| 00:05.07 | *** join/#asterisk s7r_g (~s7r@openvpn/user/s7r) |
| 00:15.43 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 00:22.29 | gartral | ok.. now I have my network working well, but I have a user that can't connect, they're trying to auth from an external, public ip |
| 00:23.08 | [TK]D-Fender | gartral: As before, show the debug |
| 00:26.03 | gartral | [TK]D-Fender: all it says is Registration from '"[scrubbed]" <sip:[scrubbed]@[scrubbed]>' failed for '[scrubbed]:5060' - Wrong password |
| 00:26.19 | [TK]D-Fender | gartral: pretty much means what it says.... |
| 00:26.21 | gartral | [TK]D-Fender: and we know, for a fact, that password is correct. |
| 00:26.33 | [TK]D-Fender | gartral: and I can't see that the user it is identifying as is right |
| 00:29.40 | gartral | [TK]D-Fender: it's set up like every other entry in sip.conf and extensions.conf.. and it doesn't work from my end, either |
| 00:29.58 | [TK]D-Fender | gartral: Sho the debug and configs masking only passwords |
| 00:30.44 | gartral | [TK]D-Fender: I'll change usernames and hostname/IPs too.. if that's ok, I'll keep it consistent |
| 00:30.48 | [TK]D-Fender | gartral: no |
| 00:31.52 | gartral | [TK]D-Fender: alright, but I'll send the link in a pm.. is that ok? |
| 00:32.02 | [TK]D-Fender | that's fine |
| 00:32.42 | gartral | i'm using people's names here.. they wouldn't be very happy with me if that became public knowledge >.> |
| 00:33.27 | *** join/#asterisk _zerick_ (~Erick@190.118.30.151) |
| 00:38.14 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 00:52.19 | *** join/#asterisk suneye (~atcmmi@119.139.133.191) |
| 01:01.15 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 01:08.44 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:08.44 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 01:18.40 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 02:14.45 | *** join/#asterisk bpietro (~pietro@82.51.236.132) |
| 02:19.44 | bpietro | hi, which is the right directive for random moh? 'random = yes' or 'sort = random' ? Found In vary doc, for me both don't work (ver. 11.5.1) |
| 02:20.50 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 02:25.42 | Penguin | It depends on version. If both don't work for you, use the one that does work. |
| 02:27.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:27.48 | ChannelZ | sort=random |
| 02:29.31 | bpietro | tnx |
| 02:42.01 | bpietro | after hard restart of server no result, still plays only first file (first of natural file order, i.e. first uploaded) oh, 2 hours of work, all WTC I resampled and no result |
| 02:43.36 | ChannelZ | how many files do you have |
| 02:43.59 | Penguin | Testing it once is not a very good sample to check how random it is. |
| 02:45.38 | ChannelZ | moh show files |
| 02:45.41 | bpietro | 24 files, calling from 2 computers (two test softphones). No other than first file |
| 02:46.00 | ChannelZ | make sure it's seeing them all.. that it has permission to read them all.. |
| 02:46.28 | bpietro | moh show files shows all 24 |
| 02:47.08 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 02:47.24 | bpietro | all files 'chown asterisk:asterisk; chmod u+r' |
| 02:47.35 | redotis | Any reason confbridge list confname doesn't show the callerid that's in sip.conf? |
| 02:50.23 | ChannelZ | It comes from the device usually |
| 02:51.44 | redotis | Did meetme use the callerid in sip.conf? |
| 02:52.37 | ChannelZ | It's not much to do about one or the other. The caller id for the channel is the caller id for the channel. |
| 02:53.28 | redotis | Hmm...ok I was using meetme with freepbx and it would list the callerid that was put in it's config. I'm trying to do the same with asterisk now. How can I get that behavior from asterisk? |
| 02:53.35 | Penguin | its config |
| 02:53.53 | redotis | I'd like the callerID that is in sip.conf to show up as the callerid in confbridge |
| 02:53.59 | redotis | confbridge list conference |
| 02:54.29 | redotis | Any idea how to accomplish that? |
| 02:58.42 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 02:58.51 | ChannelZ | What is it displaying now? |
| 02:59.24 | ChannelZ | Probably something in your dialplan is overriding it or you've got the syntax wrong or something. |
| 03:01.34 | gartral | [TK]D-Fender: thank you, and I apologize. I think I can solve the problem now that I'm awear of the root issue |
| 03:06.36 | redotis | It just shows the device name |
| 03:07.19 | redotis | what I have inside of [] in sip.conf |
| 03:15.31 | redotis | I think confbridge must only list the number portion of the caller id |
| 03:19.04 | redotis | Anyone here have experience with confbridge list? Does the callerid only return the number portion? |
| 03:24.10 | ChannelZ | yes |
| 03:24.26 | redotis | thanks channelz |
| 03:24.30 | redotis | yeah i just checked |
| 03:24.34 | redotis | i set them manually |
| 03:24.38 | redotis | for that extension |
| 03:24.45 | redotis | and it just returned the number |
| 03:24.47 | redotis | dad blasted |
| 03:25.25 | gartral | how do I unbork an asterisk system after some numb-nut went and installed Digiums' dead gui when I specifically told them NOT to? |
| 03:26.11 | ChannelZ | Dunno what all it does... presumably it alters your configs at least |
| 03:26.58 | gartral | ChannelZ: yea, I know that, and I'm going to murder the imbacile who just made me and [TK]D-Fender waste an hour running in circles |
| 03:28.08 | redotis | gartral http://extundelete.sourceforge.net/ |
| 03:28.09 | ChannelZ | the point being hopefully you had a backup of configs prior to this adventure. If not.... |
| 03:28.49 | gartral | ChannelZ: no, I was setting all this up today, I was going to back up everything after I had it all working |
| 03:29.04 | redotis | gartral http://extundelete.sourceforge.net/ |
| 03:29.06 | redotis | lol |
| 03:29.30 | gartral | redotis: I fail to see how that helps, nothing was deleted and these changed are more than several hours old |
| 03:29.47 | ChannelZ | I guess now you'll get to see how much you learned setting it up the first time! |
| 03:30.40 | gartral | ChannelZ: that doesn't help, I've go through and replaced all the configs and guess what, asterisk is acting like they aren't it's configs |
| 03:30.48 | redotis | ? |
| 03:30.55 | redotis | on overwrite is a delete is it not |
| 03:31.03 | ChannelZ | I've no idea what that thing does on install, if it's nice enough to make _bkup or similar copies of configs, or if it just switches over to realtime and maybe left your static config alone... |
| 03:31.30 | redotis | maybe i'm wrong |
| 03:31.32 | gartral | ChannelZ: I haven't thought about that, how do I detect and fix that? |
| 03:32.24 | ChannelZ | do you have extconfig.conf? |
| 03:32.34 | gartral | redotis: you're right, and EXTUNDelete takes from the journal and restores it to the working area of the disk, this is useless if the delete/corruption happened a while ago |
| 03:32.41 | gartral | ChannelZ: yes I do! |
| 03:32.51 | ChannelZ | were you using realtime config before? |
| 03:33.22 | gartral | ChannelZ: *NO* |
| 03:33.31 | ChannelZ | Then rename/remove that and see what happens |
| 03:35.20 | gartral | ChannelZ: removed and reloaded, and asterisk upon running "sip show peers" returns a big fat mind-f***ed goose egg |
| 03:36.07 | ChannelZ | quit and restart asterisk |
| 03:36.16 | ChannelZ | I'm not sure reload can 'turn off' realtime |
| 03:37.19 | ChannelZ | and if it's still not seeing your configs, check it didn't re-write asterisk.conf and point things in some other place(s) |
| 03:37.58 | redotis | dang...that sucks man. Sorry that doesn't help. I'm not too familiar with file recovery on linux but at the bottom of that page for extundelete they had other options with different programs. |
| 03:38.26 | ChannelZ | redotis: his problem is less that something got deleted but that a whole ton of shit got changed and we don't know what |
| 03:38.28 | *** join/#asterisk soy_el_pulpo (~soy_el_pu@181.66.23.150) |
| 03:38.51 | redotis | I get you but my point is if you could restore the original versions...problem solved. |
| 03:39.47 | gartral | redotis: if I were using brtfs, nilsfs, or a similar "snapshot fs" it would be exceptionally easy |
| 03:39.51 | redotis | I mean all config files are in /etc/asterisk right...or most. Ahh well...I'm prolly wrong. Thanks for the help with confbridge channelz. |
| 03:40.12 | gartral | problem is, I don't have multiple 2TB hdds to use that effectively |
| 03:40.27 | gartral | redotis: you are correct |
| 03:41.02 | gartral | oh man, my asterisk is HOSED |
| 03:41.04 | gartral | No such command 'sip show peers' |
| 03:42.08 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 03:42.19 | gartral | nnghhrrrr looks like I'm starting from scratch |
| 03:43.46 | ChannelZ | Are you saying your sip.conf looked like you left it? |
| 03:43.53 | gartral | ChannelZ: exactly |
| 03:44.20 | ChannelZ | what exactly did this person install? |
| 03:46.45 | gartral | 'core help' -> No such command 'core help' |
| 03:46.46 | gartral | ChannelZ: digiums dead web gui |
| 03:46.46 | gartral | I'm going to rip his head and prick off, reverse thir posotions and simultaniusly give him an instant gender-realignement while he suffocates.. |
| 03:49.40 | ChannelZ | as in asterisk-gui-2.1.0-rc1 ? |
| 03:50.27 | gartral | ChannelZ: yes |
| 03:52.15 | gartral | ChannelZ: are you installed on debian or ubuntu? |
| 03:52.36 | ChannelZ | ubuntu generally |
| 03:52.57 | gartral | ChannelZ: could i get your asterisk.conf for comparison? |
| 03:53.36 | rbd_ | is it possible to pass a variable name into an argument for the AGI EXEC function? |
| 03:53.37 | ChannelZ | it's stock |
| 03:54.14 | ChannelZ | http://pastebin.com/H7kEjKk3 |
| 03:54.20 | gartral | ChannelZ: i'm not so sure asterisk-gui left mine stock |
| 03:55.36 | ChannelZ | I have asterisk is a VM I can trash, let me install asterisk-gui and see what all it changes. |
| 03:57.21 | gartral | ChannelZ: thanks |
| 04:00.13 | ChannelZ | well just installing it didn't touch /etc/asterisk at all. |
| 04:00.47 | ChannelZ | just installing anyway. |
| 04:00.57 | ChannelZ | Do you know did he do "make checkconfig" ? |
| 04:01.38 | gartral | ChannelZ: I'm not sure |
| 04:01.40 | ChannelZ | actually that doesn't do a lot either. |
| 04:02.05 | Penguin | Pull out your backups and put them back where they belong. |
| 04:02.24 | ChannelZ | asterisk-gui is really just a glorified web server, it doesn't take over the system like FreePBX does. To my knowledge, and by the looks of what I"m seeing. |
| 04:02.27 | gartral | Penguin: I DIDN'T HAVE ANY! I was Just Starting! |
| 04:02.35 | ChannelZ | I think you have some other problem. |
| 04:02.47 | Penguin | Oh, then you didn't have much to lose anyway. |
| 04:04.12 | gartral | Penguin: yes I did, I had spent hours doing the cofigs |
| 04:04.27 | gartral | I was going to back them up before going to bed |
| 04:04.53 | ChannelZ | So you say you have sip.conf but in asterisk it's not loaded. what do you get if you 'module load chan_sip' ? |
| 04:05.01 | gartral | but it doesn't look like it would have fucking attered because asterisk is ignoring perfectly valid configs |
| 04:05.45 | gartral | [Oct 28 00:05:23] ERROR[2560]: chan_sip.c:31225 reload_config: Contents of sip.conf are invalid and cannot be parsed |
| 04:06.05 | Penguin | Can you pastebin the sip.conf that is is complaining about? |
| 04:07.34 | ChannelZ | He might have fucked your configs WITH asterisk-gui, but it didn't do anything by its self by just installing it. |
| 04:07.55 | ChannelZ | So yes you need to go over the configs you have and see what debauchery has been done to them. |
| 04:08.44 | ChannelZ | How much did you actually get setup? sip.conf is easy to rebuild, and if you wrote a bunch of dialplan it's probably all mostly still there |
| 04:10.52 | gartral | yea, just got new output, where is the ldap settings for the realtime shit? |
| 04:11.12 | ChannelZ | eh? |
| 04:13.29 | gartral | ChannelZ: http://paste.ubuntu.com/6316294/ |
| 04:13.46 | ChannelZ | not important |
| 04:13.54 | ChannelZ | see line 3 |
| 04:14.19 | ChannelZ | (IE you don't want it anyway) |
| 04:16.54 | gartral | ChannelZ: wait |
| 04:17.16 | gartral | ChannelZ: what does "\;!" mean? |
| 04:18.06 | gartral | ChannelZ: more importantly; do you have \;! as the very very first 3 chars of your sip.conf? |
| 04:18.24 | ChannelZ | no |
| 04:18.29 | ChannelZ | and it means garbage |
| 04:19.45 | gartral | ChannelZ: it looks like asterisk gui did enable realtime.. == MySQL RealTime reloaded. |
| 04:21.11 | ChannelZ | You said you deleted/hid extconf.conf |
| 04:22.34 | gartral | ChannelZ: I have |
| 04:23.33 | gartral | ChannelZ: I'm about to purge and reinstall asterist in it's entirety |
| 04:23.42 | Penguin | its |
| 04:23.47 | ChannelZ | not sure why realtime is running then.. unless it's seeing res_mysql.conf anyway |
| 04:24.37 | ChannelZ | but unless extconf is there and binding some of your configs to a database, that's not why it isn't working |
| 04:26.24 | gartral | ChannelZ: http://paste.ubuntu.com/6316328/ |
| 04:26.42 | gartral | there's an ls -r of my /etc/asterisk |
| 04:27.34 | ChannelZ | jesus that's like everything |
| 04:29.18 | ChannelZ | just copy all that shit somewhere else so you have it, maybe some of whatever you setup is salvageable, but start over. |
| 04:29.20 | gartral | ChannelZ: i have a snapshot as it stands now, tell me what I absolutely need for basic services and google voice and i'll start trimming away |
| 04:29.32 | Penguin | I've only got like 46 config files. |
| 04:29.50 | ChannelZ | http://burner.com/asterisk-primer/getting-started/ |
| 04:30.38 | ChannelZ | That's literally all you need from the sample configs to get going. Plus sip.conf of your own making, extensions.conf of your own making, and for Google xmpp.conf and motif.conf |
| 04:32.14 | gartral | sudo mkdir ./fucked-samps && sudo mr ./* ./fucked-samps |
| 04:32.29 | gartral | oh.. dear.. I apologize! |
| 04:32.46 | gartral | feels really smart now |
| 04:32.51 | ChannelZ | my guess is your lackie did "make samples" as suggested by asterisk-gui when it couldn't find http.conf and manager.conf, and blew over everything |
| 04:33.31 | gartral | ChannelZ: "lackie" is giving that jackass too much credit |
| 04:34.08 | ChannelZ | actually... |
| 04:34.24 | ChannelZ | that's what all your .old files are probably, the originals. |
| 04:35.11 | ChannelZ | I thought 'make samples' just copied over things but I just did it in my VM and it actually left my originals as .old |
| 04:35.25 | gartral | ChannelZ: no, those were failed experiments |
| 04:35.35 | ChannelZ | some of them anyway. The important ones.. sip and extensions |
| 04:35.44 | gartral | from when I was playing with asterisk 1.8.x |
| 04:37.55 | ChannelZ | interesting. The install script is a little wonky then. OVERWRITE defaults to y but yet it renamed a few of mine. |
| 04:38.04 | ChannelZ | shrugs and wanders off to watch Homeland |
| 04:38.32 | gartral | ChannelZ: http://paste.ubuntu.com/6316360/ look right? |
| 04:39.21 | gartral | ARRRHG |
| 04:41.42 | ChannelZ | as right as not knowing what the contents of any of them are |
| 04:42.11 | gartral | ChannelZ: something is seriously fucking wrogg |
| 04:42.12 | gartral | http://paste.ubuntu.com/6316366/ |
| 04:44.10 | ChannelZ | reload isn't a real command, that's defined in cli_aliases.conf |
| 04:44.32 | ChannelZ | module reload |
| 04:45.16 | gartral | ChannelZ: that produced alot of errors |
| 04:45.27 | ChannelZ | it usually does |
| 04:46.46 | gartral | ChannelZ: insight? http://paste.ubuntu.com/6316376/ |
