| 00:25.00 | saxa | hello, anybody knows why my * sends rtp on 5040 port if I have limited the ports in rtp.conf to be from 10000 to 10200 ? |
| 00:25.42 | saxa | I do not hear the audio anymore, but if I dial a number it shows the dialed action on the console |
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| 00:58.35 | p3nguin | seri: I cut off the fixed antenna on my DIR-601 and installed a connector for a screw-on antenna. |
| 01:01.03 | saxa | hi p3nguin |
| 01:02.11 | saxa | do you have any idea on why I see the rtp packets being sent to my phone on 5070 if i have rtp.conf set with a range of 10000-10200 ? |
| 01:02.29 | saxa | Sent RTP packet to 192.168.12.100:5078 (type 03, seq 024185, ts 019520, len 000033) |
| 01:06.46 | saxa | ok I figured it out |
| 01:06.53 | saxa | it was set on my phone |
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| 01:37.45 | Tech_Travis | I'm trying to setup * to dial out a PSTN line with exten => _NXXNXXXXXX,Dial(DAHDI/1/${EXTEN}) but all I get is a busy signal and with verbose set to 10 in the cli I get Using SIP RTP CoS mark 5. Is my exten correct for going out a PSTN line? |
| 01:39.58 | ChannelZ | Maybe |
| 01:40.19 | ChannelZ | What is DAHDI/1 ? |
| 01:43.16 | ChannelZ | oh.. actually your syntax is wrong |
| 01:43.33 | ChannelZ | You're missing the priority |
| 01:43.57 | ChannelZ | exten => [exten],[priority],[application] |
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| 01:47.09 | Tech_Travis | darn IE crashed while updating adobe, now I'm using firefox. |
| 01:47.51 | Tech_Travis | DAHDI/1 is channel 1 on my TDM410P card that the PSTN line is connected to. |
| 01:50.52 | Tech_Travis | I also left out the 1 in the exten => _NXXNXXXXXX,1,Dial..... I posted earlier |
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| 02:11.22 | Tech_Travis | Is DAHDI/1correct for having SIP phones make external calls to a PSTN line? |
| 02:12.03 | WIMPy | If that's the name of an FXO port. |
| 02:12.40 | Tech_Travis | It should be the name of the 1st port on my FXO card. |
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| 03:10.39 | Tech_Travis | when trying to make an outgoing call, regardless of the verbosity in the CLI the only message I get is Using SIP RTP CoS mark 5. Is that a configuration error in the extensions.conf? |
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| 03:22.09 | Sean-Der | [macro-hangupcall-custom] |
| 03:22.21 | Sean-Der | I think is my answer, but I can't get anything to match :( |
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| 03:53.14 | patrickod | is it possible to have res_phoneprov use a URL other than /phoneprov ? |
| 03:53.29 | patrickod | I can't find a suitable config variable anywhere |
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| 04:41.06 | woleium | hola peeps :-) |
| 04:41.31 | woleium | does anyone know of any good reading material for setting up a T1 link using digium hardware? |
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| 04:51.40 | KingDavidNYC | Hi everyone, does anybody knows if there is an asterisk-addons tar for version 10?, I don'see it |
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| 05:47.59 | kaldemar | KingDavidNYC: addons hasn't been a separate package since 1.8. it's all in one tar ball. |
| 05:50.06 | KingDavidNYC | Kaldemar: Thank you, I untared 10 and found out...cool |
| 06:01.55 | KingDavidNYC | kaldemar: what about the dahdi files? |
| 06:03.10 | [TK]D-Fender | still separate |
| 06:04.51 | KingDavidNYC | TK - should I use the same current file that I use for 1.6? |
| 06:06.05 | [TK]D-Fender | Should carry forward |
| 06:06.17 | KingDavidNYC | okay TK, thanks |
| 06:07.05 | [TK]D-Fender | ok, checkout time.... later all |
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| 08:24.33 | arni | hi, i'm getting an INCOMPLETE from my telco provider via PRI, but i dont want to (i.e. cant) dial more on the fly |
| 08:24.39 | arni | so i want to fatalise the error |
| 08:24.45 | arni | how would i do that? |
| 08:24.50 | arni | i'm thinking: |
| 08:25.05 | arni | turn the digit timeout to 0 before dialing and catch it in the invalid handler? |
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| 09:07.36 | arni | a digit timeout of 0 works - except i have no idea if i can handle it anywhere |
| 09:07.50 | arni | doesnt jump to i or t |
| 09:08.08 | wdoekes2 | does it continue after Dial then? |
| 09:08.14 | arni | nope |
| 09:08.25 | arni | goes auto fallthrough with CHANUNAVAIL |
| 09:08.39 | arni | although there would be something behind the dial |
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| 09:30.11 | arni | http://nopaste.voric.com/paste.php?f=jtqygf |
| 09:30.59 | arni | doesnt make much sense to me - telco says 28 incomplete - i trigger timeout, then its CHANUNAVAIL and then the caller gets told 16 normal call clearing |
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| 10:15.22 | itguru | I have 2 internet lines and and asterisk instance in the cloud, on one connection when the phone rings, it can be answered. On the other connection, handsets ring, but when they get picked up, they get cut off - I know it must be down to my internet connection, but I'm having a hard time figuring what, both lines are configured the same |
