| 00:00.21 | kresp0 | yes, WIMPy, enum uses dns records |
| 00:00.42 | kresp0 | Guys, the result with debug and verbose says me almost nothing: http://pastebin.com/tWz7eMWU |
| 00:00.55 | kresp0 | oh damit! |
| 00:01.10 | kresp0 | Power button pressed |
| 00:01.10 | kresp0 | The system is going down for system halt NOW! |
| 00:01.24 | kresp0 | someone has pushed the power button!! omg why? |
| 00:01.56 | WIMPy | Wire them so that you have to press both the reset and the power button simultaneousely. |
| 00:02.18 | kresp0 | oh nice idea WIMPy |
| 00:02.36 | WIMPy | Oh... |
| 00:02.48 | WIMPy | just remembers 'keys show'. |
| 00:03.37 | kresp0 | That server off means that the only one free phone booth on Madrid streets is out of service now |
| 00:03.50 | kresp0 | well free for the users |
| 00:04.58 | WIMPy | Is that some sort of advertisement? |
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| 00:10.06 | kresp0 | :D |
| 00:12.30 | kresp0 | ok is up again |
| 00:14.22 | kresp0 | leifmadsen, thank you for writing that book |
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| 00:32.55 | kresp0 | BTW WIMPy, "keys show": 0 known RSA keys. |
| 00:33.32 | kresp0 | when doing " module reload res_crypto.so", it didn't load any keys |
| 00:34.45 | WIMPy | No messages? |
| 00:34.50 | kresp0 | i can see the other peer OK when doing "dundi show peers" |
| 00:35.03 | kresp0 | chi*CLI> module reload res_crypto.so |
| 00:35.04 | kresp0 | <PROTECTED> |
| 00:35.04 | kresp0 | chi*CLI> |
| 00:35.53 | WIMPy | Yes, the ping packets don't use uncryption it seems. |
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| 00:42.47 | kresp0 | after removing the lines "include = priv" and "permit = priv" from dundi.conf, now I get "DUNDi Query EID returned no results." http://pastebin.com/PV0bzrbb |
| 00:43.13 | kresp0 | and still 0 keys with keys show |
| 00:44.55 | WIMPy | Are you sure it's the right dir? 'core show settings'. And public and private ones are in the right place and have the right names? |
| 00:46.03 | kresp0 | WIMPy: seems that you're right with the right dir |
| 00:46.16 | kresp0 | <PROTECTED> |
| 00:46.36 | kresp0 | is that the directory where I should put the keys? |
| 00:47.41 | WIMPy | I've got both types in the same dir. |
| 00:47.53 | WIMPy | So I guess that means yes. |
| 00:49.31 | kresp0 | leifmadsen, on asteriskdocs.org you say that the keys should be on /var/lib/asterisk/keys but I'm using asterisk from the asterisk.org debian repos and this dir should be /usr/share/asterisk/keys |
| 00:50.20 | WIMPy | As with everything on *nix you can change everything if you like to. |
| 00:51.06 | kresp0 | may I change that book? |
| 00:51.27 | kresp0 | :P |
| 00:51.54 | WIMPy | Your local copy. |
| 00:52.22 | kresp0 | heh |
| 00:55.56 | mzb | [TK]D-Fender: fwiw, freedoh doesn't screw up an install |
| 00:56.04 | kresp0 | you're right again WIMPy. I should modify my local copy and send the diff to the author. But I'm not that good using english, instead I tell the author and hope that someday it get fixed |
| 00:56.46 | WIMPy | For me the contents of the book are correct. |
| 00:57.04 | WIMPy | Maybe you sould tell debian guys not to mess things up? |
| 00:58.54 | kresp0 | maybe I should tell the asterisk.org guys, as the package comes from http://packages.asterisk.org/deb |
| 00:59.04 | mzb | or stop producing code that is distro-dependent? |
| 00:59.38 | WIMPy | Either do a default install or know what you're doing. |
| 01:00.24 | kresp0 | default install = download & compile every security update ? |
| 01:01.19 | WIMPy | If you were interested in updated, you wouldn;t use debian, would you? |
| 01:01.35 | mzb | nor centos for that matter ;p |
| 01:02.01 | kresp0 | I'm interested on security updates. why not use debian? |
| 01:02.25 | kresp0 | their security team does a great work |
| 01:03.05 | WIMPy | At the time you get your security fix for debian, the rest of us have long forgotten about the issue. |
| 01:03.22 | mzb | centos fanboy |
| 01:03.51 | WIMPy | Oh, and as we learned two weekends ago, don't use distro-specific kernel. |
| 01:03.58 | WIMPy | That was RH, BTW. |
| 01:04.36 | kresp0 | and what os are you using? |
| 01:05.02 | WIMPy | You mean distro? |
| 01:05.12 | kresp0 | yes |
| 01:05.17 | WIMPy | "If it matters, it's the wrong one." |
| 01:05.48 | WIMPy | I'm using good old Slackware most of the time. |
| 01:06.08 | kresp0 | and slack security team is faster that the debian one? |
| 01:06.54 | kresp0 | or how do you have your security updates? |
| 01:06.55 | WIMPy | I would have guessed, everyone is. |
| 01:07.41 | WIMPy | If you really care, you have to do it yourself anyway. |
| 01:12.03 | mzb | so 'manual security' better than 'slow security' ... doesn't sound very scalable |
| 01:12.53 | WIMPy | Either you have to do the work or have to live with what others provide you with. That's just the way it is. |
| 01:15.37 | mzb | that's to be expected |
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| 01:29.15 | mzb | so apart from having a dig at Debian, your comment was pointless |
| 01:29.38 | mzb | kresp0: in other words, Debian is a sane choice for a stable distro |
| 01:29.59 | mzb | is ALSO biased |
| 01:30.03 | kresp0 | :D |
| 01:31.41 | WIMPy | If you like to waste your time with issues that have been fixed years ago. |
| 01:38.28 | ectospasm | WIMPy: depends on the issues, really |
| 01:39.49 | ectospasm | I wouldn't use Debian stable on a server, and also use the Debian stable packages for critical software, since it is ipso facto well behind current release... |
| 01:41.19 | mzb | by definition |
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| 01:41.54 | mzb | sounds like neither of you use firewalls |
| 01:46.15 | ectospasm | mzb: heh, I intend to use some form of Debian on a router/firewall box I'm gonna make this summer. |
| 01:46.28 | ectospasm | depends on what version OSSIEM needs. |
| 01:47.17 | kresp0 | good night everyone, I'm going to sleep |
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| 02:13.27 | nny | can asterisk follow symlinks in the sound directory? |
| 02:14.41 | ectospasm | nny: I don't see why not, but I'm not in the position to test right now. |
| 02:15.17 | nny | yeah nm, that's not it. I have a file in /var/lib/asterisk/sounds/testfolder/test.ulaw, but it doesn't see it |
| 02:15.34 | nny | the file is readable by all, but not owned by asterisk |
| 02:16.46 | ectospasm | nny: is it a file, or a symlink? |
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| 02:22.39 | nny | ectospasm, using a file now, it seems to be permission issues, but the file is readable by all and the directory it is in has full perms |
| 02:24.11 | ectospasm | yeah, I don't know if Asterisk follows symlinks. |
| 02:25.00 | nny | ectospasm: yeah not a symlink now |
| 02:25.37 | nny | ectospasm: it's odd, i blew the permissions open yet nothing, looking deeper. The error is very specifc |
| 02:31.02 | mzb | ectospasm: you don't like a purpose-built distro like openwrt? |
| 02:31.11 | mzb | iirc they have an i386 build target |
| 02:32.20 | ectospasm | mzb: I'm using ddwrt on my venerable LinkSys W54GL or whatever it's called |
| 02:32.30 | mzb | hmm ... not open source |
| 02:32.38 | ectospasm | mzb: I want OSSIEM, which requires 64bit Debian |
| 02:33.27 | mzb | you could have the majority of those tools using openwrt |
| 02:33.37 | mzb | why the 64bit requirement? |
| 02:34.52 | ectospasm | mzb: OSSIEM apparently works much better in 64bit for whatever reason. |
| 02:35.18 | mzb | sounds like FUD ... although I do agree that 64bit is better than 32bit ;) |
| 02:35.51 | mzb | either way ... a router-specific OS would surely be better for the purpose? |
| 02:36.30 | ectospasm | mzb: no, the purpose is OSSIEM, with the router stuff just as a bonus. |
| 02:36.59 | ectospasm | it will be a summer-long project I'm sure. |
| 02:37.21 | mzb | hmm |
| 02:37.25 | ectospasm | The answer to why is the same as it's always been: Because it was there. |
| 02:37.38 | ectospasm | I also feel constrained by BusyBox. |
| 02:37.45 | mzb | hehehe |
| 02:38.02 | ectospasm | ...and I'm not about to roll my own BusyBox |
| 02:38.37 | mzb | download openwrt, make menuconfig, choose options, build |
| 02:38.42 | mzb | how hard is that? |
| 02:39.18 | mzb | btw: openwrt is 'there' too ;) |
| 02:39.58 | ectospasm | mzb: but OSSIEM won't run on openwrt. You seem to be missing this very crucial point. |
| 02:40.25 | mzb | I do not miss anything ... the majority of the apps that I see on their list are available as options in openwrt |
| 02:40.49 | ectospasm | majority.. not entirety |
| 02:41.20 | mzb | *AND* it creates a hardened (*cough*), *small* and *replaceable* system which is easy to test in a vm |
| 02:43.24 | mzb | combine that with LVM and you've got revisioning built into filesystem images themselves |
| 02:43.43 | ectospasm | what does this have to do with Asterisk? |
| 02:43.57 | mzb | *cackle* |
| 02:44.03 | mzb | wondered how long it would take |
| 02:44.11 | ectospasm | I thought we were in #debian |
| 02:44.27 | mzb | *you* diverted the initial comments TWICE ... I'm simply following your lead |
| 02:44.44 | mzb | <ectospasm> I wouldn't use Debian stable on a server, and also use the Debian stable packages for critical software, since it is ipso facto well behind current release... |
| 02:44.48 | mzb | <ectospasm> mzb: heh, I intend to use some form of Debian on a router/firewall box I'm gonna make this summer. |
| 02:44.53 | ectospasm | so.. it's befitting I get us back on track, no? |
| 02:44.58 | mzb | sure! |
| 02:45.13 | mzb | now that you've fizzled out ;p |
| 02:45.51 | mzb | ps: asterisk also runs on openwrt, fwiw! |
| 02:46.39 | mzb | which is handy for 3G data+voice dongles, etc |
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| 03:41.40 | BeeBuu | hello,all. how can i record the AMI events to a log file? |
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| 06:02.20 | BeeBuu | hello,all. how can i record the AMI events to a log file? |
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| 06:04.35 | shamelessn00b | [TK]D-Fender: there? |
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| 06:25.36 | schmidts | good morning |
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| 06:29.09 | stev3n | I noticed that asterisk periodically reboot, but do not see scripts in cron, what could be wrong? |
| 06:29.19 | stev3n | also, I find strange function in /etc/init.d/asterisk |
| 06:29.26 | stev3n | asterisk_run_loop() |
| 06:29.32 | stev3n | I want to control restart itself, what can I do? ) |
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| 06:33.00 | din3sh | DEBUG[17023] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) |
| 06:33.08 | din3sh | what wold this mean? |
| 06:33.11 | din3sh | *would |
| 06:35.40 | kaldemar | stev3n: such a function does not exist in the init scripts that are shipped with asterisk. change your init script. |
| 06:36.06 | kaldemar | din3sh: do you have a timing module loaded? |
| 06:37.07 | din3sh | internaltiming= in asterisk.conf? |
| 06:40.05 | din3sh | internal_timing = yes |
| 06:40.18 | kaldemar | internal_timing is not yes or -I option is not given. |
| 06:40.32 | kaldemar | see "man asterisk" on option -I |
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| 06:40.45 | din3sh | ok |
| 06:40.47 | din3sh | will do |
| 06:44.52 | din3sh | kaldemar: can you see anythg abnormal in here ---> http://paste2.org/p/1994009 |
| 06:46.12 | din3sh | double legged transfers end up with music on hold one side, voice/audio on the other |