| 04:55.12 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 05:01.39 | zendel | hi, Is there a way I could monitor the Jitter/Latency stats history[call by call basis] on IAX2 channels. I know "Iax2 show channels" give me me some info for current channels. But tha's for current(present) traffic only. |
| 05:03.44 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 05:08.27 | gartral | gives up and goes to die |
| 05:13.48 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-yiikcocmkhipkjgs) |
| 05:23.07 | *** part/#asterisk zendel (~zendel.fe@chatswood.au.ulterius.net) |
| 05:23.21 | *** join/#asterisk zendel (~zendel.fe@chatswood.au.ulterius.net) |
| 05:34.29 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:37.48 | ChannelZ | gartral: nothing looks particularly bad |
| 05:38.46 | ChannelZ | None of those errors are fatal. Those are modules trying to autoload that you don't have configs for, so they fail. It's normal. |
| 05:50.57 | gartral | ChannelZ: but sip show peers, and other commands don't work |
| 05:54.41 | gartral | ChannelZ: after recompiling and installing i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 05:54.56 | gartral | when running any veriant of asterisk -r |
| 05:56.00 | gartral | asterisk -c seems to have fixed that |
| 05:56.16 | gartral | but I'm in the same boat with a fresh install and configs |
| 06:06.29 | *** join/#asterisk gerritfromsa (~voipmagic@8ta-229-199-48.telkomadsl.co.za) |
| 06:07.52 | gartral | cries |
| 06:08.33 | gerritfromsa | About what? |
| 06:09.36 | gerritfromsa | Anyone from Africa? |
| 06:09.46 | gartral | sip show peers = no command found\ |
| 06:10.26 | gerritfromsa | Recompile asterisk, you've got missing libraries |
| 06:11.14 | gerritfromsa | Or there is no sip.conf in your /etc/asterisk folder |
| 06:11.33 | gerritfromsa | Which is more likely |
| 06:13.57 | *** join/#asterisk [1]pdonner (HydraIRC@line-1915.dyn.kponet.fi) |
| 06:14.00 | kaldemar | gartral: asterisk -r does not run asterisk. it attaches to the CLI of a running instance. |
| 06:14.44 | kaldemar | or no modules.conf that would make chan_sip.so load at startup. |
| 06:15.42 | gartral | kaldemar: is it load module chan_sip or module load chan_sip? |
| 06:16.50 | kaldemar | neither. module load chan_sip.so |
| 06:17.35 | gartral | omg |
| 06:18.10 | kaldemar | tab is your friend in CLI |
| 06:19.12 | [1]pdonner | Hi everybody. What is the easiest way to filter anonymous calls to an extension? |
| 06:23.16 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-198-184.telkomadsl.co.za) |
| 06:24.18 | [1]pdonner | Sorry if this was a FAQ. I googled on the matter and the advice was not to try anything like that if I'm a novice in the Asterisk world. |
| 06:25.04 | gerritfromsa | missed the question , please repeat |
| 06:25.28 | gartral | http://paste.ubuntu.com/6316657/ |
| 06:25.39 | [1]pdonner | What is the easiest way to filter anonymous calls to an extension? |
| 06:25.46 | gartral | I can't hear my friend, they hear me |
| 06:26.25 | gerritfromsa | gartral - most probably NAT issues , YOU are blocking rtp traffic |
| 06:28.19 | gerritfromsa | gartral , can use GotoIf( $[CALLERID(num)='anonymous'] ?true:false) |
| 06:28.58 | gerritfromsa | pdonner - can use GotoIf( $[CALLERID(num)='anonymous'] ?true:false) |
| 06:29.12 | [1]pdonner | ok I'll try that |
| 06:29.39 | [1]pdonner | 'anonymous' or '' ? |
| 06:30.12 | [1]pdonner | version 1.6.2.9 |
| 06:30.30 | gartral | [Oct 28 02:30:18] WARNING[17850]: pbx.c:9619 add_priority: Unable to register extension '_1XXXXXXXXXX' priority 1 in 'users', already in use |
| 06:37.22 | kaldemar | [1]pdonner: that depends on how anonymous calls look like when they come to your system. |
| 06:37.46 | [1]pdonner | They look like '' in CLI |
| 06:40.32 | ChannelZ | What do you mean by "anonymous calls" |
| 06:40.47 | ChannelZ | As in unauthenticated? |
| 06:41.04 | [1]pdonner | The caller doesn't give an id |
| 06:41.24 | ChannelZ | as in allowguest=yes |
| 06:42.02 | ChannelZ | Put them in a separate context and use _X. for your extension to pick up anything.. if you really want to do something with those calls |
| 06:43.07 | kaldemar | ChannelZ: methinks he meant anonymous as in anonymous by caller id. |
| 06:43.08 | gerritfromsa | pdonner : You could retrieve the LENGTH of the caller id with NoOp(${LEN(${CALLERID(num)})}) and if it is < 10 the go to true |
| 06:43.11 | [1]pdonner | Here is an example |
| 06:43.13 | [1]pdonner | [Oct 28 08:26:58] NOTICE[19601]: chan_sip.c:20325 handle_request_invite: Call from '' to extension '000972595501258' rejected because extension not found in context 'phones' |
| 06:43.56 | kaldemar | gerritfromsa: that won't work at all in his country. |
| 06:44.03 | gerritfromsa | pdonner : Hell NO - those are unauthenticated calls and could be a hacker trying to male calls ! |
| 06:44.16 | [1]pdonner | That's it |
| 06:44.43 | ChannelZ | Do you NEED/want to accept unathenticated calls for some other reason? |
| 06:44.53 | gerritfromsa | Use fail2ban to block those ip's |
| 06:44.54 | [1]pdonner | No I don't |
| 06:45.01 | ChannelZ | then put allowguest=no in sip.conf |
| 06:45.14 | [1]pdonner | Ok. I'll do that. |
| 06:45.35 | ChannelZ | in [general] |
| 06:45.38 | gerritfromsa | check you /var/log/asterisk/messages file for the ip's and block them now |
| 06:46.02 | [1]pdonner | ok I'll do that as well |
| 06:46.18 | ChannelZ | fail2ban is handy |
| 06:46.49 | [1]pdonner | I'll check how that is installed |
| 06:47.00 | gerritfromsa | Just make sure you use the latest copy of asterisk.conf with your fail2ban |
| 06:47.11 | ChannelZ | you'll need to write some rules/handlers for it yourself more than likely |
| 06:48.22 | [1]pdonner | Default for allowguest is yes. Made it no! |
| 06:48.32 | gerritfromsa | Could also add call-limit= to your sip.conf account from your service provider to limit outbound calls |
| 06:49.02 | gerritfromsa | Some default settings is bad ... |
| 06:58.10 | *** join/#asterisk charkee (~charkee@121.54.46.199) |
| 07:03.41 | [1]pdonner | allowguest=no seems to filter all calls without ids |
| 07:04.40 | [1]pdonner | Still getting some of these: [Oct 28 09:02:53] NOTICE[19601]: chan_sip.c:21812 handle_request_register: Registration from '"4058" <sip:4058@194.100.3.145:5060>' failed for '173.212.205.190' - No matching peer found |
| 07:05.45 | gerritfromsa | its the hacker again ... add deny= end permit= to sip.conf especially if you're running this on a LAN only |
| 07:05.49 | [1]pdonner | Attempts to register at number that doesn't exist. |
| 07:07.15 | gerritfromsa | these guys often use sipvicious - scans a range of accounts and guesses the passwords - range or dictionary |
| 07:11.17 | ChannelZ | Unless you've got dynamic-IP clients who need to access your system from random places, allow the IPs you need through your firewall and block the rest. |
| 07:12.17 | [1]pdonner | Is static IPs a method of hiding accounts? |
| 07:12.19 | ChannelZ | If your system does need to be somewhat open, then fail2ban is good. Make sure your peer names aren't lame (like 100 and 1002 or whatever) and you have strong passwords |
| 07:12.31 | ChannelZ | Hiding accounts? |
| 07:12.34 | [1]pdonner | I have good extension number schemes and very long passwords, |
| 07:13.15 | kaldemar | http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 07:14.19 | gerritfromsa | you can open a range of ip's : allow=192.168.1.0/255.255.255.0 fo example |
| 07:14.25 | [1]pdonner | Hiding: Avoid hacker calls without identifiable id, who then detect number and start cracking password. |
| 07:14.49 | [1]pdonner | allow=192.168.1.0/255.255.255.0 Sounds very good |
| 07:14.54 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:15.15 | gerritfromsa | remember to start with deny=0.0.0.0/0.0.0.0 |
| 07:17.30 | [1]pdonner | Ok. I'll use that. Would easily filter at least hackers from foreign countries. |
| 07:18.32 | [1]pdonner | This looks good: http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt Thanks! |
| 07:22.01 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
| 07:26.06 | gerritfromsa | the FILTER is kick-ass ! |
| 07:30.42 | *** part/#asterisk bpietro (~pietro@82.51.236.132) |
| 07:36.41 | *** join/#asterisk Tokeiito (~quassel@main.kbi.lt) |
| 07:40.30 | [1]pdonner | Can I delete the older message files at /var/log/asterisk without problems? |
| 07:44.21 | ChannelZ | sure.. but you'll want to 'logger reload' after or else it will keep writing to the orphaned files |
| 07:45.01 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:46.27 | [1]pdonner | ChannelZ: you'll want to 'logger reload' after or else it will keep writing to the orphaned file. - Thanks, I guessed that something like that might be the case, |
| 07:47.00 | gerritfromsa | use crontab or linux application called logrotate to do this in the future , and too big messages file can cause your system to crash |
| 07:47.27 | [1]pdonner | Yes, I was worried. |
| 07:50.40 | [1]pdonner | What about CLI command logger rotate? |
| 07:51.57 | gerritfromsa | can do this from bash or/and in crontab with : #asterisk -rx "logger reload" |
| 07:51.58 | ChannelZ | You can use that. But then if you're just going to cron it, you might as well just use logrotate |
| 07:52.10 | *** join/#asterisk hos7ein (~chatzilla@91.98.33.208) |
| 07:52.45 | gerritfromsa | improvement : /usr/sbin/asterisk -rx "logger rotate" |
| 07:55.13 | *** join/#asterisk mirela666 (~mirko.bra@93-87-217-142.dynamic.isp.telekom.rs) |
| 08:00.10 | ChannelZ | STAND BY, WE ARE ABOUT TO GO BACK IN TIME! |
| 08:01.24 | ChannelZ | Err no.. that's next week. Shit. |
| 08:01.30 | ChannelZ | I should be in bed. |
| 08:03.03 | carrar | yeah |
| 08:03.09 | carrar | although |
| 08:03.17 | carrar | There is always ESPRESSO in the morning |
| 08:17.37 | eirirs_ | mocha! |
| 08:17.44 | eirirs_ | (espresso+hot chocolate) |
| 08:18.36 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:20.45 | *** join/#asterisk bpietro (~pietro@2-228-138-56.ip191.fastwebnet.it) |
| 08:21.07 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:417c:8772:a236:1782) |
| 08:26.45 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 08:28.28 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:38.22 | *** join/#asterisk CeBe (~CeBe@port-92-206-113-97.dynamic.qsc.de) |
| 08:41.16 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 08:50.10 | *** join/#asterisk martinfletcher_ (~martinfle@87.237.70.109) |
| 08:51.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:52.22 | bpietro | ChannelZ, unrandom mystery resolved, error between kbd & chair ;) I've 24 files with different names and identical content |
| 08:53.38 | *** join/#asterisk eZz (~ez@unaffiliated/ezz) |
| 08:54.19 | eZz | hello |
| 08:54.49 | eZz | anybody knows, if SPA3000 (or such of) can work with pulse dialing (instead of tone) ? |
| 08:55.53 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 08:58.18 | Faustov | guys, is there a function that would return if voicemail is configured for given exten? |
| 09:06.22 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:06.42 | kaldemar | Faustov: voicemails are not configured for extensions, but devices. which do you mean? |
| 09:09.25 | *** join/#asterisk atha (~athayde@unaffiliated/athayde) |
| 09:12.18 | Faustov | kaldemar: in voicemail.conf under the default context I only got extensions listed |
| 09:13.15 | Faustov | well, in my case mailbox name equals extension, for simplicity |
| 09:13.36 | Faustov | so I guess the question is if there's a function that would run a check and return some information |
| 09:14.40 | kaldemar | Faustov: those are not extensions, they are mail boxes. which devices (in channel configuration files) have those mail boxes? |
| 09:15.38 | kaldemar | VM_INFO can be used to get the actual information. |
| 09:15.56 | Faustov | one sec |
| 09:16.09 | kaldemar | how the boxes are associated to the channels is another matter. |
| 09:16.12 | wdoekes | kaldemar: voicemails are not configured for devices either |
| 09:17.11 | Faustov | yes, vm_info will do exactly what I need |
| 09:17.12 | Faustov | thanks kaldemar |
| 09:17.24 | kaldemar | wdoekes: for example for sip they are. with the mailbox parameter. |
| 09:19.51 | kaldemar | ${VM_INFO(${SIPPEER(${CHANNEL(peername)},mailbox},exists)} |
| 09:20.43 | wdoekes | you're missing a paren |
| 09:22.02 | wdoekes | but yes, that works |
| 09:28.51 | *** join/#asterisk ghost75 (~quassel@dslb-188-105-029-200.pools.arcor-ip.net) |
| 09:36.13 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 09:36.36 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 09:45.43 | *** join/#asterisk sadtimes (~sadtimes@105-237-88-20.access.mtnbusiness.co.za) |
| 09:45.47 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 09:49.51 | *** join/#asterisk dronacid (~dronacid@vpn.openroot.de) |
| 10:07.32 | sadtimes | anyone ever have any issues where you issue a channel hangup request |
| 10:07.35 | sadtimes | or soft hangup |
| 10:07.41 | sadtimes | and the channel shows as up? |
| 10:08.08 | sadtimes | but with (none) in the application section and Data |
| 10:09.45 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 10:10.14 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 10:10.52 | *** join/#asterisk polysics (~Adium@host209-121-dynamic.51-79-r.retail.telecomitalia.it) |
| 10:11.00 | polysics | hello! |
| 10:11.15 | polysics | wow, Astricon was the bomb, I still have notes to go through after two weeks |
| 10:11.33 | polysics | question: will confbridge record start append to an existing file? |
| 10:15.48 | dym | missed astricon :( |
| 10:18.06 | polysics | I was in ATL for other things but we thought we might as well go |
| 10:18.30 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 10:18.32 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 10:24.19 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
| 10:31.23 | gerritfromsa | like atlanta ... close to charleston ... |
| 10:32.21 | gerritfromsa | was at astricon in 2009 - 10th anniversary but nothing to write home about ... |
| 10:33.08 | gerritfromsa | 10th anniversary of asterisk that is not astricon i believe this was this year |
| 10:36.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:37.46 | *** join/#asterisk felipealmeida (~user@177.157.202.198) |
| 10:38.13 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:38.52 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 10:59.30 | polysics | the t-shirt says "10" so it probably was :) |
| 10:59.46 | polysics | so, how come I can't turn OFF SIP debugging on this Asterisk instance? |
| 11:00.05 | polysics | sip set debug off is like the goggles |
| 11:00.08 | polysics | Asterisk 11 |
| 11:04.39 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 11:10.58 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-79-141.user.veloxzone.com.br) |
| 11:23.50 | *** join/#asterisk Gaiax (~gaiax@unaffiliated/gaiax) |
| 11:27.36 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 11:56.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:58.48 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 11:59.09 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 12:05.12 | *** join/#asterisk shinao1 (~blackulus@41.58.70.226) |
| 12:16.16 | *** join/#asterisk LooserOuting (~LooserOut@ip-176-198-134-194.unitymediagroup.de) |
| 12:18.18 | LooserOuting | Hi. i have a problem with receive fax. the command "fax show stats" shows a lot of "Retries Exceeded". Can i configure max retries ? |
| 12:21.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:39.42 | polysics | I can't seem to turn off SIP debugging on this machine, it's stupid :) |
| 12:40.07 | Chainsaw | polysics: That's a shame, it's very loud about stale nonces and all that jazz. |