| 10:15.34 | itguru | Can I get a hint as to where I should be looking? |
| 10:16.58 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 10:17.03 | jacc0 | good morning all! |
| 10:17.05 | jacc0 | :) |
| 10:17.12 | saxa | morning |
| 10:17.33 | jacc0 | asterisk-1.8.9.0 fails to make with -j9 on debian |
| 10:19.16 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 10:25.18 | *** join/#asterisk master_of_master (~master_of@p57B556A2.dip.t-dialin.net) |
| 10:25.33 | wdoekes2 | jacc0: -j is not supported. is fixed in trunk only. |
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| 10:49.25 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 10:49.56 | nicola_pav | hello. I want to enable dahdi_dummy with asterisk. I installed everything but when i run ldd /usr/lib/asterisk/modules/chan_dahdi.so |
| 10:50.07 | nicola_pav | i get: libpri.so.1.4 => not found |
| 10:50.15 | nicola_pav | while its found |
| 10:50.18 | nicola_pav | any hints please? |
| 10:51.14 | nicola_pav | locate libpri.so.1.4 will give me 2 paths: /usr/lib/libpri.so.1.4 and /usr/src/libpri-1.4.11.2/libpri.so.1.4 |
| 10:52.44 | wdoekes2 | ldconfig? |
| 10:53.40 | nicola_pav | wdoekes2: u r the man |
| 10:53.46 | nicola_pav | thanks a lottt :) |
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| 11:50.36 | tech | good morning, anyone around? |
| 11:52.14 | *** part/#asterisk cstachris (~chrismylo@202.182.147.104) |
| 11:59.45 | jacc0 | sure |
| 12:00.27 | tech | so i start system up. |
| 12:00.33 | tech | I have avaya <-> asterisk |
| 12:00.39 | tech | trunk group 323 and SIP phones. |
| 12:00.44 | tech | so i can make one call. |
| 12:00.56 | tech | after that the system will not allow another call. |
| 12:01.02 | tech | because it ends up using all resources. |
| 12:01.07 | tech | because it cannot detory the call. |
| 12:01.08 | tech | http://pastebin.com/hWD5V7Pa |
| 12:02.52 | jacc0 | "after that the system will not allow another call." i don't see anthing like that in your pastebin |
| 12:03.02 | jacc0 | what shoudl we see in your pastebin? |
| 12:03.36 | tech | oh.. that is the end of a phone call. |
| 12:03.44 | tech | so i make call - all goes wel.. then i hang up. |
| 12:03.49 | tech | you are seeing the hangup protion. |
| 12:03.55 | tech | but then it just keeps repeating.. |
| 12:04.00 | kaldemar | that is not a hangup. |
| 12:04.23 | kaldemar | that is a response to an OPTIONS message that is used as a keepalive/monitoring message in SIP. |
| 12:04.28 | jacc0 | why don't you post debug of where it fails? |
| 12:04.42 | tech | you are 100% correct i sent you wrong link. |
| 12:05.04 | tech | http://pastebin.com/9vFyDTMB |
| 12:05.12 | tech | that is ooh323 debug on the end of the call. |
| 12:05.25 | tech | juyst loops that command incrmenting the number after the call hangup. |
| 12:05.29 | jacc0 | wft? |
| 12:05.54 | jacc0 | just past the cli output of where it fails plz!! |
| 12:06.12 | tech | ok i will paste bin the whoel call then. |
| 12:09.14 | tech | http://pastebin.com/yj0m0EAh |
| 12:09.18 | tech | i think i see the issue. |
| 12:09.39 | tech | <PROTECTED> |
| 12:10.26 | tech | setting this publci.. |
| 12:10.34 | *** join/#asterisk n3lThon (c189effb@gateway/web/freenode/ip.193.137.239.251) |
| 12:10.45 | *** join/#asterisk elzid (~khaledhus@80.168.5.31) |
| 12:11.03 | n3lThon | hi |
| 12:11.23 | elzid | hello all - anyone know how to access a variable set in the s context from the h context after calling hangup? |
| 12:11.33 | elzid | thanks in advance |
| 12:12.04 | n3lThon | wich centos version (5 or 6) is used by latest asterisk now? |
| 12:12.22 | jacc0 | @elzid: isn't it accessable under the same name? |
| 12:12.38 | elzid | i cant seem to access it |
| 12:12.40 | elzid | for example |
| 12:12.41 | kaldemar | s and h are not contexts, they are extensions. channel variables set in an extension are accessed the same way in other extensions, they are tied to a channel as the name suggests. |
| 12:12.54 | kaldemar | elzid: show what you have. use pastebin. |
| 12:13.05 | jacc0 | ) |
| 12:13.07 | jacc0 | :) |
| 12:13.12 | jacc0 | ~pb |
| 12:13.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 12:13.23 | elzid | @jacc0 - ok |
| 12:16.15 | elzid | @jacc0: http://pastebin.com/wxR6QmGC |
| 12:16.42 | elzid | this is flawed also because when someone hangs up i dont record duration var |
| 12:17.27 | jacc0 | okay, [log] is a different context |
| 12:17.33 | kaldemar | you don't have a billsec until the call ends. |
| 12:17.39 | jacc0 | so the channel vars ar not available |
| 12:17.46 | kaldemar | jacc0: no. |
| 12:18.03 | jacc0 | okay |
| 12:18.05 | kaldemar | log being a different context has nothing to do with variable scope. |
| 12:18.14 | kaldemar | they are channel variables. contexts are irrelevant. |
| 12:18.23 | jacc0 | okay |
| 12:18.31 | elzid | @ kaldemar - I see so need to use CDR(duration)? |
| 12:18.53 | tech | tried a different soft phone and I got a different error... |
| 12:18.55 | tech | ERROR[13069]: chan_ooh323.c:1497 ooh323_set_write_format: No owner found |
