| 00:00.08 | ChannelZ | Maybe the clock on your computer is unhappy and it can't count right anymore |
| 00:00.08 | onixx99 | ChannelZ: and for the digits that work, it foes on channe.c: 4004 |
| 00:00.35 | onixx99 | ChannelZ: I rolledback to 1.4 and it worked again right away... |
| 00:01.40 | onixx99 | ChannelZ: some digits go thru, other don't... I know the panel transmit DTMF fairly quickly but it has never been a problem in 1.4.40 |
| 00:03.06 | onixx99 | ChannelZ: one big difference I note in debug is that in 1.4, the first line I see in debug is DTMF end '2' received on SIP/2299-00000001, duration 0 ms |
| 00:03.43 | onixx99 | ChannelZ: where in 1.8, I see DTMF begin '2' received on SIP/2299-00000000 |
| 00:04.03 | ChannelZ | well I suppose you could try reducing AST_MIN_DTMF_DURATION in channel.c and see if it does anything for you |
| 00:04.42 | ChannelZ | do you know what this "panel" you speak of does for dtmf length? |
| 00:05.12 | onixx99 | ChannelZ: about 50ms |
| 00:06.25 | ChannelZ | Hmm. |
| 00:07.09 | *** join/#asterisk albertoandrade (~albertoan@186.206.5.33) |
| 00:08.06 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:08.50 | onixx99 | ChannelZ: okay, I changed to 35ms |
| 00:09.44 | onixx99 | ChannelZ: now I stopped getting the channel.c:4040 and 4033 |
| 00:10.12 | onixx99 | (detected to have actual duration ...) and (has duration xx but want minimum 80) |
| 00:12.11 | onixx99 | ChannelZ: however, it still is totally inaccurate |
| 00:12.50 | onixx99 | ChannelZ: the one thing that jumps out to me is that with 1.4 all I saw was messages about "emulation" |
| 00:13.07 | onixx99 | ChannelZ: where with 1.8, I see stuff about passthrough |
| 00:26.55 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 00:29.00 | onixx99 | exit |
| 00:29.14 | *** join/#asterisk timahvo1 (~rogue@197.177.156.1) |
| 00:29.23 | onixx99 | ChannelZ: I just can't find ;-) |
| 00:47.54 | *** join/#asterisk v4x (~v4x@d209-89-85-139.abhsia.telus.net) |
| 00:49.14 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:49.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:51.10 | *** join/#asterisk Posi211 (~pdoqqq@c-98-192-65-8.hsd1.ga.comcast.net) |
| 00:53.04 | onixx99 | ChannelZ: Thanks for the advice.. I will update you if I find the problem |
| 00:53.08 | onixx99 | exit |
| 00:54.23 | Posi211 | I'm a newbe and need some help |
| 00:54.34 | Posi211 | Can I just come out and ask? |
| 00:57.23 | paulc | ~ask |
| 00:57.23 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:58.20 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 00:59.30 | Posi211 | I'm new to asterisk and loaded freepbx distro 1.8.8 and the page group feature is broken. I keep getting the same error on both systems I'm testing. Did I break this or is this a know issue? |
| 01:04.59 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 01:10.02 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:20.19 | *** join/#asterisk fr0ggie (~irc@unaffiliated/nn) |
| 01:20.58 | [TK]D-Fender | Posi211: Pastebin this error for us to look at. |
| 01:20.59 | [TK]D-Fender | ~pb |
| 01:21.00 | 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. |
| 01:21.01 | [TK]D-Fender | ^^^ |
| 01:21.49 | fr0ggie | I have an android gingerbread LG optimus behind NAT connecting to my asterisk server (not behind NAT) and one *not* behind NAT-- the local one works-- before i started fooling with my sip.conf (which i'll paste momentarily) -- |
| 01:23.09 | fr0ggie | I could call 8000 (NAT'd phone) from 8001 or 8002 (Not NAT, android phone on 8001, sipura on 8002) and it'd fail. 8000 could call either (now its broke) |
| 01:23.53 | Posi211 | I pasted it put how to I link it back to here? |
| 01:24.28 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 01:24.57 | *** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca) |
| 01:29.38 | Posi211 | OK I think I found out how to post it. http://pastebin.com/sK7kfcL4 |
| 01:30.29 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 01:39.46 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 01:40.16 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:46.33 | [TK]D-Fender | Posi211: Ok, you have a problem with some FreePBX portion of this which is not actuall an Asterisk problem. Go to #freepbx and paste the link up in there and see if someone that is better versed with it can help you there |
| 01:46.42 | [TK]D-Fender | ~freepbx |
| 01:46.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:47.02 | Posi211 | thanks for the help |
| 01:53.54 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
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| 02:34.50 | fr0ggie | Ugh |
| 02:34.57 | fr0ggie | its a pain to upload whole config dir |
| 02:37.12 | ChannelZ | eh? |
| 02:39.19 | p3nguin | Why would you be doing that, and why would it be hard if you are doing it? |
| 02:40.59 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:50.14 | fr0ggie | http://paste.pocoo.org/show/572864/ |
| 02:50.34 | fr0ggie | first is sip.conf http://paste.pocoo.org/show/572865/ is extensions.conf |
| 02:51.24 | dijib | my my |
| 02:51.49 | dijib | ten minutes later |
| 02:53.02 | fr0ggie | was hoping to put them up for http but realized port 80 is blocked so just pastebin'd |
| 02:54.26 | dijib | im on diallup right now |
| 02:57.30 | fr0ggie | fun |
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| 04:00.46 | onixx99 | ChannelZ: hello again |
| 04:04.02 | onixx99 | i managed to revord a serie of dtmf digits my asterisk 1.8 is unable to decode into a wav file |
| 04:05.20 | onixx99 | anybody willing to give it a try with 1.8 ? the sa,e used to work for me in 1.4 and it stopped when i upgraded |
| 04:12.45 | ChannelZ | no patience I guess |
| 04:15.09 | *** join/#asterisk Yourname` (~yourname@unaffiliated/yourname/x-837320) |
| 04:15.56 | Yourname` | Hi. I know this may not be the right place to ask, but my SIP provider gave me a media and signaling IP. I'm using FreePBX and was wondering what the setting should be. No username, password because they have IP based ACL. |
| 04:17.27 | ChannelZ | Dunno how it applies to FreePBX but I assume you just need to make sure your peer for them has the right IP or hostname, and leave the secret blank.. |
| 04:21.54 | Yourname` | Ok. So what about the IPs then? |
| 04:23.22 | p3nguin | You have to set the host to the signaling address. |
| 04:28.37 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-0-91.bois.qwest.net) |
| 04:34.32 | Nugget | reverse the polarity of the neutron flow. |
| 04:38.17 | [TK]D-Fender | But Egon ... I though you said crossing the streams was baaaaaaadddd ... |
| 04:48.43 | *** join/#asterisk lwizardl (~lwizardl@c-68-62-80-172.hsd1.mi.comcast.net) |
| 04:48.46 | lwizardl | hello |
| 04:52.19 | lwizardl | I know that you can use stuff like PAP2 etc, but I am looking for other features like call recording, voice malls, call waiting music, etc. So I was wondering would I need to take for example a current computer, buy a card with 2xFXO, 2xFXS modules and then just setup the software? or is there other things that are needed |
| 04:52.21 | lwizardl | ? |
| 04:53.52 | ChannelZ | not really.. the FXO/FXS card or some other ATA is the key component though for diddling in the analog world |
| 04:54.44 | lwizardl | okay thats what i thought |
| 04:55.07 | [TK]D-Fender | lwizardl: You phrased that as though those features won't work with a PAP2. They will |
| 04:55.26 | [TK]D-Fender | lwizardl: Hoever that only overs the 2 FXS part of your later port count requirement |
| 04:55.28 | lwizardl | and I have to make sure I buy from a legit dealer, and not ebay. but otherwise i would be ready for getting my pbx setup |
| 04:55.58 | [TK]D-Fender | lwizardl: I'd advise getting a PCI based solution for the FXO, and leave the FXS to ATAs |
| 04:57.23 | lwizardl | [TK]D-Fender, okay i was looking at the 410 card |
| 04:57.41 | lwizardl | TDM410 |
| 04:59.38 | [TK]D-Fender | lwizardl: That'll do... |
| 05:02.15 | lwizardl | I figured for my small business of 2 employees that would be the best for a "starter" system and then upgrade later if ever needed |
| 05:03.11 | [TK]D-Fender | lwizardl: so 1 half-loaded card, 1 ATA. Set to go and some room to expand |
| 05:06.50 | lwizardl | so you would recommend using 2 fxo modules on that card, and then a pap2 for the fxs ? |
| 05:07.05 | [TK]D-Fender | yes |
| 05:07.33 | lwizardl | over having the fxo/fxs on a single card |
| 05:08.00 | [TK]D-Fender | PCI des a functioally better job for the FXOs. For FXS, ATAs do just fine, cost far less per port, place less configuration and system load on your server and are more flexible and easily moved. |
| 05:08.08 | [TK]D-Fender | does* |