| 12:45.05 | *** join/#asterisk StaRetji (~LittleAll@178.79.6.17) |
| 12:45.12 | polysics | I have no idea why "sip set debug off" does nothing |
| 12:46.36 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:46.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:46.59 | file | what are you expecting it to do? |
| 12:47.15 | file | (ie: what are you seeing that you would expect not to see after running it) |
| 12:47.38 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:49.41 | StaRetji | folks, can someone show me example how ${HANGUPCAUSE} is used in extension to get detailed hangup cause ( Im getting FAILED now) |
| 12:50.37 | polysics | file: I would expect noSIP debug messages to be printed |
| 12:51.01 | file | the command controls output of SIP messages and some associated information about actions taken with them |
| 12:51.06 | file | it does not control core debug messages |
| 12:51.09 | polysics | also, hi Josh. Luca from Mojo Lingo here, Ben Klang's colladorator :) |
| 12:51.25 | file | I knowz |
| 12:51.27 | polysics | Astricon was THE BOMB |
| 12:51.48 | tuxx- | the NSA knows now too.. |
| 12:51.58 | polysics | I had not attended one before, it goes on the list of my obligatory conferences :) |
| 12:52.08 | file | it was a very good AstriCon |
| 12:52.20 | polysics | tuxx-: not like they did not before, I asked them for a few emails I lost in 2009 but never got an answer :( |
| 12:52.33 | tuxx- | ;-() |
| 12:52.36 | tuxx- | ;-) * |
| 12:52.37 | polysics | so, there might be some other setting that controls SIP debugging? |
| 12:52.46 | polysics | I am referring specifically to actual SIP messages |
| 12:52.55 | polysics | <--- SIP read from UDP:10.203.175.1:58677 ---> |
| 12:52.55 | polysics | SUBSCRIBE sip:usera@asterisk.local-dev.mojolingo.com SIP/2.0 |
| 12:52.56 | polysics | ets |
| 12:52.57 | file | polysics, sip set debug should control that |
| 12:53.24 | polysics | there has to be something else :) |
| 12:53.31 | file | nope |
| 12:53.33 | polysics | not a biggie |
| 12:53.33 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
| 12:54.14 | polysics | https://gist.github.com/polysics/01b49ec89f5194533437 |
| 12:54.29 | polysics | setting debug off, then de-registering and registering a peer |
| 12:54.50 | file | can't say I've seen that before... and there's no current issues open about that... |
| 12:54.58 | polysics | it is more puzzling than anything, not a big issue |
| 12:55.07 | polysics | but that SHOULD silence it, correct? |
| 12:55.10 | file | yes. |
| 12:55.48 | polysics | oh! there's a "sipdebug" setting in sip.conf |
| 12:55.51 | file | although you can override it in sip.conf |
| 12:56.07 | file | yeahhhh, the set debug off may not change that |
| 12:56.28 | wdoekes | it doesn't? that sounds .. wrong |
| 12:56.37 | file | I've never used it. |
| 12:56.54 | polysics | it does not override the sip.conf setting |
| 12:57.04 | polysics | I set it to no and now it does not debug any more |
| 12:57.20 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 12:57.32 | polysics | I *think* the correct behavior would be to simply set the initial debug status, but again, no biggie |
| 12:57.41 | polysics | ie. sip set debug off should kill it |
| 12:58.00 | file | there can be arguments either way, the person who did it probably thought the opposite |
| 12:58.04 | polysics | imho it's quite handy, saves you from firing up Wireshark for very simple debugging |
| 12:58.14 | polysics | the debugging itself, I mean |
| 12:58.21 | polysics | but no problem, now we know :) |
| 13:00.16 | wdoekes | polysics: I keep rotating pcaps around, so I can find past events.. searching through pcaps is always easier than browsing the debug log |
| 13:01.16 | polysics | wdoekes: of course that is more thorough, but things like "no available codec" are easily fixed by looking at the console |
| 13:02.12 | wdoekes | depends a bit on how cluttered your console already is ;) |
| 13:03.14 | wdoekes | but yes.. it does show more than just the socket data |
| 13:07.40 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-sleyqptwsqkzhjly) |
| 13:07.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:08.42 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:10.24 | *** join/#asterisk acidfu (~nib@unaffiliated/acidmen) |
| 13:11.18 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-sleyqptwsqkzhjly) |
| 13:15.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:20.04 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 13:22.37 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:417c:8772:a236:1782) |
| 13:25.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 13:30.14 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 13:41.00 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ejauwdlfeuzzxgme) |
| 13:41.01 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:41.01 | StaRetji | is there any way getting asterisk to write detailed hangupcause, instead only FAILED, whithout using macro? |
| 13:41.40 | [TK]D-Fender | StaRetji: What is this macro you're referring to? Got actual debug for us? |
| 13:42.48 | StaRetji | ehm, no, I was reading wiki and couldn't find example how to set asterisk to write exact hangup cause, like congestion |
| 13:44.42 | [TK]D-Fender | StaRetji: There is no magical example for channel variables. Go actually LOOK at them |
| 13:45.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-jnqlzegmbhcqccfr) |
| 13:45.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:45.30 | StaRetji | well, provider says is sending Q850 codes |
| 13:45.55 | StaRetji | and I am trying to figure out how to get them in cdr |
| 13:45.57 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 13:46.44 | [TK]D-Fender | StaRetji: So have you seen them? |
| 13:47.53 | StaRetji | [TK]D-Fender: please forgive me, seen what? |
| 13:48.21 | [TK]D-Fender | StaRetji: Those Q850 codes they're rferring to. |
| 13:48.43 | StaRetji | yes |
| 13:49.14 | *** join/#asterisk twanny796 (~twanny796@c141-181.i01-4.onvol.net) |
| 13:49.38 | twanny796 | does anyone know of a hardware device for collecting call detail recording? |
| 13:50.26 | gerritfromsa | the old pbx buffers do this via RS232 so it's possible with asterisk |
| 13:50.39 | *** join/#asterisk F-JJTH (~clement@car84-1-78-229-89-36.fbx.proxad.net) |
| 13:51.17 | [TK]D-Fender | I use a computer for that... |
| 13:51.21 | StaRetji | [TK]D-Fender: 404 NO_ROUTE_DESTINATION |
| 13:51.27 | [TK]D-Fender | It's made up of hardware... |
| 13:52.02 | [TK]D-Fender | StaRetji: You know that posting just that little snippit doesn't indicate that you'll even get your hands on it... |
| 13:52.24 | F-JJTH | Hi all, I have a question about Dial() / Transfert() |
| 13:52.45 | [TK]D-Fender | F-JJTH: ask away... |
| 13:52.48 | gerritfromsa | raspberry pi would would great |
| 13:53.27 | StaRetji | hmm, I thought there is just setting in extenstion that will tell asterisk to log q850 codes instead of just FAILED, cause FAILED can be anything |
| 13:53.35 | F-JJTH | I would use the following scheme: clientA tell phonenumber 1234 on main.mydomain.org |
| 13:53.50 | [TK]D-Fender | StaRetji: You keep talking about these cades and aren't showing exactly how they are sent. |
| 13:53.58 | F-JJTH | then main.mydomain.org will redirect the call to slave05.mydomain.org |
| 13:54.42 | F-JJTH | once the call has been redirected I would shutdown main.mydomain.org then clientA can talk to slave05.mydomain.org |
| 13:54.47 | F-JJTH | is it possible ? |
| 13:55.26 | F-JJTH | <PROTECTED> |
| 13:55.32 | StaRetji | Wiki says: It is good practice to write the cause code into the CDR: Set(CDR(userfield)=Hangupcause:${HANGUPCAUSE}) |
| 13:55.51 | F-JJTH | the goal of main.mydomain.org is to distribute the call to a farm of slave |
| 13:55.55 | *** part/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 13:55.57 | StaRetji | so, I thought putting this in extension will do the job |
| 13:56.10 | StaRetji | though, I am not sure what userfield stands for |
| 13:56.35 | StaRetji | unless field in csv file |
| 13:57.15 | [TK]D-Fender | StaRetji: You see it referenced in the CDR FUNCTION. What do YOU think? |
| 13:57.59 | [TK]D-Fender | F-JJTH: what is this redirect decision based on? |
| 13:58.06 | F-JJTH | so my question is: what shoud I use for this, Dial() or Transfert() ? the important things here is that main.mydomain.org must be totally independant once the call has been redirected |
| 13:58.10 | gerritfromsa | F-JJTH : The voice(or rtp) traffic will by default route between clientA ans slave05 but the SIP would still me with main due to the cdrs |
| 13:58.35 | gerritfromsa | You will have to force a REFER for this to work |
| 13:58.38 | [TK]D-Fender | F-JJTH: You would Transfer(), not Dial(), if using * at all... |
| 13:58.42 | StaRetji | [TK]D-Fender: now I don't know what to think, I am confused |
| 13:59.14 | [TK]D-Fender | StaRetji: If you can't wrap your head around the CDR function you have to pretty big issues. |
| 13:59.35 | [TK]D-Fender | some* |
| 13:59.37 | [TK]D-Fender | gah |
| 14:00.06 | StaRetji | well, I thought if I nest this in my dialplan |
| 14:00.15 | F-JJTH | [TK]D-Fender: decision is based on the phonenumber. slave01 can mange phone number 1110 1111 1112 1113 1114 ..... 1119 slave02 can manage 1120 1121 1122 1123.... 1129 slave03 can manage 1130 1131....1139 |
| 14:00.17 | StaRetji | it will write it to cdr |
| 14:00.53 | [TK]D-Fender | StaRetji: You say that as if the place you put it wasn't important. |
| 14:01.06 | F-JJTH | gerritfromsa: i'm using IAX2 does is works the same ? |
| 14:01.10 | [TK]D-Fender | StaRetji: You don't jsut "put stuff in your dialplan". Location and timing is important. |
| 14:01.27 | *** join/#asterisk Draecos (~Draecos@124-148-79-51.dyn.iinet.net.au) |
| 14:02.20 | *** join/#asterisk rexwin___ (~rexwin@116.202.162.78) |
| 14:03.22 | gerritfromsa | F-JJTH : Not 100% sure , but wireshark will tell you for certain |
| 14:03.52 | *** join/#asterisk CrashHD (~na@68-67-80-153.wavecable.com) |
| 14:04.05 | F-JJTH | gerritfromsa: correct |
| 14:04.17 | gerritfromsa | The IAX2/SIP signaling should be small traffic and main should be able to handle this easy , unless you're running 1000's |
| 14:04.23 | gerritfromsa | of simultaneous calls |
| 14:04.44 | F-JJTH | gerritfromsa: I plan 100 simultaneous calls |
| 14:05.57 | F-JJTH | [TK]D-Fender: any advices ? |
| 14:06.37 | gerritfromsa | This should be no problem - even for an entry level server |
| 14:07.41 | F-JJTH | the server is hosted at my home, on my personal Internet connection (that's why main server must redirect calls to multiple others servers because I have a little bandwidth here) |
| 14:07.59 | F-JJTH | is contributing to an open source project |
| 14:09.39 | gerritfromsa | Make sure the routing is in place so that the clients can route rtp traffic to one and another |
| 14:10.02 | gerritfromsa | default rtp port 10000-20000 or change it in rtp.conf |
| 14:10.05 | [TK]D-Fender | F-JJTH: IAX2 should work fine with Tansfer() |
| 14:10.18 | gerritfromsa | only need about 500 ports for 100 simultaneous calls |
| 14:10.18 | F-JJTH | the routing is done by the DialPlan of main.mydomain.org |
| 14:10.22 | [TK]D-Fender | F-JJTH: And there is no such thing as RTP there. |
| 14:10.42 | F-JJTH | IAX2 only require 4569 UDP right? |
| 14:11.02 | gerritfromsa | yes , but I would change this |
| 14:11.13 | gerritfromsa | safer to move the ports |
| 14:11.33 | F-JJTH | for spamming purpose ? |
| 14:11.42 | gerritfromsa | 4569 was the last 4 digits of Marc Spencer's telephone number .... |
| 14:11.46 | gerritfromsa | yea and hackers |
| 14:11.54 | *** part/#asterisk polysics (~Adium@host209-121-dynamic.51-79-r.retail.telecomitalia.it) |
| 14:11.54 | Penguin | It's not safer to change ports, it just makes it harder to admin it when you inherit the system. |
| 14:12.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:12.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:12.38 | Penguin | If you believe using a non-standard port makes it "more secure," I advise you to go back to network security class. |
| 14:14.25 | gerritfromsa | sniffing 4569 will definitely make it easier finding an asterisk server |
| 14:15.07 | Greenlight | It depends how you defined "more secure". Will it disgard the attemtps from people attempting on only 4569, yes. Will it stop all attacks, No. |
| 14:15.39 | Greenlight | A lot of "hackers" are just scanning for the standard ports to get the "low hanging fruit" |
| 14:15.49 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 14:15.49 | Penguin | Obscurity is not security. |
| 14:16.10 | Greenlight | As I say, it depends on your defenition. |
| 14:16.19 | Penguin | Does it make it less obvious? Sure. |
| 14:16.38 | Greenlight | The key is not to be lulled into a false sense of security. |
| 14:16.41 | Chainsaw | It shouldn't be your only defence, indeed. But I can see how it helps with tourism from the People's Republic of China. |
| 14:16.44 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 14:17.02 | Chainsaw | I really wish the great firewall was more bidirectional in nature. They don't seem to do much egress filtering. |
| 14:17.02 | gerritfromsa | software such as sipviscous for example |
| 14:17.14 | Chainsaw | sipviscous... interesting. |
| 14:17.17 | Chainsaw | Is that a fork of sipvicious? |
| 14:17.34 | Greenlight | It's thicker? |
| 14:17.36 | Penguin | But thicker |
| 14:17.38 | Penguin | Yes. |
| 14:17.40 | gerritfromsa | :) exactly |
| 14:17.40 | Greenlight | :) |
| 14:17.44 | atha | lal |
| 14:19.27 | *** join/#asterisk charkee (~charkee@121.54.46.199) |
| 14:19.47 | gerritfromsa | fail2ban then |
| 14:20.12 | Greenlight | Again, it's a step in the right direction, but not completely secure |
| 14:20.37 | Chainsaw | You would need to have the authentication set up securely enough to trigger the fail2ban 3-strikes rule, yes. |
| 14:20.54 | Penguin | Using an unusual port for a network service does make the service less discoverable in general, but this is not consistent with any security mechanism and should not be considered as security. |
| 14:20.56 | Greenlight | Well, sometimes they can get more than 3 strikes |
| 14:21.14 | Greenlight | fail2ban has to read the logs and until it does it won't block |
| 14:21.14 | Chainsaw | Greenlight: Not in my world. |
| 14:21.39 | Chainsaw | Greenlight: 1, 2, 3 *shotgun blast* |
| 14:21.40 | Greenlight | They can sometimes get 10+ attempts before fail2ban has "kicked in" |
| 14:21.43 | Penguin | Remember, security is layered. I was taught to think of security like an onion with all its layers. |
| 14:21.58 | Chainsaw | Penguin: The comparison is apt. It can easily make one cry. |
| 14:22.16 | Greenlight | remembers never to tresspass on Chainsaw's land |
| 14:23.08 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:23.08 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:23.12 | *** join/#asterisk Gaiax (~gaiax@unaffiliated/gaiax) |
| 14:27.37 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:31.14 | gerritfromsa | anyone know a good webrtc and asterisk howto url ? |
| 14:34.14 | Chainsaw | gerritfromsa: If it doesn't mention applying lots of patches or not using Firefox... it's probably incomplete. |
| 14:34.40 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 14:35.41 | gerritfromsa | will wait a bit then ... |
| 14:35.58 | glaz | hi, I'm trying to change de From: value in the SIP header with asterisk. It uses what I have in fromuser= in sip.conf but I would like to be able to specify my From: value from the dialplan, any suggestion? |