| 12:19.00 | kaldemar | elzid: if you look at the CLI when you make a call, you'll see that ${CDR(billsec)} is actually empty. your problem is not with variable access. |
| 12:20.33 | elzid | @ kaldemar - I need to solve the problem where sometimes, if I specify CDR(billsec) in the log context then the duration of the call increases significantly and therefore doesnt reflect the true call duration as experienced by the caller |
| 12:20.56 | elzid | I have experienced this but perhaps when using CDR(duration) - should this behaviour happen if using CDR(billsec)? |
| 12:21.00 | elzid | ie the problem |
| 12:21.22 | kaldemar | ahem. change exten => h,n,Goto(log,h,1) to exten => h,1,Goto(log,h,1) and try again. |
| 12:21.58 | elzid | @ kaldemar - sorry that was a typo |
| 12:22.07 | elzid | @ kaldemar - am doing it like that already |
| 12:22.47 | kaldemar | show a CLI output of a call. |
| 12:22.59 | elzid | ok |
| 12:23.03 | elzid | brb |
| 12:28.12 | elzid | @ kaldemar - I think the problem is that obviously if the caller terminates then hangup is executed and duration is not set, therefore no duration - but why would execution of the h context sometimes be delayed and thereby increasing the call duration even though the caller has hung up already? |
| 12:28.45 | elzid | does CDR(billsec) continue to increase when logic is within the h extension? |
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| 12:35.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:36.50 | elzid | guys - is there any harm in setting endbeforehexten in cdr.conf? |
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| 12:43.32 | kaldemar | elzid: what's the point in setting the duration to a variable after the dial anyway? you should be doing that in the hangup extension, if at all. you could give the CDR field directly to your AGI instead of putting it to a variable. |
| 12:44.09 | kaldemar | and i've never seen execution of h get delayed. |
| 12:45.13 | elzid | @kaldemar - yes - I was doing this to begin with but the problem was it seems that the h extension is de-prioritised and does not complete immediately after hangup of the s extension - that coupled with the fact that cdr duration still increases until h extension completes meant that we say much higher duration CDRs when infact the calls were shorter |
| 12:45.48 | elzid | but im just trying the cdr.conf tweak as mentioned above - supposedly that stops cdr billsec ticking and makes it accessible in the h extension |
| 12:46.08 | elzid | @kaldemar - can you think of any reason y I shouldnt do that from your experience? |
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| 12:48.59 | kaldemar | elzid: your system, your call. |
| 12:51.27 | elzid | @kaldemar - ok - ill give it a shot and watch how it does - thanks for your help mate |
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| 13:06.05 | jacc0 | it's snowing! :) |
| 13:08.24 | leifmadsen | jacc0: finished snowing yesterday here :) |
| 13:08.39 | jacc0 | :) |
| 13:08.46 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 13:09.06 | elzid | @jacc0 - where on earth?! |
| 13:13.47 | jacc0 | the netherlands |
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| 13:17.14 | *** join/#asterisk Jimmy890 (~chatzilla@92.60.238.0) |
| 13:22.00 | Jimmy890 | hello, I have OpenVOX a400p with asterisk 1.8.8.2 under FreeBSD 8.2, when I'm in the CLI promt command dahdi show channels doesn't show anything |
| 13:22.15 | Jimmy890 | prompt |
| 13:22.58 | Jimmy890 | any idea how to solve this problem? |
| 13:25.02 | Jimmy890 | I'm using setup described here : http://wiki.openvox.cn/index.php/OpenVox_A400P_User_Manual_for_Asterisk_1.8_on_Dahdi |
| 13:25.48 | kaldemar | Jimmy890: did you configure dahdi and chan_dahdi? if so, pastebin your configs. |
| 13:27.23 | Jimmy890 | moment, please |
| 13:29.15 | Jimmy890 | http://pastebin.com/WBT66XuY |
| 13:30.06 | Jimmy890 | I have one 2 FXO modules, 1 FXS module |
| 13:33.34 | Jimmy890 | http://pastebin.com/adn7Tuuv |
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| 13:45.37 | kaldemar | Jimmy890: and dahdi? /etc/dahdi/system.conf |
| 13:46.00 | Jimmy890 | moment please |
| 13:46.27 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:47.46 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
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| 13:51.42 | Jimmy890 | http://pastebin.com/py7iqci7 |
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| 13:52.45 | kaldemar | Jimmy890: are dahdi modules loaded? |
| 13:53.06 | Jimmy890 | I think so |
| 13:53.11 | Jimmy890 | wait sec |
| 13:57.33 | Jimmy890 | http://pastebin.com/asUaGKCq |
| 13:57.45 | Jimmy890 | it's not much informative |
| 13:57.58 | Jimmy890 | but I can't redirect from console text to file |
| 13:58.20 | Jimmy890 | dont have screen utility |
| 13:59.36 | *** join/#asterisk jsarrel (~jsarrel@66-169-32-12.dhcp.gnvl.sc.charter.com) |
| 14:00.09 | kaldemar | Jimmy890: lsmod | egrep "dahdi|wctdm" |
| 14:00.41 | kaldemar | should list both dahdi and wctdm. /proc/dahdi/1 should contain your channels along with signaling configs. |
| 14:00.53 | *** join/#asterisk Rene (~Adium@cs181081047.pp.htv.fi) |
| 14:02.24 | Jimmy890 | just a moment |