| 06:47.21 | kaldemar | i'm in such shortage of time at the moment that i can't parse through your 7k lines of debug. :P |
| 06:48.23 | din3sh | hlol |
| 06:48.46 | din3sh | nvrmind, but what could be the problem ? |
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| 06:50.04 | din3sh | simplified scenario is= 605---calls----> 601---Attended Transfer to--->603---Attended Transfer-->604: 604 only hears MOH playing while 605 can hear 604 talking. |
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| 06:53.47 | schmidts | din3sh any clients behind nat? |
| 06:54.01 | schmidts | and any clients of them snom? |
| 06:54.12 | din3sh | no nat |
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| 06:54.47 | din3sh | problem with snom yes |
| 06:55.14 | schmidts | i had a little bit different problem but its like this with some snom360 devices but this only happens after some time like 5 to 25 minutes when a call was transfered from a snom that one side suddenly hears moh while the other side still hears the normal talking |
| 06:55.54 | schmidts | but this happens also after the first transfer, which would be 605 talking to 603 and only after some time, not at the beginning |
| 06:56.34 | shamelessn00b | Hey guys, how can I stream audio out from a dialplan that is executing chanspy app? |
| 06:59.25 | BeeBuu | hello,all. how can i record the AMI events to a log file? |
| 06:59.59 | din3sh | schmidts: i have a bunch of issues with snom phones |
| 07:00.01 | din3sh | :/ |
| 07:00.36 | din3sh | one of the issues being music on hold when call is transfered from snom to an atcom, then another atcom |
| 07:00.45 | din3sh | how did you resolve your issue? |
| 07:01.41 | din3sh | the final recipient (atcom) hears moh, the number which called through the snom hears the atcom talking |
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| 07:09.11 | ChannelZ | shamelessn00b: eh? |
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| 07:10.55 | din3sh | schmidts how did you resolve your snom problem? |
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| 07:34.15 | qakhan | how long i can place timeout in queue.conf |
| 07:39.20 | BeeBuu | hello,all. how can i record the AMI events to a log file? |
| 07:39.30 | BeeBuu | no one here now? |
| 07:47.55 | kaldemar | BeeBuu: you'll have to make something that logs into AMI and listens for events, writing them into a file. |
| 07:50.57 | BeeBuu | kaldemar: i tried.but sometimes, my promgramm exited without any error message. |
| 07:52.21 | kaldemar | BeeBuu: time to fix your program... |
| 07:52.42 | BeeBuu | kaldemar: any example? |
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| 07:54.03 | kaldemar | it's goes the other way around. you show your program, tell what you think is wrong with it and someone takes a look at it and gives you suggestions on how to improve it. |
| 07:55.23 | Faustov | typo in topic |
| 07:56.00 | BeeBuu | ok,please wait |
| 07:56.11 | BeeBuu | ~paste |
| 07:56.11 | infobot | methinks paste is http://pastebin.org/ or http://bin.cakephp.org/ or http://pastebin.ca/ |
| 07:56.45 | kaldemar | Faustov: that's a step towards a he-man type of new name for the project. |
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| 07:59.29 | shamelessn00b | ChannelZ: I want to spawn a local channel that would call the chanspy app to spy on an active conference call, and stream that audio to some IP address, problem is chanspy itself is blocking |
| 07:59.46 | qakhan | kaldemar can u tell me how can i log number of count of a number |
| 07:59.48 | shamelessn00b | so I can't call EAGI etc in the dialplan after calling chanspy |
| 08:00.03 | Faustov | kaldemar: the legendary he-PBX, sounds nice ;) |
| 08:00.06 | qakhan | how many times a caller called |
| 08:01.25 | kaldemar | qakhan: i don't know what you mean by "number of count of a number" |
| 08:02.22 | qakhan | i have a T1 line. i want to log how many times a caller called a day |
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| 08:03.30 | bulkorok | qakhan: let the cdr log in to a sql-db |
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| 08:04.57 | qakhan | bulkorok here is my senario. i have setup blacklist in dialplan, i want to log CLID in separate table how many times a blacklist caller called |
| 08:05.08 | kaldemar | qakhan: oh. use a DB value for that. put Set(DB(callcounter/${CALLERID(num)})=${IF($[${DB_EXISTS(callcounter/${CALLERID(num)})}]?${INC(${DB(callcounter/${CALLERID(num)})})}:1)}) somewhere in dialplan where incoming calls land. |
| 08:05.34 | kaldemar | or something like that. |
| 08:05.54 | qakhan | i m using mysql to blacklist caller |
| 08:05.55 | kaldemar | i didn't try that monster, don't expect the braces to be 100% correct. |
| 08:06.16 | bulkorok | qakhan: instead of internal DB you can use func_odbc |
| 08:06.49 | qakhan | kaldemar and bulkorok i m using mysql to blacklist caller |
| 08:07.17 | bulkorok | ok... so you have a table with blacklisted numers |
| 08:07.22 | qakhan | yes |
| 08:07.23 | bulkorok | numbers |
| 08:07.27 | kaldemar | qakhan: you said that already. the idea is the same for whatever backend you use. the functions are different. |
| 08:07.30 | qakhan | in mysql |
| 08:07.49 | qakhan | how can i use in mysql |
| 08:07.58 | bulkorok | and you want to count how many times a blacklisted number tried to call you!? |
| 08:08.08 | kaldemar | qakhan: like bulkorok said, use func_odbc. |
| 08:08.09 | qakhan | yes |
| 08:08.20 | qakhan | yes bulkorok |
| 08:08.21 | bulkorok | then func_odbc ;-) |
| 08:08.56 | bulkorok | with func_odbc you can definie your own functions that execute a sql-statement and return some value to your dialplan |
| 08:09.35 | bulkorok | you can send the callerid to the function |
| 08:09.54 | bulkorok | i think i have sth like that... |
| 08:10.33 | bulkorok | readsql=SELECT count(*) as anzahl from blacklist where callerid='${SQL_ESC(${ARG1})}' |
| 08:11.12 | qakhan | is there any function or veriable in * for count |
| 08:11.39 | bulkorok | I don't get your question |
| 08:12.16 | qakhan | is there any function or veriable in asterisk name like count |
| 08:13.29 | kaldemar | qakhan: no. you have to do this yourself. |
| 08:13.29 | bulkorok | i don't think so... |
| 08:14.25 | qakhan | what about SetVar,"COUNTER=0"; |
| 08:14.55 | bulkorok | then you have a variable with 0 in it... |
| 08:15.08 | kaldemar | qakhan: what about it? that's just a variable that means absolutely nothing. |
| 08:15.51 | qakhan | so u mean i have to use func_odbc |
| 08:16.02 | qakhan | can u help me in it. its usage |
| 08:16.07 | kaldemar | qakhan: if you want to use mysql for this, yes. |
| 08:16.08 | bulkorok | you have to increment the var in your dialplan when the event occurs you want to count |
| 08:16.57 | bulkorok | or you use mysql for storing the variable... but you have to increment the var via the dialplan... |
| 08:17.21 | kaldemar | qakhan: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html |
| 08:18.52 | *** join/#asterisk domi (~domi@mail.tas.de) |
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| 08:47.01 | qakhan | kaldemar uniqueid is not being logged in cdr table |
| 08:51.03 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:52.53 | domi | if i dial out of a macro, the callerid sent to the called party is correct, but in the pickup-info it is wrong. any workarounds? |
| 08:54.49 | kaldemar | qakhan: was that a question? |
| 08:54.58 | domi | scenario: someone calls 490, that starts a macro, finally tha macro dials the extension 491. on 491 i can see who is calling, but everyone that gets a pickup-info has a call from 491 to 491 :/ |
| 08:55.27 | qakhan | yes |
| 08:55.37 | kaldemar | qakhan: what are you asking? |
| 08:56.45 | qakhan | i m asking there is a field in Cdr table "uniquid" |
| 08:57.19 | qakhan | now uniqueid is not being logged in "uniquid" field |
| 08:57.34 | qakhan | what could be the reason? |
| 08:58.37 | zro | so if i have a machine, and its only hooked to the internet (no pstn phone jack), and i have an account w/ teliax, and i set up iax2, I can get calls made to that teliax phone number to goto the machine, right? Thats not "IAX2 Trunking" is it? |
| 09:00.36 | *** part/#asterisk stev3n (~Stiv@sever-kirov.ru) |
| 09:00.39 | kaldemar | qakhan: i have no idea about your tables. asterisk logs "uniqueid" however. |
| 09:01.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:03.37 | qakhan | kaldemar i m using the default cdr table |
| 09:05.50 | kaldemar | there is no table by default. |
| 09:07.09 | qakhan | i mean cdr table |
| 09:07.23 | qakhan | which i made during installation |
| 09:08.30 | _abc_ | Is this kind of set worthless? http://www.ebay.com/itm/CNC-MINI-Classic-Lathe-6-Tools-in-1-Milling-Machine-Sawing-Driller-Grinder-/251024126822?pt=LH_DefaultDomain_0&hash=item3a72343366 |
| 09:08.38 | _abc_ | sorry wrong channel |
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| 09:09.23 | din3sh | O.O |
| 09:11.23 | kaldemar | qakhan: maybe your table is wrong. |
| 09:13.24 | din3sh | qakhan:which asterisk version? |
| 09:13.48 | qakhan | kaldemar where from cdr table get records. /var/log/asterisk/cdr-csv/Master.csv? |
| 09:14.02 | qakhan | din3sh i m using 1.4.38 asterisk |
| 09:14.38 | kaldemar | qakhan: your question makes no sense. csv and possible database logs are done by different modules. |
| 09:15.39 | din3sh | csv and realtime cdr are 2 different mechanisms i think |
| 09:15.43 | kaldemar | qakhan: are you using a database module to log CDR or are you using plain CSV? |
| 09:16.05 | qakhan | i m using asterisk-addon |
| 09:16.33 | kaldemar | asterisk-addon is not a CDR module. it is a bunch of modules that may or may not be in use. what are you using? |
| 09:17.24 | qakhan | i m using cdr |
| 09:18.05 | kaldemar | "cdr" is not a module either. are you using cdr_addon_mysql? |
| 09:20.07 | qakhan | look when i installed asterisk-addon i selected all modules in menuselect |
| 09:20.35 | *** join/#asterisk timahvo1 (~rogue@197.176.166.15) |
| 09:21.31 | kaldemar | qakhan: i asked if you are using cdr_addon_mysql. are you? |
| 09:21.46 | kaldemar | qakhan: if you don't know, say so. |
| 09:22.29 | qakhan | sir i dont know |
| 09:23.40 | zro | cheap itsp suggestion for screwing around? I seem to having a problem w/ teliax. |
| 09:25.11 | kaldemar | qakhan: does "module show like cdr" list you cdr_addon_mysql? |
| 09:26.33 | qakhan | i got these |
| 09:26.34 | qakhan | func_cdr CDR dialplan function 0 |
| 09:26.34 | qakhan | cdr_odbc ODBC CDR Backend 0 |
| 09:26.34 | qakhan | cdr_manager Asterisk Manager Interface CDR Backend 0 |
| 09:26.34 | qakhan | cdr_custom Customizable Comma Separated Values CDR 0 |
| 09:26.34 | qakhan | cdr_csv Comma Separated Values CDR Backend 0 |
| 09:26.35 | qakhan | app_setcdruserfield CDR user field apps 0 |
| 09:26.35 | qakhan | app_forkcdr Fork The CDR into 2 separate entities 0 |
| 09:26.36 | qakhan | app_cdr Tell Asterisk to not maintain a CDR for 0 |
| 09:26.36 | qakhan | cdr_addon_mysql.so MySQL CDR Backend 0 |
| 09:31.12 | kaldemar | qakhan: so you have the mysql module. does it log anything? how did you determine that it does not log uniqueid? what is your tabl like? |
| 09:32.41 | qakhan | its log every thing except uniqueid and userfield |
| 09:34.00 | kaldemar | show it. |
| 09:34.04 | kaldemar | ~pb |
| 09:34.04 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 09:36.26 | qakhan | here http://pastebin.com/m6Lp8YLM |
| 09:39.52 | din3sh | qakhan:which version of asterisk-addons have you compiled? |
| 09:40.21 | kaldemar | #ifdef MYSQL_LOGUNIQUEID |
| 09:40.46 | qakhan | asterisk-addons-1.4.12 |