| 14:36.28 | glaz | My sip provider uses the From: for the called ID and I can specify the user info with the P-Asserted-Identify value |
| 14:39.04 | gerritfromsa | Dialplan : SipAddHeader(P-Asserted-Identity:<sip:${YOURNAME}>) |
| 14:39.31 | gerritfromsa | Add this before the Dial() statement |
| 14:40.56 | gerritfromsa | There is also the Set(CONNECTEDLINE) options in asterisk 1.8> |
| 14:40.57 | glaz | gerritfromsa: that I'm ok with |
| 14:41.10 | glaz | it's how to change the From value |
| 14:41.25 | glaz | Asterisk sends the SIP Invite with the From: value from the username= from sip.conf |
| 14:41.36 | gerritfromsa | ip address / name |
| 14:41.52 | glaz | Humm ? |
| 14:42.12 | *** join/#asterisk serafie (~erin@nat/digium/x-enadezfhpgmnmhfr) |
| 14:42.21 | *** join/#asterisk Alex_Bkash (b4ead883@gateway/web/freenode/ip.180.234.216.131) |
| 14:42.34 | glaz | My SIP Provider doesn't support RPID |
| 14:43.24 | wdoekes | glaz: if it uses what you have in fromuser=, you should remove fromuser=. asterisk will put the callerid(num) there by default |
| 14:43.44 | gerritfromsa | concur |
| 14:44.55 | glaz | wdoekes: really, that'd be NICE. Let me try |
| 14:45.17 | Penguin | When in doubt, leave it out. |
| 14:47.24 | gerritfromsa | Was wondering how many sessions asterisk manager interface can handle - is there a limit ? |
| 14:47.34 | gerritfromsa | Calling it with PHP |
| 14:48.19 | Alex_Bkash | i am passing call through IAX. when i wait on a call it creates increasing noise. and voice is delayed |
| 14:48.26 | gerritfromsa | 100 x ExtensionState every 5 seconds works OK but what is the limit |
| 14:49.33 | gerritfromsa | Alex : Are you transcoding ? |
| 14:51.03 | Alex_Bkash | yes |
| 14:51.26 | Alex_Bkash | i am transcoding |
| 14:52.13 | gerritfromsa | g729 to something else? |
| 14:52.54 | Alex_Bkash | gsm |
| 14:53.27 | gerritfromsa | Alex : Recording as well and are you using 20ms frames or something else |
| 14:53.51 | gerritfromsa | gsm to g729 ? |
| 14:54.33 | Alex_Bkash | yes sometimes |
| 14:55.02 | Alex_Bkash | ulaw-g729 gsm-g729 g729-g729 |
| 14:55.12 | gerritfromsa | 20ms? |
| 14:56.29 | Alex_Bkash | i am not sure abt that |
| 14:57.30 | gerritfromsa | Making this call between your own servers ? or using a service provider ? |
| 14:58.07 | gerritfromsa | Using open source g729 or did you pay for it? |
| 14:58.30 | Chainsaw | rings the loaded question alarm |
| 14:58.39 | Alex_Bkash | not like that |
| 14:58.54 | Alex_Bkash | i am recieving g729 calls mostly |
| 14:59.07 | Alex_Bkash | i am calling using a sip client |
| 14:59.38 | Greenlight | You can't transcode with g729 on the * box.... |
| 14:59.42 | Greenlight | *without |
| 15:00.19 | gerritfromsa | without? |
| 15:00.42 | Greenlight | mhm |
| 15:00.51 | Alex_Bkash | i've just set priority |
| 15:00.58 | Kobaz | so what local dns does everyone use these days |
| 15:00.58 | Alex_Bkash | alow |
| 15:01.04 | Kobaz | is it still dnsmasq or something like that? |
| 15:02.43 | Greenlight | Also, your issue is "increasing noise" and "voice delay" when you wait on a call ? |
| 15:03.54 | Alex_Bkash | yes |
| 15:03.59 | Alex_Bkash | when i am silent |
| 15:04.03 | Greenlight | Can you elaborate |
| 15:04.23 | Greenlight | So, you're waiting for a call, and not on a call? |
| 15:04.33 | Alex_Bkash | i am on a call |
| 15:04.36 | Alex_Bkash | and silent |
| 15:04.45 | Greenlight | Okay |
| 15:05.02 | gerritfromsa | What handset are you using ? |
| 15:05.30 | Alex_Bkash | pc headphone |
| 15:05.31 | *** join/#asterisk navaismo (~navaismo@189.191.200.240) |
| 15:06.35 | Greenlight | So, , you're on a call from what and to what? |
| 15:07.02 | Alex_Bkash | pc to mobile phn |
| 15:08.16 | gerritfromsa | Is this a new problem or has it always existed ? |
| 15:08.29 | Greenlight | And this isn't just feedback ? |
| 15:08.52 | Alex_Bkash | i never was silent b4 |
| 15:08.52 | Alex_Bkash | :D |
| 15:09.24 | Alex_Bkash | the callee kept me waiting |
| 15:09.36 | Alex_Bkash | then i heard this weired noise |
| 15:10.54 | gerritfromsa | are you recording [MixMonitor() or Monitor()] |
| 15:12.54 | glaz | wdoekes: it works. THANKS! |
| 15:13.24 | Alex_Bkash | no i am not |
| 15:15.51 | gerritfromsa | Alex : you should try another device to trial and error this or use wireshark to monitor the rtp traffic |
| 15:16.44 | gerritfromsa | silence suppression should be switched off as well , it does not work with * |
| 15:17.24 | Alex_Bkash | ok |
| 15:20.22 | runfromnowhere | Hi all - have some pretty basic dialplan questions if anyone has a few minutes to point me in the right direction.... |
| 15:20.41 | Penguin | ~ask |
| 15:20.41 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:20.45 | Penguin | runfromnowhere: ^ |
| 15:20.52 | runfromnowhere | LOL ah, pardon me :) |
| 15:22.30 | *** join/#asterisk TSM (~the_softw@46-65-201-69.zone16.bethere.co.uk) |
| 15:23.35 | runfromnowhere | I'm tracing the route a call took through our dialplan. An incoming call was directed to (ext-queues,704,1). I can see in the logs that it proceeds through the 704 section of that dialplan properly, until it hits the "Queue" function call. I know this farms the call out to the queue, and in this case nobody in the queue responded and the caller hung up. The log then says "Spawn extension (ext-queues,704,13) exited ... |
| 15:23.41 | runfromnowhere | ... non-zero" for that particular call ID. At this point, execution of the ext-queues 704 section seems to cease? Is this expected behavior for any non-zero return? |
| 15:24.20 | [TK]D-Fender | runfromnowhere: looks normal. The caller hung up. |
| 15:24.50 | runfromnowhere | OK I'm just trying to understand it from a "program flow" perspective - because the Queue function returned -1 (I think that's the return value for a hangup) the entire execution path is aborted? |
| 15:25.16 | Penguin | If the caller hangs up, what more is there to do? |
| 15:25.30 | Penguin | And Queue() is an application, not a function. |
| 15:25.46 | [TK]D-Fender | runfromnowhere: Correct. Caller hung up. Dialplan execution ceases |
| 15:25.51 | runfromnowhere | To be specific, in this case, "determine that they didn't reach anyone and unset the call recording filename in the database" |
| 15:26.01 | runfromnowhere | OK so understanding that helps a lot |
| 15:26.08 | gerritfromsa | Can double check that in /var/log/asterisk/queue_log |
| 15:28.00 | runfromnowhere | Basically right now, a call into a queue leaves me with like 12 different database entries - one for each attempt to ring each queue member - and all of them have a recordingfile entry that doesn't actually exist. I need to make sure that recordingfile entry is properly removed when nobody is reached in the queue and thusly recording doesn't actually happen. |
| 15:30.10 | runfromnowhere | Which is why I'm pouring over debug logs and trying to get an understanding of how things flow :) |
| 15:30.22 | Greenlight | Why not start the recording once you actually connect to someone ? |
| 15:30.27 | [TK]D-Fender | runfromnowhere: Then change when that entry gets added ... for instance dialing Local channels and setting it in a Macro/Gosub upon answer. |
| 15:30.51 | [TK]D-Fender | Greenlight: CDR FIELD <- this has nothing to do with audio. |
| 15:31.14 | Greenlight | Set that at the same time as actually starting the audio |
| 15:31.34 | runfromnowhere | I could give something like that a try |
| 15:31.58 | runfromnowhere | Unfortunately (and I KNOW this is not the channel for it) we're running FreePBX so I have to work within that framework |
| 15:32.09 | runfromnowhere | Getting my fundamentals down will help me work around it |
| 15:33.24 | Greenlight | I would seem to be better to look at not setting that field prior to recording starting, than trying to clear it if recording didn't start |
| 15:34.33 | runfromnowhere | I think you're right. I can't really see a reason to set it anywhere except right next to the Monitor statement itself but I'm still not 100% sure I have the full picture, and it's a production system |
| 15:34.37 | runfromnowhere | More investigation! :) |
| 15:35.47 | Greenlight | If I remember correctly FreePBX handles all it's recording stuff via a sub/macro |
| 15:36.36 | Greenlight | Wether that's where this CDR field gets set or not, I'm unsure |
| 15:36.48 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 15:37.42 | Penguin | its <--- |
| 15:38.08 | Greenlight | Grammar Nazi! |
| 15:38.44 | runfromnowhere | The CDR field seems to get set in the dialplan via this statement: |
| 15:38.55 | runfromnowhere | Set(CDR(recordingfile)=${CALLFILENAME}.${MON_FMT}) |
| 15:39.07 | runfromnowhere | Obviously your variables may vary |
| 15:39.10 | Greenlight | Okay, and where is that in relaation to the MixMonitor ? |
| 15:39.13 | runfromnowhere | So I can definitely move it around if need be |
| 15:39.39 | Greenlight | Well, remember with FreePBX you'll have to move stuff to the "override" file; you can't edit directly |
| 15:39.41 | runfromnowhere | It's just that, due to the dialplans FreePBX builds for you, it's really hard to know for sure where it needs to be - like am I going to screw over transfers or things? |
| 15:39.44 | runfromnowhere | Yeah |
| 15:39.57 | runfromnowhere | I already overrode everything to use Monitor (heh almost typed Minotaur) for stero recording |
| 15:40.00 | runfromnowhere | Let me pastebin it |
| 15:42.33 | runfromnowhere | Greenlight: http://pastebin.com/myeJPPkd |
| 15:42.53 | runfromnowhere | You can see it's in there several places that are not directly starting recording - but that function is also called FROM many places for many reasons that I haven't untangled yet |
| 15:44.14 | Greenlight | Yea, I'm familiar with this sub - we've overridden it ourselves on a couple of installs. |
| 15:44.32 | Greenlight | So it's line 9 at first glance that appears to be where it sets it and doesn't start actually recording |
| 15:44.58 | runfromnowhere | Yeah, but it also does it in the recq subroutine which seems to be getting called fairly oftn as well |
| 15:45.32 | runfromnowhere | My gut feeling is that it should ONLY be set directly adjacent to "Monitor()" but I know I don't know enough about call transfer/queue procedures to be certain |
| 15:46.00 | *** join/#asterisk pizzahead (86127c7ebc@rider.us) |
| 15:46.03 | *** part/#asterisk pizzahead (86127c7ebc@rider.us) |
| 15:46.41 | *** part/#asterisk glaz (strke@hiro.glaciuz.com) |
| 15:47.25 | opticron | if anyone wants to review something, this could use another look: https://reviewboard.asterisk.org/r/2905/ |
| 15:48.00 | opticron | RTP fixups when tearing down P2P bridging in 12 and trunk |
| 15:48.02 | Greenlight | I can't see any major issues in removing it from the "recq" part |
| 15:49.02 | runfromnowhere | I'll give it a shot after hours |
| 15:49.17 | runfromnowhere | Right now things shouldn't be changed if I want to be a responsible employee :) |
| 15:49.27 | Greenlight | It's perhaps capturing calls which are already being recorded and have been transferred to queues |
| 15:51.29 | runfromnowhere | It looks like when the call enters the queue it's like "Oh, this queue says always record - let's set the recordingfile before we actually start recording" |
| 15:53.20 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:53.24 | Greenlight | Yea, I'd need to work out where "recq" is called from, I think you can set the Queue app to record so perhaps it works in conjunction with that |
| 15:57.15 | LooserOuting | Hi. i have a problem with receive fax. the command "fax show stats" shows a lot of "Retries Exceeded". Can i configure max retries ? |
| 15:57.38 | runfromnowhere | I think it might - anyways I think I have a lot to play around with, thanks so much for the time and help! |
| 15:57.54 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 15:59.26 | Greenlight | LooserOuting: I would imagine that if you're getting "Retries Exceeded" then the solution isn't merely to increase the max retries, but is more likely to fix the actual problem. |
| 16:02.17 | LooserOuting | Thank you Greenlight. where do i start. the fax machine ist is woking fine. so i think it had to be some compatibility issue. i can see that the rate drops down from 14.4k to 12k. after that the asterisk drop the call |
| 16:03.40 | Greenlight | Not sure I can help with that, I never put faxes through Asterisk. Just advise people to get a PSTN line for that, or use email. |
| 16:04.31 | gerritfromsa | Using Hylafax ? |
| 16:04.47 | LooserOuting | i need t38 |
| 16:05.27 | LooserOuting | these are incoming faxes through a sip line |
| 16:06.40 | gerritfromsa | Using native rxfax or HylaFax ? |
| 16:07.02 | LooserOuting | i am using receivefax |
| 16:07.16 | LooserOuting | res_fax_spandsp |
| 16:07.45 | gerritfromsa | Sorry can't help much , use Hylafax myself |
| 16:08.14 | LooserOuting | how do you receive a fax with hylafax ? |
| 16:08.43 | gerritfromsa | Yes it works great and is also Free software |
| 16:08.52 | [TK]D-Fender | [12:04]LooserOutingi need t38 <- and is that what you're using now? |
| 16:09.15 | LooserOuting | yes I do |
| 16:09.54 | [TK]D-Fender | LooserOuting: if your faxes are failing perhaps your connection is too lossy |
| 16:10.05 | *** join/#asterisk rexwin_ (~rexwin@116.202.162.78) |
| 16:10.24 | LooserOuting | gerritfromsa: are u usind the t38gateway to "bridge" between iax an sip ? |
| 16:10.35 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:10.53 | rexwin_ | hi when i try to dial an outgoing number through zoiper, i get state down, codec unknown and service or option unavailable.. |
| 16:11.05 | rexwin_ | can somebody throw light on this. |
| 16:11.18 | [TK]D-Fender | rexwin_: And what do you see in Asterisk CLI with SIP DEBUG enabled? |
| 16:11.27 | LooserOuting | [TK]D-Fender: i don't think it's my connection |
| 16:12.06 | LooserOuting | [TK]D-Fender: it works better with g711 |
| 16:12.11 | gerritfromsa | In Hylafax there was some setting to change the training ... let me see if I can find it |
| 16:13.46 | gerritfromsa | My settings : ModemType: Class1 , ModemFlowControl: rtscts , ModemRevQueryCmd: ATDF%s |
| 16:13.49 | [TK]D-Fender | LooserOuting: Really shouldn't be the case. What are you running? |
| 16:14.42 | LooserOuting | [TK]D-Fender: Asterisk 11.3.0 |
| 16:14.56 | [TK]D-Fender | LooserOuting: What fax module? |
| 16:14.59 | rexwin_ | K]D-Fender, i enabled it, what should I do next |
| 16:15.04 | gerritfromsa | It's most often the legacy fax machine in my experience , try switching error correction on/OFF |
| 16:15.33 | Penguin | rexwin_: sip set debug on; then make your call and see what comes up on the console. |
| 16:15.47 | LooserOuting | [TK]D-Fender: i am using res_fax_spandsp |
| 16:15.51 | [TK]D-Fender | rexwin_: I asked what you see.... |
| 16:15.59 | [TK]D-Fender | LooserOuting: Should be good to go then.... |
| 16:16.59 | LooserOuting | gerritfromsa: i will try to disable the error correction. |
| 16:17.29 | rexwin_ | call transaction does not exist |
| 16:17.55 | gerritfromsa | And drop baudrate to 9600 |
| 16:18.01 | rexwin_ | 401 unauthorized |
| 16:18.45 | blizzow | One of my agents made two phone calls in a row where he was transferred (once via a receptionist, once via IVR selection), the callee side of the line just went silent. I can hear his voice in the recordings up to the end, but the callee side seems to be returning silence. Asterisk is running with -vvvg and I don't see anything that in the full log that indicates why his calls just went "silent". I know it's a really gene |
| 16:19.27 | [TK]D-Fender | rexwin_: PASTEBIN <=- |
| 16:19.29 | [TK]D-Fender | ~pb |
| 16:19.30 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:19.34 | [TK]D-Fender | ^^^ |
