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 23:37.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:37.48 | sequencer | only m and x |
| 23:37.53 | p3nguin | Use whatever it has. -mo maybe |
| 23:38.06 | p3nguin | use m at least |
| 23:39.54 | sequencer | http://pastebin.com/DJiGSB2b |
| 23:40.41 | treborsux | thanks guys |
| 23:40.58 | treborsux | tomorow i need to make ftp and figure that out so these things boot quick |
| 23:40.59 | sequencer | treborsux all set with you ? |
| 23:41.09 | treborsux | then on to incoming |
| 23:41.19 | sequencer | those are simple ;) |
| 23:41.23 | treborsux | pyched i can make a call |
| 23:41.32 | treborsux | going home now |
| 23:41.37 | sequencer | alrighty |
| 23:41.45 | p3nguin | The reason you get the Failed to write frame message is because you're trying to play a file to a channel which cannot exist in the h extension. |
| 23:41.45 | treborsux | thanks guys very much |
| 23:41.52 | sequencer | you welcome :) |
| 23:42.17 | p3nguin | h is the hangup extension. You can't playback nor record in it. |
| 23:42.17 | sequencer | its not supposed to be in h |
| 23:42.33 | p3nguin | Good ole GUI fucked up something else, huh? |
| 23:42.35 | sequencer | after i added the mixmonitor my call went into h |
| 23:42.46 | p3nguin | Show me the dialplan. |
| 23:42.46 | sequencer | am not using the gui anyways |
| 23:42.58 | sequencer | thats a party now lol |
| 23:43.02 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 23:43.28 | p3nguin | I guess your GUI set autofallthrough to yes. |
| 23:43.29 | p3nguin | I would have set it to no. |
| 23:43.53 | sequencer | i thought that was a nice way to say "an error just happened " |
| 23:44.06 | *** join/#asterisk kaushal (~kaushal@14.97.57.116) |
| 23:44.08 | kaushal | Hi |
| 23:44.25 | sequencer | how do you want me to give you the dial paln, do you want it all or just the needed parts ? |
| 23:44.40 | p3nguin | I just want to see the extension where you're running mixmonitor. |
| 23:45.09 | kaushal | can someone please help me understand what is VICIDIAL and astguiclient http://astguiclient.sourceforge.net/faq.html ? |
| 23:45.35 | sequencer | u wanna see the mess ? |
| 23:45.43 | sequencer | http://pastebin.com/SDvD9U50 |
| 23:45.54 | sequencer | this is incoming |
| 23:46.02 | p3nguin | There's yer problem. |
| 23:46.41 | sequencer | yep me know |
| 23:46.46 | p3nguin | I thought I told you to never use _. as the pattern. |
| 23:46.55 | sequencer | :s |
| 23:46.59 | p3nguin | That's why it's running in the h extension. |
| 23:47.01 | sequencer | i didnt know about this one |
| 23:47.07 | p3nguin | I'll check the log. |
| 23:48.43 | sequencer | i removed the _. |
| 23:48.49 | sequencer | changed to my DID |
| 23:48.53 | sequencer | but still same issue |
| 23:49.44 | p3nguin | Okay, it wasn't you. I guess you wouldn't be surprised at how many people insist on using _. thinking it's a good pattern. |
| 23:50.18 | sequencer | of course i wouldnt |
| 23:50.19 | p3nguin | _. is not a good pattern because it matches the standard extensions of s, h, i, and t. |
| 23:50.21 | sequencer | am a programmer |
| 23:50.39 | sequencer | it matches everything and thats what am looking for ;) |
| 23:50.49 | p3nguin | No you aren't. |
| 23:50.54 | p3nguin | You think you are, but you aren't. |
| 23:51.02 | sequencer | lol wanna bet ? |
| 23:51.09 | p3nguin | I've already explained why. |
| 23:51.30 | p3nguin | So what is the new problem now that you have fixed the extension? |
| 23:51.46 | sequencer | the call is going correctly |
| 23:51.49 | p3nguin | It's not the failed to write frame thing anymore, because that was in h. |
| 23:51.51 | sequencer | let me check |
| 23:53.56 | sequencer | when i removed the mixmonitor the call went fine |
| 23:54.13 | p3nguin | I need to see some more logging if you want me to guess. |
| 23:54.28 | sequencer | yeah sure |
| 23:55.34 | kaushal | checking in again for the query ? |
| 23:55.35 | sequencer | http://pastebin.com/NmpzgfD7 |
| 23:55.50 | kaushal | can someone please help me understand what is VICIDIAL and astguiclient http://astguiclient.sourceforge.net/faq.html ? |
| 23:55.52 | sequencer | still not really going fine , i dont recieve the call on my phone |
| 23:56.29 | p3nguin | It's still doing stupid shit in h. |
| 23:56.37 | p3nguin | That tells me that you still have _. as a pattern. |
| 23:57.42 | sequencer | maybe somewhere else.. |
| 23:57.45 | sequencer | let me see |
| 23:58.05 | p3nguin | Unless you really have an h with a Goto in it. |
| 23:58.35 | p3nguin | But to me it looks like h is still trying to process a call. |
| 23:58.38 | p3nguin | And that's bad. |