| 00:01.08 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 00:03.17 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.29 | Kobaz | 3com makes phones? |
| 00:04.01 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org) |
| 00:04.07 | voipmonk | hehe |
| 00:04.42 | *** join/#asterisk nny (~scott@64.203.239.83) |
| 00:04.47 | nny | hmm |
| 00:04.55 | nny | digium has centos repos for asterisk |
| 00:05.02 | tkrn | well i bet they are rebranded |
| 00:05.20 | nny | however the dahdi one doesn't seem to want to load, complains about missing modules and fails, btu never tries ztdummy |
| 00:05.25 | nny | but |
| 00:05.31 | tkrn | it needs an propriety 3com controller of some sorty |
| 00:06.10 | *** join/#asterisk cvnet (~cvnet@dsl-69-172-67-161.acanac.net) |
| 00:06.14 | cvnet | hi all |
| 00:06.46 | cvnet | I can send fax with my * no problem, but can not receive (fax turns on but nothing comes out and it shows NG) any suggestions? |
| 00:07.28 | nny | dammit lol, so close, who compiles packages with ztdummy anyways? |
| 00:07.40 | nny | hmm |
| 00:07.45 | nny | maybe config file related let me check |
| 00:09.21 | nny | do I have to define ztdummy in /etc/dahdi/modules? |
| 00:10.45 | p3nguin_ | Hmm, isn't that like mixing a Schwinn with a Huffy? |
| 00:11.17 | nny | better question, what is ztdummy needed for? |
| 00:11.33 | nny | this is just a simple auto attendant with various numbers |
| 00:12.04 | p3nguin_ | Well, dahdi_dummy is used for timing. |
| 00:12.09 | nny | ahh |
| 00:12.12 | nny | is that the module name? |
| 00:12.19 | p3nguin_ | yes |
| 00:12.39 | nny | hmm dahdi_dummy: FATAL: Module dahdi_dummy not found. |
| 00:12.51 | p3nguin_ | Did you install dahdi? |
| 00:13.14 | nny | aye, but via digium centos repos |
| 00:13.23 | nny | I know how to compile it etc, just seeing what works with these |
| 00:13.36 | p3nguin_ | That's fine, it's still the same dahdi either way. |
| 00:13.44 | nny | hmm wonder what I am missing |
| 00:13.52 | p3nguin_ | You installed dahdi and dahdi tools? |
| 00:14.08 | nny | I commented all the modules out for the other hardware |
| 00:14.09 | nny | yeah |
| 00:14.15 | nny | since this box is hardware-less |
| 00:14.21 | nny | yum install dahdi dahdi-tools |
| 00:14.25 | p3nguin_ | Did you compile asterisk yourself? |
| 00:14.34 | nny | no that's the test heh |
| 00:14.47 | p3nguin_ | What are the dependencies for the package? |
| 00:15.17 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:15.22 | nny | not sure didn't use screen, is there a way to ask yum? |
| 00:15.42 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:15.44 | p3nguin_ | Didn't use screen? What does THAT have to do with anything? |
| 00:15.46 | *** join/#asterisk nickaugust (~anonymous@207-224-58-219.hlrn.qwest.net) |
| 00:15.55 | nny | it's already scrolled past heh |
| 00:16.12 | nny | so I can't page up to see what installed originally when I installed * etc |
| 00:16.22 | nny | nm |
| 00:16.25 | nny | yum deplist one sec |
| 00:16.38 | *** join/#asterisk elwerene (~lalala@ip-141-31-187-161.nat.selfnet.de) |
| 00:17.48 | nny | hmm what are we trying to verify here? The module seems to fail due to the dahdi_dummy module missing |
| 00:18.36 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 00:18.47 | p3nguin_ | I don't even have an asterisk package. The reason I want to know is because I expect either dahdi or zaptel to be a dependency if it was built with either. |
| 00:19.02 | p3nguin_ | If neither is a dep, then asterisk was probably not built with support for either one. |
| 00:19.06 | elwerene | asterisk as voip gateway, hearing nothing on incoming calls (behind nat) any help? |
| 00:19.47 | p3nguin_ | If that is the case, install dahdi and dahdi-tools, and then compile asterisk yourself WITH support for dahdi. Use checkinstall to roll it into an RPM, then install it. |
| 00:19.47 | elwerene | rtp and sip is forwarded |
| 00:19.59 | p3nguin_ | ~sipnat |
| 00:20.00 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:20.06 | p3nguin_ | elwerene: this ^^ |