| 14:03.59 | Jimmy890 | don't have lsmod |
| 14:04.06 | Jimmy890 | FreeBSD 8.2 |
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| 14:07.26 | Jimmy890 | but i think modules are loaded, dahdi show status shows Wildcard TDM400P REV E/F Board 1 |
| 14:07.40 | Jimmy890 | inside asterisk promt off course |
| 14:11.33 | Jimmy890 | kldstat is equivalent of lsmod |
| 14:12.18 | Jimmy890 | http://pastebin.com/tZN169mh |
| 14:14.15 | kaldemar | what output do you get in asterisk when you load the module? "core set verbose 10", "module unload chan_dahdi.so", "module load chan_dahdi.so" |
| 14:14.30 | Jimmy890 | http://pastebin.com/YU1a3nvg |
| 14:14.44 | Jimmy890 | just a moment |
| 14:21.29 | Jimmy890 | http://pastebin.com/ZPwFpDeM |
| 14:22.22 | [TK]D-Fender | Jan 30 14:07:20] ERROR[1276]: chan_dahdi.c:17939 setup_dahdi_int: File chan_dahdi.conf cannot be parsed. |
| 14:22.30 | Jimmy890 | :D |
| 14:22.31 | [TK]D-Fender | Looks like a very broken DAHDI config |
| 14:22.42 | Jimmy890 | I'm newbie |
| 14:23.07 | Jimmy890 | I just followe setup on wiki |
| 14:23.11 | Jimmy890 | followed |
| 14:23.16 | [TK]D-Fender | And running FreePBX on a BSD system. You're doing yourself NO favours... |
| 14:23.33 | Jimmy890 | :) |
| 14:23.54 | WIMPy | Does anyone have an idea, what DAHDIDNDon and DAHDIDNDoff do? |
| 14:24.09 | [TK]D-Fender | WIMPy, DND on FXS IIRC |
| 14:24.19 | kaldemar | Jimmy890: start by removing this line from chan_dahdi.conf: "#include dahdi_additional.conf" |
| 14:24.32 | Jimmy890 | kaldemar: sure |
| 14:24.41 | Jimmy890 | moment please |
| 14:25.20 | WIMPy | [TK]D-Fender: What does that mean? It's your dialplan that decides to send a call there, so what sense does a application to disable it make? |
| 14:25.33 | kaldemar | WIMPy: the same thing as the vertical service codes *78 and *79 do, i'd assume. |
| 14:25.52 | [TK]D-Fender | WIMPy, To force it I guess |
| 14:26.16 | kaldemar | afaik DAHDI DND sits between dialplan and the phone. |
| 14:29.53 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 14:29.58 | _Raptor_ | hi |
| 14:30.18 | _Raptor_ | what is the best way to strip all non-digits from a variable? |
| 14:30.40 | _Raptor_ | i want to remove the + and the - from a number like +49-123-12342342 |
| 14:32.09 | Jimmy890 | I'm sorry i have to go out, I'll be back |
| 14:32.22 | leifmadsen | _Raptor_: FILTER() function |
| 14:32.31 | Jimmy890 | thanks Kaldemar and TKD-Fender |
| 14:32.47 | leifmadsen | Set(Result=${FILTER(0-9,${thisValue})}) |
| 14:33.45 | _Raptor_ | leifmadsen: thx, works fine! |
| 14:37.11 | *** join/#asterisk Cadey (5a980315@gateway/web/freenode/ip.90.152.3.21) |
| 14:37.51 | Cadey | Hi guys, can someone with some PRI knowledge take a look at this please. My spans where find but now span 1's D-Channel keeps hopping up and down |
| 14:37.52 | Cadey | http://pastebin.com/SV0ZfR1u |
| 14:38.20 | Cadey | I have made sure the signaling is pri_cpe as the provider is set to nat |
| 14:46.13 | Cadey | its ok sorted it |
| 14:47.31 | tech | hey fellas, i have a ooh323 setup configured between my asterisk and avaya system. i am able to make calls from asterisk to avaya without issue but cannot make calls from avaya to asterisk. |
| 14:47.39 | tech | any suggestions on where to start looking? |
| 14:47.54 | tech | debug shows nothign on asterisk side which would indicate nothing is getting over there... |
| 14:48.05 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 14:48.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:48.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:48.32 | WIMPy | Cadey: So what was it? |
| 14:48.53 | tech | ok Wow this just even stranger!! |
| 14:49.13 | tech | now i can make "1" call from avaya to asterisk! |
| 14:49.40 | tech | after the first call i cannot make any "more" calls. would indicate a lack of clean hang up.. |
| 14:49.43 | tech | i need the gurus.. |
| 14:50.52 | Cadey | one of the spans was still set to pri_nat <-- not sure why but the server seemed to swap from using the dhadi-channels.conf to the FreePBX Dhadi's module |
| 14:51.06 | Cadey | but prior to the other day it was using the channels conf file |
| 14:51.27 | WIMPy | net, not nat. |
| 14:51.39 | Cadey | sorry :) |
| 14:51.44 | WIMPy | That usually gives a very clear warning, however. |
| 14:52.16 | Cadey | yeah in the full log |
| 14:52.24 | Cadey | it did say both of us think we are the server |
| 14:52.32 | Cadey | which indicates both are net |
| 14:52.43 | Cadey | however I was looking in the file and it was set to cpe |
| 14:52.48 | Cadey | so I was at a loss :) |
| 14:54.19 | [TK]D-Fender | Cadey, " pri_nat" <- so such thing |
| 14:54.33 | Cadey | it was a typo |
| 14:54.37 | Cadey | it was supposed to be net |
| 14:56.05 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 14:58.23 | Jimmy890 | Kaldemar: Thanks for pointing me in right direction, now it seems that 'dahdi show status' is showing right information |
| 14:59.06 | Jimmy890 | and dahdi show channels shows info |
| 15:00.17 | Jimmy890 | http://pastebin.com/27bMYWGj |
| 15:11.04 | Jimmy890 | And Thanks to TKD-Fender |
| 15:11.06 | Jimmy890 | :) |