| 16:19.37 | Greenlight | blitz_: Are you using directmedia ? |
| 16:19.46 | Greenlight | * blizzow |
| 16:20.09 | gerritfromsa | LooserOuting : have you changed settings in res_fax.conf ? |
| 16:20.09 | rexwin_ | one moment |
| 16:20.30 | LooserOuting | gerritfromsa: no. they are standard |
| 16:21.16 | blizzow | directmedia? |
| 16:21.23 | blizzow | I don't think so. |
| 16:21.30 | blizzow | But I don't know what it is. |
| 16:21.45 | Greenlight | It would be configured in sip.conf |
| 16:22.14 | gerritfromsa | Why don't you use a maxrate=12000 and see if it does not allow the 12k handshake |
| 16:23.05 | LooserOuting | gerritfromsa: that's also a good idea. i will test this too. |
| 16:23.35 | LooserOuting | gerritfromsa: thank you very much. |
| 16:23.55 | *** join/#asterisk Endriuska (~ray@c-67-184-148-236.hsd1.il.comcast.net) |
| 16:24.31 | gerritfromsa | LooserOuting : Can also call it from the dialplan : exten => incoming,1,Set(FAXOPT(maxrate)=12000) |
| 16:25.45 | *** join/#asterisk JamKo (~JamKo@unaffiliated/jamko) |
| 16:29.41 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:30.22 | blizzow | Greenlight: All my clients are behind my firewall. Most of them are using Zoiper as a softphone. I do notice that each extension is configured for nat=yes even though everyone is on my LAN. Could that affect call quality? |
| 16:30.33 | rexwin_ | [TK]D-Fender, http://paste.ubuntu.com/6319152/ |
| 16:30.46 | Greenlight | blizzow: No |
| 16:30.58 | Greenlight | nat=yes should be fine for most circumstances |
| 16:30.59 | Penguin | blizzow: Extensions aren't configured for nat, devices are. |
| 16:31.07 | [TK]D-Fender | rexwin_: that looks like only the end of a call. Get complete debug.. |
| 16:31.42 | Penguin | blizzow: Is your asterisk on the same LAN with those phones? |
| 16:31.44 | Greenlight | BUt if you have directmedia enabled in sip.conf then when you did a transfer it could potentially cause the issue you describe |
| 16:33.05 | gerritfromsa | blizzow : wireshark |
| 16:33.30 | Greenlight | Even with network conditions asside (same subnet etc), i've seen endpoints with bad handling of reinivtes |
| 16:33.46 | gerritfromsa | Transfer with the # key |
| 16:34.08 | gerritfromsa | directmedia=yes is the default |
| 16:34.34 | blizzow | Penguin: yes, asterisk is on the same LAN. |
| 16:35.32 | Greenlight | blizzow: If asterisk is not in the rtp path (eg your using directmedia) then that's where my betting would be where the problem is. |
| 16:35.43 | Penguin | blizzow: If asterisk is on the same subnet as the phones, you can set nat=no for each device. If you want to keep asterisk in the media path, set directmedia=no for each device. |
| 16:37.27 | gerritfromsa | Dont know much about windows but could it be it's firewall |
| 16:37.33 | Penguin | its |
| 16:37.38 | Greenlight | :) |
| 16:37.52 | ChannelZ | That is not a sentence. |
| 16:38.01 | Greenlight | What about "don't" |
| 16:38.21 | Penguin | That one hasn't worn me down yet. |
| 16:38.29 | Greenlight | And, if the endpoints are softphones on Windows, then, yes, potentially it's a firewall issue. |
| 16:39.07 | Penguin | I'm thoroughly fed up with the ignorance of its vs. it's. It's not that hard. |
| 16:39.44 | blizzow | Greenlight or Penguin: Any recommendations for keeping asterisk in the media path (assuming all phones are on the same LAN subnet as the asterisk server). |
| 16:39.58 | Penguin | blizzow: I already told you: directmedia=no |
| 16:40.05 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 16:40.07 | Greenlight | blizzow: As Penguin said, set directmedia=no |
| 16:40.30 | Greenlight | It's a quick way of ruling out a number of potential issues. |
| 16:42.16 | Greenlight | I've seen Asterisk do a lot of weird and wonderful things with reinvites, and endpoints their fair share as well. |
| 16:43.22 | Penguin | When NAT is involved, reinvites are often a problem. |
| 16:43.49 | Greenlight | Even the usual NAT issues aside, thing are not always smooth sailing. In my experience. |
| 16:43.54 | rexwin_ | [TK]D-Fender,http://paste.ubuntu.com/6319195/ |
| 16:43.59 | Penguin | I only do directmedia between devices on the same network. If NAT is involved, I forbit directmedia. |
| 16:43.59 | rexwin_ | hope that helps |
| 16:44.26 | Penguin | s/forbit/forbid/ |
| 16:45.07 | Penguin | Sometimes my dial options prevent reinvites anyway. |
| 16:45.31 | Greenlight | Yea, listening for DTMF ? |
| 16:46.21 | [TK]D-Fender | rexwin_: <--- SIP read from UDP:10.0.0.36:5060 ---> <- call sources as local |
| 16:46.27 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 16:46.30 | Penguin | That would be one reason. Another case would be when call recording is in use. |
| 16:46.41 | [TK]D-Fender | rexwin_: rexwin_ Peer audio RTP is at port 41.177.86.38:8000 <--- but advertises its RTp as PUBLIC |
| 16:46.43 | Greenlight | mhmm |
| 16:46.54 | [TK]D-Fender | rexwin_: What is 10.0.0.36 ? |
| 16:47.31 | [TK]D-Fender | rexwin_: and pastebin another call with "core set verbose 10". We are not seeing dialplan execution there... |
| 16:47.41 | Greenlight | I'm sure the code itself makse some reference to the "stars aligning" when it goes down the route of directly bridging the RTP of the two sides |
| 16:47.42 | gerritfromsa | There's your smoking gun ... |
| 16:47.43 | Penguin | Several things prevent directmedia, but if I can use it, I use it. |
| 16:51.55 | blizzow | Penguin: Greenlight: pardon my ignorance, I would think that adding directmedia=no means I'm forcing asterisk to be in the mix and I might suffer degradation in quality. I don't know why I'd want to force asterisk to stay in the audio path. |
| 16:52.31 | Greenlight | blizzow: Why would you suffer loss of quality? |
| 16:52.46 | Penguin | blizzow: You know that media works between the phones and asterisk. |
| 16:53.02 | Greenlight | These are digital frames of media. |
| 16:53.09 | Greenlight | Asterisk will not change them, merely pass them on. |
| 16:53.29 | Greenlight | And if this is on a local network, you'll not really incur any latency |
| 16:53.59 | Greenlight | Also, provided all the other conditions are met, it'll to packet-to-packet bridging, so your RTP never even hits the Asterisk code. |
| 16:54.05 | Greenlight | s/code/core |
| 16:54.05 | gerritfromsa | blizzow : Have you tried the transfer with the # key instead of the transfer key? |
| 16:54.24 | Penguin | If you have a caching web proxy on your internet gateway, do you suffer loss of words on a web page that you load because the proxy is there? |
| 16:54.36 | blizzow | gerritfromsa: I'm saying the transfer is happening on the callee side. |
| 16:55.50 | gerritfromsa | is the callee using zoiper |
| 16:55.56 | F-JJTH | Someone can point me to a example DialPlan using Transfer() please ? |
| 16:56.50 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:57.43 | gerritfromsa | Transfer() could be used as an ivr option for example : exten => 1,1,Transfer(SIP/100) |
| 16:58.11 | F-JJTH | gerritfromsa: yes, but should I use Answer() before ? |
| 16:58.12 | boom^time | What is the best method for fax detection on a sip channel? I've read that with a tdm system it can roll to a fax extension but can this be done with sip? |
| 16:58.17 | Penguin | 100 is a terrible name for a phone. |
| 16:58.26 | blizzow | gerritfromsa: I have no idea what the callee's phone system is like. |
| 16:58.37 | gerritfromsa | I agee |
| 16:58.39 | ChannelZ | Yes.. faxdetect=yes |
| 16:58.44 | Penguin | boom^time: SIP has fax detection. |
| 16:58.47 | gerritfromsa | I agree |
| 16:59.00 | ChannelZ | (in sip.conf) |
| 16:59.01 | Greenlight | blizzow: As I understood, it was *your* agent who was doing the transferring >? |
| 16:59.12 | F-JJTH | I have: exten => 0190909090122750,n,Transfer(IAX2/0190909090122750@mydomain.org) |
| 16:59.16 | karl-s | Penguin, is that fax detection actual CNG detection?? |
| 16:59.26 | Penguin | As far as I know, yes. |
| 16:59.28 | F-JJTH | but the call is not transfered |
| 16:59.29 | Greenlight | F-JJTH: Why Transfer and not Dial ? |
| 16:59.31 | karl-s | cool |
| 16:59.32 | boom^time | ChannelZ, Penguin, thanks I must have misread this I thought it was only for zapata channels (what I get for reading old voip-info) |
| 17:00.11 | Greenlight | F-JJTH: Seems like you're trying to make Asterisk behave like a SIP proxy, which it's not. It's a B2BUA. |
| 17:00.17 | F-JJTH | Greenlight: because I want the server who execute this Transfer can be offline once the Transfer has been done |
| 17:00.49 | F-JJTH | I'm trying to make Asterisk as a call distributor to other asterisk servers |
| 17:01.23 | mjordan | F-JJTH: so, a SIP proxy? |
| 17:01.34 | F-JJTH | every client are calling 1 main asterisk server. This main asterisk server transfer each call to a dozens of other asterisk servers |
| 17:01.38 | Greenlight | I think with SIP a Transfer will send back a "redirect", but really, Asterisk isn't the tool to use here for this. |
| 17:02.09 | blizzow | Greenlight: no. My agent made a call, dialed an extension via IVR and experienced silence after the IVR attendant said it was transferring. Then on the agent's next call to a different number, they got a receptionist, requested a transfer, and experienced silence. I want to do as much diagnosis of my side as possible before chalking it up to anamoly. |
| 17:02.14 | Greenlight | I've seen a few people try, and fail, to get Transfer working how it sounds like you want to. |
| 17:02.31 | F-JJTH | ok |
| 17:02.40 | Greenlight | blizzow: So, how can that be your problem? |
| 17:02.51 | Greenlight | Your side can't detect when their IVR does something. |
| 17:02.54 | Penguin | blizzow: Your agent is calling to someone else's system and there's a problem? |
| 17:02.58 | F-JJTH | so I need a main server with a lot of bandwidth because he will manage all the flow alone :( |
| 17:03.21 | Greenlight | F-JJTH: Well, it sounds like you need a SIP proxy. |
| 17:03.29 | mjordan | F-JJTH: It really does sound like what you want is Kamailio or OpenSIPs. You may want to look at those projects, as they're very good at doing what you just described |
| 17:03.35 | Penguin | I was under the impression the calls were made TO YOUR asterisk. |
| 17:03.36 | Greenlight | +1 |
| 17:03.39 | gerritfromsa | I also got it wrong ? It now sounds like the remote system has an issue |
| 17:03.48 | F-JJTH | ok, thanks you |
| 17:03.57 | Greenlight | Yea, I was under that impression to. This is clearly an issue on the remote side. |
| 17:04.04 | Penguin | If the calls are made to someone else's phone system, that isn't your problem. |
| 17:04.11 | Greenlight | Well, assuming PSTN between yourselves and them. |
| 17:05.50 | blizzow | I agree, it just seemed weird that it happened twice in a row to the same agent to different numbers. I came here to ask if anyone could think of anything at all that could make it my side (asterisk/or sip client) the cause. |
| 17:06.10 | Greenlight | Is it PSTN between yourselves and this remote party ? |
| 17:06.30 | Greenlight | (eg you've not got some direct connection via SIP to them) |
| 17:07.04 | blizzow | I have a redfone TDMOE plugged straight into the asterisk server. So it's PSTN. No SIP direct to the client. |
| 17:07.26 | Greenlight | So, the problem is at their side. |
| 17:07.36 | gerritfromsa | If the other side is using a # to transfer and you've got the big T in your Dial statement it could actually put yourself on hold and you could get silence of your moh does not kick in |
| 17:07.37 | Penguin | Once the call leaves your system and reaches the PSTN, no more changes should occur on your box. No reinvites beyond that, nothing. |
| 17:07.49 | *** join/#asterisk rexwin__ (~rexwin@180.215.67.170) |
| 17:07.54 | rexwin__ | [TK]D-Fender, http://paste.ubuntu.com/6319331/ |
| 17:08.14 | Penguin | When the call hits the far side, they can reinvite when transferring to a different phone. |
| 17:08.22 | Penguin | But that's their problem, not yours. |
| 17:08.39 | [TK]D-Fender | rexwin__: Reliably Transmitting (no NAT) to 196.41.123.184:5060: - you are registering outbound |
| 17:08.58 | gerritfromsa | It can't be a REINVITE issue - he's connected via TDM |
| 17:09.05 | [TK]D-Fender | rexwin__: Contact: <sip:s@10.0.0.5:5060> <-- and passing them your PRIVATE IP for return contact. You have not set your server up properly to work from behind NAT |
| 17:09.16 | Penguin | gerritfromsa: I said on THE FAR SIDE. |
| 17:09.24 | [TK]D-Fender | rexwin__: localnet, externaddr,nat=yes,directmedia=no <- all required for [general] |
| 17:09.31 | gerritfromsa | But the remote side's hash could trigger the transfer |
| 17:10.01 | [TK]D-Fender | rexwin__: <--- SIP read from UDP:10.0.0.36:5060 ---> <- you still haven't answered what this is... |
| 17:10.21 | Penguin | gerritfromsa: Let's get the answer to that... |
| 17:10.35 | Penguin | blizzow: Are you using any t or T option in any of your Dial()s? |
| 17:11.26 | gerritfromsa | If he's making the call then the small t actually works for the remote party |
| 17:11.42 | Greenlight | Good call, I'd not even considered that. |
| 17:12.30 | gerritfromsa | That's how Trixbox get hacked via the inbound calls |
| 17:12.33 | Greenlight | Although I can't see it fitting for the time where it happend when his user was inside an IVR |
| 17:12.49 | gerritfromsa | Good point |
| 17:12.53 | boom^time | do I need to enable t38pt_udptl=yes for faxdetect to work? |
| 17:12.58 | Penguin | no. |
| 17:13.06 | Penguin | faxdetect=yes |
| 17:13.50 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:13.53 | Penguin | And you'll want to create the 'fax' extension in the context where your call is landing. When fax is detected, it will jump to that exten. |
| 17:13.56 | boom^time | I have that enabled for the sip gateway and I'm calling my fax machine with no luck as far as it rolling over to my fax extension |
| 17:14.27 | Penguin | If you are dialing a number that goes to a fax machine, you don't need fax detection. |
| 17:14.45 | boom^time | That's the issue, I'm calling a list of a numbers and want to know when I've called a fax machine. |
| 17:14.56 | Penguin | Fax detection is used on a "voice line" where a fax can also be sent. |
| 17:15.43 | blizzow | I'm looking for T or t in my dial command and don't see anything. |
| 17:15.44 | boom^time | I'm originating a call via AMI which drops to an extension which answers, does AMD detection, but if it sees a fax machine I'd like it to roll to the fax extnension. |
| 17:15.49 | boom^time | extensions* |
| 17:16.11 | Penguin | So you call me at 12345 and get an AA where you then enter my personal extension, and you get me on the phone. Or you send a fax to 12345 and your fax machine starts sending tones instead, it jumps to the fax exten. |
| 17:16.59 | Greenlight | blizzow: That's okay, it was a long shot. The problem is at the other side. |
| 17:17.42 | Ice_Strike | How can I merge *-out.pcm and *-in.pcm recorded files? |
| 17:17.48 | Greenlight | So can faxdetect work in tandem with another application running on the channel, in your case AMD ? |
| 17:18.27 | Greenlight | Ice_Strike: sox. |
| 17:18.56 | boom^time | Greenlight, I suppose that's the question, The context still answers the call and then drops to the AMD application. |
| 17:19.14 | Penguin | The extension answers. The context just contains extensions. |
| 17:19.17 | Ice_Strike | Greenlight What about on Windows - Instead of mergeing, play both files at one? |
| 17:19.19 | Ice_Strike | once |
| 17:19.36 | boom^time | Sorry, the s extension answers |
| 17:19.37 | Greenlight | Audacity perhaps |