| 00:20.06 | nny | p3nguin_: it fails on /etc/init.d/zaptel start ... |
| 00:20.16 | nny | p3nguin_: er dahdi* |
| 00:20.20 | nny | not zaptel |
| 00:20.46 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 00:20.46 | p3nguin_ | I don't even know what you're trying to do, at this point. |
| 00:20.47 | nny | would whether or not asterisk has support at that point even matter? |
| 00:20.53 | nny | just load the dahdi_dummy module |
| 00:21.07 | nny | not even sure if this box needs it, just seeing what the repos can and cannot do |
| 00:21.29 | nny | normally (on another vm) compile + dahdi start = dummy module at least |
| 00:21.50 | nny | at least on my other vm it does |
| 00:22.12 | p3nguin_ | modprobe -l dahdi_dummy |
| 00:22.34 | nny | well |
| 00:22.44 | nny | it didn't complain, but no module in lsmod |
| 00:23.07 | p3nguin_ | There's no well. Either modprobe -l dahdi_dummy shows the module or it doesn't, |
| 00:23.16 | nny | oh it was blank sorry |
| 00:23.33 | p3nguin_ | # modprobe -l dahdi_dummy |
| 00:23.33 | p3nguin_ | dahdi/dahdi_dummy.ko |
| 00:24.03 | nny | what main directory would that be in? |
| 00:24.17 | p3nguin_ | /lib/modules/`uname -r`/ |
| 00:25.28 | nny | nothing named dahdi under there |
| 00:25.38 | p3nguin_ | Sounds like dahdi isn't installed. |
| 00:25.38 | nny | ugh |
| 00:25.41 | nny | nm |
| 00:25.55 | nny | man fuck rackspace |
| 00:26.00 | nny | 2.6.18-164.11.1.el5 2.6.32.1-rscloud |
| 00:26.10 | nny | guess which kernel is being used, and which the module is compiled for |
| 00:26.13 | p3nguin_ | Did they break something? |
| 00:29.03 | p3nguin_ | So dahdi package is built for 2.6.18, but you're running 2.6.32? |
| 00:29.04 | *** join/#asterisk cvnet (~cvnet@dsl-69-172-67-161.acanac.net) |
| 00:29.17 | nny | pretty much |
| 00:29.33 | p3nguin_ | Sounds like when I upgraded from 2.6.31 to 2.6.32. At first I couldn't figure out why dahdi was broken... |
| 00:29.44 | *** part/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 00:29.53 | cvnet | I can send fax no problem but when i try to receive it i get this --> [Feb 22 19:24:00] WARNING[2097]: chan_sip.c:5512 process_sdp: Unsupported SDP media type in offer: image 20100 udptl t38 any suggestions? |
| 00:46.48 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:46.48 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.4, 1.6.1.16, 1.6.0.24 (2010/02/18), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 00:46.50 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 00:46.57 | p3nguin_ | lol, that's potatoes and carrots! |
| 00:48.32 | *** join/#asterisk cvnet (~cvnet@dsl-69-172-67-161.acanac.net) |
| 00:57.15 | adnc | hello, i do get a "No route to destination" does it mean something in the dialplan is wrong? |
| 00:57.34 | adnc | although it is fine when looking with dialplan show 04555@ |
| 00:57.50 | *** join/#asterisk Agrajag- (~filip@c211-30-185-177.artrmn2.nsw.optusnet.com.au) |
| 00:58.23 | jaytee | is that device registered? is there an account for it in your sip.conf? |
| 00:58.49 | adnc | jaytee, the device is registered and there is an entry for the device |
| 00:59.36 | jaytee | pastebin that section of your extensions.conf, the section of sip.conf that device is in and a failed call attempt |
| 00:59.40 | adnc | it works when i call international which matches _0090 |
| 00:59.42 | jaytee | ~pb |
| 00:59.43 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 00:59.44 | p3nguin_ | No route to destination is a networking issue. |
| 01:00.21 | adnc | i thought about this aswell, since this is the configuration which as working yesterday |
| 01:00.36 | adnc | the asterisk server gets out without problems |
| 01:00.38 | jaytee | true, either the device isn't registered or asterisk doesn't know how to route it to it's address |
| 01:01.25 | p3nguin_ | its address |
| 01:01.31 | adnc | Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 01:02.00 | adnc | [Feb 23 02:02:22] WARNING[3655]: chan_sip.c:2921 create_addr: No such host: 05318890909 |
| 01:02.13 | p3nguin_ | Show us soemthing useful already. |
