| 00:04.40 | *** join/#asterisk Druken (n=jdumais@70.54.242.169) |
| 00:04.45 | Druken | evening everyone |
| 00:04.57 | Druken | anyone actually around currently? |
| 00:05.17 | Draegon | think they all out cooking |
| 00:06.09 | Druken | what is the current stable any idea? it's been a very long time.. hehe my production server is still 1.2.23 |
| 00:07.02 | Draegon | not sure, i'm kinda new, had someone built a dial plan for me, they went on vacation so trying to figure it out myself >< lol |
| 00:07.22 | Druken | oh, did you pastebin it? |
| 00:08.02 | Draegon | pastebin? |
| 00:09.30 | Druken | well, if there was something you didn't understand, you can paste the dialplan in a pastebin (program online that allows you to paste stuff for others to see) |
| 00:10.07 | Druken | that way we can see what you have, and can give you a fix, or at least a hint of what's going on hopefully |
| 00:10.10 | Draegon | ahh |
| 00:10.32 | Draegon | i kinda understand it, but trying to add a feature to bridge 2 incoming call |
| 00:10.37 | Draegon | so i dont have the code for it, lol |
| 00:11.38 | Druken | define bridge two incoming calls |
| 00:12.02 | Druken | you mean like connect the two so they can talk, like a dating service sorta thing? |
| 00:12.19 | Draegon | right now, someone call in, it put them on hold then call a list of numbers |
| 00:12.45 | Draegon | i want to make it so that instead of calling a list of number, leave them on hold, then have someone else call in, type in a code |
| 00:12.56 | Draegon | now i have the 2 channel stored in the database |
| 00:13.06 | Draegon | how do i bridge those 2 call so they talk to each other |
| 00:13.42 | Draegon | there is parking, meetme, parking has timeout, not sure if it will cause any problem. i read there was a bridge command |
| 00:13.52 | Draegon | but i cant seem to locate detail on it. lol |
| 00:14.18 | Draegon | yeah, kinda like dating service knowing the 2 channel |
| 00:14.53 | Druken | how do you intend on knowing the second channel? |
| 00:15.11 | Druken | and how does the second person know to call in and receive the call? |
| 00:15.13 | Draegon | on the first channel, i store a password + the channel |
| 00:15.21 | Draegon | then when they call in, i ask them to key in the password |
| 00:15.50 | Draegon | then i chekc the database for the password, to get the first channel, i am processing the 2nd channel |
| 00:16.09 | Draegon | how do i join the 2. i got it to do the password, verify password and everything, i just dont know how to join the 2, lol |
| 00:16.14 | wonderworld | i wanto to continue in the dialplan after a user has left a voicemail. is that possible at all? |
| 00:16.28 | Draegon | 2nd person get a page sent to him, with the password |
| 00:16.55 | Draegon | i think you have to look at deadagi wonderworld |
| 00:17.36 | wonderworld | like running voicemail from agi and using deadagi to go on afterwards? |
| 00:17.40 | Druken | Draegon: i'm sorta missing the link, sorry... perhaps if you can explain what it's gonna be used for i can have a better understanding |
| 00:18.01 | Draegon | yeah, i was reading for past few day, deadagi allow you to execute while no call exist (after they hang up) |
| 00:18.05 | Druken | wonderworld: yeah you can have the dialplan continue after the user hangs up |
| 00:18.07 | Draegon | i just dont know how it work ;) |
| 00:18.10 | Druken | just don't remember how |
| 00:18.24 | wonderworld | yes, i did that before.... |
| 00:18.27 | wonderworld | good idea |
| 00:18.38 | wonderworld | but i am not sure how to send a user into voicemail from agi |
| 00:19.07 | Draegon | right now, someone call in, press 1, it send out a page + call the person. i can make it so it just hold there, and send the page |
| 00:19.13 | Draegon | the page will have a password |
| 00:19.24 | wonderworld | the thing is..... asterisk needs to place a call to a nurse after someone has left a voicemail and must play it to her |
| 00:19.32 | Draegon | the person receive the page, will get instruction to call in and type the password |
| 00:19.44 | Draegon | with that password, i'll be able to find out what channel the other guy is on |
| 00:19.58 | Draegon | so now i just need to bridge this channel to that channel |
| 00:20.03 | wonderworld | maybe a shell-script would be more simple.... |
| 00:20.26 | Draegon | yeah shell script |
| 00:20.31 | Draegon | check if voicemail exist |
| 00:20.38 | Draegon | then call and just crobtab it |
| 00:20.42 | wonderworld | ya |
| 00:20.57 | wonderworld | there is a dialplan function for checking for voicemail, right? |
| 00:21.11 | manxpower | VMCOUNT |
| 00:21.30 | wonderworld | thanks a lot guys |
| 00:21.35 | manxpower | I think it's set by MailboxExists |
| 00:21.49 | Draegon | a=`grep "$chan" /var/spool/asterisk/voicemail/multi_dir/$vm/{INBOX,Old}/*.txt 2>/dev/null` |
| 00:21.53 | Draegon | something like that... |
| 00:22.24 | Druken | Draegon: why not instead of a password, just use an input to a meetme, and then page the meetme to the guy |
| 00:22.30 | Druken | would be a hell of alot easier |
| 00:23.05 | Draegon | im running this on a vps, trying to avoid meetme since i dont know how well the timing issue it will handle? |
| 00:23.35 | Draegon | i already ahve most thing, i just need a way to bridge 2 call, probably just need more reading, lol |
| 00:23.48 | Druken | manxpower: what's the current reccomended stable? |
| 00:24.05 | manxpower | Druken: depends on who you ask |
| 00:24.09 | Druken | http://www.voip-info.org/wiki/view/Asterisk+cmd+Bridge |
| 00:24.10 | Draegon | lol |
| 00:24.15 | Druken | well, i asked YOU :) |
| 00:24.43 | manxpower | 1.6.0.x if you don't have anything that requires 1.4.x |
| 00:24.51 | Draegon | yes! that one, lol how do i find out what version of asterisk i have? lol |
| 00:25.02 | Druken | show version on the cli |
| 00:25.02 | Draegon | i think i have 1.4 |
| 00:25.04 | manxpower | "help" in the CLI |
| 00:25.06 | wonderworld | asterisk -r |
| 00:25.18 | Draegon | hah i try show ver |
| 00:25.20 | Draegon | earlier, lol |
| 00:25.37 | Druken | manxpower: i'm upgradding from 1.2.23.... what does that tell you? |
| 00:25.50 | Draegon | crap only 1.4, is there a way to implement bridge from 1.6 to 1.4 or is upgrading the only way? |
| 00:26.31 | wonderworld | i spent two nights checking my setup for errors because i couldn't get a line out. telco told me everything was fine. |
| 00:26.47 | wonderworld | after two days i called them again, they resetted the port and the line was up ;( |
| 00:27.04 | geneticx | Hello everyone, I have a linksys sipura PAP2NA ATA and I currently have 2 phones plugged in and asterisk seems both as registered, but the ATA led light for line 2 is not on and I can't hear anything either if I pick up that line? anyone in here who had a similar problem? |
| 00:27.08 | Draegon | heh always the little thing |
| 00:27.38 | Druken | geneticx: reset the entire adaptor, and reconfigure it |
| 00:28.14 | Druken | manxpower: do i still need to have the compiling source in /usr/src/asterisk ? |
| 00:28.18 | geneticx | Drunken: Ok ill try that right now.. |
| 00:28.50 | wonderworld | the PAP2T seems to be rather nice. it works great with a old fax-machine |
| 00:28.56 | Druken | i've had that happen i don't know how many times... in my own personal experince, pap2's suck ass |
| 00:29.08 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:29.08 | wonderworld | these boxes are still expensive |
| 00:30.10 | wonderworld | Druken: what adaptor would you reccomend? |
| 00:30.47 | Draegon | Asterisk 1.4.21 as packaged in Debian Lenny has backported support for |
| 00:30.47 | Draegon | Bridge(). |
| 00:31.20 | Draegon | does that mean i can someone get bridge support on my 1.4.22? |
| 00:31.25 | Druken | wonderworld: i don't use adaptors anymore... i do voip only |
| 00:34.00 | wonderworld | i need them for faxes most of the time. many people don't want to virtualize their faxes. i never get it, but they seem to be too used to oldschool faxing. |
| 00:34.04 | Druken | why do i have a feeling i'm going to have to redo ALL my dialplans with this upgrade.... uhg,... haven't touched a dialplan in years |
| 00:34.51 | Druken | you'd be amazed on how many people ask me for my fax number.. i always tell them, don't have one. never worked right anyways... email it to me |
| 00:35.09 | Druken | 9/10 i get an ok, no problem |
| 00:36.27 | wonderworld | hehe, MY fax isn't the problem but most of my customers don't want to trash theirs.... |
| 00:36.55 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:37.24 | wonderworld | well... time to go to bed. thanks again for the help....gnight.... |
| 00:38.27 | Druken | fax is so old now anyways... how many times have ya seen someone print something off the computer, and shove it into the fax and down into the recycling bin? |
| 00:41.58 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-66-205.twcny.res.rr.com) |
| 00:42.01 | Orbixx | hates fax with a passion |
| 00:42.28 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 00:42.57 | Druken | Orbixx: you sound very passionate about that |
| 00:44.00 | Draegon | lol |
| 00:44.42 | Orbixx | Electronic documents via email or nothing. |
| 00:45.42 | Orbixx | How can I manipulate incoming callerid? |
| 00:46.04 | Orbixx | i.e. I want the callerid to be altered just before I drop the caller into a queue. |
| 00:46.12 | KyleK | set callerid? |
| 00:46.13 | Druken | just set it |
| 00:46.36 | Orbixx | It's deprecated. |
| 00:46.40 | Druken | i used to have to do that when i brought in analog lines, i had to prepend the area code |
| 00:46.42 | manxpower | You need to read The Book |
| 00:46.45 | manxpower | ~answers |
| 00:46.45 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:46.56 | KyleK | whats deprecated |
| 00:47.10 | KyleK | Orbixx: these asterisk people change stuff around very slightly a lot |
| 00:47.25 | KyleK | set debug on vs. core set debug on |
| 00:47.42 | manxpower | 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:47.48 | Druken | manxpower: have i a hope in hell on using any config from 1.2 in 1.6? |
| 00:47.56 | manxpower | That's why those Asterisk people include the UPGRADE files. |
| 00:48.02 | KyleK | :o |
| 00:48.15 | KyleK | grep -i callerid UPGRADE*.txt? ;) |
| 00:49.14 | geneticx | Drunken: That did it. Thanks very much. |
| 00:49.35 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 00:49.59 | manxpower | Druken: There are two basic types of questions here. The questions that could easily be answered by reading some docs. Those questions tend to get the asker insulted in some way. The other questions are for things that can't be easily found in the docs. |
| 00:50.26 | KyleK | lol |
| 00:51.38 | Draegon | hmm i have a .diff for bridge, i assume i have to run this thru a diff match program to replace the diff, and then recompile asterisk? |
| 00:51.51 | Druken | manxpower: yes, well, i've been demoted back to a newbie, since i haven't touched the system in years.... |
| 00:51.55 | Orbixx | What's the new 'reload' in 1.6RC? |
| 00:52.01 | Druken | forgive my forgetforness |
| 00:52.18 | KyleK | Druken: the old config is a good start but you'll probably have to make changes to each file |
| 00:52.22 | Druken | geneticx: it work? |
| 00:52.42 | geneticx | Drunken: Yes sir, flawlessly. Thanks.. |
| 00:52.56 | KyleK | Dial(SIP/somewhere,20) in 1.6, is that Dial(SIP/somewhere|20) in 1.2? |
| 00:53.28 | geneticx | Drunken: it shouldn't really take a reset to add another phone but I understand... =D |
| 00:53.46 | *** join/#asterisk lucasb (n=bussey@s154-5-252-231.bc.hsia.telus.net) |
| 00:53.56 | Druken | i've had it do that after a couple days of operation |
| 00:54.05 | Druken | and had to reset and reconfigure everything |
| 00:55.30 | Orbixx | KyleK: Correct. |
| 00:56.54 | geneticx | device seems to run a little too hot though .. |
| 00:59.22 | KyleK | hey has voicemailmain been replaced by anything? it's odbc support looks tacked on |
| 01:00.02 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 01:00.26 | Druken | geneticx: yeah, try to make sure it stands verticle |
| 01:03.40 | coppice | all you need for good cooling is a sufficiently windy day |
| 01:04.51 | geneticx | Drunken: Ok.. |
| 01:05.01 | Orbixx | Can anybody explain why Asterisk might be ignoring any following DTMF input after the first digit? |
| 01:07.11 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 01:07.17 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 01:08.04 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 01:08.46 | Draegon | dial = place call, ringing tone, bridge with current call |
| 01:08.58 | Draegon | is there a way to use that to bridge the first call to the current call? |
| 01:10.15 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 01:10.38 | Orbixx | [TK]D-Fender: Could I PM you for a little assistance, I think I have enough info for you to draw an immediate conclusion. |
| 01:10.55 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 01:11.29 | Joel | hrm |
| 01:12.19 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 01:17.37 | Orbixx | Meh, I'll throw it out in the open. |
| 01:17.44 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:19.08 | Orbixx | I have a Background() directive which only ever accepts 1 digit, even though there are extensions in the context of more than one digit. Subsequently to hitting a digit, the user usually ends up in a queue with some MOH. If I call up and hit 1 single digit, everything goes as planned - however, if I call up and hit multiple digits, the first digit is only taken into account and the rest of the digits seem to execute after the user gets put into the qu |
| 01:20.00 | Orbixx | My question is... Why? And how can I include consideration for extension numbers 2 digits long or more? |
| 01:22.24 | Druken | [Sep 7 21:22:14] WARNING[6647]: config.c:2010 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 01:23.08 | Druken | someone feel like giving me a hint to which config file?? :) |
| 01:23.29 | *** join/#asterisk Kumbang (n=epic@rusnas.paume.itb.ac.id) |
| 01:28.43 | KyleK | Orbixx: dont force that call to be just one digit? |
| 01:28.45 | Draegon | is it possible to do this? person A call in, i send him to an extension 8888. person B call in, i have that person pick up extension 8888 so now A is talking to B? |
| 01:29.12 | Orbixx | KyleK: It's not, Asterisk waits for more digits, but it doesn't take them into account even though they're dialled. |
| 01:29.30 | KyleK | Draegon: whats person A get to do while waiting for person B? |
| 01:29.40 | Draegon | music on hold? |
| 01:29.53 | KyleK | hmm that sounds like call parking |
| 01:29.54 | Draegon | and can press 1 to goto voicemail |
| 01:30.06 | KyleK | ah |
| 01:30.15 | Draegon | there is a bridge command in 1.6 |
| 01:30.21 | Draegon | but im afraid to upgrade, lol |
| 01:30.25 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 01:30.46 | KyleK | ah |
| 01:31.15 | Draegon | supposely there is a backport in debian, but im not sure how to use it |
| 01:32.03 | Draegon | if i go the parking route, can A do the usual music on hold and voicemail if they dont want to wait? |
| 01:32.30 | Draegon | i see some discussion about joining 2 channel back in 2003, im surprise there isnt an easy way to do this |
| 01:32.52 | *** join/#asterisk freakazoid0223 (n=knoppix@pool-71-246-17-206.phlapa.fios.verizon.net) |
| 01:35.49 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 01:38.51 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:40.01 | *** join/#asterisk DetroitArmz (n=DetroitA@74.204.89.249) |
| 01:43.09 | *** join/#asterisk DetroitArmz (n=DetroitA@74.204.89.249) |
| 01:44.15 | *** join/#asterisk Ta^3 (n=tacvbo@189.146.189.241) |
| 01:46.50 | *** part/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 02:13.39 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 02:14.07 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:16.07 | rossand | I'm looking for information on setting up iptables to allow sip. I learned the hard way that port 5060 (UDP in my case) was not enough - it left me able to connect but with no sound in either direction. By chance does anyone mind sharing their iptables rule(s) for the extra ports? Thanks. |
| 02:17.39 | drmessano | Whatever happened to cli_aliases? |
| 02:23.33 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 02:29.19 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 02:31.10 | jaytee | rossand, try this out http://www.voip-info.org/wiki/view/Asterisk+firewall+rules |
| 02:32.05 | rossand | jaytee: thanks. I was in the middle of testing them. Found 'em on google also. Thanks! |
| 02:33.44 | rossand | jaytee: FYI works great |
| 02:33.54 | jaytee | great! |
| 02:37.03 | jaytee | don't forget to do an iptables-save so you don't lose the changes after a reboot |
| 02:37.07 | rossand | That was a strange problem to solve. Without ports 10000:20000, it fails silently - literally. |
| 02:37.25 | rossand | Thanks. I did my changes in /etc/sysconfig/iptables (fedora box) |
| 02:37.31 | jaytee | yeah, 10000-20000 is for RTP which is the audio portion of the call |
| 02:37.54 | rossand | There are no logs telling you "hey dude, I can't set up the audio for this call" |
| 02:39.17 | jaytee | that's where sip set debug on and setting a high level of verbosity on the command line helps |
| 02:49.17 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 02:57.34 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 03:04.40 | dlynes | Anyone come across the bug in Asterisk 1.6.1.1, whereby it's trying to read /var/run/asterisk.pid, instead of where /etc/asterisk/asterisk.conf is telling it should be finding it? |
| 03:06.42 | dlynes | Even specifying -C /etc/asterisk/asterisk.conf doesn't seem to help |
| 03:08.46 | drmessano | 1.6.1.1 is old |
| 03:08.59 | dlynes | erm |
| 03:09.03 | dlynes | 1.6.1.6 I meant |
| 03:09.11 | dlynes | sorry for the confusion |
| 03:09.48 | Draegon | anyone know if it's possible ot bridge 2 channel without upgrading to 1.6? |
| 03:10.33 | dlynes | Weird...and it only happens when I don't specify -vvvvvvvvv |
| 03:10.44 | Draegon | or connect an incoming call to a call that is running already? |
| 03:10.52 | dlynes | Draegon: yes, it's done all the time |
| 03:11.09 | dlynes | Draegon: it's been possible since asterisk 1.0, I believe |
| 03:11.28 | Draegon | i cant seem to do it, how do i do it? |
| 03:11.39 | dlynes | Draegon: it's done automatically |
| 03:11.51 | dlynes | Draegon: perhaps you're not phrasing your question correctly? |
| 03:11.59 | Draegon | probably, lol |
| 03:12.02 | dlynes | Draegon: what do you mean by 'bridging'? |
| 03:12.17 | Draegon | i have 2 incoming call, 1 is current waiting and in music on hold |
| 03:12.34 | Draegon | the 2nd call coming in, i want it to talk to the first incoming call |
| 03:13.17 | dlynes | Draegon: ok, and are they related at all, in the dial plan? |
| 03:13.18 | dlynes | Draegon: i.e. how do you know to connect the two? |
| 03:13.41 | Draegon | the first incoming call, i can put it on hold and save the channel or whatever info in the database |
| 03:13.44 | Draegon | with a password |
| 03:13.52 | dlynes | Draegon: exten => _X.,1,Dial(SIP/2ndchannel) would usually do it if you want to bridge an incoming channel with an outgoing channel |
| 03:14.02 | dlynes | Draegon: assuming the incoming call is also sip |
| 03:14.10 | Draegon | the 2nd incoming call, ivr will pick it up, ask them for a password, and when they type the password it, it check against the DB to get the channel |
| 03:14.22 | dlynes | Draegon: I don't know if asterisk is capable of bridging two calls using different technology yet |
| 03:14.34 | Draegon | yeah they both sip |
| 03:15.13 | Draegon | so i just need to do Dial(Sip/channeliwanttobridge) ? |
| 03:16.34 | [TK]D-Fender | dlynes: No, he wants to INBOUND calls to bridge |
| 03:16.52 | [TK]D-Fender | Draegon: "core show application meetme" |
| 03:17.25 | dlynes | Draegon: http://www.voip-info.org/wiki/view/Asterisk+cmd+Bridge |
| 03:17.32 | dlynes | Draegon: it's new as of Asterisk 1.6 |
| 03:18.14 | dlynes | Draegon: I had mentioned dial woudl allow you to bridge inbound to outbound, not inbound to inbound |
| 03:18.14 | Draegon | yeah, i heard someone have a backport to 1.4, but i normally not good enough in linux so cant seem to figure it out |