| 17:19.44 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 17:20.00 | Penguin | ice_strike: Monitor() has the ability to join the in and out files together at the end of the call. |
| 17:20.12 | gusto | hi Penguin , how are you? |
| 17:20.17 | Greenlight | Or even better, MixMonitor does it as it goes. |
| 17:20.25 | *** join/#asterisk u0m3 (~u0m3@92.80.81.116) |
| 17:20.33 | Penguin | gusto: Not happy, but still breathing nevertheless. |
| 17:20.40 | Ice_Strike | I am using sox, I notice it is not in sync when it recorded. |
| 17:20.49 | Ice_Strike | It only happen when transfering the call |
| 17:20.54 | gusto | Penguin: what happend? |
| 17:20.55 | Greenlight | Ice_Strike: Yes, that's one of the problems you'll encounter |
| 17:21.39 | Greenlight | If the RTP stops at either side, for any reason, then the recording file also has that "gap" |
| 17:22.06 | Ice_Strike | Yep |
| 17:22.16 | Ice_Strike | What the solution to fix or workaround? |
| 17:22.21 | Greenlight | Use MixMonitor. |
| 17:22.36 | Penguin | gusto: A couple weeks ago, someone broke the right side mirror off my truck. It was hanging by the wire since then, while staying parked until I can fix it. Last night when I came in, someone broke the left side mirror off, too. So now I have two mirrors hanging by the power wires. I'm contemplating retaliation. |
| 17:23.39 | Greenlight | boom^time: That's certainly an interesting idea for detecting fax machines, i'd be interested to hear if you get it working. |
| 17:23.50 | Penguin | gusto: Passive aggression my ass, I'm going to break someone else's mirrors off. |
| 17:25.36 | Penguin | boom^time: So you're saying you want faxdetect on your box, but rather than on incoming calls like usual, you want it on outgoing calls? |
| 17:25.45 | boom^time | Greenlight, will keep you posted. But I'm confused as to when the detection is actually happening. Examples I see show an Answer() and then it should almost instantly detect because it drops to a Dial afterwards. So in theory it should detect before AMD is called |
| 17:26.12 | Penguin | Can you add any wait time before AMD()? |
| 17:26.32 | boom^time | Penguin, yes but really it should be the same thing. The call originates and when they answer it drops to the context which answers and then goes to AMD. So really it's the same thing as them calling me. |
| 17:26.36 | Greenlight | Just reading the docs now, it seems to listen to the audio for the CNG tone, so in theory it would get redirected to the fax extension *from* wherever it currently is |
| 17:26.56 | gusto | Penguin: why do you have a truck? |
| 17:27.08 | Ice_Strike | Greenlight Out of sync happen quite a lot sometime but not all the recorded files.. Sometime work fine and sometime dont work fine. |
| 17:27.11 | boom^time | Penguin, no that would cause too long of a delay before the detection which I'm trying to avoid awkward silence |
| 17:27.15 | Penguin | gusto: For transportation, just like everyone else. |
| 17:27.24 | gusto | Penguin: people can also use cars |
| 17:27.26 | Ice_Strike | Greenlight I dont have much issue on ISDN line... |
| 17:27.29 | Penguin | gusto: Same thing. |
| 17:27.32 | Ice_Strike | Worse on SIP truck though |
| 17:27.37 | Greenlight | Ice_Strike: Then use MixMonitor. That's what it's for. |
| 17:27.45 | gusto | Penguin: no, a truck seems to be bigger than a car |
| 17:27.51 | Ice_Strike | Can MixMonitor seperate the files in and out? |
| 17:27.55 | Penguin | gusto: Pretend it's a car if it makes you feel better. It's a Chevy Blazer. |
| 17:28.03 | Greenlight | Why do you need them separated ? |
| 17:28.33 | Greenlight | boom^time: Do you see calls being redirected to the fax extension ? |
| 17:28.36 | gusto | Penguin: so you have an SUV |
| 17:28.53 | Penguin | gusto: Sure. |
| 17:29.09 | Penguin | gusto: But what does that have to do with my anger over the vandalism? |
| 17:29.10 | Ice_Strike | In the extention file - I have Monitor(pcm,${CALLFILENAME},b) |
| 17:29.11 | boom^time | Greenlight, no, it just continues with AMD detection, I don't see any attempts at the fax extension |
| 17:29.32 | Penguin | AMD... the d meand detection. It isn't answering machine detection detection. |
| 17:29.54 | Ice_Strike | Yea I use AMD detection too, never perfect. |
| 17:30.03 | Greenlight | The accuracy of AMD is not the question here. |
| 17:30.09 | Ice_Strike | Ofcourse ;) |
| 17:30.09 | boom^time | The AMD detection isn't the issue, that I have working quite well. |
| 17:30.12 | Penguin | ice_strike: You use answering machine detection detection? |
| 17:30.24 | Ice_Strike | Yea |
| 17:30.25 | Penguin | boom^time: It is AMD, not AMD detection. |
| 17:30.27 | Greenlight | In theory it should redirect to "fax" from wherever it currently is. |
| 17:30.30 | boom^time | Yes, my detections are very redundant. |
| 17:30.36 | Penguin | I noticed that. |
| 17:30.39 | gusto | http://image.truckinweb.com/f/events/1004tr_sema_show_2009/27990340/1004tr_22%2BSEMA_show_2009%2Bchevy_blazer.jpg |
| 17:30.39 | boom^time | I need to detect that I'm detecting something :) |
| 17:30.59 | Penguin | Use AMDD so you can detect while you detect. |
| 17:31.11 | Greenlight | boom^time: What if you put it into say a Wait(20) for testing |
| 17:31.17 | Greenlight | Does it then detect the fax CNG tone ? |
| 17:31.22 | boom^time | Greenlight, okay I'm going to do that right now. |
| 17:31.58 | Greenlight | I'd guess that code is too old to use an audiohook, so there could be some gotchas to how it works. |
| 17:32.04 | Penguin | gusto: Not quite. |
| 17:32.10 | Greenlight | It may only listenw when two endpoints are bridged. |
| 17:33.10 | Penguin | gusto: http://en.wikipedia.org/wiki/File:98-05_Chevrolet_S-10_Blazer_LS_4-door.jpg |
| 17:33.27 | Ice_Strike | Greenlight I currently use Monitor(pcm,${CALLFILENAME},b) - do you think b flag might causing out of sync? |
| 17:33.38 | Ice_Strike | " b - Don't begin recording unless a call is bridged to another channel." |
| 17:33.47 | boom^time | still no, odd. I've added faxdetect=yes to the end of my sip gateway's configuration in sip.conf. It doesn't have to be under general does it? |
| 17:34.05 | gusto | Penguin: ah, that one seems to look ok |
| 17:34.07 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 17:34.12 | Greenlight | Ice_Strike: Potentially, although it's worded as "begin". One of the reasons people use MixMonitor and not Monitor is because the two sides can get out of sync. |
| 17:34.27 | Greenlight | boom^time: And done a sip reload ? |
| 17:34.39 | Penguin | gusto: Nothing special. V6, automatic, 4x4. |
| 17:34.40 | Greenlight | (Just double checking) :) |
| 17:34.48 | boom^time | Greenlight, yes. |
| 17:35.21 | gusto | Penguin: not a good combination. you do not need 4x4 that is not fuel efficient and the transmission has to be manual |
| 17:35.30 | Greenlight | Does it get triggered if you Dial() another peer insread of Answering and Waiti |
| 17:35.36 | Penguin | gusto: But the vehicle is not the point. If I broke off your outside mirrors, you would be upset. |
| 17:35.52 | gusto | Penguin: i have no outside mirrors |
| 17:36.11 | Penguin | gusto: If you put mirrors on your bicycle, and then I broke them off, you would be upset. |
| 17:36.31 | gusto | Penguin: maybe, but i do not put any mirrors on my bicycle |
| 17:36.31 | Greenlight | Why would you put mirrors on his bike and then break them off? |
| 17:36.43 | gusto | you see |
| 17:36.46 | Penguin | gusto: I'm going to put mirrors on your bicycle. |
| 17:36.52 | Greenlight | Seems very counterprductive. |
| 17:37.27 | Penguin | greenlight: He's trying to piss on my parade. He asked how I am, so I told him. He asked why I am unhappy today, so I told him. |
| 17:37.28 | gusto | cant you just lean out of the window? |
| 17:38.13 | gusto | Penguin: i have a solution for you... you have to get video surveilance so that you can see who is beaking your mirrors |
| 17:38.24 | Penguin | It's too late now! |
| 17:38.47 | gusto | ok, then put your car into a garage, so that no one has access to your car while unatended |
| 17:38.52 | Penguin | But I'm probably going to start breaking mirrors of everyone else who lives by me that might have broken them. |
| 17:39.39 | gusto | or do like Biden said. go outside your house and discharge a shotgun |
| 17:40.08 | Penguin | I don't think that will help. |
| 17:40.13 | gusto | no |
| 17:40.21 | gusto | but you may feel better |
| 17:40.26 | Penguin | I doubt it. |
| 17:40.52 | Penguin | If I knew for a fact who is responsible for it, I'd set fire to their car. |
| 17:40.55 | gusto | i would be pissed if someone broke my internet connection |
| 17:41.15 | Greenlight | boom^time: Any joy? Am looking at chan_sip.c now to see where the detection happens. |
| 17:41.20 | gusto | well, you have to push night shifts around your car |
| 17:41.20 | Penguin | Imagine they broke your internet connection, then you fixed it, then in two days they broke it again. |
| 17:41.31 | boom^time | Greenlight, about to test that |
| 17:42.02 | gusto | Penguin: ah, that sounds familiar to me ... that behaviour smells like the german telecom |
| 17:42.14 | Greenlight | And you 100% have a "fax" extension, and have dialplan reloaded since adding it? |
| 17:42.33 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-79-141.user.veloxzone.com.br) |
| 17:42.33 | Penguin | exten => fax,1,NoOp() |
| 17:43.09 | Greenlight | It would seem that one of the first things *sip_read(struct ast_channel *ast) does is check if we've detected a fax and then redirect |
| 17:43.13 | Penguin | core set verbose 3; make the call, see what happens. |
| 17:43.21 | boom^time | yes exten => fax,1,Verbose(Fax Detected) |
| 17:43.21 | boom^time | <PROTECTED> |
| 17:45.05 | Greenlight | You on Asterisk 11 ? (JUst making sure I'm looking at the same code that you're running) |
| 17:45.13 | boom^time | 11.5 |
| 17:45.23 | Greenlight | Ok cool |
| 17:45.36 | gusto | 11.5.1 here |
| 17:45.37 | gusto | :-) |
| 17:46.09 | boom^time | okay I switched it to Answer(), Dial(SIP/(local phone)), Hangup, still no detection. |
| 17:46.22 | Greenlight | Try without the Answer |
| 17:46.25 | boom^time | okay |
| 17:46.32 | Greenlight | Although looking at the code, I can't see why it's not being triggered |
| 17:46.51 | Greenlight | You shoulds see it in the CLI too: |
| 17:46.52 | Greenlight | ast_debug(1, "Fax CNG detected on %s\n", ast_channel_name(ast)); |
| 17:47.59 | Greenlight | Well, if you enable debug |
| 17:48.05 | Penguin | It works fine on 1.8, but I never started AMD() first. |
| 17:48.29 | boom^time | I'm not using AMD anymore now. |
| 17:48.42 | Greenlight | This all happens inside sip_read, *well* before hitting anything else. |
| 17:48.53 | Greenlight | AMD or not, should not matter in the slightest. |
| 17:48.56 | boom^time | right |
| 17:49.08 | boom^time | No difference without Answer() |
| 17:49.23 | *** join/#asterisk bchia (~Adium@75.76.51.204) |
| 17:49.26 | Greenlight | And you can hear the "fax" type tone, right ? |
| 17:49.29 | boom^time | just dials my local extension here after and I pick up to hear fax machine tones |
| 17:49.31 | boom^time | right |
| 17:49.34 | Greenlight | Hmm |
| 17:49.52 | Penguin | Where did you enable the fax detection? |
| 17:50.00 | boom^time | sip.conf |
| 17:50.23 | boom^time | under my sip gateway's info |
| 17:50.31 | boom^time | I'm going to put it under general now |
| 17:50.41 | Greenlight | Either should work fine. |
| 17:50.53 | Greenlight | Are you using out of band DTMF ? |
| 17:51.28 | boom^time | dtmfmode=rfc2833 |
| 17:51.34 | Greenlight | Hmm wonder if that's it |
| 17:51.55 | Penguin | I, too, use rfc2833. |
| 17:51.56 | Greenlight | Only going by the code alone here, but it would seem the CNG fax begin tone is a "type" of DTMF |
| 17:52.11 | Greenlight | if (f && f->frametype == AST_FRAME_DTMF) { |
| 17:52.11 | Greenlight | if (f->subclass.integer == 'f') { |
| 17:52.11 | Greenlight | ast_debug(1, "Fax CNG detected on %s\n", ast_channel_name(ast)); |
| 17:52.33 | boom^time | interesting, switching to inband momentarily |
| 17:52.47 | *** part/#asterisk bchia (~Adium@75.76.51.204) |
| 17:53.12 | boom^time | nope :\ |
| 17:55.01 | Greenlight | Is server busy or can you enable debug for a test ? |
| 17:55.25 | Greenlight | Also, what codec are you using? |
| 17:55.57 | boom^time | ulaw, I set core set debug 9 |
| 17:56.02 | boom^time | and I don't see much |
| 17:56.46 | Greenlight | Hmm what's "dsp" |
| 17:57.22 | boom^time | dsp? |
| 17:57.30 | Penguin | ~dsp |
| 17:57.30 | infobot | i heard dsp is (Digital Signal Processor) A DSP is a microprocessor designed to work with analog signals such as video or audio that have been digitally encoded. The DSP then takes these digital representations and performs operations on them. DSPs are used in video, sound, and modem technology. Intel's MMX instruction set is basically an attempt to make the Pentium processor line capable of DSP operations. This follows Intel's theory of putting all ... |
| 17:58.03 | Greenlight | Well, it seems that needs to be enabled on the channel for the detection to happen |
| 17:58.07 | Penguin | Too long, must truncate. |
| 17:59.47 | Penguin | To make fax detection work on my 1.8 system, I went into sip.conf and put faxdetect=yes in my peer entry. That's all it took. |
| 18:00.15 | Greenlight | And that works on outbound calls okay ? |
| 18:00.38 | Penguin | I never tested that case. It's unusual. |
| 18:01.22 | Penguin | Though if you gave me a fax number, I could test it. |
| 18:01.27 | Greenlight | Looking at the code, the direction shouldn't matter. |
| 18:01.51 | Greenlight | It does appear that DTMF needs to be INBAND or AUTO |
| 18:01.59 | Penguin | Mine isn't. |
| 18:02.06 | Greenlight | Yea, that's what's confusing me :) |
| 18:02.12 | Penguin | I always use rfc2833 for everything. |
| 18:02.16 | Greenlight | Although I've not got 1.8 code to hand to look at |
| 18:02.46 | Greenlight | Or, do you use T38 >? |
| 18:02.50 | Penguin | no |
| 18:02.53 | Greenlight | Hmm |
| 18:03.10 | Penguin | The ITSP doesn't support it, or I would. |
| 18:03.21 | Penguin | They probably will support it in the future. |
| 18:04.50 | Greenlight | boom^time: Can you PB the debug of an attempt ? |
| 18:05.20 | boom^time | sure |
| 18:06.28 | Greenlight | Can you keep DTMF set to inband as well for the test |
| 18:07.29 | boom^time | http://pastebin.com/KcxfSB8D |
| 18:07.38 | boom^time | shoot, need to redo with inband |
| 18:08.24 | Greenlight | For outbound SIP calls we seem to hit the following: |
| 18:08.24 | Greenlight | if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || |
| 18:08.24 | Greenlight | (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) { |
| 18:08.24 | Greenlight | enable_dsp_detect(p); |
| 18:08.24 | Greenlight | } else { |
| 18:08.25 | Greenlight | disable_dsp_detect(p); |
| 18:08.25 | Greenlight | } |
| 18:08.48 | Greenlight | So, it would appear that the DTMF mode must be INBAND or AUTO for faxdetect to work on outbound calls/ |
| 18:09.31 | boom^time | with inband http://pastebin.com/xdCEeCbL |
| 18:11.14 | Penguin | boom^time: Which way are you originating the call? In the originate command, the remote phone number first, then extension something on local? or reverse? |
| 18:12.02 | Greenlight | I can't see it actually being answered on that pb... |
| 18:12.24 | boom^time | remote phone first (this case fax machine) |
| 18:12.29 | boom^time | sorry I'll post a more indepth one |
| 18:12.39 | Penguin | I'm goint to test for myself. |
| 18:14.08 | boom^time | http://pastebin.com/p9NP1d9h |
| 18:18.45 | Greenlight | I can see the channel going up at 14:12:54, and then you hangup at 14:12:55 ? |
| 18:19.03 | Greenlight | It may need more than a second to detect it |
