| 00:02.09 | [TK]D-Fender | Mon|A|rch: I don't see what AMI has to do with anything. As for the 484, the error says it all "incomplete". Your number isn't quite what they were expecting |
| 00:02.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:08.38 | nickgaw | is google voice still free in the USA? |
| 00:15.15 | *** join/#asterisk citywok (~kvirc@208.186.203.146) |
| 00:15.56 | citywok | We're making Internation calls (currently using flowroute) in to Germany, France, and the U.K. -- Does anybody have any suggestions on what we can do to improve the callerid being displayed? Is there an alternative to simply using an ITSP in each country to terminate the calls? |
| 00:17.36 | [TK]D-Fender | What is "not good" about what you're doing now? |
| 00:22.25 | WIMPy | Yes, what does "impove" mean? |
| 00:24.11 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-160-65.red.bezeqint.net) |
| 00:37.09 | ChannelZ | His CID currently comes up as a phone sex number. |
| 00:37.41 | WIMPy | Not a baad thing if he owns that number. |
| 00:39.30 | *** part/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 00:41.52 | *** join/#asterisk Gunchedorn (~gunchedor@189.169.89.140) |
| 00:42.05 | citywok | sorry, got distracted. in many cases the callerid simply doesn't go through |
| 00:42.10 | Gunchedorn | hi |
| 00:42.13 | Gunchedorn | need tome help |
| 00:42.14 | citywok | they don't receive anything |
| 00:42.45 | WIMPy | get a local ITSP. |
| 00:42.52 | Gunchedorn | my asterisk started dying and really do not know why. tried to restore backup and now its worst |
| 00:43.15 | Gunchedorn | first, when I issue service asterisk start (elastix) it complains about |
| 00:43.24 | Gunchedorn | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 00:43.44 | Gunchedorn | but asterisk start and asterisk -vvr connects |
| 00:43.54 | Gunchedorn | what are the possible reasons? |
| 00:44.06 | Gunchedorn | oh, it happens at this line |
| 00:44.07 | Gunchedorn | g0 |
| 00:44.09 | Gunchedorn | + /bin/bash -c 'ulimit -S -c 0 >/dev/null 2>&1 ; /usr/sbin/safe_asterisk -U asterisk -G asterisk' |
| 00:44.10 | [TK]D-Fender | I'd be checking who the owner is and who * is running as. |
| 00:44.31 | [TK]D-Fender | Check the files <- |
| 00:44.35 | Gunchedorn | thanks [TK]D-Fender... its the service asterisk start command |
| 00:45.18 | [TK]D-Fender | No... check the actual owners ont he files themselves. |
| 00:45.21 | [TK]D-Fender | configs, et |
| 00:46.44 | citywok | WIMPy: yea, that's what i was afraid of. i'm afraid of the call quality riding the internet all the way from the U.S. to germany |
| 00:47.55 | WIMPy | Well, the internet was not designed for realtime applications. But that's true for all continents. |
| 00:48.02 | Gunchedorn | [TK]D-Fender is there other directory to check apart from /etc/asterisk |
| 00:48.24 | Gunchedorn | I did a chown -R asterisk:asterisk /etc/asterisk already |
| 00:48.32 | [TK]D-Fender | check the PID fil itself, /var/lib /etc/modules, etc |
| 00:48.40 | citywok | yea. depending on needs we may end up getting a private connection from the u.s. to somewhere in the EU and terminate everything from a box there. that way it's quality connection most of the way. |
| 00:49.31 | WIMPy | That might not be the interesting part of the journey. |
| 00:50.21 | citywok | possibly, but my guess is once on the right continent things should be much better. |
| 00:50.50 | WIMPy | I'd expect the intercontinental part to be the smaller issue. |
| 00:51.07 | citywok | really? interesting |
| 00:51.47 | citywok | i wonder if anybody here has any experience with delivering calls to the UK/germany/france from the U.S. and can speak to that. maybe i should try the mailing list. |
| 00:51.48 | Gunchedorn | all files correct |
| 00:51.51 | Gunchedorn | :-( |
| 00:52.12 | citywok | if that's the case i'd probably be kinda screwed. lol. |
| 00:52.21 | WIMPy | There is less competition between intercontinental carriers. |
| 00:53.02 | WIMPy | For uk/de sipgate would be the obvious one. |
| 00:53.07 | WIMPy | No idea about fr. |
| 00:55.16 | Gunchedorn | service asterisk stop goes down correctly |
| 00:55.30 | Gunchedorn | just the service asterisk start fails |
| 00:56.59 | citywok | WIMPy: thx, i'll check with sipgate and see how well their callerid handling works for the places i need |
| 01:00.39 | Gunchedorn | [TK]D-Fender> sorry to continue with this |
| 01:00.52 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 01:01.01 | Gunchedorn | but I see the safe_asterisk asking for the version *before* actually starting asterisk |
| 01:01.14 | Gunchedorn | ++ /usr/sbin/asterisk -nrx 'core show version' |
| 01:01.15 | Gunchedorn | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 01:01.31 | WIMPy | It does so to find out if it's already running. |
| 01:01.40 | Gunchedorn | oh. |
| 01:01.55 | Gunchedorn | so the error message is safely ignored then |
| 01:02.04 | WIMPy | yes |
| 01:02.11 | Gunchedorn | okay! |
| 01:02.18 | Gunchedorn | now that is not part of my issue |
| 01:03.05 | Gunchedorn | when I try to dial out, all I get is this |
| 01:03.06 | Gunchedorn | "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks") |
| 01:03.14 | Gunchedorn | I have a E1 with 20 lines |
| 01:03.20 | Gunchedorn | it was working before |
| 01:04.07 | Gunchedorn | but not anymore. I restored the backup (the one taken by elastix) |
| 01:04.13 | Gunchedorn | but now the issue is worse. |
| 01:04.30 | Gunchedorn | one sympton is core show channels |
| 01:04.47 | Gunchedorn | smpbx*CLI> core show channels |
| 01:04.48 | Gunchedorn | Channel Location State Application(Data) |
| 01:04.48 | Gunchedorn | 0 active channels |
| 01:04.48 | Gunchedorn | 0 active calls |
| 01:04.48 | Gunchedorn | 0 calls processed |
| 01:05.01 | Gunchedorn | smpbx*CLI> dahdi show channels |
| 01:05.01 | Gunchedorn | <PROTECTED> |
| 01:05.01 | Gunchedorn | <PROTECTED> |
| 01:05.16 | Gunchedorn | may I post all them here? |
| 01:05.34 | Gunchedorn | from 22 to 31 is like this |
| 01:05.37 | Gunchedorn | <PROTECTED> |
| 01:06.12 | Gunchedorn | then a TDM410P |
| 01:06.20 | Gunchedorn | <PROTECTED> |
| 01:06.20 | Gunchedorn | <PROTECTED> |
| 01:06.20 | Gunchedorn | <PROTECTED> |
| 01:06.20 | Gunchedorn | <PROTECTED> |
| 01:06.32 | Gunchedorn | asterisk show no channels and does not want to dial out |
| 01:06.35 | [TK]D-Fender | ~pb |
| 01:06.35 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:06.47 | [TK]D-Fender | Gunchedorn: do NOT flood in here like that please.... |
| 01:07.08 | Gunchedorn | sorry, new in irc |
| 01:09.39 | Gunchedorn | great! first time using pastebin |
| 01:09.40 | Gunchedorn | http://pastebin.com/raw.php?i=ieExhdLe |
| 01:15.20 | Gunchedorn | WIMPy, [TK]D-Fender... why is the trunks error? |
| 01:15.22 | Gunchedorn | <PROTECTED> |
| 01:23.34 | [TK]D-Fender | Gunchedorn: you haven't actually shown us the failed call in its entirety |
| 01:23.46 | Gunchedorn | let me pastebin it |
| 01:26.03 | Gunchedorn | here |
| 01:26.04 | Gunchedorn | http://pastebin.com/raw.php?i=XttRqg8P |
| 01:26.40 | [TK]D-Fender | Chan 1 - Far end disconnected. Reason: Network Congestion MFC/R2 call disconnected on channel 1 |
| 01:26.55 | [TK]D-Fender | Some sort of E1 issue |
| 01:27.39 | Gunchedorn | The E1 has 31 lines |
| 01:27.45 | Gunchedorn | we only have 20 lines |
| 01:28.04 | Gunchedorn | is normal to define all channels or should I define only the active ones? |
| 01:28.18 | Gunchedorn | can the issue be asterisk trying to use one of those channels |
| 01:28.35 | Gunchedorn | because calling local, works. calling cell phones, not |
| 01:28.37 | [TK]D-Fender | Gunchedorn: If you only have a fractional E1 then you need to specify those alone |
| 01:28.43 | Gunchedorn | where |
| 01:28.49 | [TK]D-Fender | Gunchedorn: But this was a Channel ! issue |
| 01:28.52 | [TK]D-Fender | 1* |
| 01:29.48 | Gunchedorn | so it can be the telco is blocking cell calls? (the ones starting here with 044) |
| 01:32.10 | [TK]D-Fender | Gunchedorn: Possible. But there is not enough actual debug at the E1 level to get a better idea of that |
| 01:34.33 | Gunchedorn | what should I look for |
| 01:34.51 | Gunchedorn | I guess I can use asterisk -vvvvvvvvvr |
| 01:35.29 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
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| 01:52.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 01:54.28 | rue_house | can anyone help me with a mgcp problem? my second gateway wont work |
| 02:04.31 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
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| 02:07.31 | DaPrivateer | OK, so if anyone can assist ... I'm trying to set up TLS with a Polycom 670 and keep getting this error: Problem setting up ssl connection: error:1409441B:SSL routines:SSL3_READ_BYTES:tlsv1 alert decrypt error |
| 02:26.37 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 02:39.23 | Gunchedorn | Leaving... thanks <[TK]D-Fender> |
| 02:46.14 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 03:09.55 | fling | exten => [TK]D-Fender,1,Hello() |
| 03:13.43 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-023-096.pools.arcor-ip.net) |
| 03:19.22 | *** join/#asterisk cheasee (~cheasee@vie-188-118-246-206.dsl.sil.at) |
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| 03:33.58 | fling | how to run a bash command from a dialplan? |
| 03:34.18 | fling | I want to recode my callrecord recorded with Monitor |
| 03:35.58 | [TK]D-Fender | "core show application system" |
| 03:36.14 | [TK]D-Fender | "core show application monitor" <- better |
| 03:36.20 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 03:43.11 | fling | [TK]D-Fender: right, I'm already monitoring and capturing sound packets into files |
| 03:43.45 | fling | [TK]D-Fender: I want to run a script aftar HangUp() that will recode my files with ffmpeg and flac |
| 03:44.39 | [TK]D-Fender | "core show application monitor" <- better |
| 03:45.26 | fling | [TK]D-Fender: ok! do I need ${MONITOR_EXEC} ? :p |
| 03:45.27 | fling | ok |
| 03:54.43 | fling | [TK]D-Fender: I need to read better :] |
| 03:56.32 | [TK]D-Fender | fling: 11 steps to go! |
| 03:57.18 | fling | [TK]D-Fender: umm? what steps? |
| 03:57.27 | [TK]D-Fender | Keep reading... |
| 03:57.34 | fling | is reading |
| 04:11.56 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 04:17.16 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 04:56.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 05:16.10 | dijib | what up y'all |
| 05:16.28 | dijib | anybody alive and up for a conf? |
| 05:23.32 | ChannelZ | Yes. It's called IRC |
| 05:27.50 | fling | ChannelZ: hey, /usr/local/bin/mixlegfiles > http://bpaste.net/show/72661/ ; /etc/asterisk/extensions.conf > http://bpaste.net/show/72662/ |
| 05:28.07 | fling | ChannelZ: am I doing it wrong? > same => n,Set(MONITOR_EXEC=/usr/locl/bin/mixlegfiles) |
| 05:29.21 | fling | ChannelZ: http://dpaste.com/890606/ |
| 05:29.43 | fling | it is recording but not running MONITOR_EXEC afterwards somewhy |
| 05:30.43 | [TK]D-Fender | <PROTECTED> |
| 05:31.05 | fling | lol! :D |
| 05:31.14 | fling | why no errors in console? |
| 05:31.45 | [TK]D-Fender | I guess it considers trying at all a "bonus" |
| 05:32.29 | fling | still not running |
| 05:32.31 | ChannelZ | it's more fun this way |
| 05:33.15 | [TK]D-Fender | flI am also not getting a good look at your files... |
| 05:33.17 | *** join/#asterisk podz (~podz@xdsl-250-108.nebulazone.fi) |
| 05:33.24 | [TK]D-Fender | And you didn't pastebin the complete call. |
| 05:33.30 | ChannelZ | what is 'mixlegfiles'? |
| 05:33.34 | [TK]D-Fender | You seem to be doing half of each step... |
| 05:33.45 | ChannelZ | oh nm I missed the link |
| 05:34.11 | podz | hi, is asterisk 1.8.14 still supported? |
| 05:35.04 | fling | [TK]D-Fender: ChannelZ: http://dpaste.com/890610/ |
| 05:35.49 | podz | I am using asterisk 1.8.14 with LDAP and the LDAP implementation is _quite_ buggy. I could help to get the code sorted out, probably. |
| 05:36.45 | [TK]D-Fender | fling: -- Executing [s@outgoing:6] Monitor("SIP/000-0000007a", ",outgoing/000/2013/2013.01/2013.01.25/2013.01.25-12:32:02_000_+700900500,m") in new stack <- lack of format is a bad start |
| 05:37.49 | fling | [TK]D-Fender: wav is the default, ok I will set it to wav |
| 05:37.53 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 05:42.23 | *** join/#asterisk din3sh (~din3sh@41.212.202.205) |
| 05:42.24 | ChannelZ | will Monitor even make that whole path of directories on its own? |
| 05:43.07 | fling | ChannelZ: yes, I have waves in the folders |
| 05:43.25 | fling | ChannelZ: and I may mix them manually with the script |
| 05:43.28 | ChannelZ | hmm ok. never tried. |
| 05:43.28 | [TK]D-Fender | still not getting a good look at your files.... |
| 05:46.51 | ChannelZ | hopefully mixlegfiles has +x ? |
| 05:47.58 | fling | [TK]D-Fender: ChannelZ: http://dpaste.com/890614/ |
| 05:49.42 | fling | outgoing/000/2013/2013.01/2013.01.25/2013.01.25-12:45:36_000_+700900500 turned into flac > http://dpaste.com/890615/ |
| 05:54.18 | ChannelZ | well I'd personally start by shoving an 'echo barf > /tmp/barf' as first in the script and see if it's executing or what |
| 05:55.54 | fling | ChannelZ: it is starting! Thanks, now I will debug it :p |
| 05:56.11 | ChannelZ | probably just a path problem. specify the path to ffmpeg and such |
| 05:56.16 | fling | right |
| 06:00.42 | fling | ChannelZ: haha > /var/spool/asterisk/monitor//var/spool/asterisk/monitor/outgoing/000/2013/2013.01/2013.01.25/2013.01.25-12:59:58_000_+700900500-in.wav-out.wav: No such file or directory |
| 06:02.24 | ChannelZ | well at least you're making progress |
| 06:09.15 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 06:17.16 | [TK]D-Fender | fling: ... // <-- |
| 06:18.08 | fling | [TK]D-Fender: yes, looks like I do not need to prefix filename with $spool variable |
| 06:18.10 | [TK]D-Fender | and yeah, the whoe doubled path |
| 06:18.29 | [TK]D-Fender | ok, bedtime, here... I think you'll have it before long... |
| 06:18.33 | [TK]D-Fender | heads out |
| 06:22.11 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 06:24.04 | fling | ChannelZ: found it! :p |
| 06:24.39 | fling | ChannelZ: first second and third args are set by asterisk, MONITOR_EXEC_ARGS is 4th arg :p |
| 06:30.43 | fling | ChannelZ: the working script http://bpaste.net/show/72676/ |
| 06:34.55 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 06:36.05 | hebber | I have checked through the documentation for Realtime SIP friends and I don't find a place to refer to a ACL. Does anyone have some experience with this? |
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| 06:59.41 | hebber | I have checked through the documentation for Realtime SIP friends and I don't find a place to refer to an ACL. Does Realtime SIP Friends support ACL? |
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| 07:02.41 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 07:05.01 | hebber | ah! I found it: SIP or IAX2 peer defined in ARA can include an 'acl' column and list the Named ACLs to apply in that column. |
| 07:11.41 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:26.41 | *** join/#asterisk schmidts (~schmidts@213162068056.public.t-mobile.at) |
| 07:26.44 | schmidts | good morning |
| 07:28.58 | ChannelZ | TGIF |
| 07:41.22 | schmidts | ChannelZ +1 |
| 07:42.53 | *** join/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 07:43.34 | ChannelZ | I'm watching Ruby the swearing parrot videos on YouTube. |
| 07:49.00 | *** join/#asterisk voxter_ (~voxter@d23-16-70-150.bchsia.telus.net) |
| 07:50.14 | *** join/#asterisk din3sh (~din3sh@41.76.47.90) |
| 07:50.36 | din3sh | hello room |
| 07:51.25 | ChannelZ | hello person |
| 07:56.12 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 08:04.38 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 08:06.43 | din3sh | I have a meetme number, pstn numbers calling from a specific country cannot key in DTMF for meetme |
| 08:07.48 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 08:14.16 | ChannelZ | how do the calls get to you? |
| 08:14.27 | ChannelZ | (SIP, PSTN, PRI..) |
| 08:17.08 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 08:21.53 | *** join/#asterisk kontinuity (~kontinuit@122.172.225.46) |
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| 08:26.35 | din3sh | PRI |
| 08:26.39 | din3sh | E1 |
| 08:26.52 | din3sh | local numbers can key in the dtmf pin code for the meetme |
| 08:27.03 | din3sh | some countries also can |
| 08:27.19 | din3sh | numbers from S.Africa seem unable to key in the pin |
| 08:27.27 | din3sh | hence get the ivalid conf pin |
| 08:27.38 | din3sh | what might be the problem? |
| 08:28.46 | srp_ | Hi, I have setup asterisk-11.2.1 on my ubuntu. When I try to register any SIP client to this asterisk server, it says (http://paste.kde.org/656072). Here is my sip.conf (http://paste.kde.org/656078). Can anyone help me with this? |
| 08:29.59 | ChannelZ | din3sh: dunno maybe their phones are jacked up and make slightly off DTMF |
| 08:31.29 | din3sh | i have tried from various phone numbers in S.A, Durban, Jo.burg etc |
| 08:31.45 | din3sh | cant surely be a tone/pulse dialing problem can it |
| 08:32.34 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 08:36.02 | ChannelZ | srp_: your peer is called "test_phone_testing" but you're trying to register as "testuser" |
| 08:41.44 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 08:42.56 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:43.23 | srp_ | ChannelZ: ok.. one sec.. will check that... |
| 08:46.09 | *** join/#asterisk tamiel (~tamiel@c-67-169-76-114.hsd1.ca.comcast.net) |
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| 08:47.54 | srp_ | ChannelZ: thanks :) its working! in one of the other setup i have installed, it says (http://paste.kde.org/656084) and fails to run. can you help with this? |
| 08:48.20 | ChannelZ | did you build from source or is this from a package? |
| 08:48.24 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:48.36 | bombev | good morning guys ;) |
| 08:48.54 | srp_ | ChannelZ: i built it from source. |
| 08:49.06 | ChannelZ | hmm |
| 08:49.25 | ChannelZ | Well unless you're using lua, you can do noload => pbx_lua.so in /etc/asterisk/modules.conf |
| 08:49.47 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 08:50.08 | ChannelZ | If you are trying to use lua, something is screwed up that I'm not sure. Never seen that before.. |
| 08:51.48 | hebber | Hi, my Verbose output is lost in CLI. My logger.conf have this parameter: console => notice,warning,error,dtmf,fax,verbose. Anyone have an idea why they don't show up. Verbose setting is set to even 200. |
| 08:52.23 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 08:53.01 | srp_ | ChannelZ: i did noload as you suggested. It worked.. :) |
| 08:53.46 | ChannelZ | srp_: treating the symptom and not the problem really - you might have some other issue with glibc or something. What OS are you running? |
| 08:53.59 | srp_ | ChannelZ: i |
| 08:54.09 | srp_ | ChannelZ: im on ubuntu 12.04 |
| 08:54.18 | ChannelZ | hmm |
| 08:59.18 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:8440:38c2:ac4b:a0f6) |
| 09:03.07 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 09:03.49 | *** join/#asterisk x4rlos (~karl@217.20.42.14) |
| 09:04.07 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 09:06.37 | *** join/#asterisk stian (~Stian@svr3.st0len.co.za) |
| 09:06.58 | stian | hello everyone |
| 09:07.33 | stian | I've got a quick question, I've set up freepbx to notify users on email when someone leaves a vm - however this suddenly stopped working |
| 09:08.56 | ChannelZ | perhaps your mailer broke |
| 09:09.09 | stian | no it did not, I just left a voicemail on my extension |
| 09:09.14 | stian | and I recieved an email |
| 09:09.25 | ChannelZ | Hurray! Everything is fixed! |
| 09:09.32 | stian | lol |
| 09:09.55 | stian | Thing is, I have our administration extension set to notify an email which is a group email |
| 09:10.08 | srp_ | ChannelZ: i'm calling an extension (2600) from a sip client and I get this error (http://paste.kde.org/656120). Here is my sip.conf (http://paste.kde.org/656114) and extension.conf (http://paste.kde.org/656126). Can you please check? |
| 09:10.16 | WIMPy | Hi din3sh. Did you get the line errors fixed? |
| 09:10.47 | WIMPy | As for DTMF: I'd suspect a misconfigured VOIP transport in the path. |
| 09:10.48 | stian | Maybe the problem actually lies with the email adress, not asterisk |
| 09:10.58 | ChannelZ | srp_: did you install the base sound set? |
| 09:11.41 | ChannelZ | core-sounds |
| 09:12.09 | stian | AH! |
| 09:12.12 | stian | I think I found it |
| 09:12.23 | stian | the email that I created through google apps |
| 09:12.30 | srp_ | ChannelZ: yes. its in /var/lib/asterisk/sounds/en/hello-world.gsm |
| 09:12.42 | stian | which is a group - wasn't allowed to recieve emails from the pbx |
| 09:12.43 | stian | :) |
| 09:13.14 | ChannelZ | srp_: does whatever user your asterisk runs as have permission to read that dir and all the files? |
| 09:13.51 | stian | YES |
| 09:13.52 | stian | fixed it! |
| 09:14.15 | stian | awesome |
| 09:15.56 | srp_ | ChannelZ: yes it has. I'm running in root and i checked it. |
| 09:18.20 | ChannelZ | hmm.. |
| 09:18.43 | ChannelZ | well unless you turned off languageprefix in asterisk.conf or really those files aren't readable, I'm not sure whats going on |
| 09:22.13 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 09:22.31 | ChannelZ | or... what does "core show settings" show for your Data directory? |
| 09:22.35 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 09:23.36 | srp_ | ChannelZ: it shows this --> /var/lib/asterisk |
| 09:24.59 | ChannelZ | ok.. and Default Language as "en" and Language prefix "Enabled"? |
| 09:27.30 | srp_ | ChannelZ: yes! |
| 09:28.01 | ChannelZ | ok then I really think it can't read the files. Did you say YOU were root looking at the files or that Asterisk is running as root? |
| 09:28.48 | ChannelZ | unless your gsm codec is broken too.. on the console if you do "module load codec_gsm" and "module load format_gsm" do you get 'already exists'? |
| 09:28.55 | srp_ | ChannelZ: Asterisk running as root |
| 09:30.42 | srp_ | ChannelZ: thanks so much .. module load format_gsm was not loaded |
| 09:31.46 | ChannelZ | uhh hmmm |
| 09:32.28 | ChannelZ | is autoload turned on in your modules.conf? |
| 09:32.59 | srp_ | ChannelZ: no... |
| 09:34.01 | ChannelZ | do that.. autoload=yes |
| 09:34.12 | srp_ | ChannelZ: okay.. |
| 09:34.21 | ChannelZ | you'll probably keep finding lots of other broken things without it |
| 09:34.45 | srp_ | ChannelZ: thanks for pointing this out :) have changed it now :) |
| 09:35.12 | ChannelZ | restart asterisk for fun. |
| 09:35.17 | ChannelZ | And good luck, I need to head off to bed |
| 09:35.29 | srp_ | ChannelZ: thanks a lot :) good night :) |
| 09:35.36 | ChannelZ | sure |
| 09:42.24 | *** join/#asterisk garymc (~chatzilla@host81-148-120-70.in-addr.btopenworld.com) |
| 09:52.25 | x4rlos | Hi all. Is there a common solution to the error: "chan_dahdi.c:4160 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway!"? |
| 09:52.52 | x4rlos | 1:1.6.2.9-2+squeeze3 |
| 09:53.02 | WIMPy | Plug in the line. |
| 09:53.12 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:658e:4925:c7be:4574) |
| 09:53.18 | WIMPy | And you should upgrade. |
| 09:53.34 | x4rlos | Well, that's what seems to be the problem somehow. The line is in (and dahdi reports OK). |
| 09:53.49 | x4rlos | We have moved from one premises to another. Different IDSN box. |
| 09:53.55 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 09:54.11 | WIMPy | So you have green light on both ends? |
| 09:54.11 | x4rlos | Yes, upgrade is next on the cards. |
| 09:54.52 | x4rlos | Yes. We are using a straight cable - though the guy who did the original install is wondering if he made a custom cable. Is this likely? |
| 09:55.28 | WIMPy | No. In all normal situations a normal patch cable fits. |
| 09:55.40 | x4rlos | (apologies if this is a stupid "how the hell would i know" kinda question - i'm a bit of a n00b with asterisk) |
| 09:55.47 | WIMPy | And if the cable is wrong, you don'g get green lights. |
| 09:55.56 | x4rlos | ah, that seems positive then. |
| 09:56.18 | WIMPy | You said you moved? Is that the same line taken to another location? |
| 09:56.24 | x4rlos | any advice you can offer trying to debug this problem? |
| 09:56.54 | WIMPy | Your Asterisk box is the same? |
| 09:57.10 | x4rlos | ah, i believe we had an ISDN line at the original site, and should be almost exactly the same that has gone into the new building. Same asterisk box and hardware entirely. |
| 09:57.25 | *** join/#asterisk thecoda (~kevin@b0fe572f.bb.sky.com) |
| 09:57.26 | x4rlos | (only ISDN terminator changed afaik) |
| 09:57.37 | WIMPy | Ok, so the configuration must be ok. |
| 09:57.57 | x4rlos | I would have thought so. |
| 09:58.02 | WIMPy | Looks to me as if the trunk hasn;t been switched to the new line then. |
| 09:58.40 | x4rlos | really? Is there anything i can attempt to prove this? |
| 09:59.01 | x4rlos | (i am going to get back on to them now). |
| 09:59.09 | x4rlos | Will see if they can confirm |
| 09:59.16 | x4rlos | (them = suppliers) |
| 09:59.47 | WIMPy | If the line is OK on all levels, but you still don't get a d-channel, the line must be ok, while there's nothing compatible at the other end. |
| 10:00.38 | x4rlos | Thanks very much for this - you have been most helpful. Lets give them a call :-) |
| 10:10.49 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 10:10.59 | *** join/#asterisk iztech (~iztech@76-246-226-131.uvs.irvnca.sbcglobal.net) |
| 10:12.25 | danfromuk | Hi. in voicemail.conf emailbody, is there a variable that contains the incoming DID that was dialled? Or can I set my own variable that contains the DID to use in voicemail.conf? |
| 10:19.42 | danfromuk | Also, is it possible to change the serveremail on a per context basis? |
| 10:21.45 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:23.22 | x4rlos | WIMPy: hmm. Providers say the line is okay :-( He's done a test call there, and says it was delivered :-l |
| 10:31.13 | x4rlos | I'm using "span=1,1,0,ccs,hdb3,crc4" with card: Digium, Inc. Wildcard TE121 single-span T1/E1/J1 card (PCI-Express) |
| 10:34.40 | WIMPy | How can a call be delivered if th interface isn't up? that's obviousely not possible. |
| 10:35.04 | WIMPy | Looks good for an E1. |
| 10:35.25 | kaldemar | and the rest of the configs? |
| 10:38.36 | x4rlos | I thought he meant the call reached the termination point. (BT box). |
| 10:40.02 | WIMPy | That's only an interface converter. No calls without working CPE. |
| 10:42.23 | x4rlos | hmmm. We have a divert on the line. So the call would never have reached the terminator anyways i dont think. |
| 10:42.27 | x4rlos | Ringing them back. |
| 10:42.37 | x4rlos | i could well sound very foolish here :-) |
| 10:43.07 | WIMPy | Oh |
| 10:43.27 | x4rlos | Outgoing calls should still work though, right? |
| 10:43.41 | WIMPy | yes |
| 10:43.44 | x4rlos | So its possible the checks he was doing were being diverted to our other line. |
| 10:43.48 | x4rlos | i guess. |
| 10:44.05 | WIMPy | Yes, I guess he checked the number, not the line. |
| 10:44.31 | x4rlos | If this turns out to be the problem, i owe you a beer. Or a dohnut. |
| 10:46.55 | WIMPy | For the jo average technician it might be an issue to check a line that can't receive calls. |
| 10:48.33 | *** join/#asterisk TSM2 (~the_softw@fw-lon1.wenn.com) |
| 10:53.58 | x4rlos | speaking to another person who is going to check again, explained the divert on the line. She says the 8 channels we have work perfectly fine. |
| 10:54.44 | x4rlos | :-( |
| 10:55.19 | [sr] | howdy |
| 10:56.02 | [sr] | i have a scenario, where i have a catch-all did's, without routes defined, and "core show channels" counts the call's, but nothing appears in CDR |
| 10:56.26 | *** join/#asterisk v0lZy (~chatzilla@84-255-194-41.static.t-2.net) |
| 10:56.28 | [sr] | does it shouldn't be counted as a call, or should it be appearing in CDR ? |
| 10:57.20 | WIMPy | Oh, only 8 channels? |
| 10:57.57 | [sr] | howdy WIMPy |
| 10:58.06 | x4rlos | Yes. We have 8 channels at this site (its a backup site). |
| 10:58.12 | WIMPy | Hi [sr] |
| 10:59.08 | WIMPy | I just wouldn;t expect a pri for only 8 channels. |
| 11:00.04 | *** join/#asterisk srp_ (~sandeep@115.115.144.218) |
| 11:00.26 | x4rlos | why not? (as i say, phone systems aren't my speciality :-)) |
| 11:01.10 | WIMPy | That's an amount where 4 BRIs would seem more likely. But I don't know BTs prices. |
| 11:01.43 | x4rlos | hmm. Well, it's something that was set up long ago, so i have no idea what kinda deal has been struck :-) |
| 11:03.21 | x4rlos | So, if the 8 channels are working fine, and it's just the D channel, what is this (signalling) channel actually being prevented from doing? |
| 11:04.14 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 11:04.17 | WIMPy | Well, actually I find that statement a little suspicious. It's not as if single channels can work or not. |
| 11:04.54 | WIMPy | There is a configured limit of channels you can use. But it's the whole line that works or doesn't. |
| 11:06.52 | x4rlos | So (here i go asking very dumb questions) why is the dchannel 16? Could it be something different? |
| 11:06.55 | srp_ | Hi, I am trying to test the working of asterisk which I have installed just now. I wanted to playback sorry.gsm file when the call is made to the extension 2600. I dont hear any sound over the softphone. The log on the asterisk server is (http://paste.kde.org/656270/). Please help |
| 11:06.56 | WIMPy | Have you tried removing power from the NT to reset it? |
| 11:07.20 | WIMPy | No, on E1 the D-channel is always on timeslot 16. |
| 11:08.56 | jacekowski | WIMPy: PRI is cheaper |
| 11:09.07 | x4rlos | I will ask them to reset the box now. |
| 11:09.36 | x4rlos | When i first spoke to BT they said "It's an ISDN E not DAS" or something like that. I have no idea if this even eans anything. |
| 11:09.39 | WIMPy | jacekowski: That surely depends on your provider. |
| 11:09.59 | jacekowski | WIMPy: with BT and everyone uses BT to provide last mile |
| 11:10.13 | WIMPy | Yes, it means it's talking DSS1 and not the old BT standard. |
| 11:10.13 | jacekowski | WIMPy: PRI hardware is cheaper as well |
| 11:10.49 | WIMPy | jacekowski: The hardware prices should be about the same. |
| 11:11.00 | jacekowski | WIMPy: not really |
| 11:11.37 | WIMPy | Well, I have to admit it's some time since I last looked at hardware prices. |
| 11:12.21 | x4rlos | Asterisk is reporting " app_dial.c:1747 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)" in the logs. |
| 11:12.26 | x4rlos | Box restarted. |
| 11:12.40 | x4rlos | Still same error every 4 econds. |
| 11:14.44 | WIMPy | jacekowski: I just took a look. It seems to depend on vendor. Sometimes 4BRI is cheaper, sometimes 1PRI. |
| 11:15.17 | jacekowski | digium wildcard £300 for PRI £400 for BRI |
| 11:16.23 | jacekowski | and to get same numer of channels on BRI it works out even worse |
| 11:16.26 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 11:16.42 | WIMPy | Off course. |
| 11:17.54 | WIMPy | But the good thin is that only parts can fail when you have multiple lines :-) |
| 11:20.11 | jacekowski | thing is, those lines come from same exchange |
| 11:20.45 | WIMPy | I wouldn't take all from the same telco. |
| 11:20.58 | jacekowski | in here you don't really have a choice |
| 11:21.10 | jacekowski | you have bt/openreach that provides the infrastructure |
| 11:21.37 | WIMPy | And lines sometimes get disconnected accidentally when setting up other lines. |
| 11:21.46 | jacekowski | and all you can really choose between is the middleman |
| 11:22.49 | WIMPy | Competition still seems far in most places. :-( |
| 11:24.52 | x4rlos | Personal experience with BT left a big scar :-) |
| 11:24.57 | WIMPy | But luck for Digium and Sangoma. |
| 11:25.18 | x4rlos | now problems prevail with this scenario (which may or may not be thir fault). |
| 11:25.39 | x4rlos | I'm gonna have to schedule a (300 quid!) engineer visit if i can't get any further. :-( |
| 11:26.18 | WIMPy | Unless your box suffered from spontaneous misconfiguration... |
| 11:26.36 | x4rlos | Solar flares do that. |
| 11:26.52 | WIMPy | That's the issue with PRIs. Yo usually don't have spare lines or hardware to change. |
| 11:27.26 | x4rlos | But even if solar flares were at work with a sprinkling of fairy dust - what could they possibly have changed that could cause this? |
| 11:27.58 | x4rlos | mind if i hastebin a few configs to have a look at? |
| 11:28.14 | WIMPy | chan_dahdi.conf should be the only thing involved at that point. |
| 11:28.22 | x4rlos | yup. |
| 11:28.30 | x4rlos | context=from-pstn |
| 11:29.53 | *** join/#asterisk srp_ (~sandeep@115.115.144.218) |
| 11:30.28 | WIMPy | pastebin it |
| 11:30.47 | WIMPy | Maybe we find the flipped electrons. |
| 11:31.09 | x4rlos | http://hastebin.com/fafuhuqali.ini |
| 11:31.45 | WIMPy | What is that? |
| 11:32.22 | x4rlos | chan_dahdi.conf |
| 11:32.39 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 11:32.43 | WIMPy | That page. No visible content, but some scripts. |
| 11:32.49 | x4rlos | huh? |
| 11:33.28 | x4rlos | really? What you using? |
| 11:33.43 | WIMPy | Phoenix :-) |
| 11:33.50 | WIMPy | And I have scripting disabled. |
| 11:34.03 | x4rlos | tbh w3m dont work with it either. |
| 11:34.07 | x4rlos | It's pretty :-) |
| 11:34.14 | x4rlos | I'll put it elsewhere, 1 sec :-) |
| 11:34.58 | x4rlos | http://pastebin.com/cim7qRUK |
| 11:35.01 | x4rlos | better? :-) |
| 11:35.16 | WIMPy | Jup |
| 11:35.50 | WIMPy | Looks ok to me. |
| 11:36.54 | x4rlos | So, is there anything i could be missing here? |
| 11:37.21 | x4rlos | Cos a kick in the balls is gonna be less painfull than getting an engineer out at £300+hourly rate and telling my boss. |
| 11:37.31 | WIMPy | switchtype, signalling and the channels are there. That should make it work. |
| 11:37.53 | x4rlos | context doesn't have a space in between the ' = ' |
| 11:38.06 | x4rlos | (yes, it's at that point i look for nothingness). |
| 11:38.14 | WIMPy | They a really don;t want satisfied customers, do they? |
| 11:38.33 | x4rlos | lol. Nope. |
| 11:39.09 | WIMPy | And it's an issue that might be possible to resolve without anyone leaving their desk. |
| 11:41.51 | x4rlos | Well, i spoke to two people now. And they are saying only thing left is to get a BT engineer out to take a look at it. |
| 11:42.00 | x4rlos | And that gonna cost me £££'s |
| 11:42.48 | WIMPy | Yes the hardest part is always to find someone who understands the situation. |
| 11:43.10 | x4rlos | will blubbering down the line help you think? |
| 11:43.12 | WIMPy | The line is up, so it's ok and connected somewhere. |
| 11:43.43 | WIMPy | But there's not the expected data on it. So where do you look? |
| 11:43.58 | WIMPy | Yes. Some fresh air might help :-) |
| 11:44.04 | x4rlos | hehehe. |
| 11:44.16 | x4rlos | Well, im going to grab some dinner from the shop round the corner. |
| 11:44.29 | x4rlos | Meanwhile, here's the dahdi system config if you wouldn't mind a quick gander. |
| 11:44.30 | x4rlos | http://pastebin.com/6YCCMsZP |
| 11:44.51 | x4rlos | I tried a couple of options but made no difference to changed back to the original. |
| 11:45.17 | x4rlos | (so changed back....) |
| 11:45.36 | WIMPy | The rest of it looks ok as well. |
| 11:45.47 | x4rlos | thanks very much for your help. |
| 11:48.49 | x4rlos | mopes to shop head in hands crying. |
| 11:51.09 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 12:13.51 | srp_ | Hi, I have installed AsteriskNOW. I have some doubts with it. Is this the right channel or is there a seperate channel for AsteriskNOW? |
| 12:15.52 | kaldemar | your doubts are likely to increase if you stay on this channel. :) try #asterisk-now |
| 12:16.18 | kaldemar | or #asterisknow, which ever it is. |
| 12:17.57 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:25.52 | *** join/#asterisk timholum1 (~chatzilla@184-158-19-224.dyn.centurytel.net) |
| 12:30.57 | timholum1 | Hello, Yesterday I had a server crash ( the whole system locked up ) I have been sifting through the logs, I noticed WARNING[10064] file.c: Unable to open agent-login (format 0x0 (nothing)): No such file or directory happend about 500 times a second with a bunch of verbose stuff as well ( here is a sample of what it was doing ) http://pastebin.com/jeZMQn89 |
| 12:31.02 | timholum1 | I have been doing googling on this, and I cant seem to find a resolution or why it happend |
| 12:31.05 | timholum1 | It almost looks to me as if a call somehow got stuck in a loop |
| 12:31.21 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 12:33.28 | Greenlight | SOunds like perhaps it was unable to access the filesystem ? |
| 12:34.19 | timholum1 | Greenlight: Maybe, but when I restarted asterisk everything seemed to work again |
| 12:34.57 | WIMPy | Format=0 looks lie a call where codec negotiation failed. |
| 12:35.04 | WIMPy | kile |
| 12:35.07 | WIMPy | like |
| 12:35.11 | WIMPy | gnah |
| 12:35.52 | Greenlight | Doesn't it look for a file to playback in any format available? That's what I was thinking the format 0x0 meant |
| 12:36.57 | WIMPy | It will always try the native format of the channel first. |
| 12:38.01 | timholum1 | If it was unable to find the correct file would it just skip it or would it keep retrying to load it? |
| 12:38.11 | Greenlight | It would just skip it |
| 12:38.27 | Greenlight | BUt not sure what the dialplan is; certainly looks like it's in some sort of loop |
| 12:38.52 | WIMPy | It would try other formats. |
| 12:39.12 | Greenlight | Yea, but if it couldn't find it at all, it would skip it. WOuldn't keep retrying |
| 12:39.31 | WIMPy | No, there's no retry. |
| 12:40.08 | Greenlight | And since there's other dialplan bits and pieces inbetween the attempts to playback that file, it suggests that there's something else going on in the dialplan. |
| 12:40.28 | Greenlight | Mind, I'd not expect that to completely lock up a system |
| 12:40.35 | WIMPy | Sure |
| 12:41.44 | timholum1 | from that it looks like it might be a loop in the agent logon for logging into the queues? |
| 12:42.13 | Greenlight | Yes, quite possibly |
| 12:42.49 | timholum1 | I am using the freepbx default one's, I will look to see if I can find it |
| 12:43.01 | Greenlight | Yea, I thought it looked freepbxy |
| 12:44.42 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 12:45.36 | Greenlight | On line 15, "GotoIf("Message/ast_msg_queue", "1?a3")" ... that seems to be where it's deciding to loop |
| 12:46.17 | Greenlight | You'd want to check what the logic is there |
| 12:48.07 | *** part/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 12:48.56 | timholum1 | Found there macro for it http://pastebin.com/ixNWJjt4 From what that message is, it looks like its probably the exten => s,n,GotoIf($["${CALLBACKNUM}" = ""]?a3) line |
| 12:50.57 | Greenlight | Yea, so that does seem somewhat silly |
| 12:51.02 | Greenlight | exten => s,n(a3),Read(CALLBACKNUM,agent-login,,,,) |
| 12:51.10 | Greenlight | We try and get CALLBACKNUM here |
| 12:51.17 | Greenlight | exten => s,n,GotoIf($["${CALLBACKNUM}" = ""]?a3) |
| 12:51.30 | Greenlight | And if it's blank, we just keep trying. Forever. |
| 12:52.51 | Greenlight | It's trying to get input from the agent |
| 12:52.58 | Greenlight | Ok |
| 12:53.12 | Greenlight | So it's a prompt where it asks them to enter their login number? |
| 12:53.14 | timholum1 | That looks like what happend, Thanks |
| 12:53.58 | timholum1 | Now I get to try to fix it :) ( and freepbx overwrites the config's when ever it gets reloaded :( |
| 12:54.49 | Greenlight | You may find this was an *effect* rather than a *cause* |
| 12:55.32 | Greenlight | Or, as WIMPy said, it could have been trying to run it on a channel where codec negotiotion failed and the channel didn't die. |
| 12:55.33 | WIMPy | Most probably. |
| 12:56.08 | Greenlight | In freepbx there's a file something like extensions_freepbx_override.conf |
| 12:56.24 | Greenlight | You can copy paste this macro there, and customsie it, without freepbx overwriting it |
| 12:56.47 | Greenlight | At the very least, i'd say add a Wait(2) in there so stop it going crazy if it happens again |
| 12:59.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:59.28 | x4rlos | WIMPy: I have just been reading this :: Supported Audio modes (1.2) ==> http://www.asteriskguru.com/tutorials/e1t1.html |
| 12:59.41 | x4rlos | It mentions that channel 16 used to be for signalling, but now it has changed? |
| 12:59.51 | x4rlos | Im not questioning your reasoning before :-) |
| 13:00.02 | WIMPy | huh? |
| 13:00.06 | x4rlos | But is this something i should possibly look at? Or am i making things up to look at? |
| 13:00.07 | WIMPy | reading.. |
| 13:00.41 | WIMPy | BT can tell you what parameters they use. |
| 13:01.27 | WIMPy | Ah, those other signalling types are not ISDN/PRI. |
| 13:01.48 | *** join/#asterisk srp_ (~sandeep@115.115.144.218) |
| 13:01.53 | WIMPy | Some weired analog emulation stuff. |
| 13:02.01 | x4rlos | ah, okay. Sorry :-) |
| 13:02.12 | x4rlos | I'm gonna have to bite the bullet and get an engineer out. :-) |
| 13:02.26 | *** join/#asterisk scubes13 (~scubes13@69.26.198.3) |
| 13:02.43 | WIMPy | Just look at the ISDn section. |
| 13:02.46 | srp_ | kaldemar: the #asterisknow channel has only 6 members. is it legit? #justwondering |
| 13:03.03 | Greenlight | It's probably just less popular |
| 13:03.41 | WIMPy | Or people have ginven up on trying to find out what's going on. |
| 13:04.10 | srp_ | the link on their channel welcome message points to some random wordpress site which has nothing! :-\ |
| 13:04.47 | Greenlight | What issue is it you're getting ? |
| 13:05.41 | kaldemar | srp_: it is valid. just less verbose channel. folks on this channel usually compile asterisk from source and don't use a GUI. if you want support on freepbx, there's also #freepbx. |
| 13:05.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:05.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:07.06 | srp_ | kaldemar: okay thanks :) I also needed some help on asterisk-gui.. is there a seperate channel for that as well? |
| 13:07.44 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:08.00 | WIMPy | #asterisk-gui |
| 13:08.17 | kaldemar | srp_: #asterisk-gui. that one is probably the wrong way to go if you want a GUI. |
| 13:08.38 | kaldemar | srp_: as in you should choose freepbx if asterisknow still has a choice. |
| 13:11.23 | srp_ | kaldemar: thanks :) Then I think i'll stick to asterisk CLI. I'm just starting with asterisk and new to it... and I thought GUI would be quick to get started... |
| 13:11.55 | WIMPy | It is, but it can become a dead end. |
| 13:12.15 | WIMPy | So if you're willing to learn, skip the GUI idea. |
| 13:12.34 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:14.37 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 13:22.29 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:25.05 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:25.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:32.37 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:44.07 | phix | WIMPy: Can you teach me? |
| 13:45.58 | Greenlight | Start with no eating yellow snow |
| 13:48.20 | *** join/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 13:48.25 | phix | what about glue? is glue ok? |
| 13:49.02 | [TK]D-Fender | Glue is for sniffing, not eating... |
| 13:49.11 | [TK]D-Fender | Amateurs.... |
| 13:49.14 | srp_ | WIMPy: i'm about to install asterisk-11.2.1 on ubuntu. Any good links to follow? the quickstart guide in asterisk.org is for asterisk-1.8! |
| 13:49.14 | phix | ah, thnx for the tip |
| 13:49.18 | phix | I will remember that :P |
| 13:50.36 | phix | [TK]D-Fender: I have an analogue phone connected to a TDM4100p, I have to hold the buttons down a fraction of a second longer than I have to when it is connected to the PSTN for asterisk / dahdi to pick up the key press |
| 13:51.05 | phix | It worked fine when I was using asterisk 1.6 and dahdi version something |
| 13:51.07 | rokjan | o/ |
| 13:51.09 | rokjan | help |
| 13:52.23 | phix | rokjan: no no me first! |
| 13:53.02 | rokjan | install asterisk, then the upgrade to centos 5.9, the following errors: |
| 13:53.03 | rokjan | kmod-dahdi-linux-2.6.1-1_centos5.2.6.18_308.24.1.el5.x86_64 from asterisk-current has depsolving problems |
| 13:53.03 | rokjan | <PROTECTED> |
| 13:55.02 | phix | ah, I use debian, I don't get those issues |
| 13:55.03 | WIMPy | phix: What? |
| 13:55.19 | *** join/#asterisk serafie (~erin@nat/digium/x-taqannwemeyyswez) |
| 13:55.24 | phix | WIMPy: teach me how to overcome my key pressing issues |
| 13:56.10 | WIMPy | srp_: Use this: http://ofps.oreilly.com/titles/9781449332426/index.html |
| 13:56.22 | WIMPy | phix: What about wanking? |
| 13:57.22 | srp_ | WIMPy: whats the recommended version of asterisk for putting it in production? because I still see a lot of docs / tutorials / blogs on 1.8 or 1.6 |
| 13:57.38 | phix | WIMPy: no no I don't need to learn how to wank, I have females for that |
| 13:57.38 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:58.15 | WIMPy | srp_: I'd go for the latest official version. |
| 13:58.33 | srp_ | WIMPy: okay.. cool.. |
| 13:58.34 | WIMPy | phix: So what other needs do you have? |
| 13:58.39 | *** join/#asterisk ManxPower (~igcewieli@10.sub-70-196-132.myvzw.com) |
| 14:00.09 | phix | WIMPy: I have a need to press a button without having to hold it down |
| 14:00.18 | phix | for an extended period of time |
| 14:00.28 | phix | and the need... the need for speed! |
| 14:00.51 | WIMPy | duct tape |
| 14:01.16 | phix | Duct tape makes you smart |
| 14:01.24 | phix | according to a TV series I watched |
| 14:02.33 | phix | any way |
| 14:03.55 | *** join/#asterisk deo (~deo@112.198.90.106) |
| 14:05.04 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:06.37 | *** join/#asterisk shadar (~eugene@88.206.118.177) |
| 14:08.47 | fling | how to replace something in variable? for example I want to replace ^8800 with +7800 |
| 14:09.58 | fling | ^ should match beginning of the line :] |
| 14:10.50 | [TK]D-Fender | phix, "relaxdtmf=yes" |
| 14:11.14 | phix | done |
| 14:11.29 | [TK]D-Fender | phix, Previously, or just now? |
| 14:11.43 | leifmadsen | fling: huh? |
| 14:11.51 | phix | how do I make it more relaxed? |
| 14:11.53 | fling | leifmadsen: Hello :] |
| 14:12.14 | fling | leifmadsen: everything works, I'm recording my calls |
| 14:12.15 | phix | relaxdtmf=smoking pot |
| 14:12.30 | ManxPower | relaxdtmf=screwupmydtmf |
| 14:12.39 | fling | leifmadsen: now when I call I first say 'The call may be recorded' :D |
| 14:12.46 | leifmadsen | coolio |
| 14:12.51 | ManxPower | fling: Asterisk does not support regular expression based replacement |
| 14:12.52 | phix | relaxdtmf=caffinatedPersonPressingButtons |
| 14:13.00 | leifmadsen | ManxPower: oh so it's that kind of day is it? throwback nick friday? |
| 14:13.05 | fling | ManxPower: ok :\ |
| 14:13.28 | phix | blitzrage: it is saturday morning |
| 14:13.37 | blitzrage | phix: you live in the future?!??! |
| 14:13.47 | phix | blitzrage: no, you live in the past |
| 14:13.48 | ManxPower | blitzrage: long story involving NAT and VMware |
| 14:13.58 | blitzrage | ManxPower: sounds about right; say no more |
| 14:14.37 | phix | say no more say no more, wink wink nudge nudge |
| 14:15.09 | blitzrage | turns up the Sneaker Pimps |
| 14:15.10 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:15.19 | ManxPower | phix: What exactly are you trying to do? |
| 14:15.31 | blitzrage | nubbie: hells ya :) |
| 14:15.40 | nubbie | ohai |
| 14:16.24 | angryuser | Hello, i have DAHDI_DUMMY/1 (source: Linux26) on machine 1 and (source: DAHDI_DUMMY/1 (source: HRtimer) on the server 2 can someone tell me what is HRtimer ? |
| 14:16.44 | phix | ManxPower: I am trying to dial a number |
| 14:16.52 | *** join/#asterisk serafie1 (~erin@76.73.167.231) |
| 14:16.55 | phix | ManxPower: but I have to hold the button for longer than usual |
| 14:17.07 | phix | otherwise dahdi / asterisk doesn't register the key I pressed |
| 14:17.37 | ManxPower | phix: is the call inbound to asterisk or outbound from asterisk? |
| 14:18.23 | phix | ManxPower: the TDM4100p card is in the same machine that asterisk is running on |
| 14:18.43 | ManxPower | phix: is the PSTN involved at all? |
| 14:19.20 | ManxPower | diagram it, i.e analog phone -> tdm410p -> pots line -> telco or something |
| 14:20.31 | phix | ManxPower: no |
| 14:20.39 | [TK]D-Fender | ManxPower, DAHDI FXS on the card |
| 14:20.52 | [TK]D-Fender | manx his issue is with the FXS end |
| 14:20.54 | phix | analog phone -> tdm4100p -> asterisk -> SIP |
| 14:21.07 | phix | FXS yes |
| 14:21.42 | phix | I bet [TK]D-Fender can fix the issue |
| 14:21.44 | ManxPower | verify you are using rfc2833 on the sip side and try increasing your rxgain by 1 - 2 db on the dahdi side |
| 14:21.50 | *** part/#asterisk mariusno (128408@diamant.ifi.uio.no) |
| 14:22.00 | phix | ManxPower: RXGain effect TDM? |
| 14:22.26 | ManxPower | phix: as it is put in chan_dahdi.conf that would be a logical thing |
| 14:23.17 | kaldemar | fling: Set(VAR=${IF($["${VAR:0:4}" = "8800"]?+7800${VAR:4}:${VAR})}) |
| 14:23.31 | phix | rxgain effects the receivers volumn when a call is in progress? |
| 14:24.00 | phix | 1:23! |
| 14:24.23 | [TK]D-Fender | ManxPower, Lets try this again. He has a DAHDI DTMF detection issue between his phone on the FXS and Asterisk. Nothing to do with SIP. At all. |
| 14:24.47 | [TK]D-Fender | ManxPower, Clear now? :) |
| 14:24.53 | *** join/#asterisk bchia (~Adium@nat/digium/x-hovdwxxqiwafeqju) |
| 14:24.57 | ManxPower | [TK]D-Fender: OK. Then I recommend increasing the rxgain on the DAHDI FXS ports. |
| 14:25.18 | [TK]D-Fender | ManxPower, Certainly a thought. |
| 14:25.20 | kaldemar | or lowering them in case the gains are too high. |
| 14:25.34 | [TK]D-Fender | Try Momma Bear's! |
| 14:25.39 | phix | [TK]D-Fender: ? |
| 14:25.45 | angryuser | Hello, i have DAHDI_DUMMY/1 (source: Linux26) on machine 1 and (source: DAHDI_DUMMY/1 (source: HRtimer) on the server 2 can someone tell me what is HRtimer ? its the new name of the soft timer ? |
| 14:25.49 | [TK]D-Fender | They're "Just Right" (tm) |
| 14:25.49 | ManxPower | kaldemar: maybe, but I'd think that would be doubled digits, not missing digits |
| 14:26.11 | ManxPower | angryuser: likely the High Resolution Timer from the Linux Kernel |
| 14:26.13 | kaldemar | ManxPower: i've observed it to cause missing ones too. |
| 14:26.39 | kaldemar | ManxPower: but checking gain levels is a good idea, yes. |
| 14:27.26 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ejozqseuyodvkjoz) |
| 14:27.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:28.37 | ManxPower | Why can't everyone just use PRI? 8-) |
| 14:29.39 | WIMPy | You're not inheaven (yet). |
| 14:30.20 | fling | kaldemar: thanks :] |
| 14:30.23 | *** part/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
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| 14:30.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:30.32 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 14:30.37 | [TK]D-Fender | @^&%^&@#%@ |
| 14:30.46 | [TK]D-Fender | Friggin Ubuntu window focus... |
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| 14:55.20 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 14:57.24 | danfromuk | Hi, after the queue cmd has been executed, how can I tell whether the caller left the queue because of a timeout and how can I tell if a caller left the queue because there were no queue members? |
| 14:59.10 | danfromuk | Never mind. ive found the solution thanks. |
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| 15:57.45 | x4rlos | WIMPy: You still there? I seem to have come to a conclusion on the problem. |
| 15:58.50 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 16:06.41 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:06.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:07.51 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:16.28 | WIMPy | x4rlos: What did you find? |
| 16:20.16 | *** join/#asterisk jeffspeff (~jeff@12.49.160.131) |
| 16:21.13 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 16:35.39 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:37.52 | x4rlos | aah, |
| 16:38.16 | *** join/#asterisk jsarrel (~jsarrel@75-136-148-3.dhcp.gnvl.sc.charter.com) |
| 16:39.56 | x4rlos | well. After speaking with Openreach for like the 4th time, it appears that they have not "transferred" the line they have been testing. They set up another line instead. Except they didn't activate it. |
| 16:40.05 | x4rlos | So the line they were testing was an old line. |
| 16:40.23 | x4rlos | So the bt guy installed the line, and tested on the stage 1 test that the 2 meg line works. |
| 16:40.38 | x4rlos | then the provider should have come back to do a stage 2 test activating the line. |
| 16:40.41 | x4rlos | this never happened. |
| 16:40.52 | WIMPy | That's about what I expected. |
| 16:42.39 | x4rlos | lol. |
| 16:43.16 | WIMPy | Did tehy fix it then? |
| 16:43.42 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 16:43.44 | WIMPy | Or did they only find out what they have to do? |
| 16:45.31 | x4rlos | it'll take 6 days to activate the line (as they forgot to activate it). |
| 16:45.37 | jsarrel | hello all, i am having some issues with the AMI being very slow to respond on a 1.4 box. top shows everything normal, what used to take less than a second to execute is now taking almost 10 seconds. any suggestions on where to look? |
| 16:46.04 | WIMPy | What? 6 days? That's ridicolous. |
| 16:46.07 | jsarrel | restarting asterisk does not help |
| 16:46.26 | x4rlos | yup. I asked if they can sort it much faster. Nice lady says she will see what she can do and get back to me. |
| 16:49.16 | WIMPy | wonders what terms and conditions BT has. At 6 days I'd expext quite some compensation. |
| 16:49.21 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:55.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:09.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:09.35 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 17:09.55 | Mon|A|rch | hey guys, I'm connecting to the AMI through cURL |
| 17:10.00 | Mon|A|rch | the call url looks kind of like this: |
| 17:10.13 | Mon|A|rch | $call_url = $this->server . "/rawman" . |
| 17:10.14 | Mon|A|rch | "?action=originate&channel=sip/10.3.1.1/91" . $next_call['phone_number'] . |
| 17:10.16 | Mon|A|rch | blah blah blah |
| 17:10.22 | x4rlos | we shall see.. |
| 17:10.23 | Mon|A|rch | what I'm wondering is |
| 17:10.36 | Mon|A|rch | I want to pass extra variables through the originate action |
| 17:10.48 | Mon|A|rch | which I've done with variable=var1=5 |
| 17:10.51 | Mon|A|rch | or something like that |
| 17:11.05 | Mon|A|rch | but the syntax for sending multiple variables (in the standard i/o version) |
| 17:11.16 | Mon|A|rch | is var1=5|var2=10|var3=... |
| 17:11.21 | Mon|A|rch | is that going to break everything? |
| 17:11.25 | Mon|A|rch | the pipes i mean |
| 17:11.34 | Mon|A|rch | can asterisk handle it if i url encode the pipe? |
| 17:11.59 | Mon|A|rch | that's a big textwall, i know |
| 17:12.01 | Mon|A|rch | sorry about that |
| 17:12.26 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 17:18.11 | Katty | Nugget: http://42ndrecipestreet.blogspot.com/2013/01/cast-iron-pizza.html |
| 17:18.33 | jpsharp | Oh, that looks tasty |
| 17:18.52 | Katty | jpsharp: it's not too bad. needs a little tweaking still. |
| 17:19.26 | Katty | the crust isn't quite as crispy/crunchy as i'd like |
| 17:19.33 | Katty | but i will figure it out |
| 17:20.30 | jsarrel | @Mon|A|rch, I did a quick search and came up with this page http://note.sdo.com/u/1410997095/NoteContent/k6xJD~jVwvCVnM00s000nH that does indeed urlencode the vars (look at Making a Call section) |
| 17:21.09 | Mon|A|rch | jsarrel, cool, I appreciate it |
| 17:21.18 | Mon|A|rch | I've been looking for that exact thing for a while |
| 17:22.47 | Nugget | mmm |
| 17:26.24 | beardy | Katty: A slice of that would have been perfect with the beer I just opened. |
| 17:28.01 | Katty | beardy: well i'll trade you 1/4 of the pizza for a beer. |
| 17:28.05 | Katty | beardy: and we'll have lunch |
| 17:28.17 | beardy | Katty: It's a deal! |
| 17:28.41 | Katty | as long as it's good beer ;> |
| 17:28.59 | beardy | Chimay Blue. |
| 17:29.55 | Katty | haven't tried that one yet. i'll see if i can find it |
| 17:29.59 | Katty | it's not seasonal is it? |
| 17:30.31 | beardy | No, not that I know. I bought in the summer, and last week so.. nope. :) |
| 17:30.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:30.54 | Katty | mkay |
| 17:30.57 | Katty | adds to grocery list |
| 17:32.16 | Katty | Nugget: did you notice the Alton Brown cool ranch seasoning post? |
| 17:32.27 | Katty | Nugget: was on reddit. |
| 17:33.13 | *** join/#asterisk serafie1 (~erin@nat/digium/x-qlnkjmpccenmalgt) |
| 17:34.56 | Nugget | nope |
| 17:35.04 | Nugget | I don't read reddit |
| 17:35.15 | Nugget | because some asshole got the username 'nugget' before I did |
| 17:35.25 | Katty | Nugget: crush up cool ranch doritos. |
| 17:35.29 | Katty | Nugget: put in pepper grinder. |
| 17:35.34 | Nugget | clever |
| 17:35.36 | Katty | very. |
| 17:36.15 | Katty | probably applies to other munchable crunchable tasty things. |
| 17:43.07 | Nugget | Katty: http://macnugget.org/crud/abtwitter.png (at bottom) |
| 17:43.56 | Katty | :> |
| 17:46.00 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 17:46.48 | x4rlos | no reply from BT. Guess will wait till monday now. Weekend time!! |
| 17:47.29 | Katty | horay for weekend time! |
| 17:48.02 | Nugget | track this weekend, but I'm carless so I'll have to get someone else to let me drive theirs |
| 17:48.28 | ManxPower | Whoo! Whoo! I'm moving to Florida! |
| 17:48.49 | Nugget | ManxPower: are you still in Alamaba? Am I remembering correctly? |
| 17:48.53 | Nugget | er alabama |
| 17:49.02 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 17:49.05 | *** part/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 17:49.32 | ManxPower | Huntsville |
| 17:49.36 | Nugget | *nod* |
| 17:50.13 | Nugget | In florida you'll be able to keep a pair of socks for more than three months without having to throw them away orange. |
| 17:50.17 | Nugget | but more bugs |
| 17:50.21 | Qwell | ManxPower: wait, you're in Huntsville? |
| 17:50.22 | Nugget | so it's a lateral move imho :) |
| 17:50.34 | ManxPower | Qwell: Yup, have been for years. |
| 17:50.40 | Qwell | How the hell have I never met you in person? |
| 17:50.52 | ManxPower | Qwell: because I don't socialize with the Digium people? |
| 17:50.59 | Nugget | used to live in Oneonta |
| 17:51.09 | Qwell | umm, okay then |
| 17:51.20 | ManxPower | nothing against them however. |
| 17:51.52 | ManxPower | I usually went to the nudist colony for social stuff. |
| 17:52.20 | ManxPower | And Mark simply would not come visit. LOL! |
| 17:52.24 | Nugget | heh |
| 17:52.39 | Qwell | wait, go back |
| 17:52.43 | Mon|A|rch | jsarrel, I have a follow-up question |
| 17:52.46 | Qwell | there's a nudist colony in Huntsville? |
| 17:52.58 | Mon|A|rch | I see that asterisk urlencodes commands it sends to the OS |
| 17:53.23 | WIMPy | There's a nudist community in the US? |
| 17:53.31 | Mon|A|rch | I'm assuming that means that it can handle urlencoded strings sent to it as actions |
| 17:53.39 | ManxPower | Qwell: "clothing optional male-only campground" a bit south of Gadsden |
| 17:54.20 | Mon|A|rch | nvm, question answered |
| 18:01.00 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:3546:2c49:3f4f:17d3) |
| 18:07.31 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:07.54 | *** join/#asterisk budster (~brhatigan@cpe-066-057-049-070.nc.res.rr.com) |
| 18:12.58 | jpsharp | ManxPower: Where in florida be you moving to? |
| 18:15.04 | ManxPower | jpsharp: Pensacola |
| 18:22.33 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 18:41.52 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
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| 18:51.58 | solitude88 | Need Help. We have an asterisk system that we were told could be run on standard cable connection but the quality was very bad so we ended up purchasing a point to point from where the asterisk system was setup to the call center but the cost is unmanagable for us so we're looking for a way to solve this |
| 18:53.31 | [TK]D-Fender | solitude88, Find another internet service |
| 18:53.57 | WIMPy | Or use phone lines if they're cheaper. |
| 18:54.13 | solitude88 | I'm in Las Vegas, cox is the best. There's century link but it always goes out |
| 18:54.42 | solitude88 | WIMPy are you saying go analog? |
| 18:54.57 | WIMPy | Off course not. |
| 18:55.01 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 18:55.10 | solitude88 | what do you mean by phone lines? |
| 18:55.21 | WIMPy | But in some areas PRIS are cheaper than the same volume of IP. |
| 18:55.41 | solitude88 | ok |
| 18:55.53 | jeev | yea i'm getting a good deal on a PRI vs a cost per port with a SIP guy |
| 18:55.59 | jeev | it's like 50 bucks more expensive for PRI |
| 18:56.02 | jeev | worth it IMO |
| 18:56.21 | solitude88 | question. Shouldnt an agent not in the same city be able to log into the system via softphone and the quality of the call be ok? |
| 18:56.39 | jeffspeff | jeev what sip provider was wanting to charge you per port? |
| 18:56.46 | solitude88 | We're thousands of dollars a month for just our internet |
| 18:56.51 | [TK]D-Fender | solitude88, Sucky internet is sucky internet |
| 18:57.01 | jeev | jeffspeff, i dont think i can say it.. i can ask him to email you if you're interestd. |
| 18:57.02 | jeev | interested |
| 18:57.06 | jeffspeff | holy crap, that's horrible for shitty service |
| 18:57.13 | solitude88 | yeah I know |
| 18:57.34 | jeffspeff | jeev, lol no. i have great service with unlimited ports with my current provider. |
| 18:57.52 | solitude88 | who is that jeffspeff |
| 18:57.58 | jeffspeff | Intelepeer |
| 18:58.21 | jeev | i'm just scared to use VOIP..., i use it but i want to support more local guys with fax and stuff |
| 19:00.25 | jeffspeff | you offer an off-site fax server on POTS lines? |
| 19:00.55 | jeev | i dont sell really.. like to the public, i offer it to my IT clients |
| 19:01.00 | jeev | i mean clients that i do IT for |
| 19:01.19 | jeffspeff | so, you put an asterisk box at each of their locations? |
| 19:01.35 | jeev | sometimes.. but if they only want fax, i give them fax to email capabilities |
| 19:02.24 | jeffspeff | you can do that over voip if your provider supports t.38 |
| 19:02.37 | jeffspeff | like mine |
| 19:03.12 | jeev | i dont know if they support it or not, i even send fax over callwithus.com sometimes very cheap routes, i do see some trouble sometimes but i also even do with PRI |
| 19:03.23 | jeev | ultimately i dont even know if my PRI provider sends the ro utes out TDM or ip |
| 19:03.53 | jeffspeff | are you in the US or canada? |
| 19:03.56 | jeev | i think toll is TDM |
| 19:03.57 | jeev | US |
| 19:04.05 | jeev | most california routes for me are TDM, outside i think is IP |
| 19:04.09 | jeffspeff | isn't callwithus in canada? |
| 19:04.11 | WIMPy | Easy enough to find out. |
| 19:04.38 | jeev | i gave callwithus as an example of someone i've sent fax over ip with. |
| 19:04.41 | jeev | they're not my PRI carrier |
| 19:04.47 | jeffspeff | oh |
| 19:05.02 | jeffspeff | i'd expect fax issues over sip with the provider in a different country. lol |
| 19:05.36 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 19:06.37 | jeev | i'm 15ms from sip.callwithus.com.. was just using as an example |
| 19:07.22 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 19:08.26 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 19:08.42 | gusto | hi |
| 19:08.44 | gusto | what's up? |
| 19:08.44 | citywok | WIMPy: sipgate closed registrations / isn't taking new customers :-\ |
| 19:08.54 | gusto | cool |
| 19:09.02 | gusto | maybe they are full |
| 19:09.04 | citywok | but it sounds like we may have figured out most of our callerid issues anyway |
| 19:10.19 | gusto | whatever, for such a big market there are only a few big names, so i see it positive when someone like sipgate just closes their regs |
| 19:11.15 | *** join/#asterisk bparker (~bp@fiveforty.net) |
| 19:11.35 | bparker | is it possible to catch DTMF tones while recording with Record() ? |
| 19:12.10 | bparker | I need to be able to stop/start my recording with a different key than * or # (and without modifying source code) |
| 19:12.50 | jeev | any as5300 users? |
| 19:28.24 | kaldemar | bparker: one on the app option allows to use any key. |
| 19:33.41 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 19:35.01 | ManxPower | solitude88: Often people who are unable to get internet service with QoS install one internet connection for data and one internet connection for SIP |
| 19:41.18 | gusto | DTMF tones when they are inband only with 711 or above |
| 19:41.21 | gusto | g711 |
| 19:45.04 | WIMPy | citywok: That's for US only, isn't it? |
| 19:46.21 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:46.22 | citywok | i told them i wanted to call from the u.s. to europe and they said sorry we don't accept new registrations. lol |
| 19:46.51 | WIMPy | You want uk and de number or just calls? |
| 19:47.15 | WIMPy | Anyway you should go to the .co.uk and .de sites. |
| 19:47.31 | citywok | we were having callerid issues to uk/de, but i think we figured it out. we got numbers for uk/de from tollfreeforwarding.com -- and have them forwarded to local DIDs |
| 19:47.53 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 19:48.15 | citywok | i think we're probably good now, the callerid is just obnoxious, i'll know on monday if we're all square and i can get away with not having to terminate with a direct .co.uk/.de site/provider |
| 19:48.42 | *** join/#asterisk ra21vi (ra21vi@122.177.53.248) |
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| 19:50.18 | ra21vi | Can I know which inbound call and dialled options were used for a given SIP account. |
| 19:52.16 | ra21vi | the scenario is, for a given IT Dept IVR, when a user enters, he is asked Ticket No. If ticket no is correct, he is connected to next available agent who uses Hardware phone to connect to asterisk using a SIP ID. I want to tell the agent the Ticket no anyhow which was punched by caller. |
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| 19:53.20 | WIMPy | SentText or put it in to the caller ID (name). |
| 19:53.42 | citywok | yea i'd do it with callerid -- that's how i tell my agents what IVR options were pressed |
| 19:54.47 | ra21vi | caller ID would be displayed in Hardware phone's display. Actually I want to get it in desktop machine of agent assigned to him, so he can open the details. |
| 19:55.20 | WIMPy | That's no longer an Asterisk question then. |
| 19:55.25 | ra21vi | but that is new to me 'SentText', I will experiemet with it |
| 19:56.47 | ra21vi | WIMPy: oh right. I was thinking with SIP. Yes that would not be really an asterisk question |
| 20:00.03 | citywok | the super ghetto way would be to email the ticket no to the agent |
| 20:00.11 | ra21vi | WIMPy: can you suggest me what would be best place to ask these questions? I am little lost |
| 20:01.11 | WIMPy | From Asterisk you can start anything via an AGI or System or whatever. What you (can) use at the client side is entirely up to you. |
| 20:01.27 | WIMPy | Especially the association between PCs and phones. |
| 20:02.03 | WIMPy | If you are lucky you may be able to connect the two. |
| 20:02.26 | ra21vi | citywok: that would be easy, :) ... |
| 20:02.28 | ra21vi | But I don't know who is sitting on which machine, so only I can map the SIP id that is used in hardphone to somescript or desktop custom app. Like I can build a small SIP client where agent would pass their SIP no, and then would get all notifications. |
| 20:02.58 | ManxPower | the easiest thing to do is use caller id name to put the ticket number |
| 20:03.13 | citywok | ManxPower: yea, we suggested that |
| 20:03.46 | citywok | you could have the agents "log in" so you know where they are sitting, and be able to associate the extension with the person with the email address |
| 20:04.45 | ra21vi | citywok: Instead of Email Address, i think SIP ID would be good. Since if other agent sits on this comp, he does not have to worry about getting logged into it |
| 20:05.10 | citywok | whatever works easiest for you |
| 20:05.29 | citywok | but a desktop application definitely isn't easy. lol |
| 20:06.26 | ra21vi | yes, but I will just use CLI on desktop initially. A simple script which can login into Asterisk (No Voice, just presence and other thing) |
| 20:07.17 | ra21vi | one of my friend asked me to look into AMI to know which SIP is connect to which call, but I think that is Management thing. Tried to find, negetive. |
| 20:08.03 | WIMPy | AMI is good for everything :-) But I don;t see the need here. |
| 20:08.16 | ra21vi | ManxPower: how can i show caller-id? Can I change the text in caller id to be displayed |
| 20:08.29 | WIMPy | But depending on your programming preferences it might still make sense to use. |
| 20:09.03 | ra21vi | WIMPy: :) I am AMI noob. Just know the name. I spent my time understanding AGI/FastAGI, still have to see what AMI does |
| 20:10.49 | ManxPower | ra21vi: Those are basic questions which you can likely find information about on voip-info, the asterisk book, or 30 million other pages. |
| 20:12.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:13.14 | citywok | ra21vi: something like this -- set(CALLERID(num)=whatever ${CALLERID(num)} |
| 20:13.24 | citywok | but yea, as manx said |
| 20:13.25 | citywok | ~book |
| 20:13.26 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:13.50 | ra21vi | citywok: :) thank you. Yes I was googling it. |
| 20:14.33 | *** join/#asterisk sfa (~root@208.66.156.12) |
| 20:14.42 | ra21vi | citywok: thanks for the http://www.asteriskdocs.org/ link. I will get a lot refernce from here. |
| 20:17.36 | citywok | yw |
| 20:18.23 | [sr] | WIMPy: |
| 20:24.48 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
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| 20:30.43 | *** join/#asterisk carrar (tim@osburn.com) |
| 20:30.47 | carrar | moo |
| 20:36.05 | *** join/#asterisk fexilal (~fexilal@unaffiliated/felixal) |
| 20:42.01 | fexilal | Hi |
| 20:42.06 | fexilal | I have a question about phone lines |
| 20:42.43 | fexilal | when you take an analogue line with 4 lines having the same number - what's the wiring like? |
| 20:43.01 | fexilal | the telco gives you a pair for each number? |
| 20:43.22 | WIMPy | That's how analog used to work, yes. |
| 20:45.10 | [TK]D-Fender | fexilal, How does a line have 4 lines? |
| 20:46.02 | [TK]D-Fender | fexilal, And if they have the same number .... what does "a pair for each number" mean when there is just one? |
| 20:46.35 | fexilal | [TK]D-Fender : I'm sorry, I meant 'analogue connection' |
| 20:47.05 | [TK]D-Fender | fexilal, Try from the beginning again being clear about kinds & qty's. |
| 20:47.15 | WIMPy | And the "each" of only one part? |
| 20:48.19 | fexilal | wow, I messed up |
| 20:49.11 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:49.34 | fexilal | does the telco give you a pair of cable for each auxillary line?* |
| 20:49.54 | WIMPy | A line is a line. |
| 20:50.08 | WIMPy | And it comes on a copper pair. |
| 20:50.23 | ra21vi | lol. he will leave |
| 20:50.33 | fexilal | okay |
| 20:50.59 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 20:54.45 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 20:55.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:56.41 | fexilal | I guess I should probably read a book before asking about BRI/PRI |
| 20:57.51 | WIMPy | 2 channels per copper pair or (30 or 24) channels per 2 or 4 wires. |
| 20:58.02 | [sr] | WIMPy: tell me if you know, i have a catch-all DID's, set to a blackhole destination, and that counts the call count of "core show channels", but nothing is saved in CDR, should it not be counted on the "core show channels", or should it be saved to CDR and it's not beeing? |
| 20:58.37 | WIMPy | What does that blackhole do? |
| 20:59.34 | WIMPy | You have the choice to log unanswered calls or not. |
| 21:00.35 | [sr] | WIMPy: i have unanswered call = yes |
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| 21:12.57 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-qeextuxwmozaflcp) |
| 21:23.27 | [sr] | WIMPy: well the blackhole (maybe its freepbx designation) does nothing with the DID'd |
| 21:23.30 | [sr] | DID's |
| 21:24.19 | [sr] | it's to prevent the number not in service if no route/DID is defined |
| 21:26.24 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
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| 21:33.55 | *** mode/#asterisk [+o nubbie] by ChanServ |
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| 21:51.34 | WIMPy | [sr]: Well if it doesn't even accept the call, there is no call to be logged. |
| 21:52.00 | [sr] | WIMPy: then core show channels should count it, right? |
| 21:52.09 | [sr] | shouldn't i mean |
| 21:52.25 | WIMPy | Right. |
| 21:52.44 | WIMPy | So what does it do? |
| 21:53.10 | WIMPy | Does it contain a NoCDR()? |
| 21:55.02 | [sr] | a CDR() does, a NoCDR() not, check: http://pastebin.com/6CiNRXgw |
| 21:57.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.17.45) |
| 21:58.32 | WIMPy | There is nothing that accepts the call. How to you see it in core show channels? It should only last milliseconds. |
| 21:59.06 | ManxPower | [sr]: you may have better luck on #FreePBX. |
| 22:00.37 | [sr] | WIMPy: http://pastebin.com/2MJLXnyd , it simply counts it as a call |
| 22:00.57 | [sr] | ManxPower: yes but i decided to ask 1st here, it seems to me to be a asterisk core stuff |
| 22:01.06 | [sr] | sorry if i'm wrong |
| 22:01.22 | WIMPy | Oh. calls processed. Not sure how much that means. |
| 22:01.31 | ManxPower | There is no commonly used term "blackhole" in Asterisk |
| 22:01.46 | WIMPy | It is definitely unanswered. |
| 22:01.49 | ManxPower | Did you configure it in the FreePBX GUI? If so, it is a FreePBX question |
| 22:02.20 | WIMPy | But it doesn't even do any intermediate speps like proceeding, progress, or ringing. |
| 22:03.01 | [sr] | WIMPy: right, no progress or ring or what so ever |
| 22:03.33 | [sr] | ManxPower: ok, i'll ask them |
| 22:03.58 | WIMPy | I've never really cared too much, when exactely CDRs are started. |
| 22:05.03 | WIMPy | But if I look at mine, I guess they really only start at a Dial or when answered. Maybe someone can give a better answer. |
| 22:06.12 | ManxPower | WIMPy: without knowing what a FreePBX blackhole does, this is all pure speculation |
| 22:06.34 | WIMPy | You can read the pastebin. |
| 22:09.01 | [sr] | i'll ask the freepbx guys also |
| 22:09.05 | [sr] | tks WIMPy |
| 22:09.17 | [sr] | i'm going to get off the PC now, my eyes are hurting! |
| 22:11.39 | carrar | I would think your butt would be sore, not your eyes |
| 22:11.47 | carrar | if you are "on" your PC |
| 22:11.58 | [sr] | :) |
| 22:12.09 | [sr] | wrong word |
| 22:12.25 | [sr] | going out, leave the pc |
| 22:12.34 | carrar | yeah don't take the PC with you |
| 22:12.44 | WIMPy | And if you sit on it backside fron, you might even get a blow job. |
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| 23:12.55 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 23:18.41 | Mon|A|rch | I'm getting a lot of originate failed responses |
| 23:18.47 | Mon|A|rch | but they're sort of random |
| 23:18.54 | Mon|A|rch | I'm never really sure where they're going to pop up |
| 23:19.00 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:19.06 | Mon|A|rch | and for some reason the CLI isn't giving me any meaningful output |
| 23:19.21 | Mon|A|rch | I can post what it's giving me |
| 23:19.26 | Mon|A|rch | but I'm sort of confused |
| 23:22.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ejozqseuyodvkjoz) |
| 23:27.14 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 23:36.41 | [TK]D-Fender | That would be a good idea |
| 23:40.42 | Mon|A|rch | when it goes buggy again I'll paste |
| 23:41.10 | Mon|A|rch | It usually only gives me "Calling on SIP Co5" or something |
| 23:41.19 | smkelly | file: hello |
| 23:41.23 | *** join/#asterisk scubes13 (~scubes13@220.sub-70-193-164.myvzw.com) |
| 23:42.53 | [TK]D-Fender | Mon|A|rch: that is a meaningles message. You'd need real SIP debug |
| 23:43.19 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 23:43.57 | Mon|A|rch | [TK]D-Fender, where do i get SIP debug from? I've used 'asterisk -vvvvvvr' |
| 23:44.05 | Mon|A|rch | is there a specific log file? |
| 23:44.11 | Mon|A|rch | or a cli command to show it to me? |
| 23:45.40 | [TK]D-Fender | "sip set debug on" |
| 23:47.00 | *** kick/#asterisk [smkelly!~file@asterisk/developer-and-muffin-lover/file] by file (smkelly) |
| 23:47.11 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 23:53.44 | fling | How to change telephone number of incoming call? it is missing starting + , so I see wrong phone number in sip client |
| 23:53.47 | smkelly | file: :( |
| 23:53.51 | fling | [TK]D-Fender: hello :p |
| 23:54.45 | [TK]D-Fender | fling: "core show function CALLERID" |
| 23:55.02 | fling | [TK]D-Fender: thanks |