| 09:41.04 | qakhan | kaldemar what? |
| 09:42.12 | kaldemar | qakhan: everything related to uniqueid in the cdr_addon_mysql code is inside that conditional. so it may well be possible that the module does not even log uniqueid. |
| 09:42.31 | din3sh | exactly kaldemar |
| 09:44.03 | qakhan | what about this http://bahjons.com/stuff/asterisk-cdr-logging-in-mysql |
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| 09:44.06 | kaldemar | you'd need to add for example "#define MYSQL_LOGUNIQUEID 1" to the file to make the condition true and recompile. |
| 09:44.17 | din3sh | qakhan:add #define MYSQL_LOGUNIQUEID in /asterisk-addons/cdr/cdr_addon_mysql.c and recompile again |
| 09:44.29 | qakhan | ok |
| 09:49.59 | *** join/#asterisk din3sh (~din3sh@41.212.201.65) |
| 09:50.19 | din3sh | yawns |
| 09:51.40 | kaldemar | that retardation has been fixed in version 1.6.0 of addons. |
| 09:58.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:00.37 | qakhan | kaldemar i did the same but still i cant get uniqueid in cdr table |
| 10:00.50 | kaldemar | show exactly what you did |
| 10:01.30 | din3sh | qakhan:add #define MYSQL_LOGUNIQUEID in /asterisk-addons/cdr/cdr_addon_mysql.c and recompile again |
| 10:01.34 | din3sh | :x |
| 10:02.08 | qakhan | din3sh yes i did the same |
| 10:03.28 | kaldemar | you must also re-install addons and reload the module or restart asterisk. |
| 10:04.15 | qakhan | yes kaldemar i all have done |
| 10:04.33 | qakhan | even i restarted mysql |
| 10:05.07 | kaldemar | show exactly what you did |
| 10:06.07 | qakhan | how |
| 10:06.56 | qakhan | i add #define MYSQL_LOGUNIQUEID on top of cdr_addon_mysql.c |
| 10:07.32 | kaldemar | use pastebin and show it. |
| 10:07.41 | qakhan | then i ran make clean then make then make install |
| 10:07.51 | qakhan | then cp /usr/src/asterisk-addons/configs/cdr_mysql.conf.sample /etc/asterisk/cdr_mysql.conf |
| 10:08.22 | qakhan | made changes in cdr_mysql.conf |
| 10:08.37 | qakhan | and reload the asterisk |
| 10:09.03 | kaldemar | "reload the asterisk"? |
| 10:09.11 | kaldemar | what command? |
| 10:10.17 | qakhan | reload |
| 10:12.24 | kaldemar | that does not reload the module. use "module reload cdr_addon_mysql.so" |
| 10:13.16 | qakhan | ok |
| 10:15.25 | qakhan | kaldemar it says Module 'cdr_addon_mysql.so' does not support reload |
| 10:15.59 | kaldemar | "module unload cdr_addon_mysql.so" and "module load cdr_addon_mysql.so" |
| 10:16.02 | din3sh | unload it, load it again |
| 10:16.38 | din3sh | btw qakhan, i use asterisk-addons.1.4.10, it works properly |
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| 10:18.40 | qakhan | din3sh i unload and load it again |
| 10:18.49 | qakhan | let see what happens |
| 10:19.03 | qakhan | do i call and check it? |
| 10:19.19 | din3sh | yes |
| 10:19.21 | din3sh | and also |
| 10:19.33 | din3sh | be sure to have loguniqueid=yes in your cdr_mysql.conf |
| 10:19.45 | qakhan | yes i have it |
| 10:19.52 | din3sh | it should work |
| 10:21.05 | qakhan | Thanks kaldemar and din3sh its working now :) |
| 10:22.09 | din3sh | been some hours we at it! :o |
| 10:22.13 | din3sh | lol |
| 10:33.43 | *** join/#asterisk kresp0 (~kresp0@213.37.151.211.dyn.user.ono.com) |
| 10:35.43 | domi | is there any way to iterate through a macros arguments? |
| 10:40.03 | din3sh | anybody using astfin/blackfin based ip04/08? |
| 10:47.04 | din3sh | it take it as a NO |
| 10:47.57 | kaldemar | domi: sure. use a While loop like this: http://pastebin.com/jC3AVFcE |
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| 11:57.29 | Kobaz | hmm |
| 11:57.55 | Kobaz | how would i set a DB entry on startup in ael... with regular vars you can use globals { foo=bar; } |
| 11:58.12 | Kobaz | but globals { DB(foo) = bar; } is a syntax error |
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| 12:00.02 | kaldemar | Kobaz: DB(foo) is missing a key. |
| 12:00.09 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-19-211.bois.qwest.net) |
| 12:01.27 | kaldemar | Kobaz: as in it should be DB(family/key)=bar |
| 12:01.33 | Kobaz | well whatever, it's not the foo part that's a problem |
| 12:01.42 | Kobaz | foo/bar/baz could be anything |
| 12:02.00 | Kobaz | Line 2, Cols: 6-6: Error: syntax error, unexpected '(', expecting '=' |
| 12:02.05 | Kobaz | the parser hits the ( and dies |
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| 12:19.18 | sawgood | Hi: normally, I have a statement in extensions.conf which = _1NXXNXXXXXX |
| 12:20.14 | sawgood | how can I still use this, but have calls to specific NPA/NXX not used .... meaning ... _1NXXNXXXXX = any destination matching (expect if you are on a list of known blocked numbers) |
| 12:20.41 | [TK]D-Fender | sawgood, the more specific ones do match first |
| 12:21.39 | sawgood | cool ... but I have a 'list' of NPA/NXX which I would like for the dialplan to look up against before then rule matching |
| 12:23.10 | [TK]D-Fender | then code up the list. |
| 12:23.52 | sawgood | 90+ NPA/NXX destinations I have in the list |
| 12:28.02 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
| 12:28.05 | nickfennell | hi all |
| 12:28.10 | nickfennell | got an issue where asterisk isn't clearing calls |
| 12:28.24 | nickfennell | Asterisk 1.8.5.0 |
| 12:28.40 | nickfennell | Is there a way (other than restarting) to clear calls? |
| 12:29.45 | wdoekes2 | channel request hangup ? |
| 12:30.56 | nickfennell | hmm that sounds like a good one |
| 12:31.45 | nickfennell | hmm what's the ID for tha channel |
| 12:32.00 | wdoekes2 | core show channels [concise] |
| 12:32.13 | nickfennell | I think I can see my stuck call (shows as Last Message Rx: Bye) |
| 12:33.49 | nickfennell | OK. Managed to clear one. Two are still stuck. |
| 12:33.54 | nickfennell | AppDial((Outgoing Line)) |
| 12:37.09 | nickfennell | restart cleared |
| 12:40.47 | *** part/#asterisk Bullmoose (~Bullmoose@71-33-19-211.bois.qwest.net) |
| 12:41.48 | Naikrovek | configuring my IP321 for a test Lync server. this.. is challenging. |
| 12:42.03 | Naikrovek | UCS4.0 for polycom is nice, though. |
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| 13:03.36 | MrTelephone | Naikrovek, you still there? |
| 13:04.03 | Naikrovek | yep |
| 13:04.10 | Naikrovek | wassup |
| 13:04.18 | MrTelephone | Hey I'm interested in your lync setup, why use that instead of asterisk? |
| 13:04.35 | Naikrovek | heh |
| 13:04.49 | din3sh | join #lync |
| 13:05.02 | din3sh | if that ever exists |
| 13:05.02 | MrTelephone | I have a hard time selling microsoft to my superiors when it comes time for license renewal. 2008 and exchange 2010 is going to cost us 13 thousand |
| 13:05.02 | din3sh | :p |
| 13:05.11 | Naikrovek | how many employees |
| 13:05.17 | MrTelephone | 150 users |
| 13:05.32 | MrTelephone | Asterisk saves us loads of cash |
| 13:05.34 | Naikrovek | cheaping out on licenses will cost you if you have that many people. |
| 13:05.47 | MrTelephone | the adminsitration is the best part of microsoft but the end user and purchaser does not see that benefit |
| 13:05.48 | Naikrovek | lync gives us things asterisk simply can't |
| 13:06.11 | MrTelephone | we fall under government pricing or it would have been 20 thousand |
| 13:06.29 | Naikrovek | do you use office? |
| 13:06.42 | MrTelephone | yeah but we buy oem's when they purchase new computers every 3-4 years |
| 13:06.50 | Naikrovek | we used to use open office but it cost us tens of thousands of dollars in lost productivity |
| 13:07.14 | MrTelephone | Good for small business but I can see how that can happen |
| 13:07.50 | din3sh | how is lync better? |
| 13:08.00 | MrTelephone | Yeah I'm wondering about that too |
| 13:08.01 | Naikrovek | that depends entirely on how you use it. |
| 13:08.09 | Naikrovek | the answer will be different for everyone |
| 13:08.25 | MrTelephone | Our office workers need email to voicemail and tone. What else can you ask for? |
| 13:08.25 | Naikrovek | it works better for us because we're already a microsoft shop and use a lot of microsoft products |
| 13:08.41 | MrTelephone | Adminsitrative tasks are improved then? |
| 13:08.44 | Naikrovek | if that's all you need, then lync will make zero sense to you no matter what i say |
| 13:08.54 | Naikrovek | i currently have an asterisk system and it works fine. |
| 13:09.30 | MrTelephone | I can see email contacts going to your phone automatically being awesome but I don't think it would take too many resources to program something to make directory.xml files from outlook contacts. |
| 13:09.39 | Naikrovek | i'm debating whether or not i should even say why we're switching, because all I ever get from the open source community is "GPL SOFTWARE CAN DO THAT YOU CUNT!" |
| 13:09.52 | MrTelephone | no please tell me, I'm very open minded |
| 13:10.05 | MrTelephone | I only discourage processes that I've tried before and never worked |
| 13:10.22 | Naikrovek | automatically update directory, as you said. and yes it can be automated without lync. |
| 13:10.28 | MrTelephone | I run open source for ISP related things but in the office we use microsoft |
| 13:10.31 | Naikrovek | mostly it's what comes with the lync client and not the phones |
| 13:10.52 | MrTelephone | so what if we made a program using MAPI to write new directory files? |
| 13:11.13 | Naikrovek | if you're familiar with the former netmeeting application, lync is a lot like that, but ... a decade older and more advanced. |
| 13:11.35 | Naikrovek | with all kinds of seamlessness that didn't exist before |
| 13:12.05 | MrTelephone | does the new netmeeting have a feature where people can watch meetings using the web? |
| 13:12.23 | Naikrovek | when I go to an appointment i have scheduled in outlook, my phone automatically marks me away and sends calls to the other IT guy (for example) |
| 13:12.32 | MrTelephone | nice |
| 13:12.43 | Naikrovek | yes, lync has pretty much everything webex has, but it's hosted in-house. |
| 13:12.58 | Naikrovek | so i can to desktop sharing with it, a whole slew of stuff. |
| 13:13.15 | Naikrovek | it fits for us |
| 13:13.31 | Naikrovek | asterisk fit us for a long time but we haven't the knowledge in-house to maintain it to the scale we're at now |
| 13:13.33 | din3sh | but lync licenses do not come cheap eithe |
| 13:13.36 | din3sh | either* |
| 13:13.41 | Naikrovek | no they don't. |
| 13:13.51 | Naikrovek | yet lync still makes a lot of sense for us. |
| 13:13.55 | MrTelephone | how come they don't have a global license that works for everything |
| 13:13.59 | MrTelephone | or do they? |
| 13:14.07 | din3sh | i think its relative |
| 13:14.22 | MrTelephone | I would consider trying it but then you have to buy new phones |
| 13:14.23 | Naikrovek | they have enterprise CALs and stuff, if that's what you mean. one CAL to cover everything. |
| 13:14.56 | Naikrovek | The lync server software is cheap, it's the CALs that cost so much. |
| 13:15.17 | Naikrovek | but even so, it's not a huge expense for us. We have almost 1,000 employees. |
| 13:15.17 | din3sh | how does the lync server connect to the pstn? |
| 13:15.24 | din3sh | digital cards? |
| 13:15.36 | MrTelephone | sip |
| 13:15.41 | MrTelephone | or t1 i bet |
| 13:15.46 | Naikrovek | SIP or through some intermediary like a cisco router or gateway device |
| 13:16.22 | Naikrovek | if you have a cisco device that can handle voice, you can just tie lync to that. or you can use sip directly. or you can use an audiocodes gateway |
| 13:16.49 | MrTelephone | sounds like a fun project to try out |
| 13:16.53 | Naikrovek | it has been |
| 13:17.15 | Naikrovek | i have a small test server going now, trying to get a polycom ip 321 to work. i'm so close |