| 18:19.08 | Penguin | I did channel originate SIP/provider/a-fax-number extension my-exten@good-context, and it doesn't detect anything. |
| 18:19.41 | Greenlight | Penguin: Are you able to change the DTMF mode for a test ? |
| 18:19.49 | Penguin | Yes. |
| 18:20.00 | Penguin | What do you want me to set it to? |
| 18:20.06 | Greenlight | inband please |
| 18:20.37 | *** part/#asterisk realbass1 (~realbass1@206.82.244.9) |
| 18:20.43 | boom^time | Greenlight, I answered the phone and then hungup because at that point it's too late for a detection |
| 18:21.10 | Greenlight | boom^time: "too late" ? |
| 18:21.23 | boom^time | it already called my local extension |
| 18:21.35 | Greenlight | It works by listening to the audio... |
| 18:21.53 | Greenlight | It's only answered, when you pickup your local extension |
| 18:22.00 | Greenlight | So it can only "begin" then |
| 18:22.06 | boom^time | I see... |
| 18:22.11 | Penguin | It's not working even with inband. |
| 18:22.15 | rexwin__ | why am getting SIP/2.0 503 Unavailable |
| 18:22.27 | rexwin__ | when trying to place calls |
| 18:22.47 | Greenlight | Penguin: Hmm how odd... thanks for testing anyway though |
| 18:22.54 | Greenlight | rexwin__: Who is giving the 503? |
| 18:23.07 | Penguin | I think it's because of two channels being created. There's the channel to the ITSP, which is making the call to the fax machine; and there's the local channel which attaches to the extension where it could jump to fax. |
| 18:23.58 | Penguin | I think it is this which causes it to not work. |
| 18:24.03 | rexwin__ | sip debug info shows that |
| 18:24.23 | Penguin | On the inbound call, it is a single channel via the peer entry with fax detection enabled. |
| 18:24.27 | Greenlight | Penguin: There's shouldn't be a Local channel |
| 18:24.37 | Penguin | There has to be. |
| 18:24.39 | Greenlight | That SIP channel should be going to the dialplan |
| 18:24.53 | Penguin | channel originate SIP/stuff extension 123@context |
| 18:24.59 | Greenlight | Yea |
| 18:25.08 | Greenlight | Unless, your "context" is creating Local channel ? |
| 18:25.16 | Penguin | I believe it will use a local channel to reach context,123,1. |
| 18:25.22 | Greenlight | It shouldn't |
| 18:25.28 | Penguin | Then what does it use? |
| 18:25.38 | Penguin | Just the core all by itself? |
| 18:25.49 | Penguin | No internal local channel? |
| 18:25.51 | Greenlight | It'll wait for SIP to answer, and then dump it directly to that extension in the dialplan |
| 18:25.54 | Greenlight | Yup |
| 18:26.07 | Penguin | I know WHAT it does, but I don't know HOW it does it. |
| 18:26.15 | Penguin | I assume the local channel is used internally. |
| 18:26.24 | Greenlight | Not from what I understand |
| 18:27.10 | Penguin | I'm going to test a call using a local channel and then check one without... just to see what I can see. |
| 18:27.43 | Greenlight | From what I can see if should listen to the RTP for the "special" DTMF tone which indicates a fax |
| 18:28.06 | Greenlight | At which point that SIP channel is send to the "fax" extension |
| 18:28.22 | Greenlight | Regardless of where it currently is, or what it's currently doing (lock contension aside) |
| 18:29.04 | Greenlight | I'm going to need to get a fax number to test with ... |
| 18:29.17 | Greenlight | And throw some extra debugging in here |
| 18:30.24 | Penguin | (510) 637-6118 |
| 18:30.40 | Greenlight | I'm accross the pond |
| 18:30.49 | Penguin | oh |
| 18:30.52 | Penguin | shakes fist |
| 18:31.07 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 18:31.58 | Penguin | greenlight: 01784 437520 |
| 18:32.08 | Penguin | or +44 (0) 1784 437520 |
| 18:32.31 | Greenlight | Cool ty - I'm not going to be annoying anyone testing to that ? |
| 18:32.41 | Penguin | Royal Holloway, University of London |
| 18:33.36 | Greenlight | Hmm that'll do |
| 18:33.49 | Penguin | If they are using a fax machine that sits on someone's desk and makes a bunch of noise when you call, they need to upgrade. |
| 18:34.39 | Greenlight | Right this is going to have to be one for the morning... need to get dinner for the kids etc... Speak again tomorrow! |
| 18:34.50 | boom^time | Greenlight, thanks for your help |
| 18:36.02 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 18:37.20 | *** join/#asterisk serafie (~erin@nat/digium/x-edsqyzoowyrfjinf) |
| 18:50.19 | rexwin__ | i am unable to make outbound calls, sip debug is http://paste.ubuntu.com/6319331/ |
| 18:53.58 | [TK]D-Fender | rexwin__: Your NAT settings are STILL all wrong as I've told you before and you have not answered my previous question on that that other device is at 10.0.0.36 |
| 18:54.33 | rexwin__ | 10.0.0.36 is the windows machine where zopier is used to call outbound numbers |
| 18:54.42 | [TK]D-Fender | rexwin__: actaully, scratch that last part, I see the UA not... |
| 18:54.46 | [TK]D-Fender | now* |
| 18:55.04 | [TK]D-Fender | rexwin__: Firs your global NAT settings as I've already told you to do |
| 18:55.09 | [TK]D-Fender | fix* |
| 18:55.24 | rexwin__ | where do i do it? |
| 18:55.31 | [TK]D-Fender | rexwin__: and you still haven't enabled verbose 10 as I also indicated |
| 18:55.51 | rexwin__ | the output is verbose 10 |
| 18:55.53 | [TK]D-Fender | [13:08][TK]D-Fenderrexwin__: Reliably Transmitting (no NAT) to 196.41.123.184:5060: - you are registering outbound |
| 18:55.59 | [TK]D-Fender | [13:09][TK]D-Fenderrexwin__: Contact: <sip:s@10.0.0.5:5060> <-- and passing them your PRIVATE IP for return contact. You have not set your server up properly to work from behind NAT |
| 18:56.08 | [TK]D-Fender | [13:09][TK]D-Fenderrexwin__: localnet, externaddr,nat=yes,directmedia=no <- all required for [general] |
| 18:56.35 | [TK]D-Fender | [12:47][TK]D-Fenderrexwin_: and pastebin another call with "core set verbose 10". We are not seeing dialplan execution there... |
| 18:56.36 | anonymouz666 | [TK]D-Fender: where is the great ~sipnat? |
| 18:56.46 | [TK]D-Fender | anonymouz666: I haven't re-writen it yet... |
| 18:56.49 | [TK]D-Fender | anonymouz666: I should.... |
| 19:02.32 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 19:04.21 | gerritfromsa | rexwin " check your bindaddr= also |
| 19:18.51 | runfromnowhere | I am full of questions today - when StopMonitor is called, what happens to the X-in and X-out wav files monitor had recorded up to that point? |
| 19:19.35 | [TK]D-Fender | runfromnowhere: Nothing unless you told it to do something afterwards |
| 19:21.01 | runfromnowhere | Ah, well I have - there's a MONITOR_EXEC application defined to mix the two together |
| 19:24.59 | [TK]D-Fender | runfromnowhere: And does it execute as expected? |
| 19:26.05 | runfromnowhere | I'm not sure yet. I'll get empirical data when I can enact my changes this evening - was wondering if anyone knew before I went straight to test :) |
| 19:26.51 | gerritfromsa | The legs would still be merged if you had the m flag |
| 19:27.14 | runfromnowhere | What's the timing on that process? |
| 19:27.34 | runfromnowhere | Like...if I call StopMonitor and then afterwards execute an "rm" command to delete the file, am I in a race condition at that point? |
| 19:28.45 | gerritfromsa | if you're concerned run sleep before the rm |
| 19:29.11 | gerritfromsa | sleep 10 && rm /var/spool/asterisk/monitor/? |
| 19:29.22 | navaismo | local channels and mixmonitor with b option has issues? |
| 19:29.45 | gerritfromsa | navaismo : how so? |
| 19:30.06 | gerritfromsa | date |
| 19:30.14 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 19:30.17 | gerritfromsa | sorry wrong window |
| 19:30.33 | navaismo | seems like using Mxmonitor with local channels and using b not generate a recording |
| 19:31.20 | runfromnowhere | gerritfromsa: Actually the 'rm' is part of the dialplan...if I jam a sleep 10 in there will it cause any problems for Asterisk? |
| 19:31.33 | gerritfromsa | could be dialplan specific |
| 19:32.11 | gerritfromsa | no it will not , the System command simply puts this in bash and carries on |
| 19:32.50 | runfromnowhere | OK so the System command spawns a new process and doesn't even care |
| 19:33.12 | *** join/#asterisk opticron (~opticron@pianoben.ch) |
| 19:33.18 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 19:33.37 | gerritfromsa | I believe thats right yes |
| 19:34.04 | gerritfromsa | I use this with home automation and it works without issue |
| 19:34.22 | *** join/#asterisk dlynes (~dlynes@d72-38-18-124.commercial1.cgocable.net) |
| 19:34.38 | runfromnowhere | Hmm, I'm going to have to tweak this specific function I think, just to make sure it does it how I want it |
| 19:34.42 | runfromnowhere | but I think I have a handle on what to do |
| 19:35.09 | *** join/#asterisk TBryant (uid12962@gateway/web/irccloud.com/x-takpnccmbdkgnwju) |
| 19:35.14 | gerritfromsa | use rm specific and not with wildcard and u should be fine |
| 19:35.29 | *** join/#asterisk serafie (~erin@nat/digium/x-wkvxnghpirqmtskj) |
| 19:35.52 | runfromnowhere | Yeah, these commands try to see if the file exists with stat, but probably I can pull the STAT out and just delete regardless |
| 19:36.30 | dlynes | I'm just wondering why if an IAX2 registration request is coming in and the username/password matches that the registration is taking hold? My IAX2 debugging shows it coming in: http://pastebin.ca/2472248 (usernames and IPs scrubbed) |
| 19:36.42 | gerritfromsa | can do the same to delete unwanted cdr |
| 19:37.24 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:38.46 | gerritfromsa | dlynes : from what did you paste this ? |
| 19:40.11 | dlynes | gerritfromsa, asterisk cli |
| 19:40.32 | dlynes | gerritfromsa, i.e. iax2 set debug peer 1234 |
| 19:40.56 | gerritfromsa | see it now |
| 19:42.44 | gerritfromsa | whats your question? |
| 19:42.58 | F-JJTH | certainly stupid question but: OpenSIP is not working with IAX protocol, right ? |
| 19:43.05 | dlynes | Seems for the iax2 debug output that locking isn't being performed for access to stdout |
| 19:43.22 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 19:43.29 | dlynes | My iax2 debug output is all interleaved between channels |
| 19:43.38 | [TK]D-Fender | F-JJTH: Of course not. OpenSIP = .... ***SIP*** |
| 19:43.57 | [TK]D-Fender | F-JJTH: there is no such thing as an IAX2 proxy. |
| 19:43.58 | F-JJTH | OpenIAX doesn't exist I guess :) |
| 19:44.11 | dlynes | gerritfromsa, why my iax2 peer connection seems to be showing up as unavailable instead of registered |
| 19:44.17 | [TK]D-Fender | F-JJTH: IAX2 is not routed protocol like SIP |
| 19:44.30 | F-JJTH | [TK]D-Fender: ok, thanks you |
| 19:44.49 | dlynes | gerritfromsa, I'm seeing the corresponding poke on both ends of the conversation as well |
| 19:45.22 | gerritfromsa | Im looking at your pastebin and dont see the unavailable , do you mean the REGREQ ? |
| 19:45.40 | dlynes | gerritfromsa, the unavailable is showing up in iax2 show peers |
| 19:46.24 | gerritfromsa | UNREACHABLE ? or Unmonitored |
| 19:46.49 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:47.22 | dlynes | gerritfromsa, unmonitored would be for specified hosts (no registration), not dynamic (registration, and thus REGREQ) |
| 19:48.36 | gerritfromsa | DO you have qualify=yes in iax.conf for this peer |
| 19:49.16 | dlynes | gerritfromsa, yes |
| 19:49.53 | gerritfromsa | and this peer shows up as UNREACHABLE? |
| 19:50.31 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:50.44 | dlynes | gerritfromsa, yes, and once it's been in that state for long enough, it switches to UNKNOWN |
| 19:51.23 | gerritfromsa | is it configured as friend,peer or user |
| 19:51.29 | dlynes | gerritfromsa, i switched them from static to try to resolve a particular problem I was having that seemed to be routing related |
| 19:51.32 | dlynes | gerritfromsa, friend |
| 19:53.14 | gerritfromsa | you could add a register= user:pass@ip to your iax.conf but this is usually not a requirement |
| 19:53.43 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:53.52 | gerritfromsa | but if a host becomes UNREACHABLE it means that the POKE did not succeed |
| 19:53.57 | dlynes | gerritfromsa, that's already in there, and it is a requirement for dynamic relationships |
| 19:54.21 | dlynes | gerritfromsa, and what is the response from a 'POKE' called? Is it 'PONG'? |
| 19:54.34 | gerritfromsa | yes |
| 19:55.10 | gerritfromsa | the qualify=yes triggers this POKE - it's a layer 7 request so it goes beyond the network |
| 19:55.18 | dlynes | gerritfromsa, ok, and if a poke is not getting a pong, what the reason for that be? the poke is traversing the firewall, but a pong isn't being generated |
| 19:55.34 | gerritfromsa | NAT must be in place |
| 19:55.36 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 19:55.57 | dlynes | gerritfromsa, as i said, the poke is traversing the firewall/NAT |
| 19:55.58 | gerritfromsa | The remote side must respond |
| 19:56.22 | dlynes | gerritfromsa, but it's not generating a PONG response to the POKE request. I can see the POKE request come in |
| 19:56.46 | gerritfromsa | The defaut I think is 2000ms it will also say UNREACHABLE if the PONG > 2000ms |
| 19:57.06 | gerritfromsa | Or it is rsponding the the wrong IP |
| 19:57.12 | dlynes | gerritfromsa, if it makes any difference, one end is asterisk 11.4.0, and the other end is 1.6.2.21 |
| 19:57.42 | gerritfromsa | Thats a good question |
| 19:57.52 | gerritfromsa | using calltokens ? |
| 19:58.50 | dlynes | gerritfromsa, calltokens? |
| 19:59.19 | gerritfromsa | You need to add the calltoken parameters to your iax.conf |
| 19:59.39 | dlynes | gerritfromsa, on the 11.4, or the 1.6.2.21 end? or both? |
| 19:59.48 | dlynes | gerritfromsa, calltoken=yes? |
| 20:00.22 | gerritfromsa | Try this : requirecalltoken=no , calltokenoptional=0.0.0.0/0.0.0.0 |
| 20:01.01 | gerritfromsa | If I remember correctly 1.6 already required this - check you iax.conf sample file |
| 20:02.48 | gerritfromsa | Good luck , I have to go sort something out ... later |
| 20:18.43 | *** join/#asterisk dxd828 (~dxd828@88-105-225-228.dynamic.dsl.as9105.com) |
| 20:26.40 | *** join/#asterisk fprior (b33c66e3@gateway/web/freenode/ip.179.60.102.227) |
| 20:35.53 | Endriuska | any freelancer around? i need help upgrading asterisk to 1.6 from 1.15 |
| 20:36.25 | WIMPy | You are searching for a historian. |
| 20:36.43 | Endriuska | ;D |
| 20:38.40 | Endriuska | what happened is, i tried upgrading php5.1 to 5.2 and it updated the entire server instead, and rolled back asterisk to 1.115 so i need somehow to upgrade it |
| 20:38.42 | Endriuska | any suggestions? |
| 20:39.30 | WIMPy | First of all I'd go for a current versions. The rest depends on how complicated your setup is. |
| 20:39.45 | Penguin | Hmm. There is no asterisk 1.15, so finding someone to do that is going to be near impossible. |
| 20:40.34 | WIMPy | I don't know anything <1.2 so I have no idea. |
| 20:41.03 | Penguin | 15>2 |
| 20:42.03 | WIMPy | But 1.2>1.15 |
| 20:42.08 | WIMPy | Depends on how you read it. |
| 20:42.09 | Penguin | no |
| 20:42.21 | Penguin | We're talking version numbers, not decimals. |
| 20:42.26 | boom^time | 1.2>1.1.5 |
| 20:42.31 | Penguin | correct |
| 20:43.37 | Penguin | 1.1, 1.2, ..., 1.9, 1.10, ... 1.19, 1.20 |
| 20:44.07 | Penguin | Now if we're talking arithmetic, 1.2 = 1.20 |
| 20:45.25 | boom^time | which is why it's counterintuitive to think 1.15>1.2 |
| 20:45.27 | Penguin | Was there an asterisk 1.1.5? |
| 20:46.06 | Penguin | It may be counterintuitive if all you think about is math. In versioning, 1.15 has always been greater than 1.2. |
| 20:46.46 | Penguin | Version, pronounced one point fifteen. |
| 20:46.49 | carrar | why upgrade to 1.6 |