| 01:02.15 | adnc | where the number is my phone number |
| 01:02.20 | adnc | p3nguin_, sure moment |
| 01:02.37 | adnc | why is the number looked up as host? |
| 01:02.50 | jaytee | his dialplan is munged |
| 01:03.16 | Agrajag- | g'day, if i want to delete all voicemail messages in a particular users voicemail (they let it fill up and will take a long time to delete them manually), is it safe to just delete all files in var/spool/asterisk/voicemail/default/<user>/INBOX? i can make sure they're not accessing voicemail when i do this. |
| 01:03.24 | *** join/#asterisk aandrade (~aandrade@189.34.121.136) |
| 01:06.53 | adnc | http://pastebin.com/d16094d01 |
| 01:07.03 | adnc | here is my extensions.conf |
| 01:07.13 | adnc | i hope there is not much wrong |
| 01:07.30 | p3nguin_ | Which extension is the problem? 04555? |
| 01:08.06 | adnc | anything that goes out from [kabeldeutschland_out] |
| 01:08.16 | nny | here's a stupid question, anyway to disable cdr in asterisk? |
| 01:08.40 | adnc | nny, you can do a noload in modules.conf |
| 01:08.46 | nny | adnc thanks |
| 01:08.57 | p3nguin_ | adnc: First of all, only like 3 of your contexts have a priority 1 in them. |
| 01:09.32 | p3nguin_ | adnc: And second, do you have a peer definition in sip.conf called [0531xxxxxxx] ? |
| 01:09.58 | adnc | p3nguin_, ahh i see, i've changed the priority, those are 1 now. |
| 01:10.09 | adnc | yes, let me show you the definition for [0531xxxxxxx] |
| 01:10.17 | p3nguin_ | So you're pasting inaccurate information... |
| 01:10.26 | adnc | p3nguin_, no |
| 01:10.26 | p3nguin_ | That's a great idea to get a problem solved. |
| 01:10.57 | jaytee | why does swiss cheese have holes? |
| 01:11.10 | Kobaz | air bubbles that pop |
| 01:11.16 | jaytee | thanks! |
| 01:11.17 | p3nguin_ | gas bubbles were produced while it was curing/hardening. |
| 01:11.20 | adnc | http://pastebin.com/d7e47ec1d |
| 01:11.28 | coppice | jaytee: its a cheat, so you get less cheese |
| 01:11.36 | Kobaz | they sell cheese by weight |
| 01:11.44 | p3nguin_ | haha |
| 01:11.57 | coppice | but it looks more impressive when its bigger, so you buy |
| 01:12.02 | jaytee | the whole idea for croutons was just a way of making money from tourist by selling them stale bread cut into cubes |
| 01:12.08 | Kobaz | haha |
| 01:12.14 | adnc | p3nguin_, thats my mistake, i did a remove which i took from a tutorial just a minute ago |
| 01:12.19 | coppice | actually gruyere has a problem that holes are disappearing |
| 01:12.42 | Kobaz | jaytee: stale garlic bread! |
| 01:12.45 | Kobaz | with butter! |
| 01:12.49 | Kobaz | my favorite |
| 01:12.57 | jaytee | trust the french to come up with that |
| 01:12.57 | coppice | I think crouton is just the french word for cretin - i.e. customer |
| 01:12.59 | p3nguin_ | adnc: make sure you can talk to proxy.kabelphone.de and reg01.kabelphone.de |
| 01:13.31 | Kobaz | crouton n 1: a small piece of toasted or fried bread; served in soup or salads |
| 01:13.39 | cvnet | my sip.conf http://www.pastebin.com/m55a21ad8 i can send fax but not receive, receiving error: [Feb 22 20:05:17] WARNING[2097]: chan_sip.c:5512 process_sdp: Unsupported SDP media type in offer: image 20100 udptl t38 |
| 01:13.47 | *** join/#asterisk Kumbang (~kumbang@167.205.24.69) |
| 01:13.48 | cvnet | any suggestions? |
| 01:14.09 | coppice | yeah. enable T.38 in your config file |
| 01:14.15 | p3nguin_ | adnc: Host reg01.kabelphone.de not found: 3(NXDOMAIN) |
| 01:14.26 | p3nguin_ | adnc: Ya think this could be a problem? |
| 01:14.33 | cvnet | coppice: T.38=yes ? |
| 01:14.36 | adnc | p3nguin_, yes, i know, but it should be reachable via the proxy |
| 01:14.38 | jaytee | lol |
| 01:14.38 | Kobaz | coppice: crouton in french means crust |
| 01:15.11 | coppice | Kobaz: you don't say :-) . In this case it seems to mean fake crust |
| 01:15.16 | adnc | register => 0531xxxxxxx@reg01.kabelphone.de:secret:0531xxxxxxx@proxy.kabelphone.de/0531xxxxxxx |
| 01:15.22 | adnc | this way i was able to connect |
| 01:15.28 | adnc | at least i think i was ;) |