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| 15:45.14 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 15:45.20 | DelphiWorld | hi |
| 15:45.21 | DelphiWorld | hi pabelanger |
| 15:45.30 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:45.40 | DelphiWorld | pabelanger: want to provision your poly? http://www.provisioner.net/ |
| 15:45.56 | pabelanger | DelphiWorld: yes |
| 15:46.09 | DelphiWorld | pabelanger: :) |
| 15:46.12 | pabelanger | before I started writing my own |
| 15:46.13 | DelphiWorld | pabelanger: see twitter:P |
| 15:46.40 | pabelanger | cool, I'll check it out |
| 15:46.49 | DelphiWorld | pabelanger: enjoi |
| 15:47.33 | DelphiWorld | pabelanger: btw this service is done by tm1000 |
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| 16:22.12 | pigpen | anybody having any luck with T.38 with a Telco (analog)---Asterisk--(sip)--Audiocodes (FXS) ----Fax Machine type setup? |
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| 17:10.20 | mazpe | I'm trying to provision a Polycom 650 with sip 2345-12600-001.sip.ld .. but for some reason from the xxx-boot.log i get "Prov|File 2345-12600-001.sip.ld was not compatible with the phone" |
| 17:10.31 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
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| 17:14.34 | tm1000 | mazpe: what application version are you trying to use |
| 17:14.43 | tm1000 | 4.0.1? |
| 17:15.31 | mazpe | tm1000: where is the actual log file: http://nopaste.info/7d0a2dfdf4.html |
| 17:15.53 | mazpe | tm1000: yes 4.0.1 |
| 17:15.54 | tm1000 | mazpe: I just need to know the version you are trying to update too. The log file will not tell me that |
| 17:16.12 | tm1000 | You have to update the bootrom to 5.0.1 |
| 17:16.14 | tm1000 | BEFORE |
| 17:17.37 | tm1000 | http://www.provisioner.net/repo_firmwares_beta/polycom/spipm/firmware/2345-12600-001.bootrom.ld |
| 17:17.40 | mazpe | 5.0.1? I only see SoundPoint IP and SoundStation IP BootROM 4.3.1 on their site |
| 17:17.43 | mazpe | ok |
| 17:17.48 | tm1000 | http://www.provisioner.net/repo_firmwares_beta/polycom/spipm/firmware/bootrom.ld |
| 17:17.57 | tm1000 | mazpe: it's beta |
| 17:17.59 | tm1000 | you wont see it |
| 17:18.22 | tm1000 | i just sent you the two links to update the bootrom |
| 17:18.27 | mazpe | downloading |
| 17:19.13 | leifmadsen | ya there is an upgrade procedure to get to 4.x. I just avoided it for now and am using 3.x still |
| 17:19.17 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 17:19.58 | mazpe | is it worth then going 4.0.1? |
| 17:20.10 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 17:21.41 | tm1000 | mazpe: yes i think so. but as leifmadsen it is sort of a chore |
| 17:22.34 | leifmadsen | has no idea if its worth it since he hasn't bothered |
| 17:22.46 | leifmadsen | look at the feature set and changes files, and see if it is |
| 17:22.59 | leifmadsen | it totally depends on what you need to do, and what the new software provides you |
| 17:24.05 | mazpe | I need it to make phone calls! lol |
| 17:24.10 | mazpe | very simple setup |
| 17:24.26 | tm1000 | 4.0.1 eliminates the lengthy boot process and updates the gui |
| 17:24.32 | mazpe | ah ok. |
| 17:24.39 | mazpe | lets give it a shot.. why not! |
| 17:25.06 | mazpe | looks nifty already |
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| 17:30.33 | tm1000 | mazpe: did the bootrom work for you? |
| 17:32.23 | mazpe | yes, it did.. thank you.. |
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| 17:33.51 | mazpe | nice gui |
| 17:35.28 | woleium | mazpe: I had a similar problem. They changed the naming scheme. you have to use the download link "SoundPoint IP, SoundStation IP and Polycom VVX Upgrader 4.4.0B" to get the new bootrom |
| 17:35.45 | woleium | oh, i see you got it working :-) |
| 17:36.19 | tm1000 | woleium: yup 4.4.0B == 5.0.1 |
| 17:36.21 | tm1000 | its lame |
| 17:36.31 | mazpe | yup |
| 17:36.56 | woleium | I'm looking for some support setting up a T1 - can anyone recommend someone that can provide remote support? |
| 17:38.09 | *** join/#asterisk The-Kernel (~Thomas@unaffiliated/the-kernel) |
| 17:38.58 | *** join/#asterisk Hanumaan (~Hanumaan@remote191-114.home.uni-freiburg.de) |
| 17:41.34 | The-Kernel | I can't seem to figure out why the astrisk is sending back a 403...can someone take a look please? http://pastebin.com/dNvLLgsa |
| 17:42.42 | file | Found peer 'vitel-outbound' |
| 17:42.45 | Qwell | The-Kernel: inbound != outbound |
| 17:43.46 | *** join/#asterisk twanny796 (~twanny@46.11.4.100) |
| 17:43.52 | [TK]D-Fender | Friend = dead. |
| 17:44.53 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v022-248.mobile.uci.edu) |
| 17:45.11 | The-Kernel | Qwell, I see that, I'm not sure what it means |
| 17:45.11 | [TK]D-Fender | And Vitelity is calling you.. from your own class C? with User-agent: Asterisk PBX <-? |
| 17:45.32 | The-Kernel | [TK]D-Fender, the SIP is being ALG'd |
| 17:46.20 | [TK]D-Fender | The-Kernel, What it means is you aren't looking at the peer it did match |
| 17:49.02 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-shocmntrvnurgyru) |
| 17:50.27 | Jimmy890 | How is sangoma A200 cards working with Asterisk? |
| 17:50.38 | [TK]D-Fender | fine |