| 20:47.01 | Penguin | In math, pronounced one point one five. |
| 20:47.24 | boom^time | Of course it has. But I see .15 as 3/20 and .2 as 1/5 |
| 20:47.27 | WIMPy | I can't read the pronounciation. |
| 20:48.48 | Penguin | But version numbers are not consistent with mathematics, fractions, decimals. You've got to get out of that mentality to understand why a version 1.15 is higher than version 1.2. |
| 20:49.14 | Penguin | In versions, 1.2 != 1.20 |
| 20:49.21 | Penguin | In math, 1.2 = 1.20 |
| 20:49.29 | boom^time | I understand versioning penguin, I was just noting how it can be counterintuitive for some. |
| 20:49.49 | Penguin | Oh, alright. |
| 20:50.08 | boom^time | I think asterisk 1.2 should be called asterisk 6/5th's Really throw em off. |
| 20:50.29 | Penguin | I understand that they look like numbers used in math. But I also understand that they aren't as they appear. |
| 20:50.33 | boom^time | That was meant to be taken facetiously. |
| 20:53.57 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
| 20:55.49 | *** join/#asterisk sectech (~sectech@192.241.140.204) |
| 21:03.35 | boom^time | can I set a variable in a context without using Set like I can a global with just BLAH=hi? |
| 21:04.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:04.15 | WIMPy | How do contexts and variables fit together? |
| 21:05.02 | boom^time | WIMPy, sometimes I forget the scope of variables in a dialplan. I like to think of a context as almost like a function |
| 21:06.19 | mjordan | If I squinted hard and tried to apply C like constructs to the dialplan, extensions would be closer to functions, while contexts would be closer to a translation unit |
| 21:07.00 | mjordan | but it's an analogy that falls apart rather quickly. |
| 21:08.30 | Penguin | boom^time: No. To the best of my knowledge, you can only set variables in extensions or in the globals section. |
| 21:08.48 | Penguin | Contexts are nothing more than a group of extensions. |
| 21:09.10 | Penguin | A sort of container. |
| 21:11.27 | Penguin | I learned a hard lesson a couple days ago regarding contexts. |
| 21:11.54 | [TK]D-Fender | Penguin: After all this time? What was it? |
| 21:12.05 | Penguin | I can define two contexts with the exact same name and the extensions from each will be "merged" into the single context name. |
| 21:13.19 | [TK]D-Fender | Penguin: Yeah, just keeps adding in as though it were continuous |
| 21:13.44 | Penguin | I had inadvertently created a context which I had already created, and in it I had a different set of extensions from what was in the original instance of it. When I ran dialplan show <context>, it shows all of the extens collectively. |
| 21:14.29 | Penguin | It took me a couple minute to stare at that trying to figure out why extensions were listed that I knew my context did not have in it. |
| 21:14.35 | Penguin | Then I found the duplicate name. |
| 21:15.20 | Penguin | It didn't cause any major problems, but I'm sure there could be a case where that could cost someone lots of money. |
| 21:16.01 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 21:16.04 | Ice_Strike | How well does Asterisk work on ESXi box, let say 100 channels? |
| 21:20.02 | navaismo | depend on how many resources you give and the processing stuff in the pbx side |
| 21:21.46 | Ice_Strike | I can give plenty of resources CPU and Ram |
| 21:22.18 | Ice_Strike | but with ESXi box - there might be slow network latency? |
| 21:22.27 | navaismo | define plenty |
| 21:23.05 | Penguin | Will a high number of calls be CPU-bound? I don't see how it could be memory-bound; the RAM gets used when the dialplan is loaded. |
| 21:24.26 | boom^time | Can you call a macro or subroutine which will jump back to extensions in the context it was called from? |
| 21:24.48 | Ice_Strike | With a VM setting, I could set to 4GB and maybe 3 cores per socket |
| 21:25.12 | Ice_Strike | E3-1290 V2 CPU |
| 21:25.29 | Penguin | boom^time: With GoSub(), absolutely. GoSub() does not require context nor extension; at a minimum, it needs a priority or label. |
| 21:26.20 | [TK]D-Fender | both jump back to where they are called from.... |
| 21:26.38 | [TK]D-Fender | Gosub requires the retun app thgouh |
| 21:26.40 | Penguin | I don't think his wording was ideal. |
| 21:26.46 | boom^time | so if I have context 'main' with the exten 1234, and call with gosub the sub 'second', and within that the exten 1234 is hit it will bounce back to the main and to that exten? |
| 21:26.53 | Penguin | I think he meant go there, not jump back. |
| 21:27.13 | [TK]D-Fender | boom^time: the continue where they left off. |
| 21:27.19 | Penguin | Next priority. |
| 21:27.25 | [TK]D-Fender | boom^time: yup |
| 21:27.29 | boom^time | main with exten 5, call gosub, within that gosub there is a background playing, someone dials 5, it will jump to main exten 5. |
| 21:27.44 | boom^time | okay perfect, thanks guys. I was using #includes and it was getting pretty nasty. |
| 21:27.47 | Penguin | Yes, but at exten 5's next priority. |
| 21:28.00 | boom^time | what do you mean its next priorty? |
| 21:28.18 | boom^time | say we are in exten s under the context main |
| 21:28.20 | [TK]D-Fender | boom^time: it is not a one-way trip.... |
| 21:28.21 | Penguin | If the sub is called at exten 5 priority 2, the point of return will be exten 5 priority 3. AKA, the next line in the extension. |
| 21:28.28 | *** join/#asterisk fprior (c9fcbea8@gateway/web/freenode/ip.201.252.190.168) |
| 21:28.29 | boom^time | exten s jumps to 'second' with gosub |
| 21:28.41 | boom^time | they dial 5 during a background, will it jump ot mains 5? |
| 21:28.56 | [TK]D-Fender | boom^time: huh? |
| 21:29.00 | Penguin | That isn't a feature of GoSub. |
| 21:29.02 | boom^time | let me try to explain it more clearly |
| 21:29.08 | [TK]D-Fender | boom^time: just try coding it |
| 21:29.08 | Penguin | If they enter 5, it will go to extension 5. |
| 21:29.14 | boom^time | [TK]D-Fender, good call. |
| 21:29.27 | Penguin | If you are in extension s... if they enter 5, it will go to extension 5. |
| 21:29.28 | fprior | Hi all, I need some new elucidation, with Asterisk and FAX, if possible |
| 21:29.33 | boom^time | if they dial 5 from within the gosub it will go to exten 5 back in the main. |
| 21:29.43 | Penguin | Not back, but to. |
| 21:30.00 | boom^time | right. okay. |
| 21:30.24 | [TK]D-Fender | boom^time: You should be very careful about your use of IVR's there |
| 21:30.25 | Penguin | If you are in extension s, and sitting in a BackGround(), entering 5 will run extension 5 in the context that you are already in. |
| 21:30.51 | [TK]D-Fender | boom^time: Because you are starting to defeat the point of gosub's, which is to pass variables, and to be able to jump back to where it was called |
| 21:30.59 | boom^time | Penguin, true, but if that background is in the subroutine called from gosub in exten s |
| 21:31.15 | Penguin | The BackGround() won't care about that. |
| 21:31.28 | Penguin | It will go to extension 5 in the current context. |
| 21:31.50 | boom^time | [TK]D-Fender, I just have a chunk of IVR that I want to reuse regularly, but I need certain extensions to be completely different depending on the context |
| 21:32.10 | Penguin | That's exactly what contexts are for. |
| 21:32.15 | [TK]D-Fender | boom^time: What "chunk"? |
| 21:32.30 | [TK]D-Fender | boom^time: And this looks like a job for INCLUDE , not Macro/Gosub |
| 21:32.43 | boom^time | [TK]D-Fender, that's what I've been using, wanted to simplify it. |
| 21:33.15 | [TK]D-Fender | boom^time: Go code what you have in mind and feel free to show us after. Right now your concepts are too vague |
| 21:33.30 | mjordan | boom^time: so examples of GoSub are in this dialplan: http://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/pbx/dialplan/configs/ast1/extensions.conf |
| 21:33.30 | boom^time | Okay, sounds good. |
| 21:33.49 | Ice_Strike | [TK]D-Fender *sometime* I get out of sync from recorded files via Monitor() (*-in.pcm, *-out.pcm) only when transferring the call. What might be causing that? |
| 21:34.05 | [TK]D-Fender | Ice_Strike: nope |
| 21:34.09 | Penguin | Subroutines are intended to leave the current spot in the dialplan execution, go somewhere else, run some other bits of dialplan, then return back to where you left off originally. |
| 21:34.19 | mjordan | shrugs |
| 21:34.20 | mjordan | best one I could come up with that actually shows a GoSub getting used |
| 21:35.41 | Penguin | And they are most useful, in my opinion, when multiple sections of dialplan need to run the exact same routines but you don't want to duplicate a whole mess of dialplan lines. |
| 21:35.42 | Ice_Strike | [TK]D-Fender What nope? |
| 21:35.49 | [TK]D-Fender | Ice_Strike: no idea |
| 21:35.54 | Ice_Strike | oh |
| 21:36.32 | Ice_Strike | Maybe RTP didnt get connected or something |
| 21:37.18 | navaismo | or bad timing source |
| 21:37.35 | boom^time | Penguin, right that's really my motivation behind it. I've been using #includes |
| 21:37.54 | navaismo | Hi fprior o/, i dont know much about FAX but ask and somebody will answer |
| 21:38.28 | Ice_Strike | navaismo yea, how do I look into that and workaround? |
| 21:42.20 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 21:46.16 | navaismo | first checking which timing source are you using... then IDK :P |
| 21:46.30 | fprior | in an asterisk fax pass-thru mode, which parameters I need to check/change to configure the environment ? |
| 21:48.46 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:50.29 | *** join/#asterisk Free99 (~Free99@pool-96-246-68-150.nycmny.east.verizon.net) |
| 21:53.37 | Ice_Strike | navaismo well I use voip sip trunk |
| 21:57.26 | Endriuska | whats the most recent asterisk version? |
| 21:57.35 | Qwell | Endriuska: /topic |
| 21:57.54 | *** part/#asterisk Endriuska (~ray@c-67-184-148-236.hsd1.il.comcast.net) |
| 21:57.54 | *** join/#asterisk Endriuska (~ray@c-67-184-148-236.hsd1.il.comcast.net) |
| 21:58.01 | Endriuska | 11.6 |
| 21:58.21 | Endriuska | how old is a 1.6 version then? |
| 21:58.51 | Endriuska | its like 2-3 years old? |
| 21:59.54 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jnqlzegmbhcqccfr) |
| 22:00.06 | Penguin | 1.6 isn't a version. |
| 22:00.18 | Penguin | 1.6 isn't even a branch, for that matter. |
| 22:03.16 | Penguin | The latest version from a 1.6.x branch is at least 1.5 years old. |
| 22:03.41 | Penguin | 1.6.0 branch went EOL three years ago. |
| 22:04.06 | Penguin | 1.6.1 branch went EOL 2.5 years ago. |
| 22:04.14 | Penguin | 1.6.2 branch went EOL 1.5 years ago. |
| 22:09.12 | Ice_Strike | what the latest version of 1.6.x ? |
| 22:09.59 | Endriuska | it came with elastix |
| 22:10.05 | Endriuska | which i need to revert it back to |
| 22:10.16 | Endriuska | i accidently upgraded to 11.5, and i need to go back 1.6 |
| 22:11.50 | Penguin | ice_strike: 1.6.what? |
| 22:13.23 | Penguin | The last version of the 1.6.x branches were: |
| 22:13.25 | Penguin | 1.6.0.28 |
| 22:13.28 | Penguin | 1.6.1.25 |
| 22:13.31 | Penguin | 1.6.2.24 |
| 22:14.03 | Ice_Strike | Cool |
| 22:14.41 | Penguin | And I never used a single one of them. Ever. |
| 22:15.08 | Ice_Strike | old pbx at work use 1.4.36 |
| 22:15.10 | Penguin | I held on to 1.4 until like 1.8.4.0-ish. |
| 22:15.38 | Penguin | I used 1.4 branch for a long time. It served me well and I learned a lot from it. |
| 22:16.15 | Ice_Strike | I could try upgrading to 1.8 but not it going to break, need to test on test box |
| 22:16.24 | Ice_Strike | not sure* |
| 22:16.32 | *** join/#asterisk dxd828 (~dxd828@88-105-225-228.dynamic.dsl.as9105.com) |
| 22:17.05 | Penguin | Unless you have special scripts and things that use specific 1.4 syntaxes, going from 1.4 to 1.8 is fairly painless. |
| 22:17.26 | Penguin | I had only minor issues when I switched. Mostly dialplan syntax changes. |
| 22:17.28 | Ice_Strike | The scripts use AMI mostly. |
| 22:19.57 | *** part/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 22:21.44 | [TK]D-Fender | Endriuska: Why would you have to go back to a branch that old exactly? |
| 22:22.49 | WIMPy | He said the e-word. |
| 22:24.54 | Free99 | hey guys, how do I check if asterisk is still talking to my database? I had to restart the db to do an upgrade |
| 22:26.50 | *** join/#asterisk monsterco (~monsterco@bas6-toronto47-1176197537.dsl.bell.ca) |
| 22:27.21 | monsterco | A bit unreleated but to be used to maintain Asterisk server - Any recommenditions for a good opensource ticketing system? |
| 22:28.49 | Free99 | monsterco: I'd really suggest osticket |
| 22:29.25 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 22:29.29 | Free99 | you may need to add some of the functions various users have made by themselves, but its rock solid and real simple |
| 22:29.40 | Free99 | simple in the good way I mean |
| 22:29.47 | monsterco | Free99 - thank - you mean there are modules? |
| 22:30.16 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:30.38 | Free99 | if you can call third-party developed php scripts modules, then yeah. the one I like best gives metrics on type of calls and stuff (all configurable, so it can focus on resolution time, etc) |
| 22:30.45 | monsterco | I think it's important for me to be able to backup, re-deploy quickly in event of crash, have support for Redhat distros, and allow for quick glance overview of tickets |
| 22:31.15 | monsterco | I see - so the third-party scripts are PAID or available on the site in a store? |
| 22:31.36 | Free99 | there are some of both, but mostly free ones in the forums |
| 22:31.45 | monsterco | I also like to integrate Asterisk with it in future - can there be a pop-up type of feature for it? |
| 22:32.26 | Free99 | monsterco, the thing is open source.. you may have to code that part yourself ;) |
| 22:32.43 | Free99 | http://opensourcehunter.com/2010/03/12/9-free-open-source-ticket-sytems/ |
| 22:32.51 | Ice_Strike | Does "show agents" command work in 1.8? |
| 22:33.12 | monsterco | Free99 - I may not mind to pay for their PAID version as well - the cloud one - does that have same features as the opensource one? |
| 22:34.01 | Free99 | dunno.. I worked for a startup to implement this, we went with free everything and I contributed the (usually minor) changes I made back if they would accept them |
| 22:34.33 | monsterco | I see - maybe a $GET or $POST to pop-up a screen? I think I used it once while back |
| 22:35.17 | monsterco | If I remember correctly e-mail address was the id rather than phone numbers - so in case of a CLID coming in, most likely we can't create a record and match it unless extensive coding is done?! is that safe to say? |
| 22:42.51 | *** join/#asterisk Preatorian (~Preatoria@546906A1.cm-12-2a.dynamic.ziggo.nl) |
| 22:50.42 | Free99 | monsterco, not sure |
| 22:50.55 | Free99 | just experiment, what's the worst that could happen |
| 22:51.22 | ghost75 | death |
| 22:51.46 | WIMPy | That's not the worst. |
| 22:52.03 | WIMPy | There's always humanitarian options. |
| 22:52.04 | Free99 | zombies are worse |
| 22:53.31 | monsterco | Free99 - time constraints - isn't that always the prob?! |
| 22:54.01 | Free99 | extensive coding is a relative term |
| 22:56.38 | Free99 | monsterco, a simple search for the callerid based on whether it's a number or email-esque address should be pretty trivial |
| 22:56.56 | Free99 | once you can get the raw data, then worry about integrating it into the ticket system |
| 22:58.04 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 22:58.05 | Free99 | have your asterisk system run an AGI (I use php) and then have the php script do a database lookup or record creation based on the details asterisk passes to it |