| 05:08.39 | [TK]D-Fender | Odds are you may move phones around a building but the server will stick around where your lines come in. |
| 05:08.59 | [TK]D-Fender | Having to directly wire hones into a centra server can be a PITA. |
| 05:09.09 | lwizardl | okay that makes sense, and i would just need to setup the IP addresses for the pap2 device |
| 05:09.24 | [TK]D-Fender | And as the evening rolls on my typing heads right on downhill |
| 05:09.41 | [TK]D-Fender | lwizardl: DHCP works just fine... generally no real need to go fixed. |
| 05:09.59 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-fzsbqxwcvtbjagrp) |
| 05:13.57 | lwizardl | and would an unlocked rtp300 work as the fxs ? |
| 05:14.34 | lwizardl | it used to be something I had when I was a vonage subscriber but canceled and used the unlock tools to fix it |
| 05:18.42 | [TK]D-Fender | ATA is an ATA... if its unlocked you should be able to set it up |
| 05:27.01 | lwizardl | cool |
| 05:27.34 | lwizardl | yeah I have access to the voip section which if was still vonage would be locked out |
| 05:27.38 | *** join/#asterisk din3sh (~din3sh@ADSL-TPLUS-71-148.telecomplus.net) |
| 05:28.39 | din3sh | hi everybody, has anyone had "486 BUSY Here" randomly on phones without any DND state? |
| 05:31.49 | lwizardl | thanks [TK]D-Fender |
| 05:32.58 | [TK]D-Fender | lwizardl: You're welcome |
| 05:34.17 | [TK]D-Fender | din3sh: Either DND (full or selective), or call being rejected by the user on the phone (eg "Reject" soft-key on a Polycom phone), or the phone is not configured to take that many calls. |
| 05:35.30 | Yourname` | p3nguin: Thanks! |
| 05:48.13 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 05:48.28 | tapout | I'm using CSipSimple on android based phones, and love it. Is there a good sip app that you guys use for iphones? I don't have an iphone to try them out... i'm helping my cousin over the phoen |
| 05:48.29 | tapout | phone |
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| 06:29.33 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:29.35 | schmidts | good morning |
| 06:34.07 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 06:56.53 | *** join/#asterisk stix (~stix@193.89.191.209) |
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| 07:37.22 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 07:37.25 | qakhan | hi all |
| 07:37.58 | qakhan | in which dir asterisk installed? |
| 07:38.26 | qakhan | like in windows every program get installed in program files.... |
| 07:40.46 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:42.39 | kaldemar | qakhan: depends on how you install. however, it is not a single directory. |
| 07:42.56 | schmidts | qakhan it depends on your system but "normally" you should find the binary in /usr/sbin/asterisk and the config files in /etc/asterisk |
| 07:43.25 | kaldemar | and modules and man pages and init script and then some. |
| 07:43.31 | Erwan | sounds in /var/lib recordings in /usr/lib |
| 07:43.32 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 07:43.43 | schmidts | Erwan or in /var/spool |
| 07:43.45 | kaldemar | "make install -n" will show what the installation does. |
| 07:45.46 | kaldemar | in case of a package install, it depends on the used package manager how to find out. |
| 07:47.28 | qakhan | i installed asterisk with ./configure them make menuselect then make then make install then make samples then make config |
| 07:52.29 | qakhan | schmidts in /usr/sbin/ asterisk is a file |
| 07:52.33 | qakhan | not a folder |
| 07:52.42 | qakhan | is it correct? |
| 07:54.49 | kaldemar | qakhan: what are you trying to do? |
| 07:55.17 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 07:55.40 | qakhan | read this |
| 07:55.41 | qakhan | To explicitly specify where to look for Asterisk, use the "--with-asterisk=" and "--with-asterisk-conf=" options. |
| 07:55.42 | qakhan | For example, if your Asterisk is installed in /usr/local/asterisk/1.6.1.10, use: |
| 07:56.18 | EmleyMoor | qakhan: Tried "which asterisk"? |
| 07:56.53 | qakhan | i am using asterisk 1.4.38 |
| 07:57.15 | EmleyMoor | That's not the question "which" answern |
| 07:57.23 | EmleyMoor | answers |
| 07:58.16 | EmleyMoor | Not "what version of" but "where is" |
| 07:58.41 | qakhan | it shows /usr/sbin/asterisk |
| 07:59.35 | EmleyMoor | qakhan: What is your purpose for wanting to know? What exactly are you trying to do? |
| 08:00.24 | qakhan | i m trying to install Asterisk Connector Bridge |
| 08:02.01 | EmleyMoor | Right - well, --with-asterisk=/usr/sbin may do what you want then... but I am not familiar with Asterisk Connector Bridge |
| 08:13.43 | qakhan | EmleyMoor read the message |
| 08:13.45 | qakhan | configure: Asterisk configuration |
| 08:13.45 | qakhan | ./configure: line 18554: /usr/sbin/asterisk/sbin/asterisk: Not a directory |
| 08:19.38 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 08:22.18 | *** join/#asterisk timahvo1 (~rogue@197.177.156.1) |
| 08:26.20 | kaldemar | qakhan: have you read http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP ? |
| 08:29.18 | kaldemar | it says that by default it installs the asterisk modules in /usr/lib/asterisk/modules. |
| 08:29.32 | kaldemar | quite a poor page btw. |
| 08:49.30 | qakhan | yes i am following that page |
| 09:01.53 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-vkwaeqparjtrutxx) |
| 09:03.42 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:13.17 | qakhan | i am getting this message |
| 09:13.18 | qakhan | chan_sip.c:8172 sip_reg_timeout: -- Registration for '1002@192.168.4.23' timed out, trying again (Attempt #146) |
| 09:17.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:28.58 | *** join/#asterisk jkroon (~jkroon@dsl-244-21-199.telkomadsl.co.za) |
| 10:03.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:13.05 | *** join/#asterisk Yourname` (~yourname@unaffiliated/yourname/x-837320) |
| 10:13.46 | Yourname` | Hi. I have a trunk that's set up over VPN. The call invites are showing my public IP, so the VPN is rejecting the calls. How can I make it so all calls that are supposed to go through the VPN based on dial pattern automatically change the IP? |
| 10:27.26 | *** join/#asterisk krotos (~d3v1l@host124-100-dynamic.15-87-r.retail.telecomitalia.it) |
| 10:27.30 | krotos | hi all guy |
| 10:30.01 | *** join/#asterisk mahiti-irc (~mahiti@119.30.39.35) |
| 10:51.13 | krotos | my provider say to me that i've to check "Frame per Tx" in g729 trasmission |
| 10:51.18 | krotos | where i can check this on asterisk? |
| 10:52.44 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 11:10.39 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-126.telkomadsl.co.za) |
| 11:11.53 | jkroon | hi guys, given that I have two SIP/ accounts that needs to have a "combined" BLF hint (ie, if either one of the two SIP/ channels has at least one active call BLF should report BUSY, or if at least one is ringing RINGING and only if neither one of the two has any action should it be IDLE - any ideas on how to do this? |
| 11:13.13 | jkroon | normally I'd just have a 123,hint,SIP/123 entry in my hints context... |
| 11:24.57 | *** join/#asterisk gusto (~gusto@nrbg-4dbe136b.pool.mediaWays.net) |
| 11:25.02 | gusto | hi |
| 11:25.12 | gusto | what is the difference between peers and registry? |
| 11:25.17 | gusto | on sip |
| 11:26.56 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 11:38.51 | kaldemar | gusto: registry is outgoing registrations from your asterisk, defined with register statements in sip.conf. peers are devices you have defined in sip.conf. |
| 11:39.31 | krotos | hi kaldemar ;)how's going on? |
| 11:39.50 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-0-91.bois.qwest.net) |
| 11:45.32 | *** join/#asterisk albertoandrade (~albertoan@187.59.73.81) |
| 11:48.08 | kaldemar | krotos: hi. same old, same old. |
| 11:49.26 | gusto | kaldemar: i have not defined any devices in sip.conf |
| 11:51.58 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:54.58 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:55.00 | kaldemar | gusto: peers, users and friends by type are called devices. |
| 11:56.25 | krotos | kaldemar: ;) "Spring is here", i'm waiting for Digium phone ..i'm little bit curious |
| 11:57.17 | lifemadison | gusto: basically, a registration is a method that an entity tells another entity where it lives on the network |
| 11:57.55 | lifemadison | for example, a device registering to Asterisk, is a device telling Asterisk *where* to call it. The "how" and authentication is done via another mechanism, in Asterisk, via the configuration of a friend or peer. |