| 01:15.37 | Kobaz | now we know your secret |
| 01:15.41 | jaytee | Heisenberg |
| 01:15.44 | adnc | but sip show registry shows a registration |
| 01:16.02 | adnc | Kobaz, thats fine ;) |
| 01:16.07 | p3nguin_ | That string makes you connecto to proxy.kabelphone.de, not to the other host name. |
| 01:16.45 | adnc | p3nguin_, can you please point me to a documentation that would describe me how to use proxies with asterisk |
| 01:16.49 | p3nguin_ | In that register statement, reg01.kabelphone.de is the domain name for your user, and it is registering to the proxy. |
| 01:17.24 | p3nguin_ | Are you installing and configuring a proxy? |
| 01:17.35 | adnc | no |
| 01:17.48 | p3nguin_ | Then you don't need to worry about that... just use the proxy's host address as the host where your calls go. |
| 01:18.07 | adnc | mhhh |
| 01:18.08 | p3nguin_ | The provider takes care of the technical part. |
| 01:18.27 | adnc | p3nguin_, should i change anything on the register part above? |
| 01:18.44 | p3nguin_ | No, the register statement is probably correct. |
| 01:18.58 | p3nguin_ | As long as you are using a 1.6 version, that is. |
| 01:19.08 | adnc | p3nguin_, 1.4 here |
| 01:19.18 | p3nguin_ | I don't think 1.4 supports domains in the register statment, so you should remove it. |
| 01:19.35 | adnc | which part? |
| 01:19.46 | p3nguin_ | register => user:secret@proxyhost/phonenumber |
| 01:19.50 | p3nguin_ | That's how it should look. |
| 01:20.24 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 01:21.29 | adnc | p3nguin_, this gives me a 404 |
| 01:22.46 | cvnet | my sip.conf http://www.pastebin.com/m55a21ad8 <-- can't receive fax, please let me know what I'm doing wrong |
| 01:23.27 | voipmonk | cvnet: whats the debug say when the call comes into Zlp |
| 01:24.43 | cvnet | voipmonk: error: [Feb 22 20:05:17] WARNING[2097]: chan_sip.c:5512 process_sdp: Unsupported SDP media type in offer: image 20100 udptl t38 <--- |
| 01:25.06 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 01:28.21 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 01:36.11 | cvnet | voipmonk: did you find any problems? |
| 01:37.23 | adnc | p3nguin_, that registration does definately not work |
| 01:41.44 | voipmonk | back |
| 01:42.35 | p3nguin_ | I really didn't know that 1.4 supported domain name in the register statement. |
| 01:43.33 | adnc | it now works |
| 01:43.37 | adnc | and it was my mistake |
| 01:44.04 | p3nguin_ | What was the mistake? |
| 01:44.16 | adnc | since i have two numbers with 0531xxxxxyy 0531xxxxxxx i mixed them up |
| 01:44.20 | *** join/#asterisk PMantis (~sswitzer@out.ewbc.com) |
| 01:44.24 | p3nguin_ | oh |
| 01:44.31 | adnc | you couldnt see them because i xx'ed them out |
| 01:44.38 | adnc | really sorry for this |
| 01:44.44 | p3nguin_ | So you're using the user:secret@proxyhost/phonenumber syntax? |
| 01:44.51 | adnc | i was giving a Dial command to the other |
| 01:44.55 | p3nguin_ | There's a reason why we don't blank usernames. |
| 01:45.04 | adnc | p3nguin_, no the initial form like this |
| 01:45.36 | adnc | register => 0531xxxxxxx@reg01.kabelphone.de:secret:0531xxxxxxx@proxy.kabelphone.de/0531xxxxxxx |
| 01:45.57 | adnc | the only thing here is that sip show registry shows the username (phonenumber) with an at sign at the end like this |
| 01:46.14 | p3nguin_ | Until someone with proper authority advises me that 1.4 supports passing the domain name in the register statement, I will continue to believe that it doesn't. |
| 01:46.30 | adnc | proxy.kabelphone.de:5060 0531xxxxxxx@ 195 Registered Tue, 23 Feb 2010 02:47:40 |
| 01:46.51 | PMantis | Hi, working with 1.6.2.0~rc2-0ubuntu1.2. The PRI stopped working tonight. Busy/congested on inbound or outbound side. "PRI Set Debug on span 2" doesn't show anything extra when a call tries to come in. |
| 01:47.10 | adnc | p3nguin_, do you think that the @ sign has a particular meaning here? |
| 01:47.35 | p3nguin_ | Yeah, I believe it is because of invalid syntax. |
| 01:47.57 | adnc | p3nguin_, good idea, i shouldnt blank usernames |