| 03:18.50 | Draegon | was trying to figure a way without upgrading to 1.6. i have a custom freepbx module that i use and afraid it might not work |
| 03:19.08 | dlynes | Draegon: if you don't want to upgrade all of your boxes to 1.6, just make sure they're upgraded to at least 1.4.24, so that they can still communicate with the 1.6 box properly |
| 03:19.27 | Draegon | [TK]D-Fender: which is better? parking or meetme? |
| 03:19.47 | [TK]D-Fender | Draegon: Depends on the circumstances about how you know which call to pick up. |
| 03:22.01 | Draegon | right now, my custom module developed by some guy, put the incoming call on hold while the system try to call a number for someone to pickup, then connect the call, during that time, they can press 1 to goto voicemail |
| 03:22.31 | [TK]D-Fender | Draegon: that isn't a 2nd INCOMING call. |
| 03:22.44 | Draegon | if i break the middle part, send it to meetme, wait for the 2nd incoming call to come in, would i still be able to send them to voicemail if they dont want to wait anymore? |
| 03:22.46 | [TK]D-Fender | Draegon: That is an OUTGOING call. Please get your story straight |
| 03:22.56 | Draegon | yes, currently it's outgoing |
| 03:23.09 | dlynes | [TK]D-Fender: I'm guessing the new Bridge app is a modified Meetme app? |
| 03:23.11 | [TK]D-Fender | Draegon: And you can already trell Dial to exit on DTMF. |
| 03:23.43 | Draegon | 1 incoming, then an outgoing, im tryng to change it so that 1 incoming, an sms is sent out to get someone to call in, which would be 2 incoming |
| 03:23.43 | [TK]D-Fender | dlynes: Not quite. It is a direct ad-hoc channel bridge given the way 1.6 got restructured |
| 03:24.05 | [TK]D-Fender | Draegon: How would the system know to match up the incoming to the 1st call? |
| 03:24.31 | Draegon | the sms that is sent out would send information needed |
| 03:24.43 | Draegon | when the 2nd incoming call come in, i ask them to key in a password |
| 03:24.55 | Draegon | it match that password to locate where that meetme/parking is? |
| 03:25.03 | [TK]D-Fender | Draegon: Then use dynamic Meetme rooms <- |
| 03:25.26 | Draegon | ohhh how does that work? |
| 03:25.40 | [TK]D-Fender | Draegon: "core show application meetme" <--------------------- |
| 03:26.44 | Draegon | oh so first incoming, create a conference, send them in there with a pin setup, 2nd call send him directly to the meetme with a pin |
| 03:26.51 | Draegon | hmm i might not be able to code that >< lol |
| 03:35.15 | Draegon | [TK]D-Fender: do you know if asterisk 1.6 is production ready yet? |
| 03:36.12 | [TK]D-Fender | Draegon: 1.6.0 is quite mature, 1.6.1 could very well be. |
| 03:38.16 | Draegon | is it pretty given that my freepbx module on 1.4 wont work on 1.6? or it shouldnt matter? |
| 03:38.41 | [TK]D-Fender | Draegon: Depends on the module. |
| 03:38.52 | [TK]D-Fender | Draegon: And this isn't #freepbx |
| 03:40.03 | Draegon | ok, thanks |
| 03:45.16 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca) |
| 03:47.07 | phix | [TK]D-Fender: what's with asterisk having namespaces on their interactive console now |
| 03:47.22 | [TK]D-Fender | phix: ? |
| 03:47.38 | phix | you need to put core infront of things that were just refered to without the vore |
| 03:47.42 | phix | core |
| 03:48.02 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:48.17 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:48.18 | [TK]D-Fender | phix: Changed quite a while back. Allows modules to prefix commands more clearly. |
| 03:49.06 | phix | yeah, I supose |
| 03:50.24 | dlynes | I like the new timestamp feature...much more useful than before |
| 03:50.36 | dlynes | They should've had that feature a long time ago |
| 03:50.49 | dlynes | Like in v1.0 or something |
| 03:51.35 | dlynes | Is there a way to override where asterisk is looking for asterisk.pid and asterisk.ctl? |
| 03:52.42 | dlynes | hrm cute....bugs.digium.com is now issues.digium.com... |
| 03:52.52 | dlynes | erm issues.asterisk.org i mean |
| 03:53.40 | Draegon | [TK]D-Fender: if my current AGI script is handling incoming call #2, can i send incoming call #1 that is in a music on hold state in another channel to the Meetme at that time when call #2 is coming in and i am processing it? (i might not be wording this right) |
| 03:53.43 | *** join/#asterisk s0lid (n=s0lid@122.53.97.197) |
| 03:54.35 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:54.41 | [TK]D-Fender | Draegon: AMI Redirect <- |
| 03:56.53 | Draegon | ok thanks, going to read up on it |
| 04:00.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:06.12 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 04:08.07 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:14.56 | *** join/#asterisk freakazoid0223 (n=mrfreaky@pool-71-246-17-206.phlapa.fios.verizon.net) |
| 04:19.44 | s0lid | anyone tried to interconnect asterisk with mera? |
| 04:20.58 | [TK]D-Fender | s0lid: Google seems to think so |
| 04:21.31 | [TK]D-Fender | loves it when people ask questions that are longer to type for us than it is to get an answer from Google... |
| 04:23.08 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 04:28.28 | phix | google is inpersonal though |
| 04:28.34 | phix | there isn't a connection you get like on IRC |
| 04:28.48 | phix | then again most ppl on IRC are fuckwits and tell you to google any way, but hey :) |
| 04:28.56 | [TK]D-Fender | phix: Yeah... Google can't tell you "fuck off you lazy bastard" :p |
| 04:30.07 | drmessano | Really now? |
| 04:30.09 | drmessano | http://tinyurl.com/l534t9 |
| 04:31.21 | [TK]D-Fender | drmessano: that isn't goolge telling you that personally... |
| 04:31.27 | [TK]D-Fender | Goolge! |
| 04:31.38 | [TK]D-Fender | goes to register "goolge.com" |
| 04:33.03 | KyleK | :o |
| 04:33.34 | KyleK | phix: jfgi is the new "not me/dunno" |
| 04:35.31 | [TK]D-Fender | KyleK: I count jfgi as my "foylb" :) Frankly some dimwits in here want you to retype entire pages of the fucking BOOK that some great people wrote and even handed out for free, just because they want someone to do it by HAND line by line for them |
| 04:36.08 | phix | [TK]D-Fender: heh |
| 04:36.15 | [TK]D-Fender | KyleK: So am I going to recite it for them? NO, I'm thinking a personal delivery is in order.... in a Jason Bourne kinda way.... |
| 04:36.28 | phix | BRB |
| 04:36.31 | phix | restart requried |
| 04:39.47 | s0lid | [TK]D-Fender: i would love to if he has na answer |
| 04:39.50 | phix | back |
| 04:40.34 | phix | who is Jason Bourne? |
| 04:42.35 | [TK]D-Fender | phix: IMMINENTLY GOOGLE-ABLE |
| 04:43.18 | phix | :P |
| 04:43.29 | phix | I know :) |
| 04:43.39 | phix | I was seeing if I could break you |
| 04:43.48 | phix | or make you crack, wathever |
| 04:43.57 | [TK]D-Fender | reaches for his katana... |
| 04:45.25 | drmessano | Well, I didnt expect a sort of spanish inquisition |
| 04:45.38 | drmessano | NOBODY EXPECTS THE SPANISH INQUISITION |
| 04:45.39 | carrar | You got spanish out of katana? |
| 04:45.43 | carrar | xWTFx |
| 04:45.59 | drmessano | OUR CHIEF WEAPON IS TFOT |
| 04:46.13 | carrar | My name is Inigo Montoya, you killed my father |
| 04:46.13 | drmessano | GOOGLE AND TFOT |
| 04:46.14 | Nugget | laughs |
| 04:46.22 | carrar | prepare to die! |
| 04:46.29 | carrar | prepair even |
| 04:46.41 | Nugget | Number eight: The Larch |
| 04:46.46 | drmessano | OUR TWOOO WEAPONS ARE GOOGLE, AND TFOT, AND A FANATICAL DEVOTION TO THE WIKI |
| 04:47.58 | drmessano | Nugget: Win |
| 04:48.42 | drmessano | CARDINAL FANG |
| 04:48.51 | drmessano | BRING ME.. THE SOFT CUSHIONS |
| 04:51.45 | p3nguin | [tk]d-fender: What's funny is that goolge is registered to google. |
| 04:57.04 | phix | Nugget: monty python? |
| 04:57.27 | Nugget | naturally. gotta pull out the Big Guns to hang with drmessano |
| 04:57.35 | phix | heh |
| 04:57.48 | phix | dcc me some eps |
| 04:58.44 | phix | so, how about that asterisk ay |
| 04:59.16 | drmessano | heh |
| 05:00.20 | drmessano | phix: I try to avoid EPS, I find TIFF files far more enjoyable |
| 05:00.58 | phix | drmessano: heh, xvid :) |
| 05:01.05 | phix | dvdrip |
| 05:02.09 | drmessano | Reminds me of one of the biggest clusterdouche head implosions I ever experienced.. Had to send a station logo to a print shop to get some shirts made. I sent them the 8MB TIFF we had, and they insisted they couldn't use it.. they needed an EPS on a 100MB MAC formatted ZIP Disk |
| 05:02.31 | drmessano | I had to start sniffing glue again to get my mind past it |
| 05:03.02 | [TK]D-Fender | makes a note to self... drmessano was sniffing glue BEFORE as well... |
| 05:03.12 | drmessano | _again_ |
| 05:03.15 | drmessano | :D |
| 05:04.30 | Draegon | hey guys, i have an phpagi script, i made it so that when i call in, i enter a password, then it park me, but after that the system hang up on me, how do i tell it to play music on hold and make that parking wait there until i tell another script to pick it up? |
| 05:04.57 | drmessano | I would break it to it gently |
| 05:07.32 | phix | heeh |
| 05:08.36 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 05:08.50 | phix | I need a tribble |
| 05:08.54 | phix | plush |
| 05:09.10 | [TK]D-Fender | phix: My mother has one... it purrs & all that.. |
| 05:09.11 | drmessano | I want a damn furby |
| 05:09.26 | drmessano | I had one of those little bastards once |
| 05:09.29 | phix | drmessano: ppfftt, tribble are way better than furbys |
| 05:09.42 | drmessano | phix: Have you ever owned a furby? |
| 05:09.47 | phix | [TK]D-Fender: ummm was that a metaphor? |
| 05:10.08 | phix | drmessano: why the hell would I when tribbles exist! |
| 05:10.08 | [TK]D-Fender | phix: No, one of those battery-operated tribbles.... |
| 05:10.19 | phix | battery-operated ay |
| 05:10.22 | phix | hmmmmm |
| 05:11.26 | drmessano | You're out of your element, phix |
| 05:11.35 | drmessano | Vladimir Ilyich Lenin!! |
| 05:13.02 | phix | indeed |
| 05:13.02 | drmessano | Furby's were awesome.. they were harder to shut up than a 6 yr old.. Duct tape, staples, screaming, short bursts with an oxy-acetylene torch... All the things you do to a kid, but the Furby is completely unfazed |
| 05:13.09 | phix | I need to shit, BRB |
| 05:18.17 | [TK]D-Fender | checkout time, later all |
| 05:18.46 | phix | heh |
| 05:20.26 | *** join/#asterisk fiddur (n=fiddur@dhcp08.textalk.com) |
| 05:23.04 | *** join/#asterisk maour_ (n=gnu@unaffiliated/maour) |
| 05:27.40 | *** join/#asterisk psiforce (n=mark@eth1426.vic.adsl.internode.on.net) |
| 05:29.02 | psiforce | hey russell, you alive atm? |
| 05:30.33 | TJNII | I remember furbies. I put red LEDs behind the eyes of mine. Made the little creepy furry thing that much more creepy. |
| 05:38.04 | phix | BBL |
| 05:45.35 | maour_ | isn't this correct ? exten => s,n,Set(COUNT=$[${COUNT} + ${ONE}]) , asterisk 1.6 give me this ! :p >> syntax error: syntax error, unexpected '+', expecting $end; |
| 05:46.02 | maour_ | prev line is exten => s,n,Set(ONE=1) |
| 05:54.48 | carrar | try Set(COUNT=$["${COUNT}" + "${ONE}"]) |
| 05:57.14 | carrar | verify COUNT is a number |
| 05:57.34 | carrar | shouldn't need quotes |
| 05:59.29 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
| 06:05.06 | maour_ | my problem solved ! by putting this on extension exten => s,n,Set(COUNT=0) |
| 06:05.38 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 06:09.38 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 06:12.45 | *** join/#asterisk zamba (n=marius@flage.org) |
| 06:17.54 | *** join/#asterisk xrmx__ (n=rm@host197-226-dynamic.1-79-r.retail.telecomitalia.it) |
| 06:20.20 | *** part/#asterisk scalex000 (n=chatzill@77puntacana02.codetel.net.do) |
| 06:24.13 | *** join/#asterisk p3nguin (i=BuhPjX1J@i.got.my.network.from.a2infotech.com) |
| 06:30.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:40.06 | *** join/#asterisk LapTop006 (n=laptop00@bean.studio442.com.au) |
| 06:47.05 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
| 06:47.51 | *** join/#asterisk [netman] (n=netman@175.Red-81-36-135.dynamicIP.rima-tde.net) |
| 06:48.22 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81.178.65.1) |
| 06:48.36 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81.178.65.1) |
| 06:59.17 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:13.19 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:20.48 | *** join/#asterisk Tim_Toady (n=moi@adsl94-108.kln.forthnet.gr) |
| 07:23.17 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 07:29.52 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-dzjellpqhdehqrbq) |
| 07:34.56 | *** join/#asterisk flohack (n=fhackenb@213.47.44.14) |
| 07:39.29 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:41.05 | *** part/#asterisk psiforce (n=mark@eth1426.vic.adsl.internode.on.net) |
| 07:45.22 | *** join/#asterisk Kevin` (n=kevin@67.52.47.69) |
| 07:46.49 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:dc56:6c40:1663:9f25) |
| 07:52.07 | Zhad | In a user declaration in sip.conf, is permit=192.168.0.0/255.255.255.0 meaningless if it isn't prepended with a deny=0.0.0.0/0.0.0.0 ? |
| 07:56.34 | Zhad | Hmm, that appears to be the case. (oops). |
| 08:01.59 | *** join/#asterisk KermitTheFragger (n=KermitTh@217.149.197.118) |
| 08:05.55 | *** join/#asterisk maour_ (n=gnu@unaffiliated/maour) |
| 08:28.38 | *** join/#asterisk Subdolus (i=dexterit@creep.bur.st) |
| 08:31.20 | *** join/#asterisk mumtazah (n=mumtazah@203.82.79.101) |
| 08:33.51 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
| 09:00.34 | Jenna | hey there ! anyone working with voicetronix hardware |
| 09:00.54 | Jenna | any ideas on debuging this chan_vpb.cc:2722 ast_module_load_result load_module(): No Voicetronix cards detected ? |
| 09:01.58 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
| 09:17.39 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 09:20.53 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:21.45 | *** join/#asterisk maskas (n=maskas@179.111.181.58.dynamic.max.com.pk) |
| 09:21.50 | maskas | hello |
| 09:22.46 | maskas | I am getting calls over IAX and transfering them out over a pri, I get this message "IAX2/2ngateway-10044 requested special control 20, passing it to DAHDI/8-1" repeatdly till the call is answered, what is it for? |
| 09:23.01 | maskas | running asterisk 1.4.26.1 |
| 09:25.27 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-66-205.twcny.res.rr.com) |
| 09:32.31 | *** join/#asterisk TimToady_ (n=moi@adsl32-126.kln.forthnet.gr) |
| 09:47.59 | *** join/#asterisk mumtazah (n=mumtazah@203.82.79.101) |
| 09:52.37 | *** join/#asterisk jgoo (n=tyyjyj@212.152.99.250) |
| 09:52.46 | jgoo | hrm, I have asterisk not loading - how can I check why? |
| 09:53.09 | jgoo | it has been loading the last few weeks, today, not, on a reboot |
| 09:53.31 | jgoo | I run asterisk by myself, but then asterisk -vvvvvvvvvvvvvvvvvvvvvr cays asterisk.ctl not present |
| 09:54.16 | *** join/#asterisk jksM (n=jks@193.189.93.254) |
| 09:54.41 | jksM | anyone knows what new in the Polycom SP 321/331 phones compared to the 320/330? |
| 09:56.23 | jgoo | for some reason now it complains about zaptel configuration... how could that be? |
| 09:57.06 | TimToady_ | jgoo ztcgf -fv |
| 09:57.33 | TimToady_ | and then asterisk -cvvv to see if tis working |
| 10:01.42 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 10:04.17 | jgoo | TimToady_, ok.. that worked... |
| 10:04.40 | jgoo | cool... so , usually it works on reboot, but 3 reboots now, nothing, and manually typing asterisk... but this worked, thakns |
| 10:04.49 | TimToady_ | then put ztcgf -fv in ur rc.local |
| 10:05.00 | TimToady_ | or generaly check how u startup asterisk |
| 10:05.26 | jgoo | ok... does everyone have to do that? |
| 10:05.52 | jgoo | ok.. also, while looking for the root of this problem, I have some questions on startup, ks or groundtart.. adn rx signaling... and a few other parameters |
| 10:06.01 | TimToady_ | depends on how u isnatll asterisk and zaptel, usualy they come with rc scripts that take care of all these |
| 10:06.42 | TimToady_ | s/isnatll/install |
| 10:07.07 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 10:07.28 | *** join/#asterisk shnootz (n=Hanan@77.73.189.210) |
| 10:08.03 | jgoo | TimToady_, I am going with Elastix now... you think that is a well maintained / well setup distro? |
| 10:08.30 | jgoo | I would want ot setup asterisk myself, but the two times I've done my own installs, about6 months apart, lots of config stuff had changed (at that time) and it #(*#@ me off :p |
| 10:08.41 | TimToady_ | no idea, never used it. I guess its just centos with * |
| 10:08.53 | TimToady_ | centos is generally a solid distro |
| 10:09.23 | jgoo | TimToady_, well,a bit more... if you put in an ISDN card, or plug in an external ISDN rack, it detects and configures it (and it worked... which shocked the hell out of me) but not using it in production |
| 10:09.46 | jgoo | TimToady_, I am asking if its configuration and asterisk setups are stable moreso (if is causes issues) |
| 10:10.04 | jgoo | I have a hangup issue on ISDN, I didn't see it as there are 6 channels, but now I see it ... hrm |
| 10:11.02 | TimToady_ | jgoo i have no experience with elastix, u can try #elastix |
| 10:12.08 | jgoo | I gathered, I was just clarifying the question anyway |
| 10:23.08 | Draegon | hey guys, if inside my agi script, i put a call on park. if i exit(0), it disconnect the call. do i basically have to run a loop and tell it to just wait wait wait until that call is picked up? and would it cost a problem? |
| 10:42.50 | *** join/#asterisk mumtazah (n=mumtazah@203.82.79.103) |
| 10:46.27 | maskas | I am getting calls over IAX and transfering them out over a pri, I get this message "IAX2/2ngateway-10044 requested special control 20, passing it to DAHDI/8-1" repeatdly till the call is answered, what is it for? |
| 10:48.09 | EmleyMoor | It indicanes that the source of the media has changed |
| 10:48.20 | EmleyMoor | indicates |
| 10:48.40 | EmleyMoor | ... at least according to a posting on the mailing list |
| 10:48.53 | maskas | umm any reason why I would be getting this message, I havent changed anything in my config and have just suddenly started getting it |
| 10:49.41 | EmleyMoor | maskas: What have you changed other than your configL |
| 10:49.42 | EmleyMoor | ? |
| 10:52.04 | maskas | emley: sorry thanks I got the error |
| 11:00.11 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 11:00.13 | iksik | hm |
| 11:00.17 | iksik | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static - is it usefull? |
| 11:15.25 | kaldemar | iksik: if you prefer handling configurations with a database, yes. |
| 11:16.05 | iksik | yes, but I already have configuration for sip accounts, extensions, and voicemail in database... and for now I don't understand what is it for |
| 11:17.55 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 11:20.00 | *** join/#asterisk maour_ (n=gnu@unaffiliated/maour) |
| 11:21.38 | kaldemar | iksik: storing configuration that cannot be stored with dynamic realtime. whole configuration files. |
| 11:21.48 | iksik | hm |
| 11:22.48 | iksik | are there any articles where I can read about asterisk real time configuration with postgres? (I've already read all of them from voip-info.org) |
| 11:22.53 | kaldemar | for example dynamic and static realtime sip peers behave differently and you can put all config under [general] in a database. |
| 11:23.14 | kaldemar | the book uses postgresql in its examples. |
| 11:23.46 | *** join/#asterisk TommyBotten (n=tommy@217-14-12-26-dhcp-osl.bbse.no) |
| 11:24.49 | garymc | anyone in here know why my GUI isnt displaying play/download call link? |
| 11:25.29 | kaldemar | sure. it's broken. |
| 11:27.50 | iksik | kaldemar, hum which book? |
| 11:28.02 | seanbright | ~tfot |
| 11:28.03 | infobot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 11:28.16 | seanbright | iksik: ^^^^^^^ that one |
| 11:28.25 | iksik | oh, ok :) |
| 11:29.00 | iksik | "Websites located on this server are currently down for maintenance. They will return shortly." |
| 11:29.00 | iksik | ;/ |
| 11:33.24 | garymc | kaldemar : know of a fix? |
| 11:34.33 | kaldemar | iksik: http://www.asteriskdocs.org/ |
| 11:35.47 | kaldemar | garymc: for what? you don't even tell what GUI you're using. besides, i know you've been told quite a few times that GUI's are not supported on this channel. |
| 11:36.23 | garymc | kaldemar: I know that im in the channel but no response for 2 days now |