| 11:58.23 | lifemadison | calls from the device are matched either by username using the user type in sip.conf, or via IP address with the peer type in sip.conf |
| 12:04.57 | *** join/#asterisk onixx99 (1000@bas1-stetherese38-2925306218.dsl.bell.ca) |
| 12:07.24 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:18.43 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:36.45 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:38.49 | *** join/#asterisk gusto (~gusto@nrbg-4dbe3153.pool.mediaWays.net) |
| 12:39.14 | gusto | does someone have a working voip telephone number? |
| 12:39.22 | gusto | i mean with register => and so on? |
| 12:39.52 | gusto | because i was getting Forbidden - wrong password on authentication for REGISTER for 'DcZETjN1TbyV3aJ5OgHh' to 'sipproxy.endesha.be-converged.com' |
| 12:39.55 | gusto | all the time |
| 12:40.21 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 12:40.26 | j0b | its pretty much says it |
| 12:40.32 | gusto | and i used the sip.conf.sample |
| 12:40.46 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-0-91.bois.qwest.net) |
| 12:40.49 | gusto | with register => as only change |
| 12:41.08 | gusto | and the username and password copied and pasted to the conf |
| 12:41.22 | gusto | so it is not possible that a password or username would be wrong |
| 12:41.47 | kaldemar | show the register statement you have now, masking the password |
| 12:42.09 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:42.54 | gusto | register => DcZETjN1TbyV3aJ5OgHh:<password>:<telnr>@sipproxy.endesha.be-converged.com |
| 12:43.25 | gusto | and register => <telnr>:<password>:DcZETjN1TbyV3aJ5OgHh@sipproxy.endesha.be-converged.com |
| 12:43.32 | gusto | both with the same results |
| 12:43.43 | gusto | Forbidden - wrong password on authentication for REGISTER for |
| 12:43.43 | gusto | '4991131042466' to 'sipproxy.endesha.be-converged.com' |
| 12:43.52 | gusto | 499... is the telnr |
| 12:44.21 | kaldemar | you should have "register => DcZETjN1TbyV3aJ5OgHh:<password>@sipproxy.endesha.be-converged.com/<telnr> |
| 12:44.26 | gusto | sipproxy.endesha.be-converged.com is a SRV record and SRV is enabled of course |
| 12:44.43 | kaldemar | the syntax is register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 12:44.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:44.46 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:44.59 | gusto | and why should extension be the telnr? |
| 12:45.05 | kaldemar | from which a simplified version in this case would be register => username:secret@host/callbackextension |
| 12:45.28 | kaldemar | gusto: that's the number you want they to call at your end when you get a call, right? |
| 12:45.37 | gusto | ok |
| 12:47.45 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 12:48.29 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 12:49.23 | gusto | well |
| 12:50.16 | gusto | so it was everything all right then |
| 12:55.12 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
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| 12:59.24 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 12:59.59 | bulkorok | hi... where can I find more informations about callfiles !? it's not in docs/callfiles.txt anymore |
| 13:00.59 | [TK]D-Fender | ~book |
| 13:00.59 | 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 |
| 13:01.01 | [TK]D-Fender | ^^^ |
| 13:01.07 | [TK]D-Fender | ~wikis |
| 13:01.08 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:01.10 | [TK]D-Fender | ^^^ |
| 13:01.53 | bulkorok | there is nothing in the book about callfiles.. |
| 13:02.06 | bulkorok | (or I didn't find) |
| 13:06.57 | lifemadison | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-14-SECT-5.html |
| 13:08.22 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+FIles |
| 13:08.38 | [TK]D-Fender | lifemadison, Seems it didn't make it's way into the 3rd ed... |
| 13:09.57 | bulkorok | ah... there is the wiki-entry... thx! and yes... it's not in the 3rd ed |
| 13:10.21 | kaldemar | callfiles are so 2nd edition. |
| 13:10.27 | lifemadison | bulkorok: true -- I'm adding a note to cover it in the 4th edition |
| 13:10.31 | lifemadison | kaldemar: that :) |
| 13:10.38 | bulkorok | perfect :) |
| 13:10.49 | lifemadison | but ya, no one uses callfiles anymore |
| 13:10.51 | lifemadison | heh |
| 13:10.59 | bulkorok | mmh.. |
| 13:11.01 | lifemadison | you either use AMI or the originate command |
| 13:11.13 | lifemadison | callfiles are kind of the old and busted way |
| 13:11.27 | bulkorok | I want to send faxes with SendFAX |
| 13:11.31 | lifemadison | if you're triggering a call like that via a script, you should be using AMI |
| 13:11.43 | lifemadison | if you're not doing it programattically, then you should just use the originate applications via the CLI or dialplan |
| 13:11.53 | lifemadison | see above |
| 13:11.59 | bulkorok | yeah... |
| 13:12.06 | bulkorok | I'm just thinkong ^^ |
| 13:12.12 | lifemadison | think about how to use AMI |
| 13:12.15 | bulkorok | lol... thinking |
| 13:12.24 | bulkorok | thinkong is great :) |
| 13:15.44 | bulkorok | in the callfiles I can set Maxretries... How can I manage this with AMI Originate!? |
| 13:16.16 | bulkorok | a new variable in the dialplan with a loop!? |
| 13:16.36 | bulkorok | hates implemnting fax... |
| 13:16.40 | lifemadison | in your program that is calling AMI probably |
| 13:17.41 | [TK]D-Fender | lifemadison, I'm having trouble finding a 3rd ed page that actually shows the AMI Originate command.... can you point it out? |
| 13:17.57 | *** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za) |
| 13:18.40 | [TK]D-Fender | lifemadison, Not seeing the old appendix minimal breakdown even... |
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| 13:21.21 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:27.04 | *** join/#asterisk TrashAmbishion (~sdicb@200.55.187.243) |
| 13:27.16 | TrashAmbishion | hi everybody |
| 13:27.29 | TrashAmbishion | somebody speak spanish |
| 13:27.35 | TrashAmbishion | i need help |
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| 14:01.22 | bobb_WU | anybody around? |
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| 14:04.16 | pabelanger | ~ask |
| 14:04.16 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:05.10 | *** join/#asterisk bio-tty (~coien@cm-84.211.83.24.getinternet.no) |
| 14:05.15 | bio-tty | if INVITE has offer with c=IN IP6 ... and answerer does only have IP4, then what response status should be used by the UAS ? |
| 14:05.21 | *** join/#asterisk bmg505 (~leon@196-209-10-223.dynamic.isadsl.co.za) |
| 14:06.43 | bobb_WU | do you have a link to postgres table create statements that are meant for 1.8? i'm only finding mysql stuff on google |
| 14:09.11 | kaldemar | bobb_WU: what was wrong with the ones i gave you esterday? |
| 14:09.28 | bobb_WU | oh i got pulled away from the computer and didn't see the link |
| 14:09.31 | bobb_WU | sorry about that |
| 14:09.39 | kaldemar | http://svn.digium.com/svn/asterisk/tags/1.8.9.3/contrib/realtime/postgresql/realtime.sql |
| 14:09.43 | bobb_WU | thanks! |
| 14:10.04 | bobb_WU | oh that is absolutely perfect |
| 14:10.06 | kaldemar | you'll find more table structures under the realtime dir for other DB's. |
| 14:10.09 | *** join/#asterisk din3sh (din3sh@41.136.240.235) |
| 14:11.28 | bobb_WU | ok and one more question: any advice on how to set up redundancy between asterisk and sip trunks? |
| 14:12.40 | bobb_WU | i ask because we have a primary SIP controller that is the gateway to the proprietary voice side which failed a week ago, so i am now looking into how to make the secondary controller into a failover gateway |
| 14:14.08 | KavanS | is 1.4 known to be buggy w/call recording? |
| 14:14.28 | KavanS | having some issues with transfered calls and recording on 1.4 |
| 14:17.52 | kaldemar | bobb_WU: that's usually done in dialplan. you can check DIALSTATUS variable after a Dial command and use another gateway if the call failed for some reason. |
| 14:19.39 | KavanS | bobb_WU, check this out - http://mikepultz.com/tag/asterisk-2/ |
| 14:19.44 | KavanS | found that last night... |
| 14:20.29 | bobb_WU | thank you both, this is exactly what i need again |
| 14:21.28 | TrashAmbishion | kaldemar question, i got Ubuntu 11 is necessary download repositor to install asterisk??? |
| 14:22.52 | TrashAmbishion | kaldemar ??????????? |
| 14:24.04 | kaldemar | TrashAmbishion: i don't understand you. |
| 14:25.48 | cloakable | attempts to translate |
| 14:26.44 | cloakable | kaldemar: TrashAmbishion has Ubuntu 11 (.04 or 10), and is presumably asking if he needs to add the Asterisk repository to install. |
| 14:27.30 | kaldemar | to install what? :) |