| 13:17.22 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 13:17.31 | Naikrovek | with the UCS 4.0.1 (and up) firmware these phones can be Lync endpoints. |
| 13:17.46 | din3sh | the licence is per extension? |
| 13:17.46 | Naikrovek | they're not fully functional Lync phones, but they're close. |
| 13:17.52 | Naikrovek | per user or per device |
| 13:18.02 | Naikrovek | depends on what makes sense. for us, it's per user. |
| 13:18.39 | MrTelephone | I've never encoutered a real case where device makes sense |
| 13:18.49 | MrTelephone | call center? |
| 13:18.52 | MrTelephone | haha |
| 13:18.54 | Naikrovek | hot desking |
| 13:18.59 | MrTelephone | how much is asterisk per user licensing? |
| 13:19.04 | Naikrovek | more users than devices? device CALs make more sense |
| 13:19.09 | Naikrovek | you know how much asterisk is |
| 13:19.11 | bipul | MrTelephone, HI |
| 13:19.20 | MrTelephone | yeah I know but we never have more users then devices for some reason |
| 13:19.29 | MrTelephone | i mean |
| 13:19.32 | MrTelephone | the other way around |
| 13:19.49 | Naikrovek | if you have 200 computers and 3 shifts of 200 people each, device CALs make WAY more sense. |
| 13:20.00 | Naikrovek | but that's not a common scenario, in my experience. |
| 13:20.04 | MrTelephone | Yeah we don't have shift |
| 13:20.29 | *** join/#asterisk serafie (~erin@nat/digium/x-pfsdelxfpjaundvo) |
| 13:21.28 | MrTelephone | Just the type of work we do. But I understand where it makes sense. I'm used to the environment where everyone has their own workstation and that's it. But at the same time if you create emails for them you still need a user cal for that, and you need a user call for 2008 because exchange is running on it. Doesn't that make device cal obsolete? |
| 13:22.15 | MrTelephone | I'm pretty sure you need a server cal and exchange cal to have a legal email user |
| 13:28.18 | din3sh | lync is evilllllll |
| 13:28.24 | MrTelephone | do you need a cal for webmail? |
| 13:28.30 | s[x] | lol |
| 13:28.39 | MrTelephone | is that device or user? |
| 13:28.44 | MrTelephone | let me call my lawyer |
| 13:29.43 | MrTelephone | He said paying the fines is cheaper than the cals |
| 13:29.44 | din3sh | then again , alcatel, avaya all also perform pretty decently |
| 13:29.53 | din3sh | why lync? |
| 13:29.59 | *** join/#asterisk mahiti-irc (~mahiti@123.49.43.157) |
| 13:30.02 | MrTelephone | because microsoft infrastructure is solid |
| 13:30.07 | MrTelephone | active directory |
| 13:30.10 | mahiti-irc | hi guys |
| 13:30.14 | MrTelephone | outlook integration |
| 13:30.22 | din3sh | asterisk also |
| 13:30.23 | din3sh | :p |
| 13:30.37 | MrTelephone | THe bad thing about microsoft is that IT jobs become expendable |
| 13:30.46 | mahiti-irc | is there anyone who can tell me how to setup ss7 |
| 13:30.46 | MrTelephone | the secretary can add a new user if needd |
| 13:31.10 | Naikrovek | yes you need a cal for webmail |
| 13:31.18 | mahiti-irc | i have installed asterisk 1.8.11.0 and libss7 1.0.2 |
| 13:31.31 | MrTelephone | naikrovek, device or user? |
| 13:31.38 | mahiti-irc | when i asked my telcom about the settings |
| 13:31.55 | mahiti-irc | they gave me three settings GT,SPC and Short Code |
| 13:32.05 | Naikrovek | MrTelephone: user. also you say "server cal and exchange cal to have legal email user" - exchange server cal is all you need. |
| 13:32.30 | *** join/#asterisk OrNix (~OrNix@l49-246-139.cn.ru) |
| 13:33.03 | Naikrovek | MrTelephone: paying the fines is not cheaper; microsoft doesn't fine you, they just make you pay up if you're caught. if you don't pay, they revoke your right to use. |
| 13:33.06 | mahiti-irc | when i look into chan_dahdi.conf, i find it requires pointcode, dpc and adjpointcode |
| 13:33.07 | MrTelephone | How sure are you about that? One of microsofts license papers said if you have less cals on 2008 than exchange you are probably in violation. |
| 13:33.17 | *** join/#asterisk darranf1990 (~crash@194.193.169.242) |
| 13:33.38 | darranf1990 | hello anyone be able to help me with a problem im having with 1.8.7? |
| 13:33.39 | Naikrovek | MrTelephone: you're confusing server CAL and exchange CAL. server CAL is for file sharing. Exchange CAL is for email users. |
| 13:34.08 | MrTelephone | yeah but the user is accessing windows server 2008 because exchange is accessing active directory and whatever else it does. |
| 13:34.26 | MrTelephone | But you would know you have way more users than we do |
| 13:34.37 | Naikrovek | MrTelephone: do you have a windows file server or samba (or some san-based thing) |
| 13:35.05 | MrTelephone | 8 file servers |
| 13:35.13 | MrTelephone | windows |
| 13:35.36 | Naikrovek | ok so you need one cal per user for windows server. if those users also have outlook (or access to OWA) then they need exchange CALs as well |
| 13:35.57 | darranf1990 | im trying to find a replacemnt dialplan for AgentCallBackLogin in 1.8? |
| 13:36.03 | WIMPy | ~ask |
| 13:36.03 | 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. |
| 13:36.13 | [TK]D-Fender | darranf1990, that was was gone in 1.6.0 |
| 13:36.16 | MrTelephone | I didn't misread that in the document but they are assuming that most places do not give out "email only" accounts |
| 13:36.20 | WIMPy | darranf1990@ ^^ if you can make it more interesting than lync, try :-) |
| 13:36.39 | [TK]D-Fender | darranf1990, "core show applications like queue" |
| 13:36.45 | Naikrovek | MrTelephone: that's how I do it. one windows server CAL for each user, one Exchange CAL for each user. One Office license, and soon one Lync client CAL and software license as well. |
| 13:36.50 | din3sh | lync is evilllllllllllll i tell youuuuuu |
| 13:37.01 | Naikrovek | maybe |
| 13:37.08 | Naikrovek | but it beats the hell out of what we have now |
| 13:37.42 | darranf1990 | we have a system where agents log onto phones on number 2199 and then that will issue that phone a 4XX number then people can phone the 4XX number the phones are registerd with asterisk on a 5XX number its a pretty bad system but ive got to upgrade and i've hit this problem :( |
| 13:37.47 | MrTelephone | I would never be able to purchase all new phones. People spend too much time travelling |
| 13:37.54 | din3sh | actually in this context asterisk provides you with a framework to build upon |
| 13:37.54 | mahiti-irc | guys i need to configure ss7 when i asked my telcom about the ss7 settings |
| 13:37.56 | mahiti-irc | they gave me three settings GT,SPC and Short Code |
| 13:38.00 | mahiti-irc | how do i set them |
| 13:38.05 | din3sh | lync is a ready off-the-shelf thing |
| 13:38.15 | Naikrovek | lol no |
| 13:38.15 | [TK]D-Fender | darranf1990, AddQueueMemeber , RemoveQueueMember, etc... |
| 13:38.17 | Naikrovek | rofl no |
| 13:38.23 | din3sh | :p |
| 13:38.23 | Naikrovek | lync is not off the shelf |
| 13:38.25 | darranf1990 | sorry all agents have there own $XX number and password and this enables them to log into diffrent phones in the office... |
| 13:38.35 | din3sh | can you develop stuff like in asterisk? |
| 13:38.43 | Naikrovek | yes. |
| 13:38.48 | Naikrovek | lync server sdk. |
| 13:38.53 | darranf1990 | will AddQueueMember allow them to hang up then be called on the 4XX number till they log off |
| 13:39.08 | din3sh | uhuh! |
| 13:39.18 | [TK]D-Fender | darranf1990, Go read the instructions |
| 13:39.40 | din3sh | why not 3cx if you want windows based pbx |
| 13:40.13 | Naikrovek | i don't want a windows-based pbx, necessarily, I want integration between my phones and everything else I have (outlook, exchange, sharepoint) |
| 13:40.26 | Naikrovek | 3cx can't do that |
| 13:42.46 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 13:43.03 | Naikrovek | when i pick up the phone, i want my IM client to show me as "busy" for example. when I go into a meeting I have scheduled in Outlook, i want phonecalls to my phone from internal extensions to go to the other IT person here. |
| 13:43.04 | Naikrovek | etc. |
| 13:43.12 | Naikrovek | and I don't want to have to fiddle with it once it's set up. |
| 13:43.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:43.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.23 | MrTelephone | Asterisk and ODBC allows for easy hotdesking |
| 13:43.34 | Naikrovek | "Easy" my ass |
| 13:43.37 | din3sh | i have tried hot desking |
| 13:43.38 | MrTelephone | lol |
| 13:44.09 | din3sh | yeah real easy |
| 13:44.10 | din3sh | ;) |
| 13:44.15 | Naikrovek | with a lync phone i can just walk up to an empty desk, plug the phone into my computer, and that phone is now my extension. no login, no logout, just works. |
| 13:44.18 | MrTelephone | I use asterisk to resell phone service and I started using ODBC for realtime account suspension that can be managed via a web interface. Actually, our whole billing system is integrated with asterisk. |
| 13:44.18 | Naikrovek | easier than that? |
| 13:44.37 | MrTelephone | When your invoice is 30 days old suspension=1 and the calls don't go out |
| 13:44.38 | darranf1990 | i've tried a few times to use the instructions but they will place the call in a queue waiting for a call to come in then "BEEP" i want them to be able to log on and then hangup but still have there logged on 4XX number |
| 13:44.55 | din3sh | MrTelephone: are you using CEL for billing purposes? |
| 13:44.55 | MrTelephone | Naikrovek, no thats hard to setup phones |
| 13:45.00 | Naikrovek | MrTelephone: that's a situation where lync would not easily work (without custom development)( |
| 13:45.05 | din3sh | i mean like attended xfer and all |
| 13:45.11 | MrTelephone | no I designed my own |
| 13:45.26 | din3sh | how you track down lost legs of attended xfers? |
| 13:45.27 | MrTelephone | I don't know what CEL is |
| 13:45.31 | din3sh | >:| |
| 13:45.39 | Naikrovek | call event logging |
| 13:45.41 | Naikrovek | i believe |
| 13:45.54 | din3sh | channel event logging |
| 13:45.58 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:45.59 | MrTelephone | I wrote a perl script on each asterisk server that writes calls to a central database |
| 13:46.01 | Naikrovek | it's like CDR but different. better in a lot of ways. |
| 13:46.08 | MrTelephone | I'll have to check it out |
| 13:46.32 | din3sh | the regular cdr has a lost xfer leg issue |
| 13:47.29 | MrTelephone | It doesn't log the original call when there is a transfer? |
| 13:49.44 | MrTelephone | Once a day it just writes the calls to the accounting database |
| 13:50.02 | din3sh | the subsequent transfered leg |
| 13:51.21 | MrTelephone | Yeah that would be a pain for diagnostics |
| 13:51.50 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 13:52.06 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:53.19 | din3sh | say 123 calls 456 and transfers(attended) to 789, theres a log for 123 to 456, and 123 to 789 |
| 13:53.31 | din3sh | but no log for 456 to 789 |
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| 13:57.18 | mahiti-irc | guys i need to configure ss7; when i asked my telcom about the ss7 settings |
| 13:57.20 | mahiti-irc | they gave me three settings GT,SPC and Short Code |
| 13:57.22 | mahiti-irc | how do i set them |
| 13:58.24 | [TK]D-Fender | [9:38] <[TK]D-Fender> darranf1990, AddQueueMemeber , RemoveQueueMember, etc... |
| 13:58.24 | watchy | anyone using polycom 4.0+ and satisfied with it? |
| 13:58.35 | [TK]D-Fender | darranf1990, Does not wait for a "beep" |
| 13:58.36 | Naikrovek | i am, on one phone. i like it. |
| 13:58.41 | Naikrovek | watchy: ^ |
| 13:58.50 | Naikrovek | web interface is SO MUCH BETTER |
| 13:59.02 | watchy | you use the phone web int? |