| 11:36.34 | garymc | its freepbx |
| 11:39.36 | *** join/#asterisk Jabka (n=jabka@hspot.sce.ac.il) |
| 11:40.12 | Druken | can someone help me with realtime in 1.6? |
| 11:42.07 | TommyBotten | Druken: Just ask the question... people will help if they are able to |
| 11:42.50 | Druken | [Sep 8 07:42:41] WARNING[6647]: config.c:2010 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 11:42.54 | Druken | what config file? |
| 11:44.47 | Jabka | I'm trying to understand the idea of LEN cards (used by Bezeq INC) , could anyone explain about it (from one side it connectes to your home wall jack from the other to PCI) can it be used by asterisk ? |
| 11:46.44 | Chainsaw | Jabka: Depends on whether it's a telephony adapter or a modem (contrary to popular belief, the two devices are quite different). |
| 11:46.51 | Chainsaw | Jabka: Do you have a datasheet for it at all? |
| 11:47.32 | Jabka | Chainsaw , no afaik it is a telephone adapter used in SYS12 exchange box |
| 11:48.26 | Chainsaw | Jabka: Knowing the chipset on the board would be helpful. |
| 11:48.33 | Chainsaw | Jabka: If you're not sure, stick it on a flatbed scanner on 100dpi and share the JPEG somewhere. |
| 11:49.16 | Jabka | Chainsaw , I can't acces it yet (it is in a telephon compney ) im thinkg about getting one but not sure if it worth it |
| 11:49.42 | Chainsaw | Jabka: Without knowing more about the hardware I can not in good conscience recommend you spend money on it. |
| 11:50.22 | Jabka | Chainsaw , Becouse of that i came here i thouth people new that peace of hardware |
| 11:50.35 | Jabka | or worked in that componey |
| 11:50.40 | iksik | hm... Attempted to update column 'useragent' - which tables should got this column? o.O |
| 11:50.49 | iksik | i can't see anything like that in examples |
| 11:58.53 | Druken | ~docs |
| 11:58.54 | infobot | docs is probably for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 11:59.10 | Druken | ~book |
| 11:59.11 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:04.02 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 12:04.24 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-3-211.w90-56.abo.wanadoo.fr) |
| 12:05.09 | merlin8282 | Hi. Is it possible to make asterisk run in a VM ? Will the hardware (TDM400P and QuadBRI and DuoGSM) be recognized ? |
| 12:05.47 | iksik | ok, i've got it... but hum, sip show users/peers, registry, doesn't show anything from CLI ... is it normal? |
| 12:06.28 | kaldemar | iksik: static or dynamic? |
| 12:06.40 | iksik | dynamic - I think :D |
| 12:07.27 | Chainsaw | merlin8282: Timer performance may be greatly impaired by running in a virtualised environment. Especially things like conferencing could be problematic. |
| 12:07.39 | kaldemar | iksik: you don't see any peers with dynamic until they register. |
| 12:07.49 | iksik | I see in DB, that my ISP gk user, was registered, cause there is some value in regseconds, and ipaddr |
| 12:08.07 | Chainsaw | merlin8282: That's if you can even persuade the host OS to let you have exclusive access to the telephony hardware in the first place. |
| 12:08.14 | iksik | but sip show registry doesn't show anything hmm |
| 12:08.16 | Chainsaw | merlin8282: That was the long answer. The short answer: No. |
| 12:09.41 | merlin8282 | Chainsaw: ok. What I am trying to do is to find out how I can compile a second version of asterisk (bristuff) on the same server (which is a production one) without being impacted (that's to say "the telephone network must continue working") |
| 12:10.06 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:10.27 | Chainsaw | merlin8282: You're free to compile it (call make) as long as you either change prefix during configure (so it installs in parallel in a completely different directory structure) or don't call make install. |
| 12:11.05 | merlin8282 | ok, thanks. |
| 12:11.31 | Chainsaw | merlin8282: Whether that's helpful depends on what you ultimately want to do of course. |
| 12:12.27 | merlin8282 | Sure. I only want to upgrade asterisk, so I am able to use Rx/Txfax apps. |
| 12:13.02 | dlynes | merlin8282: Digium's got a better solution now; they have a t.38 module |
| 12:13.40 | dlynes | merlin8282: http://www.digium.com/en/products/software/faxforasterisk.php |
| 12:14.26 | dlynes | merlin8282: the problem with rxfax/txfax is the asterisk code keeps breaking support for it |
| 12:14.38 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.22 | Chainsaw | dlynes: It's spandsp that can't decide on an API. |
| 12:15.25 | Chainsaw | dlynes: Not so much Asterisk. |
| 12:15.30 | dlynes | merlin8282: it hasn't been officially supported since 1.2.something and a very early 1.4 version... |
| 12:16.02 | dlynes | Chainsaw: some of the internal asterisk API has also changed |
| 12:16.23 | dlynes | Chainsaw: between 1.2 and 1.4 and again from 1.4 to 1.6, there was a bigger change |
| 12:16.24 | Chainsaw | dlynes: They had the decency to bump the version number when they did. |
| 12:16.50 | Chainsaw | dlynes: It's ifdef-MANIA for anything SpanDSP. |
| 12:17.43 | dlynes | Chainsaw: anyways...as I said...the faxforasterisk is a lot more painfree |
| 12:18.14 | dlynes | Or just bridge a couple of zaptel channels, too |
| 12:22.02 | Chainsaw | wonders why a Siemens Gigaset C450 consistently misses the window for a SIP qualify |
| 12:22.18 | Chainsaw | UNREACHABLE/REACHABLE, ping-pong, ping-pong, etc. |
| 12:22.58 | iksik | kaldemar, I see that my pap2t is online now... But still I can't see it under sip show peers or users ;/ |
| 12:23.55 | Chainsaw | The exact same problem this guy had, basically: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg164270.html |
| 12:24.34 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:27.57 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 12:35.45 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:38.13 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 12:40.50 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 12:45.38 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 12:51.30 | Druken | someone feel like helping me out with realtime on 1.6? |
| 12:55.54 | *** join/#asterisk d00gster (n=doughant@94.98.222.187) |
| 12:57.11 | kaldemar | Druken: you need to install res_config_mysql from asterisk-addons if you use mysql. |
| 12:58.30 | Druken | still with 1.6 realtime is an addon? |
| 12:58.41 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 12:58.57 | manxpower | ~answers |
| 12:58.58 | infobot | hmm... answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:59.21 | kaldemar | Druken: realtime is not an addon. res_config_mysql, i.e. the mysql configuration engine is. |
| 13:00.27 | *** join/#asterisk chiwawa_42 (n=jerome@ALille-552-1-99-218.w92-147.abo.wanadoo.fr) |
| 13:00.29 | chiwawa_42 | Hi ! I'm looking for ATAs and IP phones supporting any high quality codec, preferably speex. Is there anything avaible or any project for an open-spource one, that wouldn't be on the first 10 result pages from google ? |
| 13:01.27 | [TK]D-Fender | chiwawa_42: What do you consider "low quality codec"? Who are you going to be speaking to with it? |
| 13:02.25 | SuPrSluG | anything connected to an ATA is going to be 8kHz. |
| 13:02.34 | chiwawa_42 | [TK]D-Fender: gsm and sometime G.711 don't offer what I need, I.E. less than 40kbps for a decent sound quality |
| 13:03.07 | [TK]D-Fender | ChiOh, now you want better than G.711 at LOWER bitrate? |
| 13:03.30 | chiwawa_42 | yes, speeks in wideband mode |
| 13:03.33 | chiwawa_42 | speex* |
| 13:04.54 | chiwawa_42 | wich may be uncommon on asterisk as the transcoder only runs at 8kHz afaik, but it's usually flawless with freeswith or with direct connections |
| 13:05.10 | [TK]D-Fender | chiwawa_42: Nothing I can think of |
| 13:05.22 | [TK]D-Fender | ChiAgain, who are you going to be speaking with on it? |
| 13:05.23 | chiwawa_42 | np, thanks anyway :) |
| 13:05.46 | chiwawa_42 | other member on this network |
| 13:05.53 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:06.22 | [TK]D-Fender | chiwawa_42: If they're on your network, then why the BW restraint? |
| 13:06.26 | chiwawa_42 | it's a small associative (non profit) ISP, trying to build services with cuting edge open source projects |
| 13:06.35 | *** join/#asterisk dwayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 13:06.39 | chiwawa_42 | so we have mostly adsl users |
| 13:07.34 | chiwawa_42 | some users are too far form their CO to get enough bandwidth for flawless G.711 operation |
| 13:07.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.50 | SuPrSluG | There are some phones coming out w/ siren (Polycom's open sourced codec) when not sure. |
| 13:09.02 | chiwawa_42 | SuPrSluG: yeah, heard of that, but siren7 is patented |
| 13:10.07 | SuPrSluG | well CELT is the best. Can't understand why manufactures wouldn't be all over that. |
| 13:11.11 | Druken | [TK]D-Fender: heya, ltns.. how are things? |
| 13:11.32 | [TK]D-Fender | Polycom and plenty more support G.722 and is a significant improvement over G.711 at no increase in BW. If your users can't even afford 1-2 calls on their DSL then thats just plain sad |
| 13:12.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:12.26 | chiwawa_42 | SuPrSluG: yeah, CELT has a slightly better sound wuality than speex, but it's not significant on voice only, more sensible for "on hold music" and recorded announcements. But it's also more heavy on bandwidth, the minimum is 32kbps |
| 13:14.31 | SuPrSluG | Polycom is the only real choice for decent sound out of the box. |
| 13:15.18 | SuPrSluG | the HD Voice stuff is nice. Best you can get at the moment. |
| 13:15.53 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:15.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:16.10 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 13:16.50 | chiwawa_42 | thanks SuPrSluG, ill try to order some and check it out ;) |
| 13:17.10 | SuPrSluG | np |
| 13:18.00 | [TK]D-Fender | chiwawa_42: taht would be G.72 on the Polycom BTW |
| 13:18.04 | [TK]D-Fender | G722 |
| 13:18.16 | [TK]D-Fender | So 85kbps. |
| 13:20.32 | *** join/#asterisk michael-i (n=michael-@141.41.40.153) |
| 13:21.16 | michael-i | hi all! I have kind of an off-the-wall question; is there any way to force a specific DAHDI controlled port's lights to blink? |
| 13:21.27 | brad_mssw | What would cause this error: WARNING[11191]: app_dial.c:1528 dial_exec_full: Unable to create channel of type 'dahdi' (cause 0 - Unknown) |
| 13:21.40 | brad_mssw | Just upgraded to 1.6.1.6 from 1.4 |
| 13:21.57 | brad_mssw | can't get outgoing to work |
| 13:22.01 | michael-i | brad_mssw: I'd say chan_dahdi wasn't built because of unrecognized deps |
| 13:22.13 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:22.33 | brad_mssw | michael-i: 'help dahdi' shows the dahdi related calls ... would those exist if that was true? |
| 13:22.33 | SuPrSluG | did you update your dialplan too? if not look for dahdi tools |
| 13:22.55 | brad_mssw | SuPrSluG: yes, dialplan has been updated to use DAHDI instead of Zap |
| 13:23.14 | michael-i | brad_mssw: I'm wrong then, your channel is there |
| 13:23.19 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 13:24.04 | brad_mssw | 'dahdi show channel 1' shows InAlarm:0 .... so I assume the channel is ok |
| 13:24.53 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-64-34.kgldga.dsl.dynamic.tds.net) |
| 13:25.13 | brad_mssw | another channel in the same 'group' _does_ have an alarm status though ... could that be causing the issue? Even though I'm explicitly calling the dahdi channel, not the group? |
| 13:29.48 | [TK]D-Fender | brad_mssw: Show us the COMPLETe call attempt |
| 13:30.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:31.39 | brad_mssw | [TK]D-Fender: http://pastebin.com/m3ee3bb8b |
| 13:32.33 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 13:33.05 | [TK]D-Fender | brad_mssw: perhaps you need a newer DAHDI release... |
| 13:33.18 | chiwawa_42 | [TK]D-Fender: humpf, that's a bit heavy... Will keep on searching then. |
| 13:33.53 | [TK]D-Fender | chiwawa_42: Good luck... the only phone's you're likely to find using oddball codecs will likely be crap in and of themselves. |
| 13:34.15 | brad_mssw | [TK]D-Fender: I'm on 2.2.0.1 ... |
| 13:34.19 | iksik | what is a difference between USER and PEER? |
| 13:34.24 | [TK]D-Fender | ChiAnd then trying to see if * will even acknowledge the codecs is another matter |
| 13:34.26 | iksik | and FRIEND :) |
| 13:35.15 | chiwawa_42 | yeah, I think so... I'd probably be luckyier with analog cards in the IPBX and softphones for home users ;) |
| 13:35.30 | [TK]D-Fender | iksik: iksik http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer |
| 13:36.13 | [TK]D-Fender | chiwawa_42: Sounds like a waste of money & effort |
| 13:36.21 | brad_mssw | [TK]D-Fender: it's a TDM400P with 2x FXO (chan 1 & 2) and 2xFXS (chan 3 & 4) |
| 13:36.42 | brad_mssw | [TK]D-Fender: I have another identical card if you think it could be h/w related |
| 13:37.29 | [TK]D-Fender | brad_mssw: Please exit *, "dahdi_cfg -vvvv", PB your configs, restart *, "dahdi show status", "dahdi show channels", and try your call again |
| 13:38.08 | chiwawa_42 | [TK]D-Fender: research for higher quality services based on opensource isn't a waste, it's just what asterisk projects started for |
| 13:38.25 | brad_mssw | [TK]D-Fender: sure thing, hold on a sec |
| 13:39.02 | [TK]D-Fender | chiwawa_42: Depends if the journey is really more important than the destination. to me "non-profit" sounds like "doesn't HAVE money and time to waste" |
| 13:39.43 | Druken | what is the realtime replacement in 1.6 for the dialplan ? |
| 13:40.32 | chiwawa_42 | [TK]D-Fender: well, the site is written in french only, but here it is : www.fdn.fr . Our goal is to defend net-netrality and freedom of speech by proposing alternative services and encouraging people and groups to create their own (micro-)ISPs |
| 13:40.49 | manxpower | Druken: All major changes are listed in UPGRADE*.txt |
| 13:41.29 | chiwawa_42 | adressing home users rather than only geeks means we need to have some better products than the bigger ISP, and doing things in a custom manner is easy on a small network |
| 13:41.54 | brad_mssw | [TK]D-Fender: hope that's everything: http://pastebin.com/m642c7fa8 |
| 13:42.53 | chiwawa_42 | g2g |
| 13:42.58 | chiwawa_42 | see you soon ;) |
| 13:43.05 | chiwawa_42 | and thanks again for your advices ! |
| 13:43.51 | *** join/#asterisk propellerhead (n=yogurt2u@host178.190-31-154.telecom.net.ar) |
| 13:43.55 | [TK]D-Fender | chiwawa_42: theory is nice, but a common goal of FLOSS is to use commodity type tech in open and interesting ways and ensuring interoperability. Having your head tied to a PC with a headset is FUGLY at best and using uncommon codecs places transcoding loads and restricts choices. Again against the philosophy. |
| 13:44.28 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
| 13:44.33 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 13:45.13 | Druken | upgrade doesn't mention the dialplan app, realtime |
| 13:45.56 | Skeeter- | hi, im looking into upgrade my system, i need to move all voices and IVR, which was easy, but it seems that i cant find the Voicemail recording, ineed them because i would like to move them to the new system. got any clue?? |
| 13:46.02 | manxpower | Druken: then it should not have changed much |
| 13:47.08 | Druken | [Sep 8 09:46:54] WARNING[9059]: pbx.c:3170 pbx_extension_helper: No application 'Realtime' for extension (outgoing, *98, 3) |
| 13:47.08 | Druken | cries |
| 13:47.23 | [TK]D-Fender | Skeeter-: /var/spool/asterisk/BLATANT |
| 13:47.43 | manxpower | Druken: did you look at ALL the UPGRADE*.txt? |
| 13:47.50 | [TK]D-Fender | Skeeter-: /var/lib/asterisk/sounds |
| 13:47.56 | SuPrSluG | Connected to Asterisk 1.6.1.1 currently running on app1 (pid = 1602) |
| 13:47.58 | SuPrSluG | app1*CLI> re |
| 13:48.00 | SuPrSluG | realtime reload restart |
| 13:48.08 | SuPrSluG | it's there. clearly |
| 13:48.14 | manxpower | UPGRADE.txt:* The REALTIME() function is now available in version 1.4 and app_realtime has |
| 13:48.15 | manxpower | UPGRADE.txt: been deprecated in favor of the new function. app_realtime will be removed |
| 13:48.21 | Skeeter- | sounds folder was copied |
| 13:48.29 | manxpower | Druken: looks like it's mentioned to me. |
| 13:48.36 | Skeeter- | blatant are the recording for the voicemail i guess?? |
| 13:49.30 | [TK]D-Fender | Skeeter-: means you should see a folder clearly named "voicemail" and the rest should be pretty clear |
| 13:50.00 | Druken | in what version? i didn't see it in 1.6's upgrade |
| 13:50.30 | Druken | oh, my bad, i opened the wrong file |
| 13:50.52 | manxpower | Druken: If it's deprecated in 1.4 it will be mentioned in the 1.2->1.4 upgrade. If it was removed in the 1.6 upgrade it won't be mentioned, it was mentioned in the previous upgrade files |
| 13:51.34 | Skeeter- | hello |
| 13:51.36 | Skeeter- | wow |
| 13:51.40 | Skeeter- | i cant copy |
| 13:51.54 | Druken | manxpower: so what was it replaced with? :P |
| 13:51.56 | Skeeter- | oh i see |
| 13:52.04 | brad_mssw | [TK]D-Fender: see anything wrong with that pastebin? |
| 13:52.07 | Skeeter- | here is the colation /var/spool/asterisk/voicemail/default/"extension" |
| 13:52.11 | manxpower | Druken: I require dinner and drinks before I'll hold your hand. |
| 13:52.32 | manxpower | heck, the info I pasted even told you what it was replaced with. Maybe you should take a break, Druken |
| 13:52.38 | Druken | that could be arranged :) rofl |
| 13:52.58 | Druken | it says realtime() |
| 13:53.07 | manxpower | no, it says function realtime |
| 13:53.12 | [TK]D-Fender | brad_mssw: Ok, you seem to have done it all... no idea... |
| 13:53.30 | [TK]D-Fender | brad_mssw: Maybe check the physical wiring? |
| 13:53.56 | brad_mssw | [TK]D-Fender: I've only got channel 1 plugged into the wall, and it's the only channel on that card that doesn't show an alarm ... so I think the wiring is good |
| 13:54.01 | Druken | k, i am lost... i hate it when shit changes |
| 13:54.29 | manxpower | Druken: read the upgrade files, don't just grep them. read them. |
| 13:54.30 | brad_mssw | [TK]D-Fender: do you know what versions of Dahdi are compatible with 1.6.1.6 ? ... I can go to 2.2.0.2 ... or maybe back to 2.1.0.4 ?? |
| 13:54.41 | manxpower | there are other files in the doc/ directory that might he helpful. |
| 13:54.49 | [TK]D-Fender | brad_mssw: Only forward.... |
| 13:54.54 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 13:54.54 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:02.34 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 14:04.44 | Skeeter- | My FOP doesnt work the the latest stable version, i can see EXT status but not trunk/zap activity |
| 14:06.26 | manxpower | What version of Asterisk? |
| 14:06.44 | brad_mssw | [TK]D-Fender: 2.2.0.2 is no better :/ ... guess I should try the other card |
| 14:08.36 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 14:12.16 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:13.47 | *** join/#asterisk cnu (n=cnu@63.80-203-44.nextgentel.com) |
| 14:13.49 | Druken | what idiot figured the new realtime "function" would be easier to use? |
| 14:13.53 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 14:15.43 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-3-211.w90-56.abo.wanadoo.fr) |
| 14:15.57 | Skeeter- | my asterisk is at 2.5.0.3 |
| 14:16.29 | SuPrSluG | umm no. the latest is 1.6 |
| 14:17.22 | Druken | that looks like a kernal version |
| 14:17.43 | Skeeter- | sorry that was for the CLI |
| 14:17.45 | Kobaz | 2.5.. heh |
| 14:18.10 | Skeeter- | asterisk 1.6.0.10, FOP not showing trunk, for Sangoma A200(latest firmware) |
| 14:19.41 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 14:22.18 | *** join/#asterisk ronr (n=ron@82-204-104-166.fttx.bbeyond.nl) |
| 14:22.26 | timeshell | Greetings. We had an incident on Friday with our Asterisk server. We have 4 analog trunks and 2 internet trunks. We lost our internet connection and as a result, asterisk apparently hung up everything. Incoming calls on the analog lines got busy signals; none of the extensions could call each other; none of the extensions could call out on the analog lines. The internet trunks had to be... |