| 17:51.09 | Jimmy890 | I\m considering to sell OpenVOX and buy sangoma A200 :) |
| 17:51.16 | Jimmy890 | OpenVox a400p |
| 17:51.41 | [TK]D-Fender | Jimmy890, What are your actual requirements? |
| 17:52.09 | Jimmy890 | hmmm, i just started to play with Asterisk |
| 17:52.36 | *** join/#asterisk alexscott (~alexscott@2a01:e35:8b11:2e40:223:6cff:fe84:b66c) |
| 17:52.49 | Jimmy890 | i like quality |
| 17:52.57 | Jimmy890 | products |
| 17:53.20 | Jimmy890 | a400p won;'t even reply with echo test |
| 17:53.29 | Jimmy890 | only first word |
| 17:53.50 | Jimmy890 | maybe I'm just new to this :D |
| 17:58.36 | The-Kernel | they have to be the same? Inbound and outbound> |
| 17:58.37 | The-Kernel | ? |
| 18:01.32 | The-Kernel | [TK]D-Fender, you mean the config isn't looking at the peer it matched? |
| 18:06.11 | *** join/#asterisk chasing`Sol (~cS@197.132.85.222) |
| 18:06.14 | [TK]D-Fender | The-Kernel, No I mean you showed us a freind entry from your config which is not at all what the call matched. |
| 18:07.49 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-fyzazcrvxdtwhrim) |
| 18:08.28 | The-Kernel | I switched it to peer and its the same thing |
| 18:08.55 | [TK]D-Fender | The-Kernel, the "type" isn't the problem.... |
| 18:09.21 | [TK]D-Fender | The-Kernel, it matched something completely different that what you showed us. |
| 18:09.30 | [TK]D-Fender | than* |
| 18:11.32 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 18:12.32 | The-Kernel | [TK]D-Fender, there's two peers, it match the outbound when it should've matched the inbound...what I'm trying to figure out is why? |
| 18:13.04 | [TK]D-Fender | The-Kernel, You aren't even looking at the one it did match. Why do you keep asking "why"? |
| 18:13.44 | The-Kernel | [TK]D-Fender, because when you say "looking at" you lack explination |
| 18:14.19 | *** join/#asterisk alexoc (~alexoc@201.82.173.134) |
| 18:14.45 | [TK]D-Fender | The-Kernel, You gave us a pastebin of the one it didn't match. Not the one it did. You don't seem to want to show us the whole picture. |
| 18:15.35 | The-Kernel | because its SUPPOSED to match the one I showed you |
| 18:15.58 | The-Kernel | the one it matched it identical except for a name change |
| 18:16.40 | [TK]D-Fender | The-Kernel, show us all of it. |
| 18:18.15 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:18.31 | The-Kernel | sure |
| 18:18.33 | The-Kernel | hold on |
| 18:19.26 | *** join/#asterisk apatheticsheep (~ray@67.220.154.81) |
| 18:20.03 | apatheticsheep | howdy, so ... any reccomendations on sip trunking providers? |
| 18:20.38 | Qwell | ~itsplist-us |
| 18:20.39 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:20.54 | apatheticsheep | ty bot |
| 18:20.59 | apatheticsheep | and ty Qwell |
| 18:21.04 | Qwell | ty human |
| 18:22.04 | *** join/#asterisk docid (~PISSS@69.17.142.138) |
| 18:23.32 | docid | hello, how can i force a specific SIP response code be returned on hangup? ive tried PRI_CAUSE variable, as well as various codes in the Hangup() string, and it always replies 603, it needs to reply 503, how do i force this? |
| 18:23.59 | Qwell | 503 is definitely not the correct response to a hangup |
| 18:24.20 | docid | its the response i need it to send when the far side refuses the call |
| 18:24.33 | eZz | perhaps to cisco |
| 18:25.00 | docid | Asterisk 1.4, 1 pri card, caller from another sip system needs to receive back a 503 when the local pri refuses the call |
| 18:25.19 | docid | so their system keeps searching for a valid path |
| 18:26.35 | docid | and i do not have the option of altering how thier system runs |
| 18:26.49 | docid | so need to make asterisk send them the responses they want |
| 18:27.01 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 18:30.17 | Qwell | docid: You'll have to patch chan_sip then. |
| 18:30.33 | docid | so fully replace 603 with 503? |
| 18:30.44 | Qwell | no |
| 18:30.52 | docid | no way to manipulate sip response codes from the dialplan |
| 18:30.57 | docid | ? |
| 18:31.00 | docid | ok.. |
| 18:31.05 | Qwell | Why would there be? |
| 18:31.19 | Qwell | There are codes you are supposed to send in certain situations, which is what Asterisk does. |
| 18:32.42 | docid | ok, so is this a patch ill have to figgure out myself, or has somebody done this? |
| 18:34.51 | docid | ok, i think i got it, chan_sip.c doesnt seem to convoluted |
| 18:35.05 | Qwell | I am quoting you on that. |
| 18:35.09 | Katty | hello. i am not dave. |
| 18:35.13 | eZz | you can hire me and I will write this patch ;-) |
| 18:35.24 | docid | lol |
| 18:35.36 | Katty | eZz: Qwell will hire me first. |
| 18:35.44 | eZz | Katty: ;) |
| 18:35.48 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 18:35.50 | Qwell | Katty: but not for the same stuff |
| 18:36.05 | Katty | oh i didn't say it'd be for the patch.... |
| 18:36.17 | Katty | i just said i'd get hired first. |
| 18:41.50 | Katty | Qwell: did i tell you i have a new stalker now? |
| 18:41.58 | Qwell | uh oh |
| 18:42.11 | Katty | it's quite amusing. |
| 18:43.08 | Katty | he's just a baby. |
| 18:44.27 | *** join/#asterisk Rene (~Adium@cs181081047.pp.htv.fi) |
| 18:44.44 | leifmadsen | the dingo ate your baby |
| 18:45.09 | Katty | yes, i imagine riddick would maul him rather well. |
| 18:47.11 | Qwell | stalks Katty |
| 18:50.23 | Katty | Qwell: celery. |
| 18:50.31 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 18:51.42 | docid | hey, thanks for the lead Qwell, that was much easier than i expected |
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| 19:49.01 | alexoc | someone has installed successfully asterisk in vmware? |
| 19:52.56 | _Corey_ | alexoc: Many have done so |
| 19:55.09 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-wllnhrsevfdtsymc) |
| 19:56.37 | alexoc | _Corey_: Thanks, I need to decide whether to install vmware asterisk or not, we will 800 peers with possible peak of 200 simultaneous connections connections. No card will not be used. I'm doing some tests with SIPp, the results was satisfied, there is something that I consider time to virtualize the asterisk? any tips? |
| 19:57.52 | _Corey_ | alexoc: You'll find a number of blogs and articles on it if you search a bit. It's a lot easier than it used to be though... |
| 19:58.50 | _Corey_ | alexoc: I have one deployment that's about 600 users deployed on VMWare ESXi that works very well (if that helps) |
| 19:59.49 | alexoc | _Corey_: oh sure! Thanks!! 600 ligacoes simultaneas? |
| 20:00.00 | alexoc | _Corey_: oh sure! Thanks!! 600 simultaneous calls? |
| 20:00.14 | _Corey_ | 600 *users* |
| 20:00.47 | _Corey_ | simultaneous calls is lower than what you're projection is, though I don't know offhand how many |
| 20:02.04 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 20:02.11 | alexoc | _Corey_: ok.. I'm reading a lot about the issue, some say problems in the clock and drop calls. Do you have any reliable source? |
| 20:02.38 | _Corey_ | alexoc: Not offhand, sorry. |
| 20:02.49 | alexoc | _Corey_: thanks! :) |
| 20:04.37 | _Corey_ | no prob |
| 20:08.47 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-86-17.w86-208.abo.wanadoo.fr) |
| 20:29.06 | p3nguin | _corey_: Do you happen to know if I can make my 7960/7940 phones require the line key to be pressed rather than auto-selecting the last used line key when picking up the handset? I thought that was the "auto line select setting," but it is set to NO and it always uses the last line that was used. I'm using chan_sccp-b with asterisk 1.8. |
| 20:30.08 | _Corey_ | p3nguin: Sorry, it's been a while since I've used them enough. I could probably have told you if you asked me last year... :-( |
| 20:30.18 | p3nguin | :/ |
| 20:31.36 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-pfudusajzrlqljpa) |
| 20:32.12 | p3nguin | Do you know if the Ring Answer stuff works with chan_sccp-b and asterisk? I'd like a short ring from the base rather than a call waiting tone through the handset/headset speaker when a second call arrives while already on the phone. |
| 20:32.31 | p3nguin | I haven't been able to make it work, so I was curious if that is something only Call Manager can do. |
| 20:34.17 | _Corey_ | p3nguin: I used to do it with a second registration that was set to auto-answer with the SIP firmware. Not too pretty. |
| 20:37.38 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 20:37.47 | p3nguin | Maybe if I could find a comprehensive list of all settings available to use in XMLDefault.cnf.xml, I could figure out which settings to use for these things. |
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| 20:47.50 | HerdberH | hi |
| 20:53.38 | kfife | Some Jackass in Beijing is sending registration attempts at my server at about 1mbs. |
| 20:54.09 | kfife | Simply dropping at perimeter, but still chewing up a megabit of bandwidth being sent to the bit bucket |
| 20:54.22 | kfife | Fu¢kers |
| 20:55.09 | kfife | Thoughts? Toll fraud attempt, or are they trying to do some kind of overflow exploit? |
| 20:57.23 | leifmadsen | kfife: could be overflow |
| 20:57.51 | p3nguin | I would have guessed probably a toll fraud attempt. There isn't a lot more you can do in addition to ignoring packets at the firewall. |
| 20:58.22 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:58.24 | leifmadsen | 1mbps registration attempts seems excessive unless it is doing something bruteforce |
| 20:58.28 | leifmadsen | which I suppose is highly lightly |
| 20:58.31 | leifmadsen | likely* |
| 20:58.38 | kfife | I agree with you |
| 20:58.45 | kfife | Also since it's coming from Beijing |
| 20:59.01 | kfife | IF it were Nigerian, that might be a better fit for toll fruad |
| 20:59.06 | kfife | (my spidey sense tells me) |
| 20:59.14 | p3nguin | When I get toll fraud attempts, it is very fast and seems to be from a dictionary. |
| 20:59.29 | kfife | Also, the attack has been going on for a few days. |
| 21:00.03 | p3nguin | You have to ask yourself: Who did I piss off? |
| 21:00.17 | _Corey_ | kfife: I've seen them come from everywhere... Europe, africa, russia, china, and in the US (notably from the EC2 cloud). I've never seen one last for more than a couple hours from a single source IP though. |
| 21:00.27 | kfife | So the theory of operation is that the overflow abends the process, drops you into a shell, allowing priveliged access to the whole local subnet? |
| 21:00.32 | kfife | P3nguin: :-) |
| 21:00.33 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 21:00.55 | kfife | About 3 days. 222.39.89.181 |
| 21:01.13 | FLeiXiuS | How can I get CDR billsec in milliseconds? I'd rather not have to subtract time if CDR can do it for me. |
| 21:03.40 | p3nguin | I thought it was already expressed in milliseconds. |
| 21:04.24 | FLeiXiuS | p3nguin, I don't think it is... I have L() set to 5000, and Billsec reports 5 when L() is executed. |
| 21:04.32 | kfife | p3nguin: Years ago I had a Beijing IP address successfully root a windows host that had a public-facing VNC listener. VNC had a long complex password. Was only facing the net for about 12 hours. I don't think it was brute force. Not enough time. There must have been a vulnerability in VNC Server. Post-mortem forensics showed the root kit to be unbelieveably clever. |
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| 21:05.30 | *** part/#asterisk necromancer (u4538@gateway/web/irccloud.com/x-swjzxdrnveenowba) |
| 21:08.36 | *** join/#asterisk bitwize (bitwize@109.58.116.163.bredband.tre.se) |
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| 21:14.58 | *** join/#asterisk skrull (~skrull@201-34-92-158.smace701.dsl.brasiltelecom.net.br) |
| 21:17.26 | skrull | using a spa3103 with this (9090xxxxxxxxS0|[2-9]xxxxxxxS0) dialplan, when I call 909012345678, it actually calls 90901234, why is that? theres no precedence? |
| 21:17.38 | skrull | spa3102 |
| 21:21.03 | citywok | skrull: you need to share your dialplan, use pastebin |
| 21:22.18 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-iwplmuowghzjjvci) |
| 21:22.35 | skrull | http://pastebin.com/qnjNhFGv |
| 21:22.47 | FLeiXiuS | p3nguin, Any ideas on billsec? |
| 21:23.49 | citywok | skrull: your * dialplan & console output |
| 21:25.41 | *** join/#asterisk jerware (~root@www.kaylon.com) |
| 21:25.44 | jerware | hi |
| 21:25.48 | jerware | I have fax capabilities on my copy machine but no landline. Whats to set up to use fax over VoIP ? |
| 21:25.58 | jerware | err to recieve fax over voip. |
| 21:25.59 | citywok | jerware: T38 |
| 21:27.38 | *** join/#asterisk hdiogenes (~humberto@189.124.219.15) |
| 21:28.21 | Katty | LADIES AND GENTLEMEN OF THE JURY |
| 21:28.25 | Katty | i mean asterisk community. |
| 21:28.29 | Katty | you are awesome. just saying. |
| 21:29.02 | *** join/#asterisk rossand (~aross@206.191.52.37) |
| 21:29.26 | leifmadsen | Katty: thanks! ;) |
| 21:29.39 | citywok | well said |
| 21:30.59 | skrull | here we go http://pastebin.com/SZknd2yF |
| 21:31.39 | skrull | ops |
| 21:32.11 | skrull | here http://pastebin.com/4KJFwSap |
| 21:32.57 | jerware | citywok: herro, is T38 on the pbx. I mean does the printer have to support T38? |
| 21:35.27 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 21:35.38 | citywok | jerware: https://www.google.com/search?rlz=1C1CHKZ_enUS434US434&sourceid=chrome&ie=UTF-8&q=asterisk+fax+t38 |
| 21:36.26 | citywok | skrull: did you read your console output? when you make the call it prints this out: "Saindo por fixo local, |
| 21:36.39 | skrull | yes |
| 21:36.59 | citywok | so it's obviously going through ; Ligacoes locais ; xxxx.xxxx (8 digitos) |
| 21:37.05 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 21:37.24 | skrull | but theres something wrong with my spa3102 fxs dialplan because my phone is still dialing numbers and asterisk is already doing the call |
| 21:37.45 | *** join/#asterisk Goldwing (~Goldwing@84.245.46.83) |
| 21:37.49 | citywok | the problem is with yoru dialplan in asterisk. it's taking the first 8 digits and ignoring the rest |
| 21:38.30 | Goldwing | good evening |
| 21:39.19 | skrull | ok, i will try with only one extension on that context. |
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| 21:46.41 | jerware | you still need a land line connected to a T38 capable VoIP gateway. |
| 21:47.00 | jerware | That's gay the whole idea was a work around for lacking a landline. |
| 21:50.49 | citywok | jerware: lol "gay" is probably the wrong word to use in this case, because you can use T38 to connect to a voip provider that supports T38 and avoid needing a land line. |
| 21:57.15 | jerware | ingenious ! |
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| 21:58.23 | Sean-Der | Does anyone have documentation on IAXVAR? I have seen it mentioned a few times, I just can't seem to turn it up though. I am looking to pass variables with calls between two Asterisk servers via an IAX2 trunk |
| 22:01.13 | navaismo | Sean-Der: core show function IAXVAR |
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| 22:13.25 | Sean-Der | navaismo, it doesn't say how to set or fetch variables though :/ I will just guess and check though |
| 22:17.40 | navaismo | some help in this ticket https://issues.asterisk.org/view.php?id=16654 |
| 22:26.17 | WIMPy | That's old. |
| 22:27.29 | navaismo | never said its fresh example, just for reference |
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| 23:20.46 | skrull | citywok: solved, the problem was with spa dialplan, not *, thanks anyway |
| 23:21.16 | citywok | skrull: interesting, good to know. :) |
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