| 13:59.13 | watchy | thats deff not something i care about myself |
| 13:59.34 | watchy | too much trouble for me to worry about that |
| 14:00.32 | din3sh | Naikrovek: you able to configure the phones via some kinda endpoint interface in lync? |
| 14:00.35 | watchy | anyone know of a sip video client for iphone other then bria? |
| 14:02.03 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
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| 14:02.56 | Naikrovek | din3sh: phone configuration is different in lync. hard to explain. but most of the work is done in powershell. |
| 14:03.22 | watchy | whats lync? |
| 14:04.38 | watchy | blah my cc was declined by itunes cause i cancelled it and havent gotten my new one. all i wanted to do was play with bria |
| 14:04.57 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 14:04.58 | watchy | its gonna be a bad day already |
| 14:06.19 | din3sh | watchy: lync is evil! :p |
| 14:06.22 | Naikrovek | din3sh: lync uses AD and Exchange for most of its configuration. To set up a new phone you just give a user a phone number, then the user logs into the phone via the phone itself or by plugging it into a computer via USB. |
| 14:06.39 | din3sh | USB phones? |
| 14:06.49 | Naikrovek | the phone pulls its configuration from the lync server and by the time you look over to the phone it's configured |
| 14:06.54 | Naikrovek | no, not USB phones |
| 14:06.59 | Naikrovek | the phones are configured over USB. |
| 14:07.01 | Naikrovek | they CAN be |
| 14:07.07 | din3sh | oh ok |
| 14:07.11 | Naikrovek | they're PoE phones like anything else. |
| 14:07.19 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 14:08.00 | din3sh | do snom phones work well with lync? |
| 14:08.06 | watchy | man i can setup auto configing in a polycom like lync i think just using isc-dhcp and nice configs |
| 14:08.09 | mahiti-irc | no one knows ss7? |
| 14:08.29 | Naikrovek | so it would be something like "set-csuser -identity jdoe -lineuri "tel:+13215551234;ext=23456" |
| 14:08.37 | Naikrovek | ss7 is for carriers. |
| 14:08.50 | din3sh | what about auto-provisioning in lync? |
| 14:08.56 | Naikrovek | not a lot of carriers in here. |
| 14:09.00 | watchy | is lync a polycom product? |
| 14:09.05 | Naikrovek | no |
| 14:09.16 | watchy | oh its microsoft |
| 14:09.17 | Naikrovek | lync is a microsoft product, polycom makes phones that work with lync |
| 14:09.26 | mahiti-irc | Naikrovek, ok, but i am libss7 to configure my digium card |
| 14:09.28 | din3sh | lync is a unified messaging/pbx stuff by microsoft |
| 14:09.35 | watchy | i think id rather auto provision through dhcp / ftp |
| 14:09.38 | mahiti-irc | how do i setup in it? |
| 14:09.43 | Naikrovek | din3sh: if the user has a phone number assigned to them, and they log into the phone (or plug it into their computer) it's provisioned then. |
| 14:09.52 | watchy | and write an app for the phone to let someone provision it from the phones web gui |
| 14:10.21 | din3sh | how does lync integrate with sharepoint and outlook |
| 14:10.22 | Naikrovek | there are some DHCP settings you have to set up but that's a one-time thing |
| 14:10.27 | Naikrovek | din3sh: automatically. |
| 14:10.41 | watchy | but you need to run microsoft's pbx correct? |
| 14:10.50 | Naikrovek | it shows presence in outlook and in sharepoint, and lync knows how to find them both via AD |
| 14:11.00 | Naikrovek | watchy: yes, lync is that pbx |
| 14:11.09 | watchy | ah. i don't think im interested in that |
| 14:11.15 | Naikrovek | i've taken #asterisk into lync-land. sorry. |
| 14:11.26 | watchy | its cool. i always like knowing new things |
| 14:11.36 | watchy | i'm not anti microsoft by any means |
| 14:12.07 | din3sh | watchy: actually Naikrovek tried #lync, but there's probably no one in there :p |
| 14:12.15 | watchy | any corp enviroment is going to require exchange etc |
| 14:12.26 | Naikrovek | din3sh: closest thing is probably ##windows-server |
| 14:12.39 | watchy | is lync even a popular product? |
| 14:12.47 | Naikrovek | it's gaining steam fast. |
| 14:12.53 | Naikrovek | i'm about to deploy it for 1,000 users. |
| 14:13.05 | watchy | why did you decide it over asterisk? |
| 14:13.14 | Naikrovek | i already have asterisk. |
| 14:13.18 | watchy | im sure its almost idiot proof |
| 14:13.21 | Naikrovek | i need integration |
| 14:13.25 | Naikrovek | not idiot proof at all |
| 14:14.05 | Naikrovek | i have sharepoint, and outlook, and I need IM that integrates with them, and user-configurable phone routing, etc. |
| 14:14.15 | watchy | ah |
| 14:14.17 | Naikrovek | it all CAN BE DONE with asterisk, but there aren't enough hours in the day |
| 14:14.21 | watchy | well you do have alot more users then i do |
| 14:14.34 | watchy | naik: you the only it guy? |
| 14:14.39 | din3sh | lol |
| 14:14.46 | din3sh | am off guys |
| 14:14.47 | din3sh | later |
| 14:15.01 | watchy | i have about 120 users i support in a huge spread out production / office facility |
| 14:15.05 | Naikrovek | pretty much. there are some folks in our india offices but they don't really operate correctly a lot of the time |
| 14:15.10 | watchy | we use ALOT of ATA's |
| 14:15.16 | Naikrovek | they make goofy decisions that i have to fix. they have no training so i can't blame them |
| 14:15.35 | Naikrovek | they were wedged into the role they're in now and it shows. |
| 14:15.35 | watchy | also for our emergency paging system I built it out using cyberdata paging equipment for like $20k |
| 14:15.41 | Naikrovek | they haven't the skillset for what they're asked to do now |
| 14:15.45 | watchy | some company wanted $200k to do the same thing |
| 14:15.48 | Naikrovek | yeah |
| 14:16.03 | Naikrovek | time is a form of money. or maybe money is a form of time. either way, if you have time, you can save a lot of money |
| 14:16.18 | Naikrovek | and if you have money you can save a lot of time |
| 14:16.22 | watchy | cisco wanted $250k to put a phone system in here |
| 14:16.34 | watchy | i told them get fed |
| 14:17.02 | watchy | i did it for probably 50k or less. i don't remember. 3 asterisk boxes at 3 locations, lots of ATAs etc |
| 14:17.12 | Naikrovek | yeah that's high even for cisco. |
| 14:17.27 | watchy | we put phones in all of our production bunkers for emergencies |
| 14:17.32 | watchy | we build explosives |
| 14:17.44 | Naikrovek | i used to work on explosives in the USAF. |
| 14:17.55 | watchy | our bunkers have nothing but power. no phone lines etc |
| 14:18.03 | Naikrovek | as it should be |
| 14:18.08 | Naikrovek | phones spark |
| 14:18.16 | watchy | we have $1000 phones for that naikl |
| 14:18.26 | watchy | i literally pay $1000 for a red analog phone |
| 14:18.29 | watchy | its EP |
| 14:18.45 | watchy | actually its yellow not red |
| 14:19.22 | watchy | http://www.cyberdata.net/products/voip/digitalanalog/indsipphone/index.html |
| 14:19.33 | watchy | its %100 that but analog |
| 14:19.54 | watchy | ive been waiting on them to make a sip version for 2 years. but so far i haven't seen one i can purchase |
| 14:19.55 | Naikrovek | wow |
| 14:20.15 | watchy | i could eliminate about 40 ATAs with a sip version |
| 14:20.48 | Naikrovek | they make weatherproof enclosures that you can put SIP phones in |
| 14:21.15 | Naikrovek | but not weatherproof SIP phones like those analog ones |
| 14:21.23 | watchy | well |
| 14:21.32 | watchy | i gotta meet like EP rules and stuff |
| 14:21.34 | watchy | for dust |
| 14:21.49 | watchy | wrong phone could blow up the building because of dust |
| 14:22.00 | Naikrovek | yeah i gotcah |
| 14:22.16 | watchy | we have alot of RDX dust floatin around |
| 14:22.20 | Naikrovek | i bet. |
| 14:25.38 | *** join/#asterisk TheMan (~Garry@wsip-24-234-115-86.lv.lv.cox.net) |
| 14:26.20 | watchy | oh man i hate having POTS lines |
| 14:28.51 | watchy | if i knew i could pass fax across our fiber without issue i'd get rid of POTS at 2 locations |
| 14:30.16 | Naikrovek | it's pretty reliable in my experience if you use G711 |
| 14:30.59 | Naikrovek | fax needs to die though. scan & email meets legal requirements almost all the time. |
| 14:34.27 | watchy | well i am looking at t.38 over an ATA but i'm not sure how it works. |
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| 14:41.48 | Naikrovek | t38 is a fax encoding that goes over whatever protocol you use for your voip rtp data. |
| 14:41.55 | Naikrovek | s/protocol/codec/ |
| 14:42.18 | watchy | can you do POTS > asterisk > network > ata > fax |
| 14:42.28 | Naikrovek | yes. |
| 14:42.36 | Naikrovek | many do that |
| 14:42.39 | watchy | using t38 would make it pretty reliable correct? |
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| 14:43.04 | Naikrovek | t38 is the fax over voip encoding.. |
| 14:43.13 | mahiti-irc | Atleast can anyone point me to the right place to ask it? |
| 14:43.19 | Naikrovek | i don't know a great deal about it, other than for the year i used it, it worked without issue. |
| 14:44.32 | Naikrovek | mahiti-irc: ss7 question? |
| 14:45.05 | Naikrovek | i don't know of anyone in here at the moment that knows ss7 well enough to be vocal about it. |
| 14:45.17 | Naikrovek | asterisk-users email list may be able to help |
| 14:45.46 | *** join/#asterisk esrax (~hyrax@cpanel.vmlinuz.co.uk) |
| 14:45.56 | esrax | Does anyone know of a SIP trunk provider that supports re-INVITE? |
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| 14:46.47 | mahiti-irc | ok thanks Naikrovek |
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| 14:51.12 | watchy | naik: what did you use to manager ast? manual editing of files? |
| 14:54.04 | Jasnejac | hi all. I have an odd problem with upgrades of 1.8.3.3. It's on a 64-bit Debian Lenny installation. Compiling any later version from source gives no errors and claims to install correctly. However, when started DAHDI does not load properly, pbx seems not to load properly and most "core" functions are not available - in short nothing works. Anyone any ideas where to start debugging? |
| 14:55.18 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 14:55.19 | watchy | manually compiling everything or using any packages |
| 14:55.27 | Jasnejac | manual compile |
| 14:55.46 | watchy | no comp issues on dahdi? |
| 14:55.49 | WIMPy | Jasnejac: Could it be that you have multiple installations? |
| 14:56.04 | Qwell | Jasnejac: You probably aught to upgrade DAHDI as well. |
| 14:56.30 | Qwell | oh, I misread |
| 14:56.34 | Jasnejac | none - using latest versions with 1.8.3.3. and I can't see any multiples - this machine has nothing else on it except a mysql installation |
| 14:57.00 | Qwell | Jasnejac: build/install 1.8.11.1, then pastebin the entire output of asterisk -vvvvvc |
| 14:57.43 | Jasnejac | Qwell: I will do but can't at the second - this is a production server and is currently in use |
| 14:58.55 | ks3 | When I forward a voicemail with a prepended message, instead the recipient receives the prepend message as a voicemail, but no original message. And the envelope gives no indication that there's a forwarded message. Any ideas? Forwarding without prepending seems to work okay. |
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| 15:06.34 | Naikrovek | watchy: for my phones currently i use a collection of perl scripts I wrote to generate the phone configs |
| 15:08.10 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 15:08.13 | darkdrgn2k | Morning all |
| 15:08.26 | darkdrgn2k | any one ever use Cisco,Shortel or Tadiran telephone systems? |
| 15:09.09 | WIMPy | No. They are all vapourware. |
| 15:09.20 | darkdrgn2k | huu? |