| 14:22.27 | timeshell | ...deleted and the server restarted in order for the analog lines to work again. (I wasn't in the office and had to walk a non-techy through the process to delete the internet trunks, so I don't have any CLI or debug details unfornately). Anyone have any idea why Asterisk would hang up when the internet trunks are unavailable??? |
| 14:23.13 | [TK]D-Fender | Skeeter-: GUI's are NOT supported here |
| 14:23.18 | timeshell | s/unfornately/unfortunately |
| 14:23.31 | ronr | hi, it seems my agi script suddenly stopped working, all they do is set the right callerid for the channel using SET CALLERID <number>, agi debug shows it still does that, but the callerid doesn't change, how can I debug this further? |
| 14:23.34 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 14:23.42 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 14:24.12 | [TK]D-Fender | ronr: its "name" <number>, not just number |
| 14:24.35 | ronr | let's tru that |
| 14:25.47 | ronr | same result |
| 14:25.58 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) |
| 14:26.23 | [TK]D-Fender | ronr: pastebin it all |
| 14:28.53 | *** join/#asterisk maagic (n=maagic@fsck.fi) |
| 14:29.28 | Druken | manxpower: can you look over a line and tell me if i screwed it up? |
| 14:29.51 | ronr | [TK]D-Fender: http://pastebin.com/m56f0e564 |
| 14:30.05 | *** join/#asterisk huey23 (n=homygood@65.111.253.116) |
| 14:30.19 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:30.21 | ronr | asterisk ends up calling from 0107070081 |
| 14:30.56 | [TK]D-Fender | ronr: Sure doesn't look like a complete call |
| 14:31.00 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:31.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:31.48 | ronr | [TK]D-Fender: http://pastebin.com/m26ef0db5 sry, forgot some lines |
| 14:31.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:32.29 | [TK]D-Fender | ronr: and that gives no confirmation... |
| 14:32.40 | huey23 | i am getting an error message when i dial an extension from an outside line...i can provide more info if needed...can anyone help me out? http://pastebin.com/m97a40f1 |
| 14:33.21 | ronr | [TK]D-Fender: what confirmation do you mean? |
| 14:33.44 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 14:33.51 | [TK]D-Fender | ronr: I don't see you NoOping the callerid before and after |
| 14:33.56 | [TK]D-Fender | ronr: and code to back it |
| 14:34.22 | ronr | ok, back to the docs probably, weird thing is that it used to work |
| 14:34.23 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:34.27 | [TK]D-Fender | huey23: Device likely has not registered and * has nowhere to contact |
| 14:34.31 | ronr | thx anyway, need to go into a meeting now |
| 14:34.42 | angryuser | Any t.38 medem&asterisk 1.6 guru here ? |
| 14:34.46 | [TK]D-Fender | ronr: And you aren't showing the basics before looking to "go back to the docs" |
| 14:34.48 | angryuser | modem* |
| 14:35.01 | [TK]D-Fender | huey23: Go look at your peer status' |
| 14:35.12 | huey23 | [TK]D-Fender: i have reloaded...will look at peers |
| 14:36.37 | Druken | http://www.pastebin.ca/1558424 - reality check pls |
| 14:37.08 | huey23 | [TK]D-Fender: 0sip peers |
| 14:37.15 | dandre | hello, |
| 14:38.06 | [TK]D-Fender | huey23: then you'v done something very wrong |
| 14:38.08 | dandre | is there any way to accept a call on a sip phone from the ami interface? |
| 14:38.13 | Naikrovek | set up a phone system similar to his work system at work, at home, polycom phone and all, and good gravy it sounds awesome |
| 14:38.16 | huey23 | lol |
| 14:38.52 | [TK]D-Fender | dandre: You can't tell a phone to accept a call. |
| 14:39.35 | [TK]D-Fender | huey23: Wrong config file, bad permissions, massive syntax failure |
| 14:39.52 | [TK]D-Fender | Druken: Sanity fail : wheres the output? |
| 14:42.46 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-65-176.kgldga.dsl.dynamic.tds.net) |
| 14:44.17 | huey23 | [TK]D-Fender: i imagine it's syntax, reworking it, will let you know |
| 14:44.28 | Druken | how do i set the verbose level in 1.6? |
| 14:45.33 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:46.10 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:48.16 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 14:48.49 | [TK]D-Fender | Druken: core set verbose 10 |
| 14:49.12 | [TK]D-Fender | swats at Druken with a hardcover print of the changelogs |
| 14:51.34 | Druken | takes the changelogs and beats the developers with it while yelling, STOP CHANGING BASIC GOD DAMN COMMANDS!!!!! |
| 14:53.51 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 14:54.10 | djin | Hi everybody |
| 14:54.23 | djin | just a quick questing. Does queues need a timer? |
| 14:54.28 | djin | question |
| 14:55.28 | Druken | [TK]D-Fender: thanks btw... |
| 15:00.06 | djin | ztdummy is required as a timing source for MeetMe (conference calls) and is also involved in moh (music on hold) and a few other things. |
| 15:00.19 | djin | I'm trying to fdefine the 'few other things' |
| 15:00.56 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 15:01.37 | [TK]D-Fender | djin: IAX2 Trunk mode |
| 15:01.45 | [TK]D-Fender | djin: Helps with MoH sync as well |
| 15:03.01 | *** join/#asterisk sercik (n=ciccio@host116-109-dynamic.53-79-r.retail.telecomitalia.it) |
| 15:03.07 | sercik | hello! |
| 15:03.43 | sercik | someone can tell me the cheapest card? to receive and transmit on analog pstn? |
| 15:04.04 | jaytee | ~cheap |
| 15:04.05 | infobot | extra, extra, read all about it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 15:04.47 | sercik | :) |
| 15:04.50 | sercik | hi! |
| 15:05.07 | sercik | but i can't spent 500 $ |
| 15:05.26 | Katty | hugs jaytee |
| 15:05.31 | sercik | i have read on internet about x100p but i can't find |
| 15:05.36 | jaytee | hugs Katty |
| 15:05.43 | [TK]D-Fender | sercik: What do you want to do exactly? |
| 15:05.45 | Katty | jaytee: how was your weekend? |
| 15:05.50 | [TK]D-Fender | Katty: Mew. |
| 15:05.54 | *** join/#asterisk jgoo (n=r3rman@athedsl-4541648.home.otenet.gr) |
| 15:06.03 | Kobaz | sercik: you haven't looked very hard |
| 15:06.04 | Katty | [TK]D-Fender: hello (= |
| 15:06.06 | Kobaz | sercik: http://cgi.ebay.com/Asterisk-X100P-B2-FXO-PCI-For-IP-PBX-From-U-S_W0QQitemZ160331750263QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item2554845b77&_trksid=p3286.c0.m14 |
| 15:06.10 | Kobaz | two seconds of searching |
| 15:06.14 | jgoo | when I start asterisk with -cvvv ... how can I get out of the cli without killing asterisk? |
| 15:06.17 | jaytee | Katty, it was longer than usual but still too short by about 4 days :-) |
| 15:06.19 | Kobaz | sercik: the quality will suck though |
| 15:06.47 | Kobaz | jgoo: don't run it with -c |
| 15:06.53 | sercik | do you mean about voice quality? |
| 15:06.54 | [TK]D-Fender | jgoo: You can't. Means you shouldn't have started it that way for live use |
| 15:06.58 | Kobaz | sercik: yeah |
| 15:07.04 | Katty | jaytee: aww :< |
| 15:07.15 | Katty | [TK]D-Fender: curious--do you have a gym membership? |
| 15:07.17 | jgoo | Kobaz, don't run it with -c because you will end the world, or if you run it with -c it is only interactive |
| 15:07.27 | *** join/#asterisk KrisWillis (n=kris@host86-160-129-203.range86-160.btcentralplus.com) |
| 15:07.28 | Kobaz | jgoo: not me |
| 15:07.35 | Katty | or does anyone, for that matter |
| 15:07.35 | sercik | ty |
| 15:07.55 | Katty | jaytee: i've only got 3 days of work this week. taking another long weekend and renting a cabin. |
| 15:07.56 | jgoo | [TK]D-Fender, ok - I'll see if it starts the other way, it has just started complaining about zaptel config error today... nothing has changed thought for weeks |
| 15:07.57 | sercik | i saw it but i could also buy something better... |
| 15:08.06 | Kobaz | jgoo: er, never mind |
| 15:08.08 | sercik | can you give me some products... |
| 15:08.11 | Kobaz | jgoo: /etc/init.d/asterisk start |
| 15:08.16 | Kobaz | jpeeler: asterisk -rvvv |
| 15:08.22 | [TK]D-Fender | sercik: What do you want to do exactly? <------- |
| 15:08.39 | sercik | Fender hello |
| 15:08.43 | [TK]D-Fender | sercik: how many lines? What kind of volume? Personal or professional? |
| 15:09.07 | sercik | only one analog line or maybe two.. professional but with low traffic |
| 15:09.18 | sercik | is not an office... |
| 15:09.26 | Kobaz | sangoma |
| 15:09.28 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 15:09.30 | Katty | why don't you just get a dual lined phone from Sams Club |
| 15:09.34 | jgoo | Basically, hrm, sometimes on a reboot, the system works, but lately it doesn't, and it keeps dying :o |
| 15:09.36 | Kobaz | that too |
| 15:12.26 | sercik | Kobaz:? |
| 15:12.33 | sercik | this is a fake? : http://cgi.ebay.it/TDM410P-TDM400P-Asterisk-card-with-4-FXO-ports-NEW_W0QQitemZ150365889023QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item23028161ff&_trksid=p3286.m63.l1177 |
| 15:12.33 | Kobaz | sercik: sangoma |
| 15:12.54 | Kobaz | sercik: it's a clone |
| 15:12.58 | Kobaz | sercik: it will work |
| 15:13.04 | sercik | wow |
| 15:13.13 | [TK]D-Fender | chinaroby ( 102 [Punteggio di feedback compreso tra 100 e 499] ) |
| 15:13.15 | [TK]D-Fender | LOL!!! |
| 15:13.17 | sercik | on digium it costa 400 $ OOOOOOOOOOOOOOOOOOOOOOOOOOOO |
| 15:13.34 | [TK]D-Fender | Theese guys have been violating Digium's tradmarks for a LONG time now. |
| 15:14.13 | [TK]D-Fender | And spamming the WIKI |
| 15:14.13 | leifmadsen | that is so not a digium card |
| 15:14.13 | [TK]D-Fender | leifmadsen: Yeah I remember reporting this to you & qwwell :) |
| 15:14.14 | [TK]D-Fender | Qwell* |
| 15:14.16 | leifmadsen | support? non-existant |
| 15:14.44 | leifmadsen | LOL the image down a bit: Photo by ChinaRoby -- Don't Copy |
| 15:14.49 | leifmadsen | that is classic |
| 15:14.56 | sercik | i understand that this is a clone and not a digium product.. but .... digium have a very high cost.. |
| 15:16.03 | Kobaz | get rhino then |
| 15:16.13 | Kobaz | but don't complain when they release new firmware and break your card |
| 15:16.47 | sercik | this is better? |
| 15:16.48 | sercik | http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/B600_Analog_Voice_Card.html |
| 15:16.56 | sercik | what's rhino? |
| 15:17.04 | Kobaz | another vendor |
| 15:17.29 | Kobaz | sercik: that's a good card |
| 15:17.38 | Katty | sercik: if you hang aorund jameswf might show up. he works for rhino |
| 15:17.42 | ronr | [TK]D-Fender: sorry, but I just read the docs again and tried some things and I really got no clue whatsoever how NoOp would solve my problem (it's supposed to do nothiing) |
| 15:17.57 | sercik | thank you very much for informations |
| 15:18.51 | sercik | •Kobaz• i don't understand where are 4 fxo ports on that card |
| 15:18.55 | *** join/#asterisk maour_ (n=gnu@unaffiliated/maour) |
| 15:19.00 | sercik | i only see 3 holes |
| 15:19.05 | [TK]D-Fender | rorIt will PROVE what is set. I don't see a single piece of solid evidence |
| 15:19.07 | *** join/#asterisk abcdef (n=abcdef@64.92.145.104) |
| 15:19.13 | [TK]D-Fender | ronr: It will PROVE what is set. I don't see a single piece of solid evidence |
| 15:20.11 | Kobaz | sercik: two lines per port |
| 15:20.12 | huey23 | [TK]D-Fender: i believe i am getting there...it seems it was syntax but i am stuck, this syntax goes straight to voicemail: http://pastebin.com/m745fbd55 |
| 15:20.24 | sercik | ah! ok |
| 15:20.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:20.28 | Kobaz | sercik: and on the fxs, it's one line |
| 15:20.33 | ronr | [TK]D-Fender: ah, ok, so it's just for debugging, I use NoOp <text>, and prob get full variable to get <text>? |
| 15:20.40 | sercik | fxo can be connected to analog phone line? |
| 15:20.54 | [TK]D-Fender | ronr: you get whatever you shove in there |
| 15:21.00 | [TK]D-Fender | sercik: Yes |
| 15:21.00 | Kobaz | fxo takes a phone cable from the wall, it "recieves dialtone" |
| 15:21.17 | sercik | Kobaz you explain very good... |
| 15:21.36 | sercik | i can receive and also send on fxo ports |
| 15:21.38 | sercik | ?? |
| 15:21.50 | [TK]D-Fender | huey23: Instead of doing what? Looks fine to me... |
| 15:22.39 | Kobaz | sercik: fxo is like a faxmodem.. when the phone rings, the pc will pick it up... if you want to make a call, it will pick up and send digits |
| 15:22.41 | ronr | [TK]D-Fender: yeah, but obviously NoOp CALLERID will print out CALLERID, I'll have to get a value into some variable somehow right? (sorry if I'm asking something stupid but I think I'm missing something really obvious here) |
| 15:23.02 | huey23 | [TK]D-Fender: step 9 should play financegreeting2 but it goes straight to voicemail u5009 |
| 15:23.14 | Kobaz | sercik: if you want to hook a regular analog phone to your pc... you will need an fxs port... fxs provides dialtone |
| 15:23.20 | [TK]D-Fender | ronr: Yes, it means you don't seemt o have a grasp on how to call functions or use variables... not a good sign. This is dialplan 101 |
| 15:23.27 | sercik | Kobaz |
| 15:23.29 | [TK]D-Fender | ronr: "core show function CALLERID" |
| 15:23.39 | sercik | i want to use voip phone on internal side |
| 15:24.01 | sercik | bur i want to be able to make outgoing calls on analog pstn and not only through adsl |
| 15:24.02 | Kobaz | sercik: then you need just fxo, on the pc side... but that card is a good combo |
| 15:24.04 | [TK]D-Fender | huey23: -- Executing [2@default:1] Goto("Zap/1-1", "finance_group|2|1") in new stack <- look where you are jumping to. Youa re doing this explicitly in your GOTO |
| 15:24.21 | sercik | brb |
| 15:24.33 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 15:24.34 | ronr | [TK]D-Fender: yeah, I made all this almost 2 years ago and didn't look at it again until it stopped working just now |
| 15:24.47 | huey23 | [TK]D-Fender: i never made the connection mr. obvious |
| 15:24.58 | [TK]D-Fender | ~[TK]D-Fender |
| 15:24.59 | infobot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 15:25.07 | [TK]D-Fender | Yup, that's me... |
| 15:25.10 | huey23 | :P |
| 15:25.41 | [TK]D-Fender | huey23: Hint : we hide it in the BIG PRINT |
| 15:25.58 | huey23 | [TK]D-Fender: right, reading is a virtue |
| 15:26.36 | abcdef | Hi, i'm trying to get my snom 360 to show the correct caller id phone number. it is registered as an extension to a trixbox/asterisk box, and the caller id currently shows up like "555-555-1234@192.168.1.10". That format might be correct for anonymous sip, but this phone is registered to a PBX, so the outbound dialing doesn't work. I suppose changing the outbound dialing might be another option. Anyone know the right solution here? |
| 15:26.37 | Naikrovek | leifmadsen: just posted this on your blog, but then remembered you were here also: http://chart.apis.google.com/chart?chs=225x225&cht=qr&chl=blahblahblah |
| 15:26.48 | *** join/#asterisk Tim_Toady (n=moi@adsl320-230.kln.forthnet.gr) |
| 15:28.23 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 15:28.47 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:28.54 | ronr | [TK]D-Fender: ah... you mean using NoOp in the dialplan... I just kept rereading the docs on NoOp in the AGI docs... now it makes sense |
| 15:29.21 | leifmadsen | Naikrovek: :) nice! That as the link I was trying to find! I knew google had an API to make those too |
| 15:29.26 | leifmadsen | thanks! |
| 15:29.35 | abcdef | sercik: FXO allows you to use POTS (plain old telephone service). I use a sangoma remora with FXO ports and have 4 regular phone lines plugged in for incoming/outgoing calls. |
| 15:30.10 | sercik | abcdef ty |
| 15:30.21 | sercik | i have a doubt about pots |
| 15:30.29 | sercik | i used to call it pstn |
| 15:30.35 | sercik | is the same? |
| 15:31.55 | abcdef | sercik: don't know the terminology for sure, but a quick wikipedia check makes me think pstn is the phone company network where as POTS is the service they provide to you as an end user |
| 15:32.27 | sercik | so essentially is the same thing |
| 15:32.49 | abcdef | sercik: you'll need hardware to do the POTS service, I use sangoma who is typically pretty helpful too with their support. digium is another option |
| 15:33.01 | sercik | i know |
| 15:33.08 | abcdef | k |
| 15:33.12 | sercik | i'm actually searching for hardware |
| 15:33.37 | sercik | but i only need 2 fxo |
| 15:34.02 | abcdef | each sangoma fxo port can support 2 actual phone lines |
| 15:34.58 | sercik | but 24 needs 12 holes in one pci card?... is there enough space?? |
| 15:35.40 | abcdef | each pci slot accomodates 4 phone lines. they use an RJ-22 jack, not RJ-11. it's the same but smaller |
| 15:36.07 | abcdef | they give you rj11-rj22 cables or you can make your own |
| 15:36.37 | sercik | can i order a sangona with only 2 fxo ports?? |
| 15:37.02 | abcdef | should be able to... i'd just call em up and get the part number or email them |
| 15:37.16 | abcdef | i've been buying direct, but telephony depot i know has them too |
| 15:37.39 | abcdef | the number of options is crazy, so having someone give you the exact part number is useful |
| 15:37.45 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 15:38.25 | sercik | i understood |
| 15:38.37 | sercik | i can order that card and add from 2 to 24 ports |
| 15:38.42 | abcdef | so anyone know where in asterisk that caller id string format gets set for my voip phone? |
| 15:39.02 | brah | ~pastebin |
| 15:39.03 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:40.09 | abcdef | brah: was that a suggestion for me? |
| 15:40.23 | brah | Nope, I needed to steal the pastebin line for another channel. |
| 15:40.24 | brah | :) |
| 15:40.33 | abcdef | ok :) |
| 15:44.00 | timeshell | How does chan_sip deal with IP channels using FQDN when a DNS server or query is unable to resolve, and as a result is unable to get an IP, and as a result cannot connect? |
| 15:44.02 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 15:44.41 | superbeef | Anybody know what this means? DEBUG[6584] channel.c: Generator got voice, switching to phase locked mode |
| 15:44.41 | superbeef | <PROTECTED> |
| 15:45.54 | timeshell | The result I experienced on Friday is that all our analog trunks became unavailable both inbound and outbound. |
| 15:46.40 | timeshell | This concerns me because IP trunks being unavailable should not affect analog trunks. |
| 15:46.54 | timeshell | Also, all our internal extensions became unresponsive to the server. |
| 15:46.55 | djin | [TK]D-Fender: thanks. Got called away for a moment |
| 15:46.55 | huey23 | [TK]D-Fender: i got it working, thanks for the help |
| 15:47.52 | superbeef | lol |
| 15:50.14 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:50.46 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:51.15 | *** join/#asterisk AlexJ^ (n=chatzill@pdpc/supporter/student/alexj) |
| 15:51.38 | AlexJ^ | hello... can anyone help me with an installment of sccp on asterisk? |
| 15:52.47 | *** join/#asterisk spenguin[work] (n=penguin@59.162.86.164) |
| 15:53.01 | *** join/#asterisk jtrimmer (n=jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 15:53.33 | abcdef | timeshell: I would guess that DNS glitch had a domino effect. I assume you fixed the DNS, restarted, and things resumed as normal? |
| 15:54.14 | jtrimmer | anyone with some analog truck experience who could give me a hand. I updated today and I don't know if that was the cause or not but now I can receive calls just fine but if I try to make an outgoing call it tells me all circuits are busy. |
| 15:54.40 | spenguin[work] | hey, would it be ok if two PCI cards are using the same IRQ address 3w-9xxx, wcte11xp |
| 15:54.58 | spenguin[work] | as per /proc/interrupts |
| 15:55.20 | Naikrovek | spenguin[work]: well linux seems to be okay with it |
| 15:56.08 | spenguin[work] | Naikrovek: would it affect the performance of the cards? |
| 15:56.30 | Naikrovek | spenguin[work]: don't think so... |
| 15:56.39 | Naikrovek | why are they on the same IRQ |
| 15:56.51 | Naikrovek | did they just come up on the same IRQ? |
| 15:57.00 | Naikrovek | hasn't really had to deal with IRQ issues for 15 years |
| 15:57.35 | spenguin[work] | Naikrovek: yeah I got that from /proc/interupts |
| 15:57.41 | abcdef | the last time I was suspicious of IRQ sharing issues it all turned out to be another problem |
| 15:58.09 | Naikrovek | yeah the x86/amd64 architectures are tolerant of multiple devices with the same irq |
| 15:58.39 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:58.40 | Naikrovek | don't get confused by the 'amd' in the 'amd64', intel's 64-bit architecture failed, so they use amd's |