| 01:49.07 | adnc | p3nguin_, well, i use "Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on pbx (pid = 3623) |
| 01:49.58 | p3nguin_ | You lost me. |
| 01:50.19 | p3nguin_ | I was with you up to the "I use" part. |
| 01:50.35 | jaytee | you had me at hello |
| 01:53.01 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 01:53.37 | JT | is there an obvious way to ring multiple numbers at once, but have the callerid set differently for one of the destination numbers? |
| 02:01.18 | voipmonk | u can ring them one after the next and set a new cid for the next ring |
| 02:01.28 | adnc | officially my provider kabeldeutschland doesnt give away voip, they sell a box which does sip. in germany we extract the registration information and make it work with asterisk, i see that via sip the user-agent is transferred as asterisk to the operator. is it possible to set the user-agent string? |
| 02:01.52 | p3nguin_ | adnc: yes |
| 02:01.54 | voipmonk | yes check the sip.conf sample |
| 02:02.01 | adnc | ok |
| 02:02.24 | p3nguin_ | I bet you could guess the setting. |
| 02:02.37 | adnc | user-agent="My Agent"? |
| 02:02.44 | p3nguin_ | no hyphen |
| 02:02.49 | p3nguin_ | and no quotes |
| 02:02.51 | adnc | or useragent="My Agent" |
| 02:02.54 | adnc | ok, cool |
| 02:03.07 | adnc | do you think there are other possibilities they could find out it is not there box? |
| 02:03.25 | p3nguin_ | As long as you provide the same info, probably not. |
| 02:03.29 | adnc | any reason why i shouldn't change this |
| 02:03.41 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 02:04.12 | p3nguin_ | If I wanted to mimic the proprietary device, I would try to use every value that it uses. |
| 02:04.32 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 02:04.52 | p3nguin_ | It's probably not against any rules, either way. |
| 02:05.12 | p3nguin_ | Sorta like the MAC clone feature on home network appliances. |
| 02:05.19 | adnc | i think so aswell, but |
| 02:05.31 | adnc | you never know how they see it |
| 02:05.33 | p3nguin_ | It could be in the terms, and you could get terminated. |
| 02:05.43 | adnc | that would be good |
| 02:05.59 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 02:07.21 | adnc | AVM FRITZ!Box Fon WLAN 7270 54.04.80 (Dec 4 2009) |
| 02:07.29 | adnc | thats what they send as User-Agent |
| 02:07.45 | p3nguin_ | All that? |
| 02:08.03 | adnc | yes, all that, i just took a sip-package with wireshark |
| 02:08.26 | *** join/#asterisk nickaugust (~anonymous@207-224-59-105.hlrn.qwest.net) |
| 02:08.30 | p3nguin_ | That's a lot compared to Elite SMTA 6011S 00032 or Cisco-CP7940G/8.0. |
| 02:08.32 | KingDavidNYC | Hello everybody! |
| 02:08.50 | adnc | p3nguin_, yes, theres is sending the Cisco ua |
| 02:11.46 | KingDavidNYC | I have a question regarding call files, anybody here able to capture dialstatus? |
| 02:12.08 | p3nguin_ | Everyone is able to "capture" dialstatus. |
| 02:12.21 | p3nguin_ | Verbose(${DIALSTATUS}) |
| 02:12.46 | KingDavidNYC | penguin, from a call file, and get the dialstatus in the dialplan?n |
| 02:12.59 | KingDavidNYC | sorry, p3nguin |
| 02:14.33 | KingDavidNYC | here is what happens: when call initiated with a call file, dialstatus is cleared just before control is branched to the dialplan it points to |
| 02:14.41 | *** part/#asterisk PMantis (~sswitzer@out.ewbc.com) |
| 02:17.09 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 02:18.49 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net) |
| 02:19.37 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
| 02:27.05 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 02:28.10 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 02:30.12 | JT | voipmonk: they must be rung simultaneously |
| 02:30.19 | JT | i know it can be done, freepbx will do it |
| 02:31.14 | voipmonk | then find out how freepbx does it and replicate that code :) |
| 02:31.35 | JT | the code it creates is horrible and hard to trace though :/ |
| 02:31.40 | voipmonk | you can do it! |