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| 15:31.29 | StaRetji | folks, I have strange problem with asterisk 1.8 and a2billing. I can call local extensions, answer, all is okay. For some reasone, calling my mobile number, I receive a call and answer, but my sip account keeps ringing and even go troigh failover trunks |
| 15:31.39 | StaRetji | never understands the call is answered |
| 15:31.42 | StaRetji | any tip? |
| 15:39.06 | leifmadsen | StaRetji: trace the call (I assume it's via SIP) and see if the far end is sending any indication that call has been answered. Look at the console output of Asterisk and understand what it is doing and why it is continuing in the dialplan. |
| 15:39.20 | leifmadsen | your statement is quite vague and there isnt' really anything in particular anyone can point you to. |
| 15:39.34 | leifmadsen | you need to gather debugging information as a first step |
| 15:39.52 | StaRetji | I set debug 4 and verbose 4 |
| 15:39.58 | StaRetji | firstly, thx for answer |
| 15:40.08 | StaRetji | console doesn't complain at all |
| 15:40.30 | leifmadsen | look at the sip trace then and see why the other end isn't sending any indication that the line was answered |
| 15:40.33 | StaRetji | I'm banging my head several hours, googled around |
| 15:40.33 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 15:40.49 | leifmadsen | I bet the other end isn't sending something asterisk need to mark the call as answered |
| 15:41.17 | StaRetji | thx, let me try |
| 15:42.30 | *** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 15:44.56 | StaRetji | hm |
| 15:45.00 | wdoekes2 | StaRetji: are you sending out an RFC1918 ip in the Contact: in the INVITE? |
| 15:45.07 | StaRetji | I tried calling one of did numbers in same server |
| 15:45.45 | StaRetji | it also rings and do not understand that call is being answered. After I hangup, I get callback, weird |
| 15:45.49 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 15:45.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:45.51 | StaRetji | from the number I was calling |
| 15:46.08 | StaRetji | all I see in console is locally/bridginh |
| 15:46.33 | StaRetji | wdoekes2: thx for reply, I'm afraid I dont understand |
| 15:46.58 | StaRetji | I used how to install asterisk and same for a2b, excpected it will work out of the box |
| 15:47.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v005-016.mobile.uci.edu) |
| 15:47.55 | wdoekes2 | (hrm.. never mind the contact bit..) |
| 15:48.07 | wdoekes2 | in any case, sounds like your invites are going out, but you're not getting the replies |
| 15:48.13 | wdoekes2 | sip set debug on |
| 15:48.16 | StaRetji | yes |
| 15:48.20 | StaRetji | I did, debug 4 |
| 15:48.30 | StaRetji | very strange, no complaints |
| 15:48.31 | wdoekes2 | *gone* |
| 15:49.29 | StaRetji | I called with my sip account 9991 to did number, it rings on other side (did number attached to 9992) I answer, but 9991 keeps ringing |
| 15:49.37 | *** join/#asterisk seloha (~seloha@79-75-72-181.dynamic.dsl.as9105.com) |
| 15:49.40 | StaRetji | I hangup, 9991 keeps ringing |
| 15:49.50 | StaRetji | after some time, 9992 calls 9991 |
| 15:49.52 | StaRetji | wtf lol |
| 15:51.37 | koffel | <PROTECTED> |
| 15:51.53 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:52.29 | StaRetji | have tried menu checkconfig to see if you installed correctly? |
| 15:53.36 | StaRetji | urvg4: ping mate |
| 15:54.39 | koffel | yes |
| 15:54.47 | koffel | it says looks ok |
| 15:55.17 | koffel | * Checking if HTTP static support is enabled: OK |
| 15:55.17 | koffel | * Checking if manager is enabled: OK |
| 15:55.17 | koffel | * Checking if manager over HTTP is enabled: OK |
| 15:55.17 | koffel | <PROTECTED> |
| 15:55.17 | koffel | <PROTECTED> |
| 15:56.17 | StaRetji | well, type your ip address instead of linux-MS..... |
| 15:56.20 | StaRetji | ifconfig |
| 15:56.36 | koffel | i did |
| 15:56.41 | koffel | still same |
| 15:57.12 | *** join/#asterisk din3sh (din3sh@41.136.246.13) |
| 16:02.43 | [TK]D-Fender | koffel, #asterisk-gui <-- has its own channel as it is not supported here. |
| 16:02.49 | [TK]D-Fender | koffel, And do not flood in IRC. |
| 16:02.50 | [TK]D-Fender | ~pb |
| 16:02.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:02.52 | [TK]D-Fender | ^^^ |
| 16:03.50 | koffel | sorry tkd they at asterisk-gui never answer |
| 16:04.07 | [TK]D-Fender | koffel, it is virtually unsupported |
| 16:04.35 | koffel | okay what is supported gui wize in here |
| 16:05.17 | [TK]D-Fender | In HERE? None |
| 16:05.20 | leifmadsen | none |
| 16:05.34 | leifmadsen | #asterisk is for vanilla Asterisk configuration and support |
| 16:05.39 | [TK]D-Fender | koffel, #freepbx is fairly well supported... in their own channel |
| 16:05.46 | [TK]D-Fender | #freepbx |
| 16:06.18 | koffel | well is it hard to manaul config asterisk? |
| 16:06.44 | Qwell | ~book |
| 16:06.44 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:06.45 | Qwell | Not at all. |
| 16:06.53 | StaRetji | koffel: , I find harder to use gui lol |
| 16:07.33 | koffel | so i have to buy a book for asterisk lol |
| 16:07.57 | TheMan | no, don't have to buy it... just read it online |
| 16:08.02 | TheMan | but it is a great resource either way |
| 16:08.05 | [TK]D-Fender | No, you just have to read... which you are clearly already allergic to. |
| 16:08.16 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:08.27 | koffel | tkd i been up straight for 2 weeks |
| 16:08.53 | [TK]D-Fender | koffel, Time to become a Korean gamer... |
| 16:09.30 | koffel | llol |
| 16:09.33 | coppice | is "up straight for 2 weeks" an ad for viagra? |
| 16:09.47 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:16.31 | *** part/#asterisk seloha (~seloha@79-75-72-181.dynamic.dsl.as9105.com) |
| 16:22.14 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:23.48 | anonymouz666 | I really don't understand why this happens... when you sniff lets say eth0 (10.10.10.10) with wireshark, you watch the outgoing packets in the own interface that you are monitoring you see lost packets (from 10.10.10.10 to other address). Is this a fail at application layer (asterisk) missing the packets and then reflecting into network layer? I could not think other reason for that |
| 16:24.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:25.21 | *** join/#asterisk twanny796 (~twanny@85.232.219.136) |
| 16:26.50 | anonymouz666 | 172.30.130.11 14708 172.30.136.70 28946 0x21E5DA31 ITU-T G.711 PCMU 5727 674 (10.5%) 1139.91 3.93 0.09 X |
| 16:27.03 | anonymouz666 | I sniffed this traffic in eth0 172.30.130.11 |
| 16:27.21 | anonymouz666 | how could be possible this 10.5% of loss? |
| 16:27.33 | anonymouz666 | confused |
| 16:29.27 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 16:30.58 | anonymouz666 | coppice: viagra would be 4 hours not 2 weeks. |
| 16:31.03 | anonymouz666 | :P |
| 16:31.58 | [TK]D-Fender | anonymouz666, No... that's just the point where you're supposed to contact the doctor :) |
| 16:32.05 | coppice | it wouldn't be an ad without a little exaggeration |
| 16:35.06 | *** join/#asterisk celord (~celord@201.192.210.58) |
| 16:37.14 | *** join/#asterisk seloha (~seloha@79-75-72-181.dynamic.dsl.as9105.com) |
| 16:38.14 | LiuYan1 | hi, I need advice to build a large asterisk server(s). we're going to setup asterisk with 450 SIP users + 15 E1 trunks, can 1 single asterisk server (with a mordern hi-end hardware server) fit such requirement? |
| 16:39.24 | *** join/#asterisk kpettit (~kpettit@99-116-144-138.lightspeed.hstntx.sbcglobal.net) |
| 16:41.26 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 16:42.03 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 16:44.35 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 16:45.02 | talntid | one of my clients is saying that they are recieving voicemails ~2-3 days after they are left... anyone heard of this? |
| 16:45.07 | talntid | seems pretty much impossible to me |
| 16:45.16 | talntid | because time is synced properly on the server |
| 16:46.28 | *** join/#asterisk steve-o_ (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 16:48.37 | steve-o_ | Does anybody know why a call delivered to a sip phone from directory() wouldn't time out and go to voicemail if no answer? Dialing the same device directly it goes to vm if no answer. |
| 16:50.06 | MrTelephone | LiuYan1, what is the system for? You might want to use asterisk as your voicemail and gateway appliances and use a sip proxy to handle registrations. I'm trying to build an asterisk cluster using dialplan but it is difficult. |
| 16:51.38 | MrTelephone | I wouldn't be comfortable doing more than a 4 port t1 card in one dual xeon server. You could try 2 servers, 2 4-port e1 cards and see what the load is like. |
| 16:52.19 | MrTelephone | With that amount of call volume it would be best to spread it out over at least 4 servers. If you had 2 servers go down you would still have 8 e1's active. |
| 16:53.44 | LiuYan1 | currently, we want to integrate it with tradition system, so we won't run complecated applications on asterisk, just accept incoming call from PSTN(E1) and route it to SIP phones |
| 16:54.56 | MrTelephone | Is there 30 channels on an E1? |
| 16:55.03 | leifmadsen | 32 technically |
| 16:55.05 | LiuYan1 | yes |
| 16:55.07 | leifmadsen | 30B + 2D |
| 16:55.09 | [TK]D-Fender | steve-o_, First Directory() does not dial SIP devices directly. Next how it dials and what timeouts are attached have nothing to do with Directory(). |
| 16:55.23 | coppice | 30B + 1D and a sync channel |
| 16:55.27 | leifmadsen | ah right |
| 16:55.30 | MrTelephone | 30 voice channels then? |
| 16:55.33 | leifmadsen | yes |
| 16:55.33 | [TK]D-Fender | steve-o_, And the thought of VM following a call... again, that's more dialplan that Directory() has nothing to do with |
| 16:55.38 | MrTelephone | 80kbytes times 30 x 4 = ? |
| 16:55.58 | ChannelZ | No one told me there would be math. |
| 16:56.05 | [TK]D-Fender | MrTelephone, Where are you getting 88kB from? |
| 16:56.06 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:56.14 | MrTelephone | iptraf |
| 16:56.37 | MrTelephone | I can verify again, isn't g711 around 80kilobits |
| 16:56.38 | MrTelephone | sorry |
| 16:56.42 | MrTelephone | not kilobytes |
| 16:56.48 | [TK]D-Fender | And your E! is internet data used for SIP? |
| 16:56.51 | [TK]D-Fender | E1* |
| 16:57.12 | LiuYan1 | only voice |
| 16:57.20 | LiuYan1 | connected to tradition systems |
| 16:58.03 | MrTelephone | I'd like to see some statistics on someone who is running 15 e1's on one machine. |
| 16:58.25 | steve-o_ | [TK]D-Fender: I am setting up my first asterisk system so still learning. Directory() requires the vm and dialing contexts. So should I likely be looking in the [localsets] (in my case) context for the problem? |
| 16:58.55 | [TK]D-Fender | steve-o_, You should be looking in the contexts you are pointing things to and the dialplan you put there. |
| 16:59.10 | *** join/#asterisk qakhan (~qakhan@180.178.160.130) |
| 16:59.30 | LiuYan1 | MrTelephone: lol, do you mean that's very heavy to running 15 E1s on 1 single machine? |
| 16:59.46 | [TK]D-Fender | MrTelephone, And no... G.711 is 64kbit. RTP @ 20ms brings it to 85kbps of IP traffic... but that's IP. |
| 17:00.11 | [TK]D-Fender | And encapsulated as described |
| 17:00.15 | qakhan | i want to setup a table in mysql which has live agents. which agents are logged in in a queue and which agents are in call |
| 17:00.21 | qakhan | please help |
| 17:00.36 | MrTelephone | LiuYan, you would have a heart attack the first time that system fails. 450 people with no phone. |