| 15:59.25 | bmoraca | can anyone give me any guidance as to why I am getting this error with a config file that i've used a hundred times with Polycom 330 phones: Loaded application sip.ld successfully, errors 0x120. |
| 15:59.43 | Naikrovek | pastbin the whole line and the surrounding lines |
| 16:00.54 | *** join/#asterisk t_j (n=tj@tomjudge.vm.bytemark.co.uk) |
| 16:01.02 | *** join/#asterisk feder (n=feder@host.190.15.192.172.static.itcsa.net) |
| 16:01.37 | t_j | hi, anyone know of any good free SIP did providers in the US and UK, I have googled but wanted some personal recomendations |
| 16:01.39 | feder | Hello, sorry to bother, asterisk doesn't seem to respect my codec priority :( |
| 16:01.52 | Naikrovek | feder: explain |
| 16:02.00 | feder | I have a callcentric account |
| 16:02.03 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 16:02.23 | Naikrovek | feder: and it uses 711 when you want 729 or something |
| 16:02.32 | feder | 10 g729 licenses and allow=g729&ulaw in the trunk, all the calls drop to ulaw |
| 16:02.48 | Orbixx | How can I record calls received via a queue? |
| 16:03.07 | Naikrovek | feder: are you sure your provider can use 729? turn on sip debug and place a call then pastebin the result |
| 16:03.23 | Naikrovek | bmoraca: can you pastebin the log? |
| 16:03.40 | *** join/#asterisk dysinger (n=tim@166.129.114.8) |
| 16:03.43 | Naikrovek | bmoraca: feel free to sanitize anything you wish |
| 16:03.52 | bmoraca | Naikrovek: that's the only relevant line from the log, but i'll go ahead and pastebin it |
| 16:04.00 | Naikrovek | bmoraca: okay |
| 16:04.45 | feder | Naikrovek: I'll try to, the server is quite busy now It may be difficult to get a clean trace, I'll try to be back in minutes |
| 16:05.11 | Naikrovek | feder: yeah don't do it while it's under heavy load |
| 16:05.11 | *** join/#asterisk el_critter (n=critter@200.8.188.225) |
| 16:05.47 | timeshell | abcdef I ended up deleting the internet trunks as we had one of our major fiber cables cut and were out of internet for much of the day. However, this is very bad as I cannot have the asterisk server fail because a internet voip trunk goes down. |
| 16:07.07 | bmoraca | Naikrovek: http://www.pastebin.org/16166 ...yes, I realize that I don't have the phone-specific bootrom or sip images...i have the combined ones. i've tried upgrading to the new software versions with the standard cfgs included, but it doesn't fix the problem...i cannot get these phones to boot. |
| 16:07.20 | bmoraca | my 550s, though, work just fine |
| 16:07.36 | feder | Naikrovek: can I debug sip to a file? |
| 16:07.38 | timeshell | But, I agree with your conclusion that it probably had something to do with the DNS cascade effect. Chan_sip should have a check in it to make sure that it is only connecting when the DNS resolution is successful and not impact the rest of the server when is not. |
| 16:08.21 | abcdef | timeshell: agreed. I haven't seen that happen. are your dns servers remote over the fiber that broke? I have local dns servers on my network. while in an ideal world it shouldn't matter, I wonder if that is a simple fix to the issue--installing a local dns server. |
| 16:08.46 | Naikrovek | bmoraca: that's the error seen when the update can't be applied because you're already up to the latest version |
| 16:08.53 | Naikrovek | bmoraca: there are MUCH newer firmwares out there, btw. |
| 16:08.54 | abcdef | timeshell: getting an empty answer may be better than no response |
| 16:08.55 | timeshell | abcdef Yes, the asterisk server was using DNS servers provided by our upstream provider on the other side of our fiber link. |
| 16:09.16 | bmoraca | Naikrovek: like I said, I tried the newer firmwares and the issue persisted. |
| 16:09.36 | Naikrovek | feder: it will log to /var/log/asterisk/full and to the console, but a busy system will utilize the disk enough to cause issues, possibly |
| 16:09.44 | timeshell | abcdef Perhaps that could be a work around but Asterisk really should be able to deal with such a scenario. An unavailable trunk should not hang the entire asterisk process. |
| 16:09.57 | abcdef | timeshell: asterisk may be in some dns loop waiting for a response and never getting anything. seems ridiculous but if you need a solution today... it probably works |
| 16:10.03 | timeshell | And actually, it didn't hang the asterisk process per sey, but the analog lines |
| 16:10.50 | timeshell | Asterisk took the analog lines off hook. When calling in on the lines, they returned busy. So, it wasn't just that it wouldn't answer them, it actually apparently took off hook as busy. |
| 16:11.16 | Naikrovek | bmoraca: it is downloading properly, it says that the image is identical to current version, and that is the cause of the error code - think of that one as an informational error |
| 16:11.55 | Naikrovek | bmoraca: the phone works fine, yes? |
| 16:11.58 | bmoraca | Naikrovek: right, however the phone is in a continuous reboot |
| 16:12.00 | bmoraca | Naikrovek: no |
| 16:12.06 | Naikrovek | bmoraca: okay |
| 16:12.09 | Naikrovek | i didn't see that before |
| 16:12.41 | sercik | hello and goodbye thanks for informations and help |
| 16:12.49 | Naikrovek | bmoraca: you may want to put the correct bootrom on there, looks like the one it sees isn't correct for the phone |
| 16:12.55 | timeshell | abcdef Also, Asterisk shouldn't be talking to DNS directly... it should be the OS network subsystems taking care of that. |
| 16:13.09 | Naikrovek | timeshell: applications talk to DNS all the tiem |
| 16:13.33 | Naikrovek | timeshell: though the normal way is as you described |
| 16:13.48 | timeshell | Naikrovek So you're saying that Asterisk directly asks DNS for a resolution and then gives the resolved IP to socket connection request? |
| 16:14.06 | Naikrovek | timeshell: i'm saying it's possible, not that asterisk does it |
| 16:14.12 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 16:14.12 | timeshell | Naikrovek I know it's possible. |
| 16:14.18 | el_critter | Hi, I'm having troubles creating PSTN outgoing calls, incoming calls are fine, in fact, CONGESTION status for outgoing calls is reseted by an incoming call. |
| 16:14.56 | timeshell | My point is, it may not matter if there is a local DNS server or not. If it cannot resolve, it may end up with the same issue. |
| 16:15.00 | Chainsaw | el_critter: How are you getting to the PSTN? |
| 16:15.36 | timeshell | And it still concerns me that chan_sip could affect chan_dahdi in that way. |
| 16:15.45 | el_critter | Chainsaw: digium TDM400P |
| 16:15.52 | timeshell | directly or indirectly... whatever it is, it should not have interfered with the analog trunks. |
| 16:15.52 | Chainsaw | el_critter: Okay, zaptel or DAHDI? |
| 16:16.12 | el_critter | Chainsaw: dahdi |
| 16:16.27 | el_critter | Chainsaw: dahdi 2.2.0.2 |
| 16:16.27 | Chainsaw | el_critter: Did you set up your country mode correctly? |
| 16:16.30 | timeshell | el_critter I know what the problem is |
| 16:16.33 | timeshell | It's a bug |
| 16:16.40 | timeshell | Just a sec, I'll find the patch link for you. |
| 16:16.42 | hardwire | so right now skype for asterisk beta is limited to a single channel eh? |
| 16:17.16 | Qwell | hardwire: no and no |
| 16:17.25 | el_critter | Chainsaw: yes, country is ok |
| 16:17.29 | hardwire | wow.. I'm working on old info Qwell |
| 16:17.35 | hardwire | just checked out digiums site |
| 16:17.44 | Chainsaw | el_critter: Okay, then I will now transfer you to timeshell *hold music* |
| 16:17.44 | Qwell | It is now out of beta, and it was not limited. |
| 16:17.46 | hardwire | the beta appears to be over. |
| 16:17.59 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:18.01 | hardwire | Qwell: good chance I can't use my beta keys now eh. |
| 16:18.05 | Qwell | correct |
| 16:18.07 | el_critter | hahaha |
| 16:18.19 | timeshell | Qwell Would you mind looking at my comments over the morning and comment? |
| 16:18.31 | hardwire | Qwell: ok.. works for me. I just buy more channels and use a single account then eh? |
| 16:19.13 | Qwell | hardwire: I don't know how it works, tbh |
| 16:19.16 | timeshell | Qwell, There seems to be a bug that causes IP trunks to interfere with analog trunks when the IP trunks are unavailable. |
| 16:19.17 | hardwire | thats fine. |
| 16:19.23 | hardwire | Qwell: that appears to be how it works. |
| 16:19.30 | hardwire | tubbeh. |
| 16:19.40 | Qwell | Did you just call me fat? |
| 16:19.46 | hardwire | tbh. |
| 16:19.53 | hardwire | :P |
| 16:21.27 | timeshell | el_critter https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=14577 |
| 16:21.39 | timeshell | el_critter https://issues.asterisk.org/view.php?id=15429 |
| 16:21.54 | el_critter | timeshell: thnx, let me check that |
| 16:25.27 | ccesario | hi... |
| 16:26.06 | ccesario | somebody can say me if this is "correct" ... http://pastebin.com/m2908a007 ... or can be a possible problem/bug ? |
| 16:26.22 | feder | Naikrovek: http://pastebin.com/d7a8af345 |
| 16:27.54 | garymc | so anyone know why my sound stops working after a while? |
| 16:27.57 | bpgoldsb | Anyone know any good alternatives to CDR? |
| 16:27.58 | garymc | Anyone help me out herE? |
| 16:28.06 | Naikrovek | garymc: define "a while" |
| 16:28.13 | Naikrovek | feder: i'm looking |
| 16:28.15 | garymc | about 20 mins |
| 16:28.24 | garymc | or after playing a few songs |
| 16:28.34 | *** join/#asterisk gunthr (i=gunthr@mail.ericksontech.com) |
| 16:28.35 | Naikrovek | on hold? |
| 16:28.44 | garymc | infact i think its when it goes off when idle maybe |
| 16:30.28 | Naikrovek | feder: it is indeed listing PCMU (G.711u) before any other codec. can you pastebin your trunk config? feel free to strip the IPs if you like |
| 16:31.18 | Naikrovek | garymc: does the sound come back when you make a noise on your side |
| 16:32.52 | feder | Naikrovek: http://pastebin.com/d40caa0c8 |
| 16:33.00 | garymc | Naikrovec : Just realised im in the wrong channel :( sorry |
| 16:33.12 | Naikrovek | garymc: np dude |
| 16:34.05 | Naikrovek | feder: try multiple allow= lines |
| 16:34.10 | Naikrovek | so, |
| 16:34.14 | Naikrovek | allow=g729 |
| 16:34.14 | raden_work | is there a way if i know another call coming on on another extension i could dial something on my phone to pick it u p |
| 16:34.16 | Naikrovek | allow=g711 |
| 16:34.28 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:34.35 | Naikrovek | raden_work: *8? can't remember |
| 16:34.37 | raden_work | just like one of the girls walked out of the office and her phone ringing i wish i could pick it up from mine |
| 16:34.37 | feder | Naikrovek: let's seeeeee |
| 16:34.42 | Naikrovek | raden_work: yes there is a way i think |
| 16:35.17 | Naikrovek | raden_work: you using FreePBX? ** may do it |
| 16:35.59 | Naikrovek | feder: i mistyped, s/g711/ulaw/ but multiple lines |
| 16:37.30 | feder | Naikrovek: Freepbx doesn't allow me to do so, It converts the multiple allow lines in one |
| 16:37.37 | Naikrovek | feder: ah okay |
| 16:37.50 | Naikrovek | feder: hang around for [TK]D-Fender, he can help you |
| 16:37.57 | *** join/#asterisk errotan (n=errotan@62.201.122.227) |
| 16:37.58 | Naikrovek | sorry i couldn't help |
| 16:38.23 | Naikrovek | feder: you'll know when [TK]D-Fender shows up, he starts yelling at people. have those same pastebin links ready |
| 16:38.27 | feder | Naikrovek: Dude, you've been great, I really really thankyou for your time |
| 16:38.34 | Naikrovek | feder: no problems |
| 16:38.59 | Naikrovek | [TK]D-Fender is extremely knowledgeable, he'll figure it out. Qwell maybe also can help |
| 16:39.09 | Naikrovek | just hang out for a bit and they'll show up |
| 16:39.29 | feder | Naikrovek: thx |
| 16:41.38 | Qwell | Qwell usually doesn't respond when his name is mentioned. |
| 16:43.00 | raden_work | how do i send call waiting to a phone |
| 16:43.24 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 16:43.26 | raden_work | this wifi phone i have supposebly supports call waiting but if someone using it it goes to a busy signal |
| 16:46.40 | bmoraca | ARG |
| 16:46.56 | bmoraca | now i'm getting errors 0x100 from files from a working system that i KNOW WORK! |
| 16:48.02 | manxpower | bmoraca: polycom? |
| 16:48.12 | bmoraca | yes |
| 16:48.16 | bmoraca | polycom 330 |
| 16:48.28 | *** join/#asterisk maour_ (n=gnu@unaffiliated/maour) |
| 16:48.31 | bmoraca | it's making my hair turn grey...i've NEVER had this problem with a 330 before until now |
| 16:48.40 | manxpower | bmoraca: Usually that's a typoe, missing " is my most common cause of that, but other things like . or , or = screwed up. |
| 16:48.52 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 16:49.00 | bmoraca | manxpower: i copied the sip.cfg file from a working system...no changes at all |
| 16:49.40 | bmoraca | i haven't touched the sip.cfg file since i built the provisioning engine i use for polycoms...and i've never had this problem before |
| 16:50.39 | Katty | hi |
| 16:50.56 | manxpower | bmoraca: there are many more files than just sip.cfg |
| 16:51.11 | manxpower | the onces I usually screw it up on are MAC.cfg or MAC-phone.cfg |
| 16:51.38 | bmoraca | arg |
| 16:51.41 | bmoraca | i think i may have found it |
| 16:51.51 | manxpower | looks at bmoraca |
| 16:52.04 | bmoraca | i hate polycoms :( |
| 16:52.25 | bmoraca | well, i hate provisioning them, anyway |
| 16:53.06 | bmoraca | nope |
| 16:53.08 | bmoraca | nevermind |
| 16:53.12 | bmoraca | that didn't do anything |
| 16:54.36 | hardwire | cries |
| 16:55.33 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 16:55.46 | bmoraca | what the hell does this mean? "Loaded application sip.ld successfully, errors 0x100." does polycom have a list of error codes or something? |
| 16:56.09 | t_j | anyone using ipkall.com for incomming SIP dids? |
| 16:56.18 | t_j | bmoraca: good luck with that |
| 16:57.18 | t_j | bmoraca: we have found it means the config is bad or it cant dl a config |
| 16:58.07 | t_j | last time i saw it was when we tried to use a /23 subnet on the latest firmware, which cant handle it for some reason and was not able to contact the boot server after the app loaded |
| 16:58.14 | bmoraca | the config works perfectly with the 550s i have and i can't find any errors in it |
| 16:59.14 | KrisWillis | Hi guys, is there any official documentation available for Asterisk? I can only find the O'Reilly downloadable book, and the knowledge base over at the Digium website... |
| 16:59.14 | jgoo | anyone know of completely free DID provider with free US -> US DIDs? |
| 16:59.55 | t_j | jgoo: i'm current looking at ipkall.com |
| 17:00.04 | leifmadsen | they work |
| 17:00.22 | jgoo | t_j, they don't configure their webserver to allow without www. |
| 17:00.47 | leifmadsen | KrisWillis: the O'Reilly book is pretty much the most comprehensive documentation for Asterisk, other than the doc/ directory in the asterisk source -- you could also try voip-info.org, although information there may be incomplete or obsolete |
| 17:00.50 | hardwire | KrisWillis: thems good info. |
| 17:00.53 | t_j | jgoo: what do you mean? |
| 17:01.09 | KrisWillis | leifmadsen: Thanks |
| 17:01.14 | KavanS | does anyone monitor their SIP peers with nagios/etc.? |
| 17:01.20 | jgoo | t_j, you can only access the page at www.ipkall.com not ipkall.com |
| 17:01.36 | t_j | jgoo: oh |
| 17:02.20 | jgoo | t_j, so they are a DID -> Sip service with free numbers |
| 17:02.26 | leifmadsen | yes |
| 17:02.30 | leifmadsen | not termination though |
| 17:02.59 | jgoo | Yeah. still interesting |
| 17:03.02 | leifmadsen | if you want US DID -> US DID, then use Google Voice |
| 17:03.25 | jgoo | leifmadsen, thanks for reminding me, I have my invite to setup still, but I bet they limit DIDs |
| 17:03.41 | [TK]D-Fender | is now stuffed. Indian food = win |
| 17:03.52 | leifmadsen | indian++ |
| 17:04.01 | t_j | leifmadsen: know of someone that has free DIDs (+incomming calls) and a pay as you go type termination service? |
| 17:04.08 | leifmadsen | nope |
| 17:04.37 | *** part/#asterisk AlexJ^ (n=chatzill@pdpc/supporter/student/alexj) |
| 17:04.43 | [TK]D-Fender | leifmadsen: 7$, no tax 4 pots ordered a-la-carte, 4 breads (puri & bhatura), entree of pakora, and 4 free desserts.... |
| 17:04.53 | leifmadsen | wtf, amazing |
| 17:04.56 | leifmadsen | I want that now! |
| 17:05.17 | bmoraca | why does polycom say that Polycom 330s support 3.1.2 firmware and then say in the same document that if you're using 3.1.2 firmware, you can't use Polycom 300 or 500 phones? |
| 17:05.38 | manxpower | because the 300 is not the same as the 330 |
| 17:06.07 | leifmadsen | ^^^ |
| 17:06.17 | leifmadsen | 300 and 500 are older models with less memory |
| 17:06.30 | leifmadsen | even the 300 is not the same as the 301 |
| 17:06.56 | t_j | bmoraca: we have issues with that firmware on 330's they run out of ram and crash :( |
| 17:07.22 | bmoraca | that's weird, because in the configs, it looks as if they're referencing the product line, rather than specific phone models |
| 17:08.08 | t_j | bmoraca: when you dl the lastest firmway from the website it should include versions compatiable with all models in it from what I can remember |
| 17:08.13 | feder | [TK]D-Fender: Hello, sorry to bother you, I cant set codec priority in asterisk. I try to call with g729 and it drops to ulaw even when both peers support g729 |
| 17:08.25 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:08.44 | bmoraca | t_j: I'm not having issues with any other phones except the 330s. |
| 17:08.44 | manxpower | feder: that is normally that way it works. ulaw is normally chosen over g729 |
| 17:08.55 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:08.55 | [TK]D-Fender | feder: So fix the order. |
| 17:09.14 | feder | I have Allow=g729&ulaw in the trunk |
| 17:09.26 | t_j | bmoraca: we run 3.1.2.0392 on about 150 330's mostly fine |
| 17:09.35 | feder | I have g729 licenses |
| 17:09.42 | [TK]D-Fender | feder: those should be on separate lines |
| 17:09.47 | jgoo | [TK]D-Fender, in your experience have you ever seen two extensions start dialing the same extension? These are PAP2Ts... I am thinking... some wires must be crossed, I cannot think how this has suddenly started happening |
| 17:09.53 | t_j | bmoraca: except for already mentioned memory crashing issue |
| 17:09.57 | [TK]D-Fender | feder: And if you want g.729 then you should SPECIFY it exclusively |
| 17:10.13 | bmoraca | t_j: that's wonderful...but mine will not do anything except go into a constant reboot loop, and I cannot figure out why |
| 17:11.01 | feder | [TK]D-Fender: Freepbx doesn't allow me to put it in 2 lines, It joins them. I want to be able to use ulaw if I run out of g729 licenses |
| 17:11.07 | t_j | bmoraca: do your provisioning server logs show the phone pull the files? |
| 17:11.14 | [TK]D-Fender | feder: Yes, it does. |
| 17:11.15 | bmoraca | yes |
| 17:11.44 | bmoraca | this is the error i'm getting: 0908153146|app1 |4|00|Loaded application sip.ld successfully, errors 0x120. |
| 17:11.44 | t_j | bmoraca: then the only thing i can think of is that there is corrupt mac.cfg or something |
| 17:12.06 | feder | [TK]D-Fender: at least Freepbx 2.4.1.0 doesn't |
| 17:12.34 | [TK]D-Fender | feder: It does, you're not trying hard enough |
| 17:12.36 | Naikrovek | feder: if you have 729 as a priority, it won't fall back to ulaw when you run out of licenses. those calls will just fail |
| 17:12.40 | [TK]D-Fender | feder: and this isn't #freepbx |
| 17:12.41 | t_j | seems to remember that 0x1XX where config errors |
| 17:13.00 | KyleK | is there a button i can press while dialing on an spa3102 to have the number go out? like dial 111 then press star? |
| 17:13.00 | t_j | bmoraca: have you looked in the admin guide i think its in there somewhere... |
| 17:13.08 | bmoraca | nope |
| 17:13.14 | bmoraca | admin guide doesn't list the error codes |
| 17:13.58 | feder | ok |
| 17:14.19 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 17:15.09 | *** part/#asterisk el_critter (n=critter@200.8.188.225) |
| 17:15.20 | Druken | http://www.pastebin.ca/1558609 |
| 17:15.32 | Druken | what am i overlooking in my dialplan? |
| 17:15.35 | *** join/#asterisk el_critter (n=critter@200.8.188.225) |
| 17:16.45 | bmoraca | and nothing gets written to mac-app.log...this is friggin wonderful. |
| 17:17.20 | t_j | bmoraca: whats the subnet mask you are using? We had problems with non standard ones |
| 17:17.31 | bmoraca | using a /24 |
| 17:17.34 | t_j | humm |
| 17:17.59 | t_j | have you tried regenerating mac.cfg and mac-phone1.cfg? |
| 17:18.17 | bmoraca | i haven't been able to find any issues with either file |
| 17:18.36 | t_j | will those files work on another model? |
| 17:19.04 | bmoraca | yes, my 550s boot without issue |
| 17:19.29 | bmoraca | the only differences between the two are that the 550s have the number of line keys set to 4 and the 330s have it set to 2 |
| 17:19.50 | [TK]D-Fender | Druken: exten => _X.,15,Set(row_BLOCKED="${REALTIME(blocked,realtime,${CALLERIDNUM}${PEER_customer_id})}") <- var not valid in 1.4+ |
| 17:19.57 | [TK]D-Fender | Druken: and you aren't showing OUTPUT again |
| 17:20.18 | [TK]D-Fender | Druken: And you're adding garbage quotes |
| 17:20.23 | *** join/#asterisk kyoshi (i=4a65e1b7@gateway/web/freenode/x-yqdcqwqyxaopiqvj) |
| 17:20.44 | *** join/#asterisk Meaulnes (n=LeGrandM@nat.finelight.com) |
| 17:21.30 | jgoo | zaptel.conf and fxsks=1-4 -- does that have to be in zaptel.conf if I have a 4port isdn card using mISDN? |
| 17:21.37 | Druken | garbage quotes? |
| 17:21.51 | Druken | oh... hmm... i see them now |
| 17:21.55 | jgoo | **fxoks I mean |
| 17:22.13 | *** join/#asterisk youngproguru (n=youngpro@smtp.deltasoniccarwash.com) |
| 17:22.27 | jgoo | also bchan=1-2 -- I ask, because the system works.. but my zaptel.conf is empty |
| 17:22.33 | jgoo | does mISDN override this? |
| 17:22.37 | Druken | [TK]D-Fender: you see anything else wrong with it otherwise? |
| 17:22.57 | Meaulnes | Has anyone ever seen an issue wherein a ZAP call will experience a loss of audio without disconnecting the call? PRI debugging seems to indicate that a disconnect was received, but the call is not disconnected. Asterisk 1.4.17 |
| 17:23.35 | kyoshi | Using Asterisk RealTime, for priority "1", my App is "Dial", my AppData is let's say "SIP/${EXTEN}@31241255", normally in the Dial function i can specify a timeout such as "Dial(SIP/${EXTEN}@7475518761,10)" but how do I do this in RealTime? |
| 17:23.37 | [TK]D-Fender | Druken: Again stop wasting time showing just code, show its EXECUTION too |
| 17:23.56 | [TK]D-Fender | Druken: We should ne be GuesSING what you may have screwed up. go generate some ERRORS for us to look at. |
| 17:24.11 | [TK]D-Fender | Druken: Because if everything is fine then you're just wasting our time.. |
| 17:24.28 | Druken | http://pastebin.ca/1558620 |
| 17:25.17 | Druken | well, that's just it... i'm not getting an ERROR so to say, it doesn't appear to be getting the incoming DID from the callerid(num) for the initial lookup in the database |
| 17:27.24 | KyleK | hey is there any guides out there for organizing internal extensions? |
| 17:27.42 | jgoo | KyleK, there are no task based guides I've seen |
| 17:27.52 | jgoo | or goal oriented docs |
| 17:28.28 | KyleK | ah |
| 17:30.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:32.13 | *** join/#asterisk xpot-mobile (n=james@mx0.synergyconsultant.net) |
| 17:32.22 | bmoraca | do there exist any applications which can verify the integrity of polycom config files? |
| 17:33.39 | manxpower | bmoraca: Any XML validate program should work, shouldn't it? |
| 17:33.54 | bmoraca | for structure, yes, but not for content... |
| 17:35.06 | raden_work | anyone know of a good CRM |
| 17:35.43 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 17:35.46 | dustybin | i spent the whole day getting my asterisk server to call my friends and play random messages :D |
| 17:35.53 | dustybin | using .call files :D |
| 17:37.47 | [TK]D-Fender | Druken: exten => _X.,3,Set(row_DID="${REALTIME(dids,did,${CALLERID(num)})}") <- where do i see this table & mapping? |
| 17:38.15 | dustybin | is that regex? |
| 17:38.31 | [TK]D-Fender | dustybin: no |
| 17:38.40 | dustybin | ok |
| 17:40.25 | [TK]D-Fender | Druken: And what did I tell you about those useless quotes? |
| 17:42.43 | *** join/#asterisk davidandgoliath (n=David@out.clearnet.com) |
| 17:43.29 | Druken | yeah i noticed that after i pasted the pastebin... it's been corrected |
| 17:43.39 | jgoo | dustybin, I emailed my 'friend' a dialer that set his computer to repeatedly dial his mobile phone, back in the modem days, aaah it was fun. It happened randomly, he never knew WHO was in HIS house, dialing his mobile. I luled |
| 17:44.58 | Druken | the DID table is on my mysql server, it's a very simple one, 4 fields, DID, customer_id, Assigned, note |
| 17:45.31 | Druken | dids => mysql,general,dids |
| 17:46.17 | dustybin | jgoo: lol |
| 17:46.36 | *** join/#asterisk Dibbler (n=Dibbler@87.194.103.72) |
| 17:49.01 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:50.06 | Katty | stretches |
| 17:50.29 | [TK]D-Fender | Druken: I see no reference that you can use REALTIME ad-hoc for any table yuo want. |
| 17:50.51 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:51.04 | Druken | well, it used to work... unless they changed it to restrict, it still should... |
| 17:51.48 | *** join/#asterisk twanny796 (i=lus@sdf-eu.org) |
| 17:52.12 | beek | Hello everyone. |
| 17:52.20 | twanny796 | Hello |
| 17:52.45 | *** join/#asterisk errotan (n=errotan@62.201.122.227) |
| 17:52.46 | jaytee | hi beek |
| 17:52.50 | beek | hi jaytee |
| 17:52.57 | Katty | hugs on beek |
| 17:53.27 | beek | hi katty! |
| 17:53.37 | Katty | :> |
| 17:53.46 | twanny796 | trying to get my Z100P card working |
| 17:53.55 | twanny796 | X100P |
| 17:54.45 | twanny796 | insmod wcfxo should do it , yes? |
| 17:55.33 | Katty | has a sore from too much salt :< |
| 17:55.47 | KyleK | can I make my SPA3102 require an asterisk calling it to send a password? |
| 17:56.25 | Tim_Toady | twanny796 first u modprobe zaptel or dahdi and then the specific module |
| 17:56.31 | *** join/#asterisk sjobeck (n=Adium@66.178.159.242) |
| 17:56.37 | [TK]D-Fender | Druken: I highly recommend you use a normal agnostic function for DB like func_odbc, etc |
| 17:57.18 | beek | Katty: Marguerittas? |
| 17:57.22 | *** part/#asterisk sjobeck (n=Adium@66.178.159.242) |
| 17:57.28 | Katty | chips |
| 17:58.02 | Katty | and it's right on the side so it hurts anytime i do anything >.< |
| 17:58.29 | dustybin | turns on his War Dialer script |
| 17:58.31 | beek | Ask your dentist for a bottle of Peredex. That stuff heals any mouth sores. |
| 17:58.33 | twanny796 | Tim_Toady : modprobe zaptel : Cant locate module zaptel !! |
| 17:58.42 | Katty | beek: wow, really? |
| 17:58.52 | Katty | calls dentist |
| 17:58.55 | Tim_Toady | twanny796 dahdi then |
| 17:58.55 | beek | I always keep a bottle in the medicine chest. |
| 17:59.15 | Druken | you sure you wana admit you get frequent mouth sores? |
| 17:59.18 | twanny796 | Tim_Toady : same err |
| 17:59.34 | Katty | Druken: are you overly sensitive about that subject? |
| 17:59.35 | beek | Druken: It heals the burns caused by corn-on-the-cob. |
| 17:59.36 | Tim_Toady | twanny796 have u got dahdi installed? |
| 17:59.57 | Druken | mmm, corn on the sob.... too damn i can't have that no more... :( |
| 18:00.06 | Druken | wife can't eat corn... |
| 18:00.08 | Katty | beek: i get them from too much citrus too |
| 18:00.08 | Druken | cries |
| 18:00.10 | twanny796 | I don't know about dahdi |
| 18:00.25 | beek | Wait... SHE can't eat corn but what is stopping you? |
| 18:00.25 | Katty | there's gotta be somethin wrong with me |
| 18:00.38 | Katty | beek: probably one of those support things |
| 18:00.46 | Druken | beek: the fact she loves it, and i won't eat it in front of her.. |
| 18:00.47 | Katty | beek: he doesn't keep it around so she won't feel tempted. |
| 18:00.56 | Katty | Druken: that's very sweet of you. |
| 18:00.58 | Tim_Toady | twanny796 in order to get ur hardware working u need dahdi-linux and dahdi-tools: http://www.asterisk.org/downloads |
| 18:01.05 | Katty | Druken: i'm sure she appreciates it a great deal. |
| 18:01.08 | beek | Druken, send her out on a shopping spree and while she's gone... |
| 18:01.44 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
| 18:01.53 | twanny796 | Tim_Toady : I'm trying to run asterisk on ipcop, from Berlios |
| 18:01.56 | Katty | starts making list for cabin trip. |
| 18:02.11 | Katty | it's amazing all the crap i want take with me to a cabin. you'd think i'd want to get away from all of it |
| 18:02.35 | Tim_Toady | twanny796 thensearch if it has some packages for dahdi or zaptel (dahdis older name) |
| 18:02.55 | Druken | [TK]D-Fender: do we know who wrote the function of realtime? |
| 18:03.23 | *** join/#asterisk wonderworld (n=ww@mue-88-130-95-064.dsl.tropolys.de) |
| 18:03.34 | twanny796 | Tim_Toady; thinking there's something with modprobe |
| 18:03.41 | jaytee | Is Riddick going to the cabin too? |
| 18:04.03 | Tim_Toady | twanny796 try depmod -a and then modprobe |
| 18:04.17 | *** join/#asterisk x86 (n=porteb1@p3m/member/x86) |
| 18:04.21 | *** join/#asterisk uski (n=uski@nor75-27-88-178-184-116.fbx.proxad.net) |
| 18:04.33 | wonderworld | hey i am having problems with asterisk calling some fax machines. the second the fax on the other side picks up our call, asterisk hangs up on it and says"congestion". i can call the fax with my cell just fine and hear the fax sounds.... |
| 18:05.05 | uski | hi; i am looking for a way to allow a user to enable/disable something ("do not disturb") using his phone by calling a special extension. I can't find a way to set a non-volatile variable and to check its status. any idea/example? thanks |
| 18:05.22 | twanny796 | Tim_Toady ; still cant' locate module zaptel |
| 18:05.27 | wonderworld | uski: use the asterisk DB |
| 18:05.45 | twanny796 | should it be just zaptel or zaptel.o or zaptel.o.gz? |
| 18:05.47 | uski | wonderworld, ok I'll look into it, thanks |
| 18:05.48 | Tim_Toady | twanny796 then its not installed |
| 18:06.34 | twanny796 | Tim0_Toady; zaptel.o.gz is in /lib/modules/2.4.31/misc |
| 18:06.59 | twanny796 | maybe its out of modprobe path? |
| 18:07.02 | Tim_Toady | twanny796 is that ur running kernel? |
| 18:07.23 | twanny796 | how do i check that, |
| 18:07.32 | Tim_Toady | uname -r |
| 18:07.45 | bmoraca | wtf...my polycom 330s aren't downloading the included configuration files from mac.cfg...what would cause that? |
| 18:08.05 | Naikrovek | bmoraca: do the logs show the attempt |
| 18:08.08 | [TK]D-Fender | Druken: Nope |
| 18:08.09 | twanny796 | nope ist' 2.4.34 |
| 18:08.26 | bmoraca | Naikrovek: no...neither TFTP logs nor syslog nor boot logs show an attempt |
| 18:08.28 | Tim_Toady | twanny796 then boot with 2.4.34 or install zaptel for 2.4.31 |
| 18:08.40 | Tim_Toady | oups the other way arround |
| 18:08.41 | Naikrovek | bmoraca: then they don't get far enough to require a config |
| 18:08.50 | Naikrovek | bmoraca: are they booting successfully? |
| 18:08.50 | [TK]D-Fender | twanny796: You never confirmed what you were RUNNING |
| 18:09.08 | Zhad | which files are they asking for? |
| 18:09.09 | Druken | [TK]D-Fender: s'ok, i checked the source code, emailed him... i'm also skimming the code.. see what i can find out |
| 18:09.12 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 18:09.25 | [TK]D-Fender | twanny796: Please specify your *, zaptel/DAHDI, kernel and OS version |
| 18:09.37 | [TK]D-Fender | Druken: Just use proper tools for the job. |
| 18:09.43 | twanny796 | the prob is that I cannot compile anyting on ipcop |
| 18:09.49 | [TK]D-Fender | Druken: Realtime si for * configs, not your own ad-hoc stuff |
| 18:09.56 | bmoraca | Naikrovek: no...they get to the "running sip.ld" and then reboot. |
| 18:10.22 | Zhad | remembers having that briefly with some 501s |
| 18:10.28 | twanny796 | can I move themods form /lib/modules/2.4.31/misc to /lib/modules/2.4.34/misc ;) |
| 18:10.39 | Naikrovek | bmoraca: it's the sip.ld file that requires the [mac].cfg. have you tried later firmware? if you have, i'd call polycom |
| 18:10.48 | Naikrovek | bmoraca: polycom is awesome about returns/replacement |
| 18:10.50 | Zhad | I think it was caused by the .cfg format not matching the firmware |
| 18:11.17 | bmoraca | Naikrovek: it downloads mac.cfg no problem |
| 18:11.23 | Zhad | was based on the xml shipped with 3.x.x and the firmware was 2.4.x iirc. |
| 18:11.26 | bmoraca | Naikrovek: it's just the included files that are not downloaded. |
| 18:11.27 | Tim_Toady | twanny796 NO |
| 18:11.43 | Naikrovek | bmoraca: okay. did you try new firmware? |
| 18:12.05 | Zhad | That's what happened to me, check the mac.cfg against the one shipped with the firmware you are running. |
| 18:12.06 | [TK]D-Fender | twanny796: Zaptel MUST be compiled for your running kernel |
| 18:12.07 | bmoraca | Naikrovek: several versions, yes. 2.2.2 as well as 3.1.2 as well as several bootloader versions |
| 18:12.22 | Naikrovek | bmoraca: how many phones are affected |
| 18:12.33 | bmoraca | Naikrovek: every single 330 I try |
| 18:12.45 | bmoraca | i've tried 4 different phones |
| 18:12.46 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:12.55 | bmoraca | my 550s, though, work perfectly with the exact same config files |
| 18:13.01 | Naikrovek | bmoraca: call polycom. i've never even heard of that problem until now |
| 18:13.14 | Naikrovek | bmoraca: config files vary from major SIP version to another |
| 18:13.23 | Naikrovek | 2.x.x used a different format than 3.1.x |
| 18:13.42 | Naikrovek | i'm using 3.1.3.0439 if it helps |
| 18:13.44 | bmoraca | Naikrovek: i'm aware of that. however, the 550s always at least attempt to download the included files. the 330s NEVER do |
| 18:14.02 | bmoraca | Naikrovek: mind uploading your config files? |
| 18:14.17 | Naikrovek | bmoraca: np. one sec |
| 18:14.48 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 18:15.11 | twanny796 | Tim_Toady : any workarounds to find drivers for 2.4.34?? |
| 18:15.25 | *** part/#asterisk huey23 (n=homygood@65.111.253.116) |
| 18:16.05 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 18:16.11 | Zhad | bmoraca> replace one of the <mac>.cfg files with the 000000000.cfg files from the firmware that you're running and see if the phone starts (and which files it fetches). |
| 18:16.14 | Tim_Toady | twanny796 i have no idea how ipcop works, maybe it provides some precompiled packages |
| 18:16.52 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 18:17.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:18.25 | twanny796 | Tim_Toady : thanks, will search |
| 18:18.52 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 18:19.27 | bpgoldsb | I'm trying to do some reporting on activity on my Asterisk installation. CDR doesn't seem to give me the information I need. Anyone know other solutions for logging the DAHDI/SIP channel that a call came in on, and the DAHDI/SIP channel that it was connected to? |
| 18:20.03 | KyleK | do something in the dialplan |
| 18:20.27 | bpgoldsb | I'm working on doing that, but I feel like an under-the-hood solution would be better. |
| 18:20.30 | *** join/#asterisk rollot (n=rollotom@96.245.38.145) |
| 18:20.31 | [TK]D-Fender | bpgoldsb: CDR is all you've got unless you log your own as KyleK suggested |
| 18:20.36 | bpgoldsb | And I don't know if someone has already tackled that. |
| 18:21.02 | rollot | Can anyone recommend a good provider for unlimited outbound (US48)? |
| 18:21.47 | KyleK | set(cdr(userfield)) = "channel number here" |
| 18:22.44 | bpgoldsb | I suppose I'll take a look at that. Thanks. |
| 18:25.53 | carrar | hawt: http://seclists.org/fulldisclosure/2009/Sep/0039.html |
| 18:27.46 | bmoraca | it seems to be working now...but it's plugged in to a different switch...maybe it doesn't like my Cisco 3550 switches... |
| 18:28.44 | bmoraca | uhg that's going to piss me off |
| 18:29.02 | ccesario | somebody can say me if this is "correct" ... http://pastebin.com/m2908a007 ... or can be a possible problem/bug ? |
| 18:29.20 | carrar | bmoraca, is that a POE switch? |
| 18:29.26 | ccesario | increasing the verbose I get a warning... http://pastebin.com/m5d5c6d74 |
| 18:29.37 | bmoraca | carrar: yep |
| 18:29.47 | carrar | bmoraca, force the POE to always on |
| 18:30.01 | carrar | not auto |
| 18:30.14 | jaytee | On a midnight dark and dreary, whilst I pondered weak and weary |
| 18:30.16 | carrar | I remember reading something about that |
| 18:30.38 | carrar | something inline power on |
| 18:30.52 | Xetrov` | quoth the server 404? |
| 18:31.50 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 18:32.55 | bmoraca | carrar: sounds about right...let me go attempt this and see what happens |
| 18:32.59 | bmoraca | uhg |
| 18:33.56 | carrar | bmoraca: http://www.trixbox.org/forums/vendor-moderated-forums/polycom/cisco-3550-24-port-poe-switch-and-polycom-430s-phone-reboots |
| 18:34.01 | carrar | int range fa0/2 - 24 |
| 18:34.02 | carrar | power inline delay shutdown 20 initial 300 |
| 18:37.20 | [TK]D-Fender | jaytee: Oh, once upon a midnight dearie I woke with something in my head. I couldn't escape the memory of a phone call and of what you said |
| 18:39.24 | jaytee | "For the love of God, Montressor!!!!" |
| 18:42.37 | Druken | [TK]D-Fender: -- Executing [7058123271@incoming:3] Set("SIP/64.34.181.47-08271490", "row_DID=did=7058123271,customer_id=00000000001,assigned=000001,note=Toll Free,") in new stack |
| 18:42.37 | Druken | there's your reference :P |
| 18:44.39 | *** part/#asterisk twanny796 (i=lus@sdf-eu.org) |
| 18:46.12 | *** join/#asterisk Micc (n=dotirc@c-98-225-59-171.hsd1.wa.comcast.net) |
| 18:46.43 | *** join/#asterisk sjobeck (n=Adium@66.178.159.242) |
| 18:48.26 | *** part/#asterisk sjobeck (n=Adium@66.178.159.242) |
| 18:53.46 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:55.08 | bmoraca | carrar: yeah, i found that and it works |
| 18:55.14 | wonderworld | is there a way to make dahdi ignore incoming isdn msns? |
| 18:55.55 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:56.30 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 18:57.49 | carrar | sweet |
| 18:58.00 | Naikrovek | bmoraca: poe switch? really |
| 18:58.09 | bmoraca | Naikrovek: yep...PoE switch was the issue |
| 18:58.13 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-jgfiejifbdrpaeuz) |
| 18:58.14 | Naikrovek | bizarre |
| 18:58.21 | Naikrovek | is that for a particular switch or just any cisco switch |
| 18:58.29 | Naikrovek | reads back |
| 18:58.29 | came0 | hey I'm looking to integrate my asterisk server with some web based software I wrote. is AJAM still the recomended way of doing this? I'll need to get voicemail information as well as initiating calls, ect. |
| 18:58.45 | Naikrovek | so did setting it to always on fix it or did you just not use POE |
| 18:58.54 | bmoraca | Naikrovek: probably any Cisco PoE switch... |
| 18:59.16 | bmoraca | Naikrovek: loosening the restrictions on the power settings fixed it |
| 18:59.22 | Naikrovek | cool beans |
| 18:59.25 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 18:59.28 | Naikrovek | update to the very latest firmware |
| 18:59.42 | Naikrovek | use the combined one, then switch to split for bootrom and sip |
| 18:59.56 | Naikrovek | split will bring the phones up faster once they boot |
| 19:00.01 | Naikrovek | actually |
| 19:00.11 | bmoraca | Naikrovek: I did upgrade and i am using split finally |
| 19:00.15 | Naikrovek | ok cool |
| 19:00.18 | Naikrovek | split works better |
| 19:00.24 | bmoraca | yep |
| 19:00.51 | Naikrovek | phones take much less time to boot on split |
| 19:01.02 | carrar | yeah using combined is dog slow |
| 19:01.13 | carrar | and uses more memory |
| 19:01.21 | carrar | avoid that |
| 19:01.40 | Naikrovek | no one else knows wtf i'm talking about. polycom offers their firmware in two downloads; one single .ld file, many many megs in size, containing firmware for all their phones |
| 19:01.41 | Naikrovek | or |
| 19:02.03 | carrar | Naikrovek, most in here do I believe |
| 19:02.06 | Naikrovek | multiple, small .ld files, one file for each phone model. downloading split will make it so the phone only downloads the small .ld file it needs, and nothing else |
| 19:02.12 | Naikrovek | well a month ago, i wouldn't have |
| 19:02.21 | Naikrovek | now it's archived somewhere |
| 19:02.41 | carrar | anyone who does what they are supposed to do and read the Admin guide knows :) |
| 19:02.46 | Naikrovek | that's true |
| 19:02.54 | Naikrovek | but how many actually read the manuals before they attempt it |
| 19:03.00 | carrar | heh |
| 19:03.08 | carrar | 1:1231412312 |
| 19:03.26 | Naikrovek | lol |
| 19:03.28 | Naikrovek | about that |
| 19:03.29 | Naikrovek | yes |
| 19:04.00 | came0 | anybody have experience using AJAM to access the AMI? |
| 19:04.01 | Katty | wooo!!! |
| 19:04.05 | Katty | my jinx package came in!! |
| 19:05.13 | carrar | flow modeling? |
| 19:06.55 | jaytee | I use the smaller ld files on my Polycom phones. |
| 19:07.06 | Naikrovek | jaytee: they work better |
| 19:07.09 | Naikrovek | much |
| 19:07.43 | jaytee | less to download when reconfiguring the phone |
| 19:11.51 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 19:14.33 | KyleK | [Sep 8 12:14:09] WARNING[16466]: pbx.c:3839 __ast_pbx_run: Channel 'SIP/missy-spa-0a1902a8' sent into invalid extension '2004' in context 'internal', but no invalid handler |
| 19:15.05 | KyleK | i think that error is gramatically wrong :-/ |
| 19:15.39 | KyleK | (i know why its reporting that, but the error text is making me go "huh".) |
| 19:16.43 | *** join/#asterisk korihor (n=korihor@190.77.83.180) |
| 19:20.18 | *** join/#asterisk d00gster (n=doughant@94.98.197.67) |
| 19:20.44 | [TK]D-Fender | KyleK: Souds remarkably straightforward to me... |
| 19:22.20 | wonderworld | why would i get span 1 got hangup request, cause 17 (busy) when calling a fax machine that isn't busy? |
| 19:28.08 | *** join/#asterisk brezular (n=brezular@adsl-dyn128.91-127-119.t-com.sk) |
| 19:28.10 | KyleK | but no invalid handler? |
| 19:28.23 | KyleK | oh i guess somewhere to go when things screw up? |
| 19:28.27 | bmoraca | KyleK: that means you don't have an extension set up for handling invalid extensions |
| 19:29.43 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:30.11 | KyleK | ah |
| 19:33.34 | [TK]D-Fender | wonderworld: Care to actually show us the call. System status, etc? |
| 19:35.50 | *** join/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 19:36.17 | spck | for some reason my asterisk box stopped generating queue member status events, has anyone else run into this? |
| 19:36.28 | uski | anyone uses mbrola with festival with asterisk? I can't manage to get festival to see my mbrola voice |
| 19:36.38 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:36.47 | *** part/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 19:38.02 | Skeeter- | i need to convert a .wav into .wav formart that asterisk can use in the IVR |
| 19:38.28 | Skeeter- | cant figure out how to make it work, tired the built in converter and other software |
| 19:38.39 | KyleK | really? |
| 19:38.55 | [TK]D-Fender | Skeeter-: if the builtin converter could read it you wouldn't NEED to convert |
| 19:39.09 | KyleK | convert it to single channel 8000hz sample rate 16bit samples |
| 19:39.15 | Tim_Toady | Skeeter- convert it to 8000Hz with sox |
| 19:39.21 | Skeeter- | i did that |
| 19:39.28 | [TK]D-Fender | Skeeter-: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 19:39.31 | Skeeter- | i think i miss the PCM encoded part |
| 19:39.52 | Skeeter- | tried everythingon that page |
| 19:40.09 | [TK]D-Fender | Skeeter-: http://audacity.sourceforge.net/ |
| 19:40.10 | *** join/#asterisk brezular (n=brezular@adsl-dyn128.91-127-119.t-com.sk) |
| 19:40.30 | Defraz | I have a PRI into my asterisk server. When I dial a menu that requires a selection (DTMF) if i do it slow it seems to work pretty good but when I do it fast it never picks up the digits. |
| 19:40.41 | Tim_Toady | and where is the problem then? |
| 19:40.46 | Defraz | So for example if I dial my bank. |
| 19:41.07 | Defraz | well, on a normal land line I can just punch away and it seems to work well. |
| 19:41.12 | [TK]D-Fender | defdo it slow from what to what? |
| 19:41.16 | Defraz | even on my cell phone. |
| 19:41.18 | [TK]D-Fender | Defraz: do it slow from what to what? |
| 19:41.51 | Defraz | Like I dial my bank and it asks for my account number, I have to hold the keys down longer and wait in between a second or two. |
| 19:41.52 | KyleK | Defraz: what are you calling through asterisk with? ATA or VoIP phone? |
| 19:42.04 | Defraz | I am using a polycom 501 |
| 19:42.28 | KyleK | is dtmf inband or out of band? |
| 19:42.38 | Defraz | I have tried a grandstream and via an Handy Tone 502 and a linksys pap2t |
| 19:42.38 | [TK]D-Fender | Defraz: What card, what settings? |
| 19:42.56 | Kobaz | i should check out pbx_lua |
| 19:43.22 | Defraz | rfc2833 |
| 19:43.30 | Defraz | so that would be out of band right? |
| 19:44.05 | Defraz | I have a Cisco 1760 with a pri card in it and it talks sip to the Asterisk box. |
| 19:45.31 | Kobaz | hmm |
| 19:45.36 | Kobaz | what the heck is call token, in iax |
| 19:45.45 | Kobaz | the config sample doesn't say much about it |
| 19:47.48 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:48.41 | [TK]D-Fender | Defraz: then your Cisco has issues, or is not OOB to * |
| 19:48.53 | [TK]D-Fender | (Actually that may still imply "issues" |
| 19:49.01 | Defraz | OOB? |
| 19:49.07 | [TK]D-Fender | Out Of Band |
| 19:49.11 | Defraz | got it |
| 19:49.15 | Skeeter- | Fender: Audacity worked on the first try |
| 19:49.26 | Defraz | hmm I have rfc2833 set on the cisco. |
| 19:49.43 | Defraz | seems to do the same thing with my SIP providers too. |
| 19:50.27 | bmoraca | why is your PRI going through your Cisco 1760 in the first place? |
| 19:52.04 | Defraz | Well, I have 2 asterisk servers, and if one server died I wanted it to fail over to the other. |
| 19:52.24 | Defraz | and I had a cisco router when using call manager and the pri so I thought, hey that would be a good backup. |
| 19:53.44 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 19:56.33 | Naikrovek | Defraz: if you don't have any dahdi hardware just virtualize the server and store the vdisk on a raid array |
| 19:57.01 | *** join/#asterisk Tim_Toady (n=moi@adsl113-199.kln.forthnet.gr) |
| 19:57.09 | Naikrovek | not sure what kind of budget you have, but if you can afford two servers you can afford to run freenas(free) or openfiler(free) and have some virtual machine hosts. |
| 19:57.56 | bmoraca | i wonder if freenas has an iscsi target... |
| 19:57.56 | Micc | can asterisk dialplan handle millions of extensions? |
| 19:58.12 | Naikrovek | bmoraca: yes |
| 19:58.17 | Naikrovek | openfiler certainly does |
| 19:58.21 | bmoraca | is it relatively fast? |
| 19:58.22 | Naikrovek | uses openfiler |
| 19:58.41 | Naikrovek | bmoraca: i get gigE speeds to openfiler (around 100-110MBps) |
| 19:58.49 | p3nguin | Has it improved in the last year? |
| 19:58.57 | Naikrovek | try it and tell me |
| 19:58.59 | Naikrovek | i'm new to it |
| 19:59.06 | p3nguin | The last time I wanted to use it, it wasn't working well for me. |
| 19:59.10 | Naikrovek | it has been flawless in the last two months |
| 19:59.12 | Naikrovek | not a single issue |
| 19:59.22 | p3nguin | Sounds good enough to test it again, then. |
| 19:59.22 | Naikrovek | even through a power failure it came up and resumed its work |
| 19:59.30 | Naikrovek | i don't even have a keyboard or monitor on it |
| 19:59.37 | p3nguin | of course |
| 19:59.43 | Micc | What is openfiler? |
| 19:59.49 | p3nguin | a NAS os |
| 19:59.50 | bpgoldsb | If I want to import a var from another channel I would use import. But how do I get a list of all active channels so I can choose one to import? |
| 20:00.06 | Naikrovek | Micc: http://www.openfiler.com/ |
| 20:00.10 | Micc | what protocol does it use? nfs? |
| 20:00.11 | bmoraca | hmmm...this could be useful |
| 20:00.26 | p3nguin | smb/cifs last I looked |
| 20:00.29 | Naikrovek | bmoraca: NFS, CIFS (windows file share), FTP, HTTP, uh, iSCSI |
| 20:00.59 | Naikrovek | i bought a big disk array for $500 from my brother in law, and now use it as a SAN with openfiler |
| 20:01.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 20:01.16 | p3nguin | You've got a SAN on your NAS? Nice! |
| 20:01.17 | bmoraca | i've been looking for a cheap iscsi SAN...this might fit the bill...i wonder how flexible it is... |
| 20:01.34 | Naikrovek | ah okay NAS |
| 20:01.38 | bpgoldsb | Cheap for a person, or cheap for a SAN? |
| 20:01.39 | Naikrovek | network attached storage |
| 20:01.54 | Naikrovek | bpgoldsb: cheap for a san. if you have hardware and disks already, it's free |
| 20:02.01 | bmoraca | cheap for a SAN |
| 20:02.28 | Naikrovek | i use it for iSCSI |
| 20:02.31 | Naikrovek | works great |
| 20:02.33 | p3nguin | I currently have a FreeNAS box, but I'm going to have to check Openfiler again. |
| 20:02.40 | bpgoldsb | I've been toying around with building a SAN for my company for a while |
| 20:02.43 | bmoraca | i looked at the StarWind iscsi target software...but it's slower than shit and their sales people won't stop calling me |
| 20:02.50 | bpgoldsb | Using DRBD/LVM/iSCSI/Heartbeat |
| 20:03.02 | Naikrovek | if freenas does what you need i'd use that. I tried freenas but it wouldn't install because i have more than 2gb of ram in that server |
| 20:03.09 | Naikrovek | which is a problem for freenas apparently |
| 20:03.09 | bpgoldsb | or if not iSCSI, ATAoE |
| 20:03.23 | Naikrovek | iscsi is basically a network harddrive cable. |
| 20:03.32 | Naikrovek | you can only plug it into one server, |
| 20:03.35 | bpgoldsb | If you're going to build a SAN, I suggest NOT going with something like Openfiler etc |
| 20:03.36 | Naikrovek | etc |
| 20:03.44 | Naikrovek | bpgoldsb: explain |
| 20:04.01 | bmoraca | they're not good for mission critical :P |
| 20:04.07 | bpgoldsb | Because when you start using a SAN, you start needing it to be reliable very quick. If you can't fix virtually any problem with a SAN, you're effed. |
| 20:04.23 | bpgoldsb | My point is if you don't understand all the underlaying technology, it's a bad idea |
| 20:04.37 | p3nguin | iSCSI just uses IP networking. |
| 20:04.39 | Kobaz | or if you have a support contract |
| 20:04.47 | bpgoldsb | Because when Openfiler et all doesn't have a button to fix... Bye. |
| 20:05.05 | bpgoldsb | Sure, I guess I'm just used to my company which doesn't like support contracts. |
| 20:06.22 | Kobaz | if you need mission critical storage. you need a support contact where you can call a say 'broken, fix!' and they will show up at your door in an hour |
| 20:06.34 | bpgoldsb | Or know how to fix it yourself. |
| 20:06.48 | Kobaz | but expect to pay several thousand a year for a contract like that |
| 20:06.50 | Kobaz | or that too |
| 20:07.12 | bpgoldsb | I feel very happy with my experiences with DRBD, Heartbeat and LVM to have a product thats bother cheaper and more reliable for the same price. |
| 20:08.53 | Kobaz | i've been using linux raid1 with lvm on top |
| 20:08.58 | *** join/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 20:09.26 | Naikrovek | uses his openfiler for backup. Windows uses it for the "previous versions" feature |
| 20:09.35 | Naikrovek | so if it goes away, i still have it on tape |
| 20:09.44 | Naikrovek | uses Cybernetics hardware for mission critical stuff |
| 20:09.49 | Naikrovek | that IS an endorsement |
| 20:09.53 | Naikrovek | they rule |
| 20:10.05 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:12.05 | grandpapadot | uses rsync ... never had a problem ... ever |
| 20:12.14 | Kobaz | heh |
| 20:13.49 | Naikrovek | well when you need to have previous versions of network share files, and you don't want to be a part of every single file restore, rsync won't cut it |
| 20:13.53 | Naikrovek | but yes, rsync is cool |
| 20:13.55 | Naikrovek | very cool |
| 20:14.02 | *** part/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 20:14.41 | grandpapadot | Naikrovek: source one dir, dest another based on date/time, done. |
| 20:14.47 | Naikrovek | i need to upgrade the disks in my primary NAS device too... |
| 20:15.00 | Naikrovek | grandpapadot: i know how to use it, but for what I use openfiler for, it won't cut it |
| 20:15.26 | Naikrovek | not because it can't cut it, because i don't want to be involved every time anyone in india or the US offices accidentally deletes a file or wants to see a previous version |
| 20:15.43 | Naikrovek | i believe in automation |
| 20:16.46 | Naikrovek | why should i have to be involved when i can write a solution (or employ an existing solution) that does the work for me? i have better things to do |
| 20:18.32 | Naikrovek | this is why PBXs were created. i don't want lily tomlin and her smartass comments at my office listening to all my phone calls |
| 20:18.53 | Naikrovek | so we automate that |
| 20:20.07 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 20:23.30 | *** join/#asterisk dysinger (n=tim@166.187.255.96) |
| 20:24.02 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-75-14.pskn.east.verizon.net) |
| 20:24.56 | IBC_jkenney | I know this isn't the hylafax room but nobody is ever in there |
| 20:25.20 | IBC_jkenney | i want to setup hylafax that if a specific modem gets a fax it sends it to a specific e-mail |
| 20:25.20 | Naikrovek | anyone know how to make my paging and intercom work across a trunk? |
| 20:26.12 | grandpapadot | Naikrovek: Polycom phones? |
| 20:26.12 | Naikrovek | grandpapadot: yes |
| 20:26.20 | Naikrovek | of course, what kind of wanker do you think i am? :) |
| 20:27.13 | Naikrovek | i use freepbx, it hink i could just add an outbound route for those extensions |
| 20:27.19 | *** join/#asterisk wonderworld (n=ww@mue-88-130-101-031.dsl.tropolys.de) |
| 20:27.28 | Naikrovek | so it passes the *805XX to that server |
| 20:27.43 | Naikrovek | instead of trying a local extension |
| 20:27.48 | Naikrovek | will figure it out |
| 20:27.50 | wonderworld | i am having problems with compiling latest 1.4 i compiled and installed libri, dahdi, dahdi-tools andd asterisk |
| 20:28.02 | wonderworld | after that i have no chan-dahdi in asterisk available? |
| 20:28.05 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:29.01 | *** join/#asterisk KyleK (n=Kyle@64.114.61.6) |
| 20:29.02 | *** join/#asterisk Joel (i=jjshoe@75.85.173.90) |
| 20:29.21 | [TK]D-Fender | checkout time, BBIAB |
| 20:30.46 | grandpapadot | Naikrovek: You using the Alert-Info header? |
| 20:30.57 | Naikrovek | yeah |
| 20:31.01 | Naikrovek | already figured it out |
| 20:31.02 | Naikrovek | hehe |
| 20:31.08 | grandpapadot | cool. |
| 20:31.09 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-145-59.dsl.stlsmo.sbcglobal.net) |
| 20:31.12 | Naikrovek | i gotta stop using /me |
| 20:31.16 | Naikrovek | it's annoying even to me |
| 20:32.08 | Naikrovek | i just added an "outbound route" for all you freepbx lurkers, and just specified the one extension i have at home in the route. when i dial it, it passes the call to my pbx at home and redials it there |
| 20:32.14 | Naikrovek | instant intercom |
| 20:32.22 | LemensTS | hey all. Im writing a calling program in phpagi and need to keep track of call times and stuff for billing purposes and when to cut there service off. Ive never done cdr's before, should i just keep the default text cdr's, or is there a reason to go with mysql cdr's? |
| 20:32.29 | Naikrovek | i should add a password to that trunk so no one can intercom me at home except me |
| 20:32.44 | grandpapadot | Naikrovek: Ya, that's how we do call groups, works great. |
| 20:32.57 | Naikrovek | yeah i have several paging groups set up here, it's awesome |
| 20:33.28 | Katty | sips tea. |
| 20:34.04 | beek | is enjoying a glass of iced tea. |
| 20:34.24 | asterwiki | LemensTS: mysql cdr offers good user-interface and tools to do call analysis, makes obtaining n=billing details easier as well; |
| 20:34.52 | IBC_jkenney | peek o boo |
| 20:34.57 | Naikrovek | throws his iced tea against the wall, then feels bad and goes to watch terminator 2 in the bedroom |
| 20:35.02 | *** join/#asterisk propellerhead (n=yogurt2u@host18.190-138-101.telecom.net.ar) |
| 20:35.05 | Naikrovek | i see youuuu |
| 20:35.18 | *** join/#asterisk TimToady_ (n=moi@adsl164-83.kln.forthnet.gr) |
| 20:35.19 | Naikrovek | just kidding. i'd dump it in the toilet |
| 20:35.26 | IBC_jkenney | anyone know how to set a individual |
| 20:35.28 | Naikrovek | doesn't throw; just watch him play basketball |
| 20:35.35 | IBC_jkenney | fax number in hylafax |
| 20:35.43 | IBC_jkenney | to send faxes to a different e-mail |
| 20:35.45 | LemensTS | asterwiki: thx |
| 20:35.54 | Naikrovek | IBC_jkenney: i have no experience with hylafax, can't help ya. |
| 20:36.10 | IBC_jkenney | i checked the hylafax room heard crickets |
| 20:36.42 | came0 | IBC_jkenny: I used to run a hylafax server here.. what are you trying to do? |
| 20:37.07 | *** join/#asterisk twanny796 (n=chatzill@85.232.204.228) |
| 20:37.29 | IBC_jkenney | if someone gets a fax on modem A send it to email address b |
| 20:37.42 | twanny796 | any links to precompiled zaptel moduls for linux 2.4.34? |
| 20:37.43 | IBC_jkenney | if someone gets a fax on modem b send it to email A |
| 20:37.53 | Naikrovek | twanny796: linux 2.4? why? |
| 20:38.17 | twanny796 | Naikrovek: running asterisk on ipcop |
| 20:38.29 | Naikrovek | what's ipcop |
| 20:38.41 | Naikrovek | ah |
| 20:38.43 | Naikrovek | googles it |
| 20:38.46 | twanny796 | Naikrovek: firewall |
| 20:38.54 | IBC_jkenney | came0 did you get that? |
| 20:39.03 | twanny796 | Naikrovek: www.ipcop.org |
| 20:39.07 | Naikrovek | twanny796: yeah |
| 20:39.19 | Naikrovek | twanny796: you use this for your main corp/home firewall? |
| 20:39.22 | wonderworld | i am having problems with compiling latest 1.4 i compiled and installed libri, dahdi, dahdi-tools andd asterisk |
| 20:39.25 | wonderworld | after that i have no chan-dahdi in asterisk available? |
| 20:39.31 | came0 | IBC_jkenney: yeah let me think about it for a second |
| 20:39.37 | twanny796 | Naikrovek: for home, yep |
| 20:39.38 | IBC_jkenney | ok just checking |
| 20:40.03 | Naikrovek | twanny796: k. things like this are why virtual machines were created |
| 20:40.27 | Naikrovek | twanny796: i've not compiled against that kernel. I skipped 2.2 and 2.4 as i was out of the linux world then |
| 20:40.32 | Naikrovek | that's about 12 years of time i think |
| 20:41.04 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:41.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:41.04 | Naikrovek | twanny796: you may have better luck on digium.com forums |
| 20:41.27 | Naikrovek | twanny796: when you rebuild that machine, if you do, install esxi server, then add a virtual machine for ipcop and another for asterisk. |
| 20:41.36 | Naikrovek | then you can upgrade them indepenently and they can run on the same hardware |
| 20:41.42 | Naikrovek | i am a BIG believer in virtualization |
| 20:42.01 | grandpapadot | thinks running asterisk on a firewall is bad, m'kay? |
| 20:42.13 | Naikrovek | yeah probably not a good idea but he didn't ask how to take it off |
| 20:42.17 | twanny796 | Naikrovek: frankly, I don't like virtual machines |
| 20:42.41 | Naikrovek | twanny796: why not? I have a dozen guests running on a single $5k server at work, and they run great perform great |
| 20:43.20 | twanny796 | Naikrovek: but that's a $5k machine not a $200 PIII |
| 20:43.25 | twanny796 | ;) |
| 20:43.33 | dustybin | at last, a nice simple easy to read, compile asterisk howto as non-root user |
| 20:43.35 | dustybin | http://www.thinkdebian.org/archives/828 |
| 20:43.44 | Naikrovek | on 32bit machines you will see a tiny (<1%) performance degredation, on 64-bit hardware there is no performance penalty at all, actually since the drivers are so much simpler, they seem to perform better |
| 20:43.53 | came0 | IBC_jkenny: I cant remember how I did it... something in the setup.modem i think |
| 20:43.54 | grandpapadot | twanny796: You will probably have better luck with 1.2 and those iterations of Zaptel on a 2.4 kernel ... |
| 20:43.57 | Naikrovek | twanny796: aah yeah. that would be a good reason |
| 20:44.10 | grandpapadot | *asterisk 1.2 |
| 20:44.13 | Naikrovek | twanny796: may i suggest a single Cisco ASA 5505 for $400 if you want a super firewall? |
| 20:44.33 | Naikrovek | and remote VPN access to your home network |
| 20:44.34 | heedly | Naikrovek: the complicated drivers are at some level |
| 20:45.23 | Naikrovek | heedly: yeah the virtual drivers have to go through the host machine drivers, but even those are pretty damn slim. ESXi 4.0 is like 32MB total; pretty slim |
| 20:45.36 | twanny796 | Naikrovek: prob I will be trying the berliOS distro which promises both world |
| 20:45.46 | Naikrovek | twanny796: cool |
| 20:45.58 | Naikrovek | i've never used dahdi at all, so i can't even say how to compile them against 2.4 |
| 20:45.58 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 20:46.04 | Naikrovek | surely there must be a tutorial out there somewhere |
| 20:46.30 | *** part/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 20:46.43 | twanny796 | Naikrovek: first of all I'm trying to get my X100P working, seems to need to load the zaptel driver first |
| 20:47.11 | iksik | is it possible to dial extension from CLI ? |
| 20:47.13 | grandpapadot | twanny796: You're installing an X100P and Asterisk on your firewall? Is this a business? |
| 20:47.42 | Naikrovek | iksik: core show application originate is what [tk]d-fender usually says |
| 20:47.48 | Naikrovek | grandpapadot: home |
| 20:48.10 | twanny796 | grandpapadot: no, dangerous? |
| 20:48.41 | Naikrovek | twanny796: well, firewalls are kinda meant to do firewall only, one in port from internet, one out port to each network they firewall; that's it |
| 20:48.53 | iksik | Command 'core show application originate' failed |
| 20:49.01 | grandpapadot | twanny796: for fun at home, I say go for it, for business, bad mojo ... |
| 20:49.09 | Naikrovek | if you misconfigure, any apps running on the firewall may not be protected |
| 20:49.26 | Naikrovek | iksik: core help show application originate maybe? whatever the help command is |
| 20:49.33 | Naikrovek | iksik: what asterisk version are you running |
| 20:49.41 | iksik | 1.6.1.1 |
| 20:52.07 | Naikrovek | iksik: in the asterisk CLI, type: help core show application Originate |
| 20:52.08 | IBC_jkenney | ok i will keep looking thanks |
| 20:52.25 | Naikrovek | i don't have asterisk 1.6 anywhere, so I can't verify here before i open my mouth on IRC |
| 20:52.41 | iksik | help core show application Originate |
| 20:52.41 | iksik | No such command 'core show application Originate'. |
| 20:52.42 | iksik | ;] |
| 20:53.03 | beek | IMAP VM storage is an all-or-nothing proposition isn't it? e.g., do all my users have to be using IMAP storage or can I be selective about it? |
| 20:53.52 | p3nguin | iksik: 1.4 doesn't have that, either. |
| 20:54.03 | iksik | hm, hm |
| 20:54.09 | Naikrovek | iksik: ask [tk]d-fender when he comes back on |
| 20:54.10 | Naikrovek | he'll know |
| 20:54.12 | KyleK | beek: its coded all or nothing in 1.6.1 |
| 20:54.17 | Naikrovek | he's answered that many times |
| 20:54.39 | iksik | hehe, ok |
| 20:54.45 | KyleK | answered what? |
| 20:54.57 | Naikrovek | KyleK: how do place a call from Asterisk CLI |
| 20:55.14 | Naikrovek | maybe it's not possible |
| 20:55.23 | beek | KyleK: Thanks. That's what I feared. A choice would be nice but I'll just play with a test server for a while before commiting the whol' enchalata. |
| 20:55.31 | Naikrovek | i've seen lots of people ask about it and [tk] is always right there with an answer that i've never read |
| 20:56.08 | Kobaz | it seems asterisk is constantly leaving odbc connections open, and creating more and more, until the server reaches it's limit |
| 20:56.15 | KyleK | Naikrovek: type originate in the cli |
| 20:56.20 | Naikrovek | KyleK: ah |
| 20:56.29 | Naikrovek | iksik: type originate in the cli |
| 20:56.30 | KyleK | the originate is a 1.6 thing |
| 20:56.33 | beek | iksik: Isn't that an AMI command? |
| 20:56.33 | Naikrovek | yes |
| 20:56.41 | Naikrovek | Kobaz: suck |
| 20:56.48 | iksik | hah |
| 20:56.51 | Naikrovek | Kobaz: that sucks, i mean |
| 20:56.51 | Kobaz | it would be nice if Originiate was also a dialplan application |
| 20:56.54 | iksik | originate works :P |
| 20:56.57 | iksik | but now what? D |
| 20:56.58 | iksik | :D |
| 20:57.00 | iksik | originate dial ? |
| 20:57.04 | beek | iksik: As an application? |
| 20:57.05 | Naikrovek | iksik: help originate |
| 20:57.56 | p3nguin | kylek: It can't be, since I have it in 1.4. |
| 20:57.57 | iksik | hmmm, wierd usage description ;P |
| 20:58.11 | KyleK | ah |
| 20:58.38 | p3nguin | iksik: http://www.voip-info.org/wiki/view/Asterisk+cli+originate |
| 20:59.10 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 20:59.45 | *** join/#asterisk wonderworld (n=ww@mue-88-130-101-031.dsl.tropolys.de) |
| 21:00.36 | wonderworld | i am desperate. i can't make asterisk 1.4 work with dahdi and a b410p |
| 21:01.10 | Naikrovek | wonderworld: i would help you if i could, but i've never used any physical asterisk hardware |
| 21:01.27 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:03.40 | Naikrovek | wonderworld: [tk]d-fender can probably help you when he comes on, but you'll need to be able to show him error messages and config files and the like via pastebin.ca |
| 21:04.00 | Naikrovek | he's kinda rough but he's smart and can help you if you state your question when he returns |
| 21:06.33 | kn0x | any reason why these polycom would subscribe, but not REGISTER? |
| 21:06.49 | grandpapadot | kn0x: you're seeing the notification subscription (mwi, hints) |
| 21:07.05 | grandpapadot | kn0x: one sec |
| 21:07.23 | kn0x | Found peer '733' for '733' from 192.168.1.33:5060 |
| 21:07.29 | kn0x | but it is not registering |
| 21:07.40 | kn0x | SIP debug only shows it subscribing |
| 21:08.01 | kn0x | Found peer '733' for '733' from 192.168.1.33:5060 |
| 21:08.03 | kn0x | oops |
| 21:08.09 | kn0x | Creating new subscription |
| 21:08.18 | Kobaz | hmmm |
| 21:08.28 | Naikrovek | kn0x: do you have a password on the phone or something |
| 21:08.44 | kn0x | just the polycom default 456 |
| 21:08.47 | Kobaz | there's something wrong with BASE64_ENCODE/DECODE |
| 21:08.52 | Naikrovek | no not that one |
| 21:08.56 | iksik | erm.... i've got wierd error |
| 21:08.58 | iksik | Channel 'SIP/7012-b8269d20' sent into invalid extension '7050' in context 'default', but no invalid handler |
| 21:09.03 | iksik | what does it mean? :| |
| 21:09.04 | Kobaz | strings encoded with BASE64_ENCODE are not properly decided with BASE64_DECODE |
| 21:09.25 | Kobaz | iksik: it means extension 7050 doesn't exist in the default context |
| 21:09.28 | p3nguin | Hopefully the originate discussion isn't too far passed... How do I find a channel to originate FROM? The guide suggests zap/1/123456 as an example, but I only have SIP channels. Not sure how to find a SIP channel to originate. |
| 21:09.45 | grandpapadot | iksik: means that a call was sent to exension 7050 in your default context but 7050 didn't exist and neither did the special invalid 'i' exension |
| 21:10.05 | Naikrovek | kn0x: i'm talking about the reg.1.auth.password= line in the phone config file |
| 21:10.11 | iksik | hmm |
| 21:10.38 | Kobaz | iksik: your sending a call to an extension@context that doesn't exist |
| 21:11.02 | kn0x | yes |
| 21:11.08 | kn0x | Naikrovek: yes i have a password |
| 21:11.14 | iksik | but this extension exists, and have exactly the same context like before (in sip.conf file)... but now it's moved into pgsql |
| 21:11.18 | iksik | hm |
| 21:11.22 | p3nguin | iksik: How do I determine a SIP channel to originate from? |
| 21:12.17 | iksik | p3nguin i've tried: originate SIP/extension_number extension extension_number |
| 21:12.25 | dustybin | are SMS messages supported in asterisk? |
| 21:12.29 | iksik | and it works ;] |
| 21:12.37 | p3nguin | I'll try that. |
| 21:12.50 | p3nguin | AH, it does. |
| 21:13.52 | p3nguin | iksik: Extensions are not configured in sip.conf, but are in extensions.conf instead. If the extension cannot be reached, it's probably because there is not one configured. |
| 21:14.11 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 21:14.21 | grandpapadot | dustybin: no |
| 21:14.54 | dustybin | grandpapadot: what is smsq ? |
| 21:15.31 | iksik | damn, I don't get it ;/ |
| 21:16.00 | p3nguin | Q: Is it possible to use the sms_app over zap without the .call file? |
| 21:16.01 | p3nguin | A: in newer versions of asterisk there is smsq - a tool that sends sms from the command line. |
| 21:16.23 | p3nguin | dustybin: http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 21:16.38 | p3nguin | iksik: I'll try to help you. What's the problem? |
| 21:16.39 | grandpapadot | dustybin: http://lmgtfy.com/?q=asterisk+smsq |
| 21:16.44 | *** join/#asterisk amazinzay (n=amazinza@67.108.187.186.ptr.us.xo.net) |
| 21:16.59 | iksik | p3nguin this line [2009-09-08 23:17:00] WARNING[21352]: pbx.c:3831 __ast_pbx_run: Channel 'SIP/7012-b8269d20' sent into invalid extension '7050' in context 'default', but no invalid handler |
| 21:17.33 | bmoraca | these damn polycom phones will be the death of me |
| 21:18.01 | iksik | now i'm using postgres database to keep my sip accounts... there are only 3 extensions, before when it works on files everything was ok, but now it does'nt ;/ |
| 21:18.08 | p3nguin | iksik: Seems clear to me. You have dialed an invalid extension '7050' in context 'default' |
| 21:18.10 | iksik | extensions.conf is not changed at all |
| 21:18.27 | iksik | only sip accounts was moved, dialplan is the same |
| 21:19.17 | p3nguin | I run mine from regular files instead of a database, so I don't know why that happens to you. |
| 21:19.27 | amazinzay | iksik: try running dialplan show 7050@default |
| 21:20.13 | iksik | There is no existence of 7050@default extension |
| 21:20.23 | p3nguin | If there is no explicit 7050 in the default context, that's going to be the result. |
| 21:20.50 | iksik | but i don't get it, why he is trying to call it via default context ;/ |
| 21:20.50 | p3nguin | If you are doing matching rather than having definitive ext numbers, that is. |
| 21:21.01 | p3nguin | What context are you trying to dial? |
| 21:21.04 | iksik | http://pastebin.com/m11814dc8 |
| 21:21.33 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 21:21.34 | dustybin | look at this monster |
| 21:21.34 | dustybin | http://www.polycom.eu/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip670.html |
| 21:21.49 | iksik | and 7050 belongs to DLPN_OutDPResellerPOST context |
| 21:22.23 | ayeso | Is there anything negative that can happen if a 'reload' is issued from the CLI on a live system? |
| 21:22.30 | p3nguin | Then originate the call to 7050@DLPN_OutDPResellerPOST |
| 21:22.58 | amazinzay | ayeso, it should not affect the calls currently connected on the system. |
| 21:23.10 | *** join/#asterisk ketema (n=ketema@turtle.ketema.net) |
| 21:23.30 | dustybin | I am so close to buying this, |
| 21:23.33 | dustybin | shall i buy it? |
| 21:23.33 | dustybin | http://www.pcwb.co.uk/catalogue/item/A0467565?cidp=Froogle |
| 21:24.16 | amazinzay | iksik: in your setup for the phone, what extension do you specify for that phone? |
| 21:24.58 | iksik | of this 7050? |
| 21:25.08 | amazinzay | no, the phone you are calling from |
| 21:25.13 | p3nguin | The way his macro looks, if he would dial 7050, extension 050 would be called. |
| 21:25.16 | iksik | it's my cellphone ;) |
| 21:25.32 | amazinzay | how are you connecting to the system? are you calling from the outside? |
| 21:25.42 | iksik | i'm trying to setup incoming calls route for _91XXXX into that extension |
| 21:26.01 | iksik | and i'm testing it with my cellphone |
| 21:26.03 | amazinzay | ok, so you are coming in through? PRI, SIP tunk, Analog? |
| 21:26.06 | p3nguin | Then you had better create an extension for it. |
| 21:26.10 | iksik | SIP Trunk |
| 21:26.11 | ayeso | Has anyone ever seen the voicemail application not hangup calls when the caller is gone? |
| 21:26.12 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 21:26.43 | amazinzay | ok so the config for that sip trunk, do you have a context explicitly declared? |
| 21:26.59 | iksik | everythink is ok with dialplan, cause nothing was changed here... something is wrong with database query or hmm, or something :D |
| 21:27.11 | iksik | everything* :P |
| 21:27.16 | p3nguin | I think it's the dial plan of extensions.conf. |
| 21:27.20 | p3nguin | That's my opinion. |
| 21:27.29 | *** join/#asterisk digilink_ (n=digilink@tn-76-5-159-175.sta.embarqhsd.net) |
| 21:27.37 | p3nguin | That, and not understanding how extensions are dialed. |
| 21:27.47 | iksik | extensions.conf wasn't changed... before (when I was using only files) everywhing was ok |
| 21:27.48 | amazinzay | you said you moved to a database based config |
| 21:27.50 | iksik | amazinzay hmm |
| 21:27.58 | iksik | amazinzay yes, but only sip accounts |
| 21:28.09 | amazinzay | so that sip trunk is going to be in that |
| 21:28.18 | iksik | yes |
| 21:28.31 | iksik | and all settings of trunks was moved into database |
| 21:28.54 | amazinzay | so in that database table there should be a line that specifies that particular trunk, on that line there should be a setting for context |
| 21:29.07 | amazinzay | if it is not set it will default to the default context |
| 21:29.10 | *** join/#asterisk ZX81 (n=Matt_Rid@121.74.243.11) |
| 21:29.16 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 21:29.19 | amazinzay | you need to change that row in the table and set the context |
| 21:29.30 | iksik | amazinzay it's set to: DLPN_OutDPResellerPOST |
| 21:29.49 | iksik | just like before ;/ |
| 21:30.11 | *** part/#asterisk ZX81 (n=Matt_Rid@121.74.243.11) |
| 21:30.12 | p3nguin | What does dialplan show 7050@DLPN_OutDPResellerPOST say? |
| 21:30.41 | iksik | There is no existence of 7050@DLPN_OutDPResellerPOST extension |
| 21:30.58 | p3nguin | I believe it when it says that. |
| 21:31.28 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 21:31.29 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:31.36 | iksik | yeah, like before he says that no one is connected... there was just missing rtcache-something in sip.conf to make sip show peers works |
| 21:31.49 | iksik | mabe here is some simillar problem? :| |
| 21:33.54 | amazinzay | you said you are trying to match _91XXXX into the extension.... what happend if you put in dialplan show 917050@DLPN_OutDPResellerPOST |
| 21:34.00 | iksik | hm, when I calling it from my extension |
| 21:34.03 | iksik | Executing [0947175060@DLPN_DialPlanIN:1] Goto("SIP/7012-b82053c8", "default,7050,1") in new stack |
| 21:34.03 | iksik | <PROTECTED> |
| 21:34.19 | iksik | Got SIP response 603 "Decline" back from IP |
| 21:34.24 | iksik | it seems to work :| |
| 21:34.54 | amazinzay | are both your users and peers mapped to the same database table? or do you have seperate ones |
| 21:35.08 | iksik | it's one database |
| 21:35.12 | iksik | and one database table |
| 21:35.15 | amazinzay | ok |
| 21:39.39 | amazinzay | does that SIP response happen every time you try? in other words, are you sure it is from this call and not just a response to a SIP notify? |
| 21:40.38 | iksik | hm, wait, I'll try to call my extension from outside |
| 21:40.52 | *** join/#asterisk el_critter (n=critter@200.8.188.225) |
| 21:41.02 | iksik | and it works ;/ |
| 21:42.27 | iksik | ok, i'm to tired today... ;/ |
| 21:42.40 | amazinzay | if you call 0947175060 from your extension again does it give you the same SIP error? |
| 21:45.14 | amazinzay | My queues aren't reporting agent state. I have a queue set, but it always says "(agent not in use)" When I run queue show from the CLI. Anyone know what I might be doing wrong? |
| 21:46.43 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:48.32 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 21:48.48 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:03.34 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 22:07.53 | *** join/#asterisk rj45 (n=rj45@c-68-46-128-93.hsd1.pa.comcast.net) |
| 22:10.37 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 22:11.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:12.03 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
| 22:12.39 | *** join/#asterisk SparFux (n=raoul@e182031040.adsl.alicedsl.de) |
| 22:12.50 | *** part/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
| 22:18.11 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 22:20.30 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:20.45 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:26.56 | *** part/#asterisk amazinzay (n=amazinza@67.108.187.186.ptr.us.xo.net) |
| 22:29.32 | *** join/#asterisk justsomedood (n=justsome@mail.serverplus.com) |
| 22:30.18 | justsomedood | anybody know why in our CDR records, in the dst column we'd occasionly get "sw-XXXX-XXXX" (where the X's are numbers) instead of the extension dialed? |
| 22:35.57 | tgunr | In the docs the example for originate is originate Zap/1/123456 extension 400@greeting but what the hech is the syntax for <tech/data>? What is Zap in there? |
| 22:36.26 | manxpower | Zap, SIP, H323, Skinny, Local, MGCP are all "Tech"s |
| 22:36.53 | manxpower | in your example 1/123456 is the "data" |
| 22:37.22 | tgunr | hmm, so if im testing I presume it is my extension name? |
| 22:37.48 | manxpower | tech is a device, not an extension. |
| 22:38.15 | manxpower | The Local/ channel will let you to "tech only" things with the dialplan. Think of it as sort of a "loopback" interface for phone calls. |
| 22:38.32 | tgunr | ok, will try it with Local |
| 22:39.26 | manxpower | You could originate Local/401@greeting and instead of going off somewhere the call will hit the dialpan at exten => 401,1,Whatever in the [greeting] context. |
| 22:41.47 | tgunr | nothing but help messages, what I am really trying to find is what is being match when I try to originate a call to my SIP provider as everything but my internal extensions are failing with 'extension not found' |
| 22:42.01 | tgunr | match=matched |
| 22:44.24 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 22:46.42 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 22:48.08 | *** join/#asterisk ketema (n=ketema@turtle.ketema.net) |
| 22:50.43 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:01.42 | tgunr | is there anyway to debug what is trying to being matched in the dialplans? |
| 23:02.03 | *** join/#asterisk pzn (n=pzn@187.23.90.124) |
| 23:02.42 | pzn | Hi, How can I set a dialplan for when someone dial "0" it will get one of the availabla lines from sip/101 sip/102 and sip/103 ? |
| 23:14.15 | bmoraca | woooo |
| 23:14.15 | bmoraca | freakin ay |
| 23:16.22 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:16.53 | bmoraca | polycom phones finally upgraded and problem-free! |
| 23:17.49 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-65-176.kgldga.dsl.dynamic.tds.net) |
| 23:19.30 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 23:21.46 | p3nguin | When I hang up my phone after talking to someone, a message pops up on the console: == Spawn extension (phones, <number I called>, 2) exited non-zero on 'SIP/101-00756d40' |
| 23:21.56 | p3nguin | What does the whole non-zero thing indicate? |
| 23:23.00 | *** join/#asterisk PanicMan (i=SS7@122.102.33.80) |
| 23:23.28 | PanicMan | unable to config IAX compression, any helper care to help |
| 23:24.31 | PanicMan | when i use TrunkMTU=1240, its not working :( all packets are tramsitting with 65bytes instead of 1240 |
| 23:25.16 | PanicMan | everyone is sleeping :( |
| 23:25.52 | tgunr | snores loudly |
| 23:27.34 | PanicMan | eheh |
| 23:28.06 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
| 23:30.58 | *** part/#asterisk PanicMan (i=SS7@122.102.33.80) |
| 23:33.58 | ricko73 | steps on the crickets |
| 23:34.25 | p3nguin | Also, rtp.c:786 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.63.41.218 |
| 23:34.47 | p3nguin | That client address is that of my termination provider, but the claim the problem is on my end. |
| 23:34.56 | ricko73 | it's a warning |
| 23:34.56 | p3nguin | they claim, rather |
| 23:35.03 | ricko73 | not an error |
| 23:36.08 | ricko73 | http://lists.digium.com/pipermail/asterisk-users/2003-April/002875.html |
| 23:36.18 | p3nguin | I want it elimitated, since it will eventually fill my logs. |
| 23:36.29 | ricko73 | I want a pony |
| 23:37.06 | p3nguin | I meant the comfort noise thing... I don't care about the non-zero exit, just wondered what it means. |
| 23:37.06 | Kobaz | omg pwnies |
| 23:37.30 | ricko73 | ~google |
| 23:37.30 | infobot | from memory, google is http://lmgtfy.com/?q=google |
| 23:37.37 | Kobaz | OMG PWNIES |
| 23:38.03 | Kobaz | http://teeblog.org/wp-content/uploads/2009/06/omg_pwnies.jpg |
| 23:38.08 | drmessano | http://lmgtfy.com/?q=recursion |
| 23:43.21 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 23:51.07 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 23:51.08 | manxpower | ~answers |
| 23:51.08 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 23:51.36 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 23:59.21 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.104) |