| 02:32.49 | JT | i was hoping someone knew a good way to do it |
| 02:33.00 | JT | freepbx may not be doing it the best way |
| 02:36.06 | LemensTS | if i DIAL(SIP/1234&SIP/1235) and sip 1234 answers it, how can I know that sip 1234 answered it for use as a variable? |
| 02:39.55 | x86 | ${DIALSTATUS} |
| 02:40.07 | x86 | oh... |
| 02:40.11 | x86 | hmm |
| 02:40.29 | x86 | not only do you want to know if it was answered, but which party answered... |
| 02:40.34 | x86 | ${DSTCHANNEL} iirc |
| 02:40.48 | x86 | check the wiki for channel variables, very useful info there |
| 02:42.01 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 02:42.01 | *** mode/#asterisk [+o file] by ChanServ |
| 02:42.37 | KingDavidNYC | what about chan_spool.c? I see that it sends dialstatus too |
| 02:44.47 | *** part/#asterisk nny (~scott@64.203.239.83) |
| 02:46.29 | LemensTS | x86: ${CDR(dstchannel)} gets me SIP/2222222222-086821b8...thats closer than ive gotten. Anyway to just have it show the 2222 part? |
| 02:46.58 | LemensTS | besides triming between / and = |
| 02:47.01 | LemensTS | i mean - |
| 02:47.04 | x86 | LemensTS: core show function CUT |
| 02:47.26 | x86 | and, iirc, SUBSTR |
| 02:47.36 | x86 | or check the wiki for examples of those |
| 02:49.14 | LemensTS | awesome thanks for the help ive been looking for a while |
| 03:05.21 | *** join/#asterisk ectospasm (~weechat@24.42.221.223) |
| 03:09.06 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 03:12.13 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 03:17.32 | Katty | bmoraca_work: ping |
| 03:19.17 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 03:21.22 | LemensTS | x86: I can use that to find the destuser on parked and transfered calls too, that was a major help |
| 03:21.29 | LemensTS | MAJOR |
| 03:22.31 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 03:23.33 | *** join/#asterisk xmitter (~xmitter@72.11.81.253) |
| 03:24.06 | x86 | LemensTS: no prob man :) |
| 03:42.21 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org) |
| 03:45.26 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 04:02.05 | adnc | Katty, hi ;) |
| 04:03.09 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:06.12 | LemensTS | when i hangup from a sip to sip call it will initiate a DeadAGI script successfully. If i call sip to sip, and try to transfer it to parking lot, it hangsup the one call and goes thru the same hangup code and starts to launch the DeadAGI but it doesnt go into it. |
| 04:08.51 | *** join/#asterisk Jhirley (~Jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 04:10.04 | sbrath | I'm having an issue sending DTMF over SIP/IAX phones, I can connect a phone via another PBX trunked into the Asterisk over a TE110P and the DTMF to the PSTN works ( the asterisk is the gateway to the PSTN ) |
| 04:10.22 | sbrath | I've tried rfc, and info, but not sure what else to try. |
| 04:10.56 | sbrath | It works for most DTMF apps, but not for a call your credit card company, and enter your card # ..... That fails |
| 04:12.42 | *** join/#asterisk enyawix (~enyawix@adsl-179-2-85.bna.bellsouth.net) |
| 04:13.22 | enyawix | is the the wrong place to ask phone wire questions? |
| 04:13.41 | sbrath | what you want to know? |
| 04:13.55 | sbrath | red to red and green to green :) |
| 04:14.49 | ManxPower-work | sbrath: increse toneduration |
| 04:15.36 | sbrath | But I think I've proven that it's not the TE110P card, as I can initiate a call from my Merlin that's trunked into the ASterisk, and those calls transmit the DTMF corectly. |
| 04:15.39 | sbrath | ? |
| 04:16.17 | sbrath | The TE110P I'm adjusting duration on is the only exit to the PSTN, the Merlin egresses into Asterisk to get to the PSTN. |
| 04:16.39 | sbrath | I might note that all the phones are wideband as well |
| 04:18.44 | sbrath | The phone I'm testing on now thou is a Softphone over IAX2 |
| 04:19.07 | sbrath | zopier, but it's failing simmilar to the office SIP phones. |
| 04:22.17 | *** join/#asterisk mick_laptop (~mick@clamwin/admin/mickhome) |
| 04:22.24 | mick_laptop | hi everyone |
| 04:23.04 | leifmadsen | hi dr. nick! |
| 04:32.19 | *** join/#asterisk jayesh (~jay@122.172.124.160) |
| 04:34.34 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
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| 04:46.31 | ChannelZ | Inflammable means flammable? Who knew!? |
| 04:49.10 | leifmadsen | lol |
| 04:59.08 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-zkcmzcxgipfxnwrn) |
| 05:01.03 | *** join/#asterisk alexsea7 (~Sheeju_Al@122.169.210.138) |
| 05:02.47 | ChannelZ | hmm.. wasn't there some way to forward a voicemail to someone else, or am I imaginging things? |
| 05:04.33 | *** part/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 05:07.20 | *** join/#asterisk Jak_o_Shadows (~Fake@121.209.35.170) |
| 05:11.11 | LemensTS | at line 73 it should be going into the script like on like on line 36...anyone know why? |
| 05:11.14 | LemensTS | http://pastebin.ca/1806415 |
| 05:14.31 | alexsea7 | asterisk shutdown issues when used with DAHDI, how to figure out the exact reason? |
| 05:14.37 | ChannelZ | is your AGI seeing something different and terminating for some reason? |
| 05:14.50 | LemensTS | on line 34 its complaining about running DeadAGI on live channel, but it doesnt complain when you do the Parked call |
| 05:17.59 | LemensTS | ChannelZ: onlything i can tell is that line 34 says the channel isnt hangup and it works properly. apparently the transfer-to-park is hanging the channel up differently than how a sip-to-sip call hangs it up |
| 05:18.06 | *** join/#asterisk Znuff (~ibm86@suge-ma.apoi.linge-ma.ws) |
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| 06:10.25 | alexsea7 | guys I am facing some asterisk shutdown after updating asterisk and dahdi. These are some warnings I got before shutdown.. |
| 06:10.25 | alexsea7 | [Feb 23 16:00:03] WARNING[8487] res_agi.c: If you want to run AGI on hungup channels you should use DeadAGI! |
| 06:10.25 | alexsea7 | [Feb 23 16:00:03] WARNING[8487] file.c: Failed to write frame |
| 06:10.26 | alexsea7 | [Feb 23 16:06:51] NOTICE[4791] cdr.c: CDR simple logging enabled. |
| 06:10.26 | alexsea7 | [Feb 23 16:06:51] NOTICE[4791] loader.c: 143 modules will be loaded. |
| 06:10.26 | alexsea7 | [Feb 23 16:06:54] WARNING[4791] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. |
| 06:10.26 | alexsea7 | [Feb 23 16:06:54] NOTICE[4791] chan_iax2.c: The option 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly' |
| 06:10.27 | alexsea7 | [Feb 23 16:06:55] ERROR[4791] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 06:10.41 | alexsea7 | can anyone please help.. |
| 06:18.00 | *** join/#asterisk s2krish (~chatzilla@113.199.172.205) |
| 06:18.27 | s2krish | I have install Asterisk GUI 2.0, and configured successfully. trying make checkconfig. it success everything |
| 06:18.38 | s2krish | I tried from browser, get nothing but 404 |
| 06:18.46 | s2krish | anyone already tried? |
| 06:20.18 | idespinner | <PROTECTED> |
| 06:20.43 | s2krish | it's in my localnetwork, asterisk server IP is 192.168.1.253 |
| 06:20.57 | s2krish | i I tried like http://192.168.1.253:8088/asterisk/static/config/index.html |
| 06:21.04 | s2krish | as shown in make checkconfig |
| 06:21.09 | idespinner | gotcha... |
| 06:21.25 | idespinner | we need to double check your config files |
| 06:21.42 | idespinner | in... well i think its http.conf in /etc/asterisk? |
| 06:21.49 | idespinner | the web one... |
| 06:21.58 | idespinner | which defines the asterisk root |
| 06:23.02 | s2krish | this is what I have in httpd.conf |
| 06:23.03 | s2krish | [general] |
| 06:23.05 | s2krish | enabled=yes |
| 06:23.06 | s2krish | bindaddr=0.0.0.0 |
| 06:23.08 | s2krish | bindport=8088 |
| 06:23.09 | s2krish | enablestatic=yes |
| 06:23.34 | s2krish | In manager.conf, I have: |
| 06:23.35 | p3nguin_ | I guess no one ever heard of pastebins. |
| 06:23.36 | s2krish | enabled = yes |
| 06:23.38 | s2krish | webenabled = yes |
| 06:23.39 | s2krish | port = 5038 |
| 06:23.41 | s2krish | bindaddr = 0.0.0.0 |
| 06:23.42 | s2krish | [astercc] |