| 17:00.41 | steve-o_ | [TK]D-Fender: I was confused because when I am dialing within that context calls do go to VM but when using directory() they don't. I will look further. Thanks. Let me know if you have further thoughts... |
| 17:01.08 | [TK]D-Fender | steve-o_, Only thought is "look at the call". That's what we're not seeing. The call. And configs. |
| 17:01.25 | MrTelephone | steve-o directory needs to know the context for the directory. If you are using odbc voicemail user storage you have to set list in directory=yes |
| 17:02.20 | MrTelephone | Is TDMoe still supported, can any seniors endorse the fact that it works will? |
| 17:02.24 | MrTelephone | s/will/well |
| 17:02.32 | steve-o_ | [TK]D-Fender: MrTelephone: Thanks guys....I need to dig deeper. |
| 17:02.57 | MrTelephone | steve-o, isn't it Directory([context])? |
| 17:03.21 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:03.45 | qakhan | [TK]D-Fender can you help me |
| 17:03.46 | [TK]D-Fender | MrTelephone, TDMoE is flakey, and any chump stuck with a RedFone gets what they deserve.... |
| 17:04.04 | steve-o_ | MrTelephone: Directory(vm-context[,dial-context,options]) |
| 17:04.45 | MrTelephone | Fender Bender, ever use tdmoe? |
| 17:04.54 | MrTelephone | oops sorry |
| 17:04.59 | MrTelephone | didn't see you respond |
| 17:06.39 | MrTelephone | Big companies are using TDMoe over radio links and selling the gear for 13,000. *cough* adtran |
| 17:07.11 | LiuYan1 | MrTelephone: thank you, maybe i need take a look at clustered asterisk servers |
| 17:07.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:08.04 | qakhan | i want to setup a table in mysql which has live agents. which agents are logged in in a queue and which agents are in call |
| 17:08.12 | din3sh | LiuYan1:http://www.red-fone.com/images/case_studies/intuit-large-callcenter.pdf |
| 17:08.44 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:08.57 | MrTelephone | LiuYan1, consider going with a cisco appliance with 16e1's and asterisk to handle the sip registrations and what not. It will be expensive though |
| 17:09.52 | MrTelephone | din3sh, how does the cluster work |
| 17:10.45 | din3sh | load balancing across servers using ultramonkey |
| 17:11.34 | qakhan | din3sh can u tell me |
| 17:11.37 | qakhan | i want to setup a table in mysql which has live agents. which agents are logged in in a queue and which agents are in call |
| 17:12.01 | LiuYan1 | qakhan: as far as I know, there's no way to setup live agents |
| 17:12.44 | LiuYan1 | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue |
| 17:14.25 | din3sh | qakhan: how do you intend to have live dynamic data in a table? |
| 17:14.28 | talntid | not out of the box, maybe, but I have it on mine |
| 17:14.42 | talntid | not in the exact same configuration he is asking for |
| 17:15.38 | qakhan | din3sh i want to see live agents in queue and agents who are in call |
| 17:17.17 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:18.05 | *** part/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:18.54 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:23.08 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:23.32 | LiuYan1 | din3sh: thanks, that's good case to study, 1000 agents, wow! |
| 17:23.54 | din3sh | 48 E1s they claim |
| 17:23.55 | din3sh | lol |
| 17:23.55 | MrTelephone | I'm curious how you do cross-asterisk calling? |
| 17:27.15 | din3sh | am curious too |
| 17:27.16 | din3sh | :p |
| 17:29.13 | MrTelephone | the authentication part does not work right when you try and relay calls to another asterisk system via SIP |
| 17:29.15 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:29.59 | MrTelephone | So a T1/E1 trunk between the 2 boxes seems like a good idea |
| 17:32.04 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:32.34 | TheMan | Any of you here use the new G100 or G200 gateways? |
| 17:32.48 | [TK]D-Fender | Whose? |
| 17:32.55 | TheMan | Digium's |
| 17:32.58 | TheMan | they just came out |
| 17:33.27 | TheMan | We just picked up a G100 to test with |
| 17:33.33 | TheMan | looks nice |
| 17:35.15 | TheMan | nice to have the ability to put the PBX in a location physically separate from where the PRI comes in |
| 17:35.28 | TheMan | and/or give failover options to your PRI connections |
| 17:35.37 | din3sh | MrTelephone: servers configured as load balancers have a virtual ip, phones point to the load balancers which in turn route traffic to the asterisk servers |
| 17:35.57 | [TK]D-Fender | TheMan, I'd be interested in seeing what it sells for... |
| 17:36.07 | TheMan | its about $1200 msrp |
| 17:36.11 | TheMan | for a single port |
| 17:36.14 | [TK]D-Fender | For 1 port? |
| 17:36.14 | [TK]D-Fender | eek |
| 17:36.25 | [TK]D-Fender | well.. that's MSRP.. |
| 17:36.30 | [TK]D-Fender | I'll wait till it hits retail |
| 17:36.40 | TheMan | it should have already |
| 17:36.53 | TheMan | we just ordered one from Distribution |
| 17:36.58 | TheMan | picked it up yesterday |
| 17:37.00 | din3sh | TheMan: have you tried digium phones too? |
| 17:37.03 | TheMan | not yet |
| 17:37.08 | TheMan | curious, though |
| 17:37.17 | TheMan | we're Polycom and Aastra partners already |
| 17:37.33 | din3sh | digium phone looks kinda polycom ones |
| 17:37.47 | TheMan | if you think about it, price wise (on the G100) it's not much more than core hardware would cost |
| 17:37.57 | TheMan | $700 or so for the PRI card with echo canceller |
| 17:38.07 | TheMan | plus the hardware for an * box to run it |
| 17:38.17 | TheMan | its all solid state, no moving parts |
| 17:38.31 | TheMan | 1u height, but a little narrower than standard 1U rackmount |
| 17:38.41 | MrTelephone | din3sh, sip realtime, when a phone registers it registers with all servers? |
| 17:38.50 | serafie | TheMan: They are exactly half-width, so you can mount 2 side-by-side. |
| 17:38.57 | TheMan | ah, IC |
| 17:39.05 | serafie | and, yes, they are nice! |
| 17:39.18 | TheMan | the interface is pretty... :) |
| 17:39.27 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 17:39.29 | serafie | Heh, yes it is. |
| 17:39.37 | din3sh | MrTelephone: realtime is in another cluster |
| 17:39.38 | din3sh | :p |
| 17:39.39 | TheMan | my only complaint is no IAX support |
| 17:39.45 | din3sh | i mean the db |
| 17:40.00 | serafie | TheMan: send in your feedback! |
| 17:40.00 | TheMan | SIP and T1/E1 only |
| 17:40.13 | TheMan | where, to Digium directly? |
| 17:40.29 | serafie | pengler@digium.com |
| 17:40.30 | TheMan | I'd LOVE to see IAX support on the G100/G200 boxes |
| 17:40.35 | MrTelephone | the load balance registers phone 1 on aterisk 1, phone 2 goes to asterisk 2, phone 1 calls phone 2 and the load balancer sends the invte to asterisk 1 where phone 2 is not. Am I missing something? |
| 17:40.41 | SaRSAeOL | not to interrupt but can anyone give me some suggestions on how to get header compression on even the first call in sip |
| 17:40.44 | serafie | or malcolm@digium.com |
| 17:40.57 | SaRSAeOL | so like without IAX trunking |
| 17:41.19 | din3sh | registers with the floating load balancer IP |
| 17:41.32 | din3sh | not the actual asterisk ip |
| 17:41.43 | TheMan | serafie, I'll do that |
| 17:41.59 | din3sh | the load balancers have a list of asterisk IPs to which they forward traffic |
| 17:42.35 | din3sh | so instead of having multiple asterisk ips, there's only 1ip (which is the load balancer floating ip) |
| 17:43.42 | MrTelephone | I get that part but I thought it does round robin when it sends the message to the asterisk ips. |
| 17:44.42 | din3sh | try the setup and you tell me |
| 17:44.44 | din3sh | :D |
| 17:48.10 | *** part/#asterisk steve-o_ (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 17:53.01 | MrTelephone | http://www.voip-forum.com/asterisk/2006-01/multiple-sip-registrations/ |
| 17:53.04 | MrTelephone | I have this problem but incoming |
| 17:56.45 | *** join/#asterisk geogeek (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 17:57.10 | SaRSAeOL | any suggestions on how to get header compression working with codec g723 and not using IAX? |
| 17:57.17 | MrTelephone | din3sh, I think my problem is that I'm using rtcache and so asterisk doesn't check the central database for peers |
| 17:58.21 | anonymouz666 | MrTelephone: what if you discover that asterisk will freeze the main thread if can't resolv a DNS peer hostname? |
| 17:58.52 | anonymouz666 | this issue is in Asterisk since day-1 of chan_sip, I guess. |
| 17:58.55 | din3sh | try setting the rtcache=no |
| 17:58.57 | din3sh | to test |
| 18:00.20 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 18:00.24 | MrTelephone | How did you get around it anonymouz? |
| 18:01.02 | WIMPy | SaRSAeOL: What header compression? And what does the codec have to do with it? |
| 18:01.38 | anonymouz666 | it isn't fixed. We have to live with it. Some will say to install a local cachenameserver, but then we have a TTL stuff etc. |
| 18:03.01 | MrTelephone | Does it crash the thread for calling or your entire machine? |
| 18:03.13 | anonymouz666 | using dnsmgr I think its just a matter of changing the thread will get stuck |
| 18:04.37 | anonymouz666 | it does not crash, the thread just wait for an answer that never comes, in the meanwhile the rest of chan_sip waits too and SIP devices does not work anymore. |
| 18:05.14 | SaRSAeOL | WIMPy: I'm now looking into sigcomp for compression of signaling, g723 was just mentioned as the preferred codec due to lower bandwidth usage than say 711 |
| 18:05.41 | anonymouz666 | g723 sucks. |
| 18:06.27 | anonymouz666 | you can have a payload of 5.3 or 6.3 in some devices, and when you add the overhead of other protocols there is almost no difference if you compare to g729 |
| 18:07.43 | WIMPy | sigcomp may be used for sip, but I don't see it with rtp. |
| 18:08.51 | *** join/#asterisk albertoandrade (~albertoan@186.206.4.118) |
| 18:09.09 | din3sh | SaRSAeOL: why do you want to compress in the first place? |
| 18:09.37 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 18:10.06 | SaRSAeOL | to be honest I'm not sure… theres a bandwidth concern |
| 18:10.40 | SaRSAeOL | WIMPy: i think ftp may be included… voip-info seems to say it is defined in rfc3320 |
| 18:10.45 | SaRSAeOL | not ftp… rtp |
| 18:11.31 | anonymouz666 | RTP 3550 |
| 18:11.33 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:11.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:11.34 | WIMPy | rtp is not mentioned in rfc3320. |
| 18:12.51 | din3sh | bandwidth? on what type of connection? IPLC? |
| 18:13.10 | SaRSAeOL | yeah as i look for myself i see that… i was going off of this statement from voip-info "SIP and RTP signalling compression is defined in SIGCOMP, RFC 3320." sorry |
| 18:13.51 | MrTelephone | does your regserver field get filled in when somoene registers? I can't get it to work. |
| 18:14.20 | din3sh | SaRSAeOL:bandwidth? on what type of connection? IPLC? |
| 18:18.41 | rrittgarn | Anyone run into confbridge only recording 44bytes of data for a conference call? I've got an entry for each conference call, but no data in it. I haven't tried setting the recording path manually, just trying to use the defaults first |
| 18:20.22 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-93-170.rev.ibsinc.com) |
| 18:26.03 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 18:26.57 | SaRSAeOL | sigcomp was evidently a workable solution… thanks all, sorry for the convoluted mess that was my question |
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| 18:33.45 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 18:54.49 | *** join/#asterisk angryuser (4ee09885@gateway/web/freenode/ip.78.224.152.133) |
| 18:56.25 | angryuser | Good day, i have installed asterisk 10.2.1 with one sangoma 2xE1 card, the voice was perfect until yesterday, i have not found any network issues, however wireshark dump shows me 100% out of seq packets, any ideas why ? |
| 18:57.06 | angryuser | It's a bug, or something else ? |
| 18:57.39 | angryuser | also wrong timestamp 100% |
| 18:57.42 | devmikey | can anyone rec a good voip did provider |
| 18:57.48 | devmikey | sip |
| 18:58.50 | angryuser | devmikey: Country ? |
| 18:59.00 | angryuser | Or at least continent |
| 19:00.27 | devmikey | usa |
| 19:00.48 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:01.52 | angryuser | have no idea |
| 19:02.23 | *** join/#asterisk TheMan (~Garry@wsip-24-234-115-86.lv.lv.cox.net) |
| 19:03.02 | WIMPy | ~itsplist-us |
| 19:03.02 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 19:03.34 | devmikey | SHould I expect to have to port forward sip ports? |
| 19:03.54 | devmikey | If my phone is behind a NAT? |
| 19:04.12 | angryuser | devmikey: and your server is on public ? |
| 19:04.57 | devmikey | currently I use a 3rd party provider, yes |
| 19:05.03 | WIMPy | devmikey: Usually no |
| 19:05.21 | devmikey | so they shouldn't be behind a NAT |
| 19:06.54 | angryuser | devmikey: you have only one phone, or more than one at the same location ? |
| 19:07.12 | angryuser | (connecting to the same provider) |
| 19:07.14 | devmikey | 1 phone |
| 19:07.19 | devmikey | just me |
| 19:07.43 | angryuser | devmikey: if your router has a bit of brains, you dont need to forward |
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| 19:27.32 | devmikey | well i cant even do an echo test. i'm using the codec they sat to |
| 19:27.38 | devmikey | say* |
| 19:29.06 | watchy | my t.38 cisco ATA is here |
| 19:29.10 | watchy | now how do i use it |
| 19:29.11 | gusto | i even got voip working on a NAT'ed umts on vodafone / germany |
| 19:31.46 | watchy | i need to negotiate me a raise at work |
| 19:34.05 | angryuser | watchy: go to #samary |
| 19:34.17 | angryuser | ;) #salary |
| 19:36.31 | gusto | in todays economical situation with high growth of inflation a pay rise would be a good option |
| 19:37.05 | *** join/#asterisk TheMan (~Garry@wsip-24-234-115-86.lv.lv.cox.net) |
| 19:37.20 | watchy | well with my skillset and how much I have saved the company in the 3 years i've been here, i don't see why it would be out of the question |
| 19:38.09 | watchy | i have saved around 300k or so by doing everything in house and using 0 outside contractors for things they initially wanted to do by out sourcing to other contractors |
| 19:38.36 | gusto | you should talk to them about that |
| 19:38.43 | gusto | there is nothing we can do about it |
| 19:38.51 | watchy | you can hug me |
| 19:38.58 | carrar | We can format your hard drive |
| 19:39.06 | carrar | Lets start with that |
| 19:39.15 | watchy | man i just ate lunch and i ate too much catfish |
| 19:39.24 | watchy | probably gonna die from mercury now |
| 19:39.25 | leifmadsen | any == too much |
| 19:39.34 | watchy | you don't like fried catfish? |
| 19:39.49 | SaRSAeOL | watchy: mmmmm |
| 19:39.58 | leifmadsen | nah |
| 19:40.08 | carrar | Does it taste like cat or fish? |
| 19:40.11 | watchy | i'm in arkansas. catfish is the local delicacey |
| 19:40.33 | watchy | next week is all you can eat crawfish |
| 19:40.38 | coppice | if catfish is the local delicacy you need to move |
| 19:40.44 | watchy | haha |
| 19:40.45 | devmikey | lol |
| 19:40.58 | watchy | this is arkansas, its not like we have much else |
| 19:40.59 | carrar | with or without BP Oil on it? |
| 19:41.05 | Naikrovek | when I was a kid I broke up all the thermometers in the hous and played the mercury. all the tiem |
| 19:41.10 | watchy | without we aren't near the ocean |
| 19:41.34 | watchy | most of the catfish at places around here are locally farmed |
| 19:41.35 | carrar | not too far though |
| 19:41.49 | gusto | how many americans are here? |
| 19:42.12 | watchy | me. probably the rare one |
| 19:42.13 | coppice | I can see the ocean from where I'm sitting, although its a bit dark. |
| 19:42.21 | gusto | i see this channel as the one with the most americans concentration here |
| 19:42.22 | watchy | coppice: where you located |
| 19:42.27 | gusto | maybe US citisens like to talk |
| 19:42.34 | carrar | 311,591,917 |
| 19:42.36 | coppice | here vvvvvv |
| 19:43.36 | coppice | to be accurate I can't see an ocean. I can see a sea |
| 19:43.46 | carrar | Can you see Russia? |
| 19:44.02 | coppice | I'm not Sarah Palin |
| 19:44.11 | watchy | hah |
| 19:44.16 | gusto | but sarah palin is hot |
| 19:44.25 | gusto | i would shake her |
| 19:44.30 | watchy | what your nuts |
| 19:44.30 | gusto | but she is dumb, of course |
| 19:45.04 | coppice | hot? them's some low standards you got there |
| 19:45.15 | gusto | why? |
| 19:46.15 | coppice | I guess she'd be more attractive with a gag |
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| 19:48.30 | gusto | what is a gag? |
| 19:48.40 | *** join/#asterisk Kl4m (~kelam@gw2.noc1.sys-tech.net) |
| 19:48.43 | coppice | something to shut her up |
| 19:49.01 | gusto | yes |
| 19:49.08 | gusto | agreed |
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| 20:19.12 | rrittgarn | Anyone have any experience with confbridge and recording calls? |
| 20:22.46 | Kl4m | I have an issue with a Polycom IP 601: Since about 2 weeks, the client complains that she can't transfer some incoming calls. I tried dialing out to SIP,IAX2,or DAHDI for the transfer and the result is the same: "Call accepted by [...]" or such depending on protocol, then immediate hangup without answer. I have no idea what to try next |
| 20:23.22 | Kl4m | ^ I have iax.conf:notransfer=yes and sip.conf:canreinvite=no as well |
| 20:24.07 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 20:27.55 | watchy | any changes made? |
| 20:31.17 | Kl4m | no changes afaik (can't tell about ITSP) |
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| 20:33.42 | irule | anyone ever done a 3com 3100 entry phone? I gave it power, red light flashes and I dont see a new IP address on the network :s |
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| 20:41.43 | hypknight | <Local/1000@from-queue-cc28;2>AGI Rx << Extension '<path to script>/customerSetAgentInfo.agi.php' not present. |
| 20:41.43 | hypknight | <Local/1000@from-queue-cc28;2>AGI Tx >> 510 Invalid or unknown command |
| 20:42.04 | hypknight | anyone seen Asterisk throw an error with a PHP AGI file like this? |
| 20:42.11 | hypknight | I've verified the path |
| 20:42.15 | hypknight | verified the permissions |
| 20:42.22 | hypknight | verified the shebang line |
| 20:42.32 | hypknight | and there's another AGI script running in the call path that works perfectly |
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| 21:00.55 | hypknight | more specific question: if I'm inside of the [from-internal-custom] context and executing AGI, how do I do what I need to do, and then allow the script to continue executing along [from-internal] as it normally would? |
| 21:02.09 | WIMPy | When an AGI finishes, the dialplan continues. |
| 21:05.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:07.13 | hypknight | that's what I thought... then this is definitely killing it... |
| 21:07.49 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 21:08.55 | *** join/#asterisk kresp0 (~kresp0@213.37.151.211.dyn.user.ono.com) |
| 21:10.04 | kresp0 | hi WIMPy, everyone |
| 21:12.11 | jaytee | I have a strange intermittent problem with no audio that seems to affect only outbound calls and calls from extension to extension. I've been unable to capture sip debug output so far on a failed call with no audio today while testing but I've had several failed calls like that. This is an Asterisk 1.6.2.20.2 running on Centos 5.7 This client's network has Dell Powerconnect managed switches |
| 21:12.11 | hypknight | this thing is driving me nuts... noone has seen this before? |
| 21:12.11 | jaytee | and I've verified that the CoS queues are giving RTP the highest priority but I don't think the switches are properly configured and handling the traffic flow. Anyone here ever had an issue with Dell Powerconnect switches? |
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| 22:01.23 | irule | what ports do I need to forward from my adsl modem to *? I would like to register my sip phone through the internet |
| 22:06.23 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 22:12.53 | fprior | Hi, I've a GrandStream GXW4108 FXO Gateway; when dialplan execute Hangup(), FXO line change to idle but on remote phone of caller there is only silence. Any idea ? |
| 22:16.18 | *** join/#asterisk Agro (~Agro@108-79-20-223.lightspeed.hstntx.sbcglobal.net) |
| 22:16.26 | Agro | Is there anyone here familiar with using spandsp? |
| 22:17.17 | [TK]D-Fender | What "remote phone"? |
| 22:17.30 | Agro | ? |
| 22:17.52 | Chainsaw | Agro: I use the spandsp-based fax functionality, yes. |
| 22:17.57 | Chainsaw | Agro: With Asterisk 10.3.1 |
| 22:18.05 | Agro | Have you coded with it, in C? |
| 22:19.34 | Agro | @ Chainsaw |
| 22:20.38 | Chainsaw | Agro: Nope, only used it through the Asterisk fax application. |
| 22:20.59 | Agro | Ahh... Do you have any tips on tone detection or at least writing a function to do it with spandsp? |
| 22:21.06 | fprior | [TK]D-Fender: as 'remote phone' I'm refer to phone's caller. sorry for ambiguous temrs. |
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| 22:21.57 | Agro | @ Chainsaw :| |
| 22:22.06 | [TK]D-Fender | fprior: and if you plug an analog phone of your own in parallel with the GXW and pick it up, is the line indeed free? |
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| 22:47.43 | fprior | [TK]D-Fender, I've checked what you have suggested: line never hangup from telephony provider. So, if you call me, we speak, I hangup and you forgot phone "open", you will stay connected all time. Is normal this ? |
| 22:48.19 | GG111 | fprior: this was normal in the PSTN days |
| 22:48.25 | [TK]D-Fender | fprior: Then your GXW has issues |
| 22:49.11 | [TK]D-Fender | fprior: Easily confirmed by just answering on an analog phone, hanging up, and picking up and getting tone. |
| 22:49.30 | WIMPy | GG111: In the VERY early days of the PSTN that is. |
| 22:50.19 | GG111 | WIMPy: depends where you live, I recall that still happening in the 90's, if some called you, and did not hang up well, you could not make a call. |
| 22:50.56 | GG111 | and you chould just hear they're stupid chatter and shout " helllo !! hey!! hang up the phone!!" |
| 22:51.34 | fprior | [TK]D-Fender, in my case: answering on an analog phone, hanging up, and picking up and continue call |
| 22:51.36 | WIMPy | The 90s of which century? |
| 22:52.29 | kresp0 | WIMPy, I remember that, less than 20 years ago here in spain. |
| 22:52.39 | GG111 | WIMPy: how old are you ? |
| 22:52.53 | WIMPy | 41 |
| 22:52.57 | GG111 | I guess you were'nt a teenager in the 90's |
| 22:53.03 | GG111 | bingo :) |
| 22:53.08 | kresp0 | xD |
| 22:53.22 | WIMPy | I know that from the 70s. |
| 22:53.37 | kresp0 | on which country? |
| 22:53.38 | WIMPy | Ok, that's not the beginning, but a long time ago. |
| 22:53.46 | GG111 | Well , I grew up in Isreal and we had it there, in the 90's |
| 22:53.55 | GG111 | *Israel |
| 22:54.19 | GG111 | Don't know, perhaps they still have in in analog lines there. |
| 22:57.02 | fprior | so, or I'm in a bad place or I traveled back in time. |
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| 23:52.36 | talntid | are call files deleted upon completion of the call, or when it starts? |