| 14:28.10 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 14:28.31 | *** join/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 14:28.45 | cloakable | Asterisk, I guess! |
| 14:28.56 | *** part/#asterisk ajg (~inaki@wah.dreamchaos.net) |
| 14:29.18 | kaldemar | ubuntu does have some version of asterisk in its own repositories. you can use those, add the digium repository or some other repository that has asterisk or install from source, which does not require any repositories to be added. |
| 14:29.52 | cloakable | Just a bucket of manual dependency resolving :) |
| 14:30.37 | kaldemar | or a few apt commands if the default repositories are there. :) |
| 14:30.59 | cloakable | TrashAmbishion: You can install Asterisk from the Ubuntu repositories, use the Digium ones to get the latest stable, or install from source. |
| 14:31.24 | cloakable | kaldemar: True, but would apt-get build-dep get all the dependencies for the latest Asterisk? ;) |
| 14:32.42 | kaldemar | cloakable: probably. but google gives a gazillion results for commands that install them with a simple query. |
| 14:33.03 | cloakable | Mmmm |
| 14:33.19 | cloakable | I plan on using the Digium repos myself :) |
| 14:33.58 | Andee | ubuntu asterisk-addons are all borked |
| 14:34.00 | *** join/#asterisk asilva (~somebody@c95336b9.virtua.com.br) |
| 14:34.05 | asilva | Hello to everyone!! |
| 14:34.06 | Andee | last time i check |
| 14:34.20 | cloakable | Well, digium it is then :) |
| 14:35.23 | asilva | Can someone help me out on something here.. i have a tandberg videophone, the first line is to the vc system, and the second line is configured for asterisk/SIP. first problem, from time to time the videophone looses its registration(unregister) and also keeps appearing this messages on asterisk cli handle_request_subscribe: Sending fake auth rejection for device <sip:VF-GRC-Rodrigo@vcs.unesp.br>;tag=004b5680e50eba25 . ifsomeone could help me o |
| 14:35.24 | asilva | ut i appreciate! |
| 14:38.10 | asilva | anyone ? |
| 14:45.12 | asilva | ?? |
| 14:48.06 | chuckf | does it loose its connectoin to the vc system as well? |
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| 15:14.04 | bobb_WU | is there an example realtime peers table def somewhere? i don't see it in http://svn.digium.com/svn/asterisk/tags/1.8.9.3/contrib/realtime/postgresql/realtime.sql |
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| 15:35.04 | kaldemar | bobb_WU: http://svn.digium.com/svn/asterisk/tags/1.8.9.3/contrib/realtime/mysql/ |
| 15:36.08 | bobb_WU | guess i'll have to remove all the enums, thanks kaldemar |
| 15:36.55 | bobb_WU | hmmm i think i did that and saved it somewhere already |
| 15:38.25 | krotos | there is another way that is not iptables , for making asterisk listen on two ports ? (5061 and 5060)? |
| 15:38.37 | Qwell | krotos: no |
| 15:38.54 | [TK]D-Fender | krotos, Or set up a proxy in front |
| 15:39.14 | krotos | ok ;) thankyou |
| 15:40.15 | *** join/#asterisk kareena (~kareena@unaffiliated/kareena) |
| 15:40.21 | kareena | hi |
| 15:40.29 | bobb_WU | kaldemar: is there a way i can contribute this to asterisk? i did go through and create types so the column values can be enumerated |
| 15:40.33 | kareena | any one familiar with vocalcom to asterisk? |
| 15:41.12 | kareena | who know hermes pro? |
| 15:41.40 | Qwell | ~polls |
| 15:41.40 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 15:42.33 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:43.22 | kareena | ~ask |
| 15:43.22 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:44.45 | kareena | i want to connect hermes pro dialogic card with asterisk digium card E1 |
| 15:45.01 | [TK]D-Fender | kareena, You have our premission. |
| 15:45.27 | [TK]D-Fender | permission even. |
| 15:48.21 | kareena | thank you |
| 15:49.39 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 15:51.41 | cloakable | haha |
| 15:53.56 | [TK]D-Fender | kareena, http://www.urbandictionary.com/define.php?term=metaquestion |
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| 15:56.14 | kareena | its hard to explaine in english :( |
| 15:56.19 | kareena | no one speak french? |
| 15:56.22 | [TK]D-Fender | Yes |
| 15:56.36 | kareena | u speak french? |
| 15:56.43 | [TK]D-Fender | Evidement |
| 15:56.47 | kareena | cool |
| 15:56.52 | kareena | je peux parler en francais? |
| 15:57.07 | kareena | j'ai un serveur vocalcom |
| 15:57.27 | [TK]D-Fender | kareena, Si tu trouves que t'es incapable de bien expliqer votre problem... y-a pas de point de t'a-mene ici non? :p |
| 15:57.35 | kareena | ok |
| 15:57.43 | [TK]D-Fender | kareena, Juste a-dire... |
| 15:57.43 | kareena | bon ecoute |
| 15:57.46 | [TK]D-Fender | kareena, Alors.... |
| 15:58.04 | kareena | j'ai un serveur vocalcom etait brancher a une gateway |
| 15:58.19 | kareena | j'ai elever la gateway et j'ai placer un serveur asterisk avec une carte digium |
| 15:58.37 | kareena | j'ai configurer le fournissuer d'access sur asterisk |
| 15:58.59 | kareena | maintenant quand je passe les appel sur vocalcom les appel arrive pas sur asterisk |
| 15:59.38 | [TK]D-Fender | kareena, VC etait direct au E1 avant de mettre * dans milieu? |
| 16:00.08 | [TK]D-Fender | kaldemar, Alors * c'est une intermediare maintenant? |
| 16:00.13 | [TK]D-Fender | kareena, ^ |
| 16:04.34 | kareena | oui |
| 16:04.42 | kareena | c'etait connecter a deux E1 |
| 16:07.57 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:08.12 | [TK]D-Fender | * c'est mainenant devant just une.... ou les deux? |
| 16:08.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:08.44 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
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| 16:09.16 | *** join/#asterisk bmg505 (~leon@196-209-84-176.dynamic.isadsl.co.za) |
| 16:10.53 | kareena | non directement sur asterisk avec la carte digium |
| 16:12.32 | [TK]D-Fender | kareena, Avec les 2 E1? |
| 16:13.15 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:13.43 | *** part/#asterisk TJ_stamm (~TJ_stamm@c-98-213-150-32.hsd1.il.comcast.net) |
| 16:14.00 | kareena | non j'ai une carte asterisk 1 E1 |
| 16:14.44 | kareena | la gateway etait quitum |
| 16:15.02 | [TK]D-Fender | <kareena> c'etait connecter a deux E1 |
| 16:15.07 | [TK]D-Fender | kareena, No-claire... |
| 16:15.33 | [TK]D-Fender | BRB |
| 16:16.08 | kareena | avec la QUINTUM TENOR DX2060 2xPRI |
| 16:16.15 | kareena | etait connecter a deux E1 |
| 16:16.24 | kareena | mais avec la carte digium juste 1 E1 |
| 16:17.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-092.mobile.uci.edu) |
| 16:18.45 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 16:19.46 | Dovid | which command lets me play a sound file and jump in the file? |
| 16:22.00 | [TK]D-Fender | kareena, ok, si t'as un problem de connectivity veuiller aller dans CLI aves "sip set debug on" et faire un pastebin d'une appel qui ne marche-pas. |
| 16:22.01 | [TK]D-Fender | ~pb |
| 16:22.01 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:22.02 | [TK]D-Fender | ^^^ |
| 16:22.13 | [TK]D-Fender | Dovid, PlaybackControl |
| 16:22.22 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-092.mobile.uci.edu) |
| 16:22.43 | kareena | ok |
| 16:22.59 | Dovid | thanks. |
| 16:23.28 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 16:32.36 | woleium | 'ning :-) |
| 16:33.06 | woleium | has anyone had probeems with * emiling old vm messages? |
| 16:36.50 | [TK]D-Fender | * doesn't e-mail old mesages. It e-mails ones that arrive. |
| 16:37.02 | [TK]D-Fender | And only if you set it up to. And your MTA |
| 16:39.45 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
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| 16:52.48 | Kobaz | what would be a possible reason why when attended transferring someone, the caller will not get any audio once the call is transfered (and this is all sip, no nat, and canreinvite=no) |
| 17:00.43 | [TK]D-Fender | Kobaz, Same pysical LAN, no routing between, just switches? |
| 17:04.03 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 17:07.39 | Kobaz | just switches |
| 17:07.47 | Kobaz | and it's *random* |
| 17:07.56 | Kobaz | and they are nice polycom phones |
| 17:08.45 | Naikrovek | sounds like firmware bug perhaps |
| 17:09.00 | Kobaz | hmm, could be |
| 17:09.05 | Naikrovek | iono |
| 17:09.06 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 17:09.08 | Kobaz | i think i'm running like 3.2 on this box |
| 17:12.38 | woleium | [TK]D-Fender: sorry, I gat called afk |
| 17:13.00 | woleium | That's what I thought, but this system is sending old messages out |
| 17:13.26 | woleium | also, they are being sent without the envelope, but envelope is set |
| 17:13.27 | [TK]D-Fender | woleium, Or it's finally catching up to old messages stuck in queue |
| 17:13.45 | [TK]D-Fender | because * doesn't look back on the apst for this |
| 17:13.49 | [TK]D-Fender | past* |
| 17:13.51 | woleium | I believe that * was set to send messages for a while before the MTA was configured correctly |
| 17:14.09 | Qwell | is the time configured correctly? maybe the mail queue held them until the time was == now |
| 17:14.11 | woleium | so I guess it's probably a postfix queue thing |
| 17:15.04 | woleium | to the `postqueue` :-) |
| 17:16.03 | woleium | "Mail queue is empty" :-( |
| 17:17.10 | woleium | Qwell: I thought it may be a clock thing. It's set correctly now, but who knows what it was set to when the messages were recorded. |
| 17:17.32 | woleium | I guess I could look at timestmaps in var/spool |
| 17:37.22 | *** join/#asterisk VinceAntw (~VinceAntw@91.176.2.140) |
| 17:38.12 | VinceAntw | Hi everyone, I have a DAHDI problem, can anyone help me? |
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| 18:00.53 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:01.51 | leifmadsen | ~ask |
| 18:01.52 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:02.09 | leifmadsen | oh nevermind, he waited a whole minute |
| 18:05.14 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 18:13.18 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 18:14.32 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
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| 19:15.02 | *** join/#asterisk Vince-0 (~AndChat@41-132-156-188.dsl.mweb.co.za) |
| 19:16.21 | *** join/#asterisk VinceAntw (~VinceAntw@91.176.2.140) |
| 19:17.41 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 19:23.19 | VinceAntw | Hi everybody, I have a problem with DAHDI, can anyone help me? |
| 19:24.05 | navaismo | ~ask |
| 19:24.05 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:24.15 | [TK]D-Fender | VinceAntw, If you actually stick around for more than a minute and actually show us the problem... perhaps... |
| 19:25.43 | VinceAntw | I have a TDM400P analog card and running asterisk 1.8.10, dahdi 2.6.0 |
| 19:27.53 | VinceAntw | when I make a call with a telephone connected to the card and the other party disconnects it plays a beep-beep-beep tone but that tone is much to fast |
| 19:28.15 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:29.11 | VinceAntw | I live in belgium and it should be 425/500, 0/500 but it is much much faster, something like 425/125, 0/125 |
| 19:31.12 | VinceAntw | in /etc/dahdi/system.conf I have set the zone to "be" and the wctdm driver is loaded with opermode=BELGIUM |
| 19:32.30 | VinceAntw | (when I make a SIP to SIP call, the beep sound is correct) |
| 19:33.35 | kaldemar | set your country in indications.conf |
| 19:36.57 | urvg4 | hi,got a problem with moh first caller starts at the beginning of the moh but subsequent callers get the middle or tail end of the moh |
| 19:37.35 | urvg4 | how do I fix this? |
| 19:38.37 | VinceAntw | kaldemar: in that file the country is also set to "be" |
| 19:38.41 | *** join/#asterisk SteveWilliams (~chatzilla@59.162.182.218) |
| 19:39.03 | SteveWilliams | Hi All! I have a noob question. I am trying to write a small phpagi script which when called by a browser, calls a phone number. please help! here is the pastebin http://pastebin.com/TiZirehw |
| 19:39.22 | *** part/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 19:39.34 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 19:41.31 | [TK]D-Fender | SteveWilliams, You understanding of AGI is backwards |
| 19:42.00 | [TK]D-Fender | SteveWilliams, AGI is a means of processing a CALL. You can't just call a script from a webrowser like that. AGI means taking IO from *..... |
| 19:42.32 | [TK]D-Fender | SteveWilliams, Asterisk calls teh AGI dialplan app which in turn executes some script redirecting STDIN, STDOUR, and STDERR. |
| 19:42.37 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 19:42.59 | [TK]D-Fender | SteveWilliams, And that script is again calling a dialplan app that's supposed to take a caller that is already processing and then call out. |
| 19:43.27 | [TK]D-Fender | SteveWilliams, What you seem to be looking for is a way to trigger * to call out all by itself. That would be done with an AMI Originate or a Call File. |
| 19:44.02 | SteveWilliams | [TK]D-Fenderthanks for clarifying. i will google for it now. thanks again. |
| 19:54.00 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
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| 19:58.39 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:58.39 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:03.17 | *** part/#asterisk SteveWilliams (~chatzilla@59.162.182.218) |
| 20:04.52 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 20:06.20 | *** topic/#asterisk by mjordan -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:14.30 | *** join/#asterisk jsjc (~Adium@51.Red-79-146-23.dynamicIP.rima-tde.net) |
| 20:16.53 | *** join/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com) |
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| 20:26.15 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 20:38.02 | VinceAntw | is it rude to repost my question when it is not solved? |
| 20:40.00 | p3nguin | If you have posted it in the last hour or so, or if there was no significant window scroll since the last post, I wouldn't do it again now. |
| 20:40.16 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:40.23 | p3nguin | If there was scroll and/or ample time has passed, go ahead and say it again. |
| 20:40.52 | VinceAntw | ok thank you |
| 20:41.23 | ChrisInSydney | p3nguin, but then again, others, like myself, may have joined and missed out on the challenge |
| 20:41.32 | VinceAntw | I have a TDM400P analog card and running asterisk 1.8.10, dahdi 2.6.0 |
| 20:41.42 | ChrisInSydney | not that I am offering any help |
| 20:41.49 | VinceAntw | when I make a call with a telephone connected to the card and the other party disconnects it plays a beep-beep-beep tone but that tone is much to fast |
| 20:42.08 | ChrisInSydney | I think thats a regional tone thingy |
| 20:42.08 | VinceAntw | I live in belgium and it should be 425/500, 0/500 but it is much much faster, something like 425/125, 0/125 |
| 20:42.42 | VinceAntw | in /etc/dahdi/system.conf and /etc/asterisk/indications.conf I have set the zone to "be" and the wctdm driver is loaded with opermode=BELGIUM |
| 20:42.59 | VinceAntw | (when I make a SIP to SIP call, the beep sound is correct) |
| 20:43.08 | *** part/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 20:43.19 | ChrisInSydney | hmm |
| 20:43.40 | ChrisInSydney | SIP is OK, but the DAHDI isn't |
| 20:43.48 | VinceAntw | yes |
| 20:45.41 | ChrisInSydney | I think I have seen this elsewhere |
| 20:45.46 | ChrisInSydney | some time ago |
| 20:46.13 | VinceAntw | oh thats good news (I think) |
| 20:46.58 | ChrisInSydney | http://jkroon.blogs.uls.co.za/it/voip/south-africa-and-isdn |
| 20:47.14 | ChrisInSydney | has some stuff in there |
| 20:48.15 | ChrisInSydney | Hope it helps |
| 20:48.35 | ChrisInSydney | must run. paid work to do :-) |
| 20:48.49 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 20:49.12 | VinceAntw | ok thanks |
| 20:49.31 | VinceAntw | I will check that out |
| 20:50.01 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 20:50.52 | *** join/#asterisk johno12345 (~chatzilla@cpc5-rawt2-2-0-cust25.10-2.cable.virginmedia.com) |
| 20:51.00 | *** join/#asterisk ^^netmax (~netmax@is.linux-administrator.com) |
| 20:51.03 | johno12345 | evening |
| 20:51.45 | johno12345 | wonder if anyone could help... |
| 20:52.00 | navaismo | ~ask |
| 20:52.00 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:52.16 | johno12345 | its probably me being stupid, but i'm trying to setup a mock SIP provider box |
| 20:52.41 | johno12345 | so that i can connect up a second box to act as a local PBX |
| 20:53.04 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:53.07 | johno12345 | i've got extensions created on the first box, trunk setup to connect to said extension on the second |
| 20:53.28 | johno12345 | i call the pseudo number and it goes down the trunk |
| 20:53.37 | johno12345 | but callerid isn't being transferred |
| 20:54.04 | johno12345 | Received an unknown call with DID set to is the message in the console |
| 20:54.28 | johno12345 | anyone any ideas where i'm going wrong |
| 20:55.25 | johno12345 | console on the second box that is |
| 20:55.35 | p3nguin | johno12345: http://pastebin.com/Ag7tknm2 |
| 20:55.57 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:56.27 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:56.27 | johno12345 | and that goes in sip.conf? |
| 20:56.36 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 20:56.37 | johno12345 | on each box that is |
| 20:59.01 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 20:59.04 | cj | moo |
| 20:59.14 | cj | where should I go to register a did |
| 20:59.18 | cj | or figure out what that means? |
| 20:59.37 | cj | carrar: you know these things, right? |
| 21:00.12 | cj | carrar: my customer wants voice mail and centurylink told them to talk to me, since I stoled their account from clink. |
| 21:01.02 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 21:01.03 | p3nguin | Do you need a DID, or do you need voice mail? They are completely different things. |
| 21:01.06 | cj | in order to route the call to the clink circuit for a few rings and then switch over to the vm system on PICKUPFAIL events, I hear I need to blah blah DID blah blah |
| 21:01.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:01.36 | cj | I can handle voicemail at sip0.colliertech.org |
| 21:01.52 | cj | or add a cname for vm.colliertech.org or whatever |
| 21:02.20 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 21:02.45 | grandpapadot | Hey guys, is there a way to tune sip fax detection? About 1 out of every 10 calls gets the fax tones erroneously, 1.8.latest ... |
| 21:03.14 | p3nguin | A DID is a phone number. |
| 21:03.22 | p3nguin | ~did |
| 21:03.22 | infobot | somebody said did was Direct Inward Dialing, or just a phone number |
| 21:04.28 | fprior | Hi all: I've one * 1.4 box than use LinkSys SPA400 as FXO gateway; I would update to * 1.8; spa400 are not compatibles with 1.8 and I won't waste this HW. My idea is leave one server with 1.4 as "spa400 proxy" and forward all calls to 1.8 box. Is a better/good solution ? |
| 21:04.51 | [TK]D-Fender | fprior: What gives you the impression that it won't work on 1.8? |
| 21:06.18 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:06.43 | fprior | [TK]D-Fender, unfortunately I've tried. spa400 is discontinued. |
| 21:07.32 | [TK]D-Fender | What does "discontinued" have to do with not working with *? This sounds outright crazy |
| 21:07.38 | p3nguin | If it speaks SIP, it isn't incompatible with Asterisk. |
| 21:08.12 | johno12345 | anyone any ideas on my SIP trunk to extension issue? |
| 21:08.32 | p3nguin | There's no such thing as a SIP trunk. |
| 21:08.50 | p3nguin | It's just a peer. I gave you a sample configuration on asterisk-to-asterisk via sip. |
| 21:08.57 | Qwell | ~book |
| 21:08.57 | 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 |
| 21:09.14 | *** join/#asterisk albertoandrade (~albertoan@187.112.141.167) |
| 21:09.45 | johno12345 | ok peer then, i'm using freepbx which calls them trunks - as do most providers |
| 21:10.03 | p3nguin | ~freepbx |
| 21:10.04 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:10.05 | Qwell | johno12345: Then you need to ask in #freepbx |
| 21:10.41 | johno12345 | thanks I will do... |
| 21:10.43 | *** join/#asterisk Russ (~russ@67.139.9.146) |
| 21:12.35 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:12.52 | fprior | p3nguin, [TK]D-Fender : we have already discuss spa400 problem here, search 20:00.31 in http://ibot.rikers.org/%23asterisk/20120118.html.gz |
| 21:13.17 | p3nguin | (1607.38) <p3nguin> If it speaks SIP, it isn't incompatible with Asterisk. |
| 21:13.26 | p3nguin | So... does it do SIP or not? |
| 21:14.29 | Qwell | So, the answer didn't change from last time. What's your point? |
| 21:14.56 | fprior | p3nguin, yes, spa400 does SIP |
| 21:15.36 | p3nguin | Then it still works with Asterisk. Asterisk speaks SIP. The SPA-400? speaks SIP. They will work together. |
| 21:16.16 | fprior | @Qwell, my question is "My idea is leave one server with 1.4 as "spa400 proxy" and forward all calls to 1.8 box. Is a better/good solution ?" |
| 21:16.27 | Qwell | no, that's a stupid idea |
| 21:16.51 | fprior | p3nguin: as described in past "I can do one call fine. the second call return "Got SIP response 503 "Service Unavailable" back from 192.168.0.xxx:5060", "SIP/spa400e-000001b2 is circuit-busy " |
| 21:17.42 | fprior | @Qwell, why ? |
| 21:17.55 | Qwell | ~asterisk versions |
| 21:17.56 | Qwell | ~asteriskversions |
| 21:18.07 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:18.12 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:19.06 | [TK]D-Fender | fpriorin group asterisk-es (http://tinyurl.com/8xyp5nm) the problem was discussed but with no solution; in the opinion of someone, there is "something" different in Asterisk 1.8 not compatible with LinkSys <- Guess how much we trust the opinions of unnamed individuals in some other channel |
| 21:19.24 | [TK]D-Fender | fprior: And I also see no rason * would be responsible fo a device rebooting. |
| 21:19.34 | [TK]D-Fender | fprior: And you ahve failed to show us debug and configs |
| 21:19.56 | [TK]D-Fender | fprior: You are taking ssecond-hand advice from a very shallow corner of the gene pool... |
| 21:20.51 | *** join/#asterisk gatty (ajg@2a01:348:11b:beef:b7:5a3a:77ae:530) |
| 21:21.57 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:24.44 | p3nguin | moved on 2.37 months ago. |
| 21:26.52 | fprior | [TK]D-Fender, I will collect debug information and come back. thanks. |
| 21:31.50 | [TK]D-Fender | fprior: One would think that with the assessment you walked int he door with that you'd already have a ton of it to justify the plans you wanted our advice on. Or not.... |
| 21:32.45 | [TK]D-Fender | fprior: I'd recommend coming in with a nice full set of configs from both sides, call debug, the works.... |
| 21:35.11 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-zohwwuornkowizku) |
| 21:35.37 | rdegges | Hey all, I'm running the lastest 1.8 release, and was wondering something about reloading asterisk extensions.conf using the dialplan reload command. |
| 21:35.38 | urvg4 | getting this issue with moh audio stream,first caller starts moh and subsequent callers connect to the middle and tail end of this same moh stream(depending on how long it been on) rather initiating a new moh stream on connect |
| 21:35.51 | urvg4 | how do I fix this? |
| 21:35.53 | rdegges | I've got the #exec functionality enabled in my asterisk.conf, and in my extensions.conf I've got an #exec include. |
| 21:36.22 | rdegges | However, when I run 'dialplan reload', asterisk doesn't pick up my #exec include and run it. Is there a way to force asterisk to run it without restarting asterisk? |
| 21:36.47 | gatty | urvg4: why is that a problem? that's how most moh works - many people even use multicast to distribute a single stream between pbx nodes or endpoints. |
| 21:37.18 | Qwell | rdegges: what does it look like? |
| 21:37.32 | *** join/#asterisk onixx99 (1000@bas1-stetherese38-2925306218.dsl.bell.ca) |
| 21:37.35 | rdegges | Qwell: #exec </var/lib/asterisk/modules/path/to/script.py> |
| 21:37.41 | Qwell | no <> |
| 21:37.43 | rdegges | :o |
| 21:37.47 | rdegges | Ok, let me give that a try. |
| 21:37.54 | rdegges | I usually use <> around my #include s |
| 21:38.14 | urvg4 | gatty: I want each new caller to initiate a separate moh audio stream, not join an existing one |
| 21:38.31 | onixx99 | Hello all, I need some help from somebody who is on 1.8... I'am having a DTMF decoding problem which I did not have on 1.4 before I upgraded yesterday |
| 21:39.03 | onixx99 | I have a fast dtmf sequence that my 1.8 just can't decode |
| 21:40.24 | onixx99 | I would really appreciate if someone could use Read to play the wav file I recorded over and see if it catches the 16 digits |
| 21:49.20 | rdegges | Hey Qwell, same thing :( |
| 21:49.27 | rdegges | Won't run the #exec on 'dialplan reload' |
| 21:49.46 | blitzrage | rdegges: did you enable it in asterisk.conf ? |
| 21:49.50 | rdegges | Yeah. |
| 21:49.53 | blitzrage | by default it is disabled |
| 21:49.56 | blitzrage | and you restarted asterisk? |
| 21:50.07 | blitzrage | requires a restart to make it active |
| 21:50.31 | rdegges | Didn't restart Asterisk, but i have: clearglobalvars=yes |
| 21:50.32 | blitzrage | rdegges: core show settings |
| 21:50.38 | blitzrage | Executable includes: Disabled |
| 21:50.51 | blitzrage | rdegges: that really has nothign to do with what you're trying to enable |
| 21:50.57 | rdegges | blitzrage: oh :( |
| 21:51.00 | rdegges | Yeah, you're right. |
| 21:51.06 | rdegges | It's disabled. damn |
| 21:51.10 | blitzrage | you need to restart to make anything in asterisk.conf active |
| 21:51.13 | rdegges | I was hoping I wouldn't have to restart Asterisk. |
| 21:51.14 | blitzrage | not realod |
| 21:51.17 | rdegges | blitzrage: I see :x |
| 21:51.18 | blitzrage | reload* |
| 21:51.23 | blitzrage | rdegges: you do -- |
| 21:51.24 | p3nguin | core restart now ! |
| 21:51.26 | rdegges | But once it's active, I can dialplan reload to re-scan my #exec includes? |
| 21:51.33 | blitzrage | yes |
| 21:51.36 | blitzrage | that's the point |
| 21:51.37 | rdegges | Gotcha. |
| 21:51.41 | rdegges | Makes sense. |
| 21:51.52 | rdegges | Thanks for your help ^^ |
| 21:52.03 | blitzrage | when you dialplan reload, it'll run the exec, and whatever stdout from the exec happens will be read in as if it were read from a flat file |
| 21:52.12 | rdegges | Right. |
| 21:52.23 | rdegges | I suppose I thought asterisk.conf would be read on dialplan reload |
| 21:52.31 | leifmadsen | uhhh no |
| 21:52.37 | leifmadsen | asterisk.conf is a core asterisk configuration file |
| 21:52.40 | leifmadsen | it's not a dialplan file |
| 21:52.52 | rdegges | Doesn't reload it on a 'module reload' either, though. |
| 21:52.56 | leifmadsen | of course |
| 21:53.01 | leifmadsen | CORE asterisk configuration |
| 21:53.08 | rdegges | Yah, didn't realize that. |
| 21:53.09 | leifmadsen | it is read on asterisk start up |
| 21:53.14 | leifmadsen | and no other time |
| 21:53.17 | leifmadsen | it's not part of a module |
| 21:53.31 | leifmadsen | asterisk.conf --> configuration for asterisk |
| 21:53.39 | leifmadsen | sip.conf --> configuration for sip module |
| 21:53.57 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 21:54.15 | onixx99 | does the speed of my linux computer could be the cause of poor inband dtmf detection |
| 21:54.37 | leifmadsen | probably not |
| 21:54.43 | leifmadsen | poor dtmf information is likely the cause |
| 21:55.03 | leifmadsen | the method you use greatly determines dtmf reliability too |
| 21:55.05 | rrittgarn | Anyone have a moment to help me out with a Sound file issue. PB of the issue is: http://pastebin.com/MzGT2gGg |
| 21:55.14 | p3nguin | I know that System() will block, but does SHELL() block if run inside a Set()? |
| 21:55.22 | leifmadsen | p3nguin: probably |
| 21:55.30 | leifmadsen | it's waiting for a return value |
| 21:55.47 | *** join/#asterisk serafie (~erin@nat/digium/x-azanqkwvvqpkpcqx) |
| 21:56.22 | leifmadsen | and with that, I'm out |
| 21:56.23 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:56.25 | onixx99 | leifmadsen : would you have a moment to try to help ? I'm puzzling if I should log a bug or if my hardware is issue |
| 21:57.28 | *** join/#asterisk NephFL (434e9576@gateway/web/freenode/ip.67.78.149.118) |
| 21:58.13 | NephFL | if I have a digial card and a call comes in, and I then set it to connect out to another number, is there a way for the telco to connect that number directly rather than using two channels? |
| 21:58.59 | NephFL | or is that default functionality? |
| 22:01.42 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:05.44 | *** part/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-zohwwuornkowizku) |
| 22:06.19 | onixx99 | p3nguin : not sure what the topic is but adding & at the end of your command, within a shell script should return immediately |
| 22:06.34 | onixx99 | you could call that script with Shell() |
| 22:06.44 | p3nguin | Right. |
| 22:06.53 | p3nguin | I'm not running any scripts, though. |
| 22:07.10 | p3nguin | And it's SHELL, not Shell. |
| 22:07.14 | onixx99 | well then I'm out of context... wasn't following |
| 22:07.29 | p3nguin | Don't worry. It was just a question about the function. |
| 22:08.01 | patrickod | I'm having an issue with call files not working with Polycom phones. |
| 22:08.28 | [TK]D-Fender | p3nguin: And as Leif mentioned, both are blocking. You could always bacground whatever you're calling if you want... but the usual reason for calling the function is the expectation of a return value for which it'd have to be blovking for most cases |
| 22:08.29 | patrickod | I'm trying to set the channel to a working SIP extension but asterisk says the calls can't go through because circuits are busy |
| 22:08.47 | [TK]D-Fender | patrickod: Polycom knows absolutely nothing of call files. |
| 22:08.47 | patrickod | if I use a softphone with the same credentials everything works |
| 22:09.15 | patrickod | [TK]D-Fender: I realise that the phone shouldn't come into this, that Asterisk should be phone agnostic but I can only reproduce this on Polycom phones |
| 22:09.20 | [TK]D-Fender | patrickod: Show us the real peer, call file and attempt with SIP DEBUG enabled |
| 22:09.42 | [TK]D-Fender | ~pb |
| 22:09.42 | 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. |
| 22:09.44 | [TK]D-Fender | ^^^ |
| 22:09.49 | p3nguin | It has nothing to do with the phone. |
| 22:09.55 | onixx99 | p3nguin : would you know if anybody that comes in here would be able to help me work on a DTMF detection issue in 1.8 |
| 22:10.09 | [TK]D-Fender | Do hurry on providing backup. I'm out the door in a few minutes. |
| 22:10.24 | p3nguin | onixx99: Maybe. I can't speak for everyone, though. |
| 22:10.27 | rrittgarn | Anyone have a moment to help me out with a Sound file issue. PB of the issue is: http://pastebin.com/MzGT2gGg |
| 22:11.03 | NephFL | did you guys get my question? |
| 22:11.10 | onixx99 | p3nguin : it seems dtmf detection is much different in 1.8 vs 1.4 |
| 22:11.20 | p3nguin | rrittgarn: Show me the result of ls -l /var/lib/asterisk/sounds/en/custom/Zimmerman/ZimmCommercialbusiness.* |
| 22:11.44 | patrickod | [TK]D-Fender: I'm seeing sip 404's on the invite. I presume this is the problem |
| 22:11.48 | NephFL | not trying to nag, just not sure if I need a +v or something |
| 22:12.14 | p3nguin | nephfl: If you said it, people can read it. |
| 22:12.54 | p3nguin | That doesn't mean they have read it or that they will read it, but that they CAN read it. |
| 22:12.55 | [TK]D-Fender | NephFL: "transfer=yes" <- for your channels. If the provider supports 2BCT it will hand them off |
| 22:13.10 | [TK]D-Fender | ~2bct |
| 22:13.14 | [TK]D-Fender | Hmmmm... |
| 22:13.15 | NephFL | so, it is default if the provider supports it? |
| 22:13.19 | [TK]D-Fender | uriously absent |
| 22:13.22 | [TK]D-Fender | c even |
| 22:13.49 | [TK]D-Fender | NepNo, it isn't default, and the provider has to. Not all sinalling types support it (switch types that is) |
| 22:14.14 | rrittgarn | nothing found p3nguin... |
| 22:14.35 | p3nguin | rrittgarn: and ls -l /var/lib/asterisk/sounds/custom/Zimmerman/ZimmCommercialbusiness.* |
| 22:15.06 | rrittgarn | also nothing... |
| 22:15.22 | p3nguin | rrittgarn: That's a clue. |
| 22:15.25 | rrittgarn | yeah... |
| 22:15.38 | p3nguin | rrittgarn: Are you using the language prefix? |
| 22:15.40 | [TK]D-Fender | ~2bct |
| 22:15.40 | infobot | [~2BCT] 2BCT (2 B-Channel Transfer) allows a call coming in over DAHDi and back out again to the same telco to be handed off freeing the channels from your circuit. To enable this (if your carrier supports it) add "transfer=yes" to your channel conifigurations. |
| 22:15.48 | rrittgarn | p3nguin: no |
| 22:16.00 | [TK]D-Fender | ~2bct |
| 22:16.01 | infobot | [~2BCT] 2BCT (2 B-Channel Transfer) allows a call coming in over DAHDi and back out again to the same telco to be handed off freeing the channels from your circuit. To enable this (if your carrier supports it) add "transfer=yes" to your channel configurations. |
| 22:16.03 | [TK]D-Fender | better |
| 22:16.26 | rrittgarn | p3nguin: /usr/local/share/asterisk/sounds/Zimmerman is the path to the files it would seem |
| 22:16.45 | p3nguin | rrittgarn: That's non-standard, but expected on a debian-based packaged asterisk. |
| 22:17.08 | rrittgarn | which is what this box is |
| 22:17.34 | p3nguin | rrittgarn: So... ls -l /usr/local/share/asterisk/sounds/custom/Zimmerman/ZimmCommercialbusiness.* |
| 22:18.33 | [TK]D-Fender | Ok, time's up over here..... back later.... |
| 22:18.42 | rrittgarn | nothing there, however without the local the path is correct |
| 22:19.13 | p3nguin | rrittgarn: You mean /usr/share/asterisk/sounds/custom/Zimmerman/ZimmCommercialbusiness.* ? |
| 22:19.41 | rrittgarn | /usr/local/share/asterisk/sounds/Zimmerman has files |
| 22:19.51 | p3nguin | Then update your dial plan accordingly. |
| 22:20.01 | rrittgarn | <PROTECTED> |
| 22:20.06 | rrittgarn | so which is the sym link |
| 22:21.15 | p3nguin | Your dial plan is using custom/Zimmerman/ZimmCommercialbusiness.* |
| 22:21.15 | p3nguin | So show me the permissions on (path to sounds)/custom/Zimmerman/ZimmCommercialbusiness.* |
| 22:21.43 | rrittgarn | yes on the dial plan |
| 22:21.48 | rrittgarn | and sure sec |
| 22:22.21 | carrar | cj |
| 22:22.25 | carrar | sorry I missed you |
| 22:22.28 | carrar | just got back home |
| 22:22.43 | carrar | was out having some PINK SLIME |
| 22:22.48 | carrar | thats good stuff |
| 22:23.21 | p3nguin | vomits a little bit |
| 22:24.07 | rrittgarn | http://pastebin.com/aQPUyRE3 |
| 22:24.30 | p3nguin | Not what I asked for. |
| 22:25.53 | rrittgarn | http://pastebin.com/ne9BT4af |
| 22:26.21 | p3nguin | Why are all these files +x ? |
| 22:26.37 | p3nguin | And why are they writable by everyone? |
| 22:26.39 | rrittgarn | saw "permissions issue" and 777'd the directory |
| 22:27.10 | p3nguin | Way to admin that box! |
| 22:27.25 | p3nguin | Next time, don't do that. |
| 22:27.26 | rrittgarn | its a production box... i just wanted it working first |
| 22:27.46 | p3nguin | If you're doing shit like that on a production box, you're lucky you still have a job. |
| 22:27.57 | rrittgarn | thanks |
| 22:28.51 | p3nguin | The error indicated to me that it was looking for a ulaw file. Fix all the permissions that you ruined, and then convert your files to ulaw. |
| 22:29.54 | rrittgarn | how did it work previous to today though would be my next question? dumb luck? |
| 22:30.12 | p3nguin | What was changed between the time it did work and today? |
| 22:30.39 | rrittgarn | thats what I'm trying to figure out. Only changes that were made to my knowledge was NFS was installed |
| 22:30.45 | rrittgarn | which shouldn't break that directory |
| 22:31.19 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:31.38 | p3nguin | I've had problems exactly like this before. If a file exists as, let's say, .wav but the channel is using ulaw and trying to find a .ulaw file, asterisk should transcode the file it has to the codec it wants... but doesn't. |
| 22:32.01 | p3nguin | I do not know why it doesn't just transcode. |
| 22:32.21 | rrittgarn | my thoughts exactly |
| 22:32.58 | NephFL | i dont see a setting in dahdi-channels for facilityenable but when i test it, I only see one channel in use even though I am dialing in and out... |
| 22:39.54 | p3nguin | I want to ensure received callerid numbers contain only numbers. Should I use FILTER() to filter out any letters, potentially leaving it blank, or should I use REGEX() to find out if it contains letters and deal with it in a different manner if it does? |
| 22:41.11 | rrittgarn | by the way p3nguin, voicemail sounds aren't working either, and those definitely didn't get changed... |
| 22:41.30 | p3nguin | rrittgarn: Which asterisk version are you using? |
| 22:43.33 | rrittgarn | 1.8.8.1 |
| 22:43.39 | p3nguin | core show settings |
| 22:44.02 | p3nguin | Look at the Language prefix value. Is it enabled or disabled? |
| 22:44.19 | rrittgarn | yeah |
| 22:44.26 | rrittgarn | (enabled) default being en |
| 22:44.40 | p3nguin | Make sure your files are in the sounds/en/ path. |
| 22:44.52 | p3nguin | not just sounds/ path. |
| 22:45.08 | rrittgarn | including custom? |
| 22:45.47 | p3nguin | Every sound file that you are trying to play in dial plan which does not use a full explicit path to the sound file must be under the sounds/en/ path, not just the sounds/ path. |
| 22:46.13 | rrittgarn | all the VM prompts are in /en and not working. |
| 22:46.25 | bmoraca_work | has anyone implemented rate limiting in Asterisk before? |
| 22:46.51 | p3nguin | rate of what? |
| 22:46.59 | bmoraca_work | CPS |
| 22:47.08 | bmoraca_work | or, specifically, calls per minute, really |
| 22:47.32 | bmoraca_work | i want to limit the number of international call attempts my customers are making to help mitigate a potential issue |
| 22:49.59 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
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| 22:54.11 | onixx99 | bmoraca_work : trying to mitigate risk of attack ? |
| 22:54.18 | bmoraca_work | yes |
| 22:54.35 | onixx99 | bmoraca_work : have you tried fail2ban ? |
| 22:54.52 | bmoraca_work | i don't have fail2ban loaded on this box, but that might be the easiest... |
| 22:55.04 | onixx99 | bmoraca_work : works great for me |
| 22:55.05 | bmoraca_work | my concern is overhead, though. f2b on my DNS server generates a lot of it. |
| 22:55.15 | bmoraca_work | how many calls per day do you see through your system? |
| 22:55.32 | p3nguin | How is fail2ban going to help where a legitimate caller is making too many calls? |
| 22:55.45 | onixx99 | bmoraca_work : not much... home system. but I get 2-3 attacks per week |
| 22:55.50 | bmoraca_work | ahh |
| 22:56.20 | p3nguin | For a legitimate caller making allowed calls, but too many of them, I don't see how fail2ban is the right tool. |
| 22:56.31 | bmoraca_work | p3nguin: i agree with that |
| 22:56.46 | p3nguin | You'll probably have to use GROUP() and GROUP_COUNT(). |
| 22:56.53 | bmoraca_work | yeah, that's what i was thinking |
| 22:57.16 | p3nguin | That's how I limit concurrent calls to and from the ITSP. |
| 22:57.20 | onixx99 | p3nguin : good point. |
| 22:57.37 | p3nguin | Set(GROUP()=inbound-limit) |
| 22:57.44 | p3nguin | GotoIf($[${GROUP_COUNT(inbound-limit)} > 6]?overlimit) |
| 22:58.02 | p3nguin | (overlimit),Congestion() |
| 22:58.16 | p3nguin | They are allowed 6 concurrent inbound calls. |
| 22:58.26 | bmoraca_work | can a channel be part of multiple groups? |
| 22:58.35 | bmoraca_work | i don't think it can be |
| 22:58.39 | p3nguin | For that, you need to use categories. |
| 22:59.44 | bmoraca_work | i don't know that categories will help if i need to keep track of the TOTAL number of calls for a peer as well as the number of intl calls, independent of each other |
| 22:59.51 | p3nguin | Set(GROUP(something)=inbound-limit) |
| 22:59.55 | bmoraca_work | i suppose i could use astdb |
| 23:00.02 | p3nguin | GotoIf($[${GROUP_COUNT(inbound-limit@something)} > 6]?overlimit) |
| 23:00.17 | p3nguin | Channels can only be in one group, but can be in multiple categories in that group. |
| 23:00.20 | p3nguin | It's really the only way. |
| 23:00.39 | bmoraca_work | so they can be in multiple categories in the group? |
| 23:00.45 | p3nguin | Correct. |
| 23:01.02 | p3nguin | Set(GROUP(foo)=inbound-limit) |
| 23:01.04 | p3nguin | Set(GROUP(bar)=inbound-limit) |
| 23:01.19 | p3nguin | Group inbound-limit, categories foo and bar. |
| 23:01.20 | bmoraca_work | gotcha |
| 23:01.27 | bmoraca_work | says it can be in one group per category |
| 23:02.01 | bmoraca_work | ok, that should work ok...except in the instance when the attack just makes one call at a time |
| 23:02.02 | p3nguin | Maybe I'm expressing the restriction incorrectly, but you can't have it in more than one group. |
| 23:02.31 | bmoraca_work | per "core show function GROUP", it says specifically "each channel can be a member of exactly one group per category" |
| 23:02.32 | p3nguin | You can't have it in GROUP()=inbound-limit and GROUP()=some-other-limit |
| 23:02.50 | bmoraca_work | right |
| 23:02.56 | bmoraca_work | because that's not defining a category |
| 23:03.25 | p3nguin | Oh, you're saying you think you could do GROUP(foo)=inbound-limit and GROUP(bar)=some-other-limit |
| 23:03.32 | bmoraca_work | yes |
| 23:03.35 | p3nguin | That's probably correct. |
| 23:04.04 | bmoraca_work | in fact, it says i could even probably do GROUP(foo)=limit and GROUP(bar)=limit |
| 23:04.15 | bmoraca_work | which would be great |
| 23:04.20 | p3nguin | Right, that's what I was saying before. |
| 23:04.27 | bmoraca_work | because "limit" is dynamically read from a database as the call is set up |
| 23:04.48 | p3nguin | I just used the inbound-limit group in my example rather than the limit group. |
| 23:05.19 | bmoraca_work | so using that and astdb to track CPS, i could have it turn off international calling if it's above a certain rate |
| 23:05.25 | p3nguin | I have inbound-limit for calls coming in, outbound-limit for calls goin out, and call-limit for calls between phones. |
| 23:05.34 | p3nguin | Absolutely! |
| 23:06.09 | p3nguin | At least concurrency. |
| 23:06.10 | bmoraca_work | both inbound and outbound are the same group in my scenario, but i want a separate group to ahve a separate limit for international |
| 23:06.51 | bmoraca_work | now to implementation...maybe AGI would be a better option for this than pure dialplan |
| 23:07.28 | bmoraca_work | easier to modify my database and easier to send myself an email in the event of a breach |
| 23:07.37 | bmoraca_work | to the psuedocode machine! |
| 23:08.06 | p3nguin | Set(GROUP(foo)=limit), GotoIf($[${GROUP_COUNT(limit@foo} > ${DB(limit/international)}]?overlimit) |
| 23:08.37 | p3nguin | Don't forget to debug my typos. |
| 23:09.13 | p3nguin | <PROTECTED> |
| 23:09.50 | bmoraca_work | i'm thinking international limit should be (overall limit)/3 |
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| 23:56.09 | onixx99 | exit |