| 06:23.44 | s2krish | secret = astercc |
| 06:23.45 | s2krish | read = system,call,log,verbose,command,agent,user |
| 06:23.47 | s2krish | write = system,call,log,verbose,command,agent,user |
| 06:23.47 | JT | ~pb |
| 06:23.48 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 06:23.48 | s2krish | deny=0.0.0.0/0.0.0.0 |
| 06:23.50 | s2krish | permit=192.168.1.253/255.255.255.0 |
| 06:24.41 | s2krish | p3nguin tx |
| 06:24.58 | p3nguin_ | Nah, I'm in IL. |
| 06:25.52 | idespinner | s2krish, anything in /var/lib/asterisk/static-http ? |
| 06:25.58 | idespinner | cd to /var/lib/asterisk/static-http and do an ls |
| 06:26.20 | p3nguin_ | Why do you need to cd in order to ls? |
| 06:26.29 | p3nguin_ | Why can't you just ls /var/lib/asterisk/static-http ? |
| 06:26.40 | idespinner | laziness... |
| 06:26.54 | idespinner | no real reason honestly |
| 06:26.55 | p3nguin_ | What's laziness? |
| 06:27.00 | Dovid | lol |
| 06:27.04 | s2krish | ajamdemo.html astman.css astman.js config docs index.html prototype.js |
| 06:27.28 | p3nguin_ | Your way certainly isn't lazy, since you have to type more. |
| 06:27.29 | idespinner | i'd tell ya but i cant bring myself to do it... |
| 06:27.38 | idespinner | sometimes more is less |
| 06:30.40 | *** join/#asterisk fiddur (~fiddur@192.121.104.121) |
| 06:32.55 | idespinner | s2krish, you dont have a prefix... |
| 06:33.10 | idespinner | min w/o a [prefix is http:/[ipaddresss]/static/config/index.html |
| 06:33.47 | s2krish | idespinner, thanks |
| 06:33.53 | idespinner | works? |
| 06:33.53 | s2krish | worked http://192.168.1.253:8088/static/config/index.html |
| 06:34.33 | idespinner | that is a pretty crazy url for a single purpose HTTP daemon |
| 06:37.01 | s2krish | yes, i think make checkconfig need to be patched. |
| 06:39.28 | idespinner | well i mean more, all requests to the http daemon should redirect to the url, so you can just do http://[ipaddress]:8088 |
| 06:39.45 | idespinner | well actually it does on this build i have... |
| 06:40.17 | p3nguin_ | You could stick a redirector in the root directory. |
| 06:40.42 | idespinner | that would be an excellent default IMHO |
| 06:41.03 | idespinner | on the ABE versions, it appears to be default though.... |
| 06:41.09 | p3nguin_ | Yeah, I don't quite understand the reasoning behind the way it is. |
| 06:41.59 | s2krish | yea, would nice to have 8088 should be redirected, |
| 06:43.32 | vader-- | do any of you guys have your asterisk boxes open to the internet so phones from the outside can connect in? |
| 06:43.44 | p3nguin_ | of course |
| 06:44.00 | p3nguin_ | Not much good for an internet phone system to not be on the internet. |
| 06:44.15 | vader-- | didn't know if you knew of any sip vulnerbilities |
| 06:44.44 | p3nguin_ | The biggest one is easy-to-guess username/secret. |
| 06:44.45 | *** join/#asterisk kamh (~kamh@xdsl-1814.wroclaw.dialog.net.pl) |
| 06:44.54 | idespinner | just dont use the extension name as the password... |
| 06:45.03 | p3nguin_ | extension name? |
| 06:45.10 | p3nguin_ | like 4534? |
| 06:45.11 | idespinner | extension number** |
| 06:45.13 | idespinner | yes |
| 06:45.19 | p3nguin_ | People do that? |
| 06:45.24 | fiddur | Yep :) |
| 06:45.26 | idespinner | err yea... |
| 06:45.30 | fiddur | I did on my first installation |
| 06:45.35 | idespinner | i aswell |
| 06:45.35 | fiddur | although, it was firewalled.... |
| 06:45.39 | s2krish | that's true, people have to be educated to use tough password. |
| 06:45.44 | p3nguin_ | WHY?! |
| 06:45.56 | idespinner | because i didnt know any better at the time... |
| 06:46.06 | s2krish | :P |
| 06:46.15 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 06:46.18 | fiddur | p3nguin_: easy to tell folks what to set on their phones... and I trusted the firewall.... |
| 06:46.26 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 06:46.34 | p3nguin_ | You couldn't come up with a better password than the phone number used to reach the devices? |
| 06:46.53 | p3nguin_ | Not even "secret" or "password" ?! |
| 06:46.54 | idespinner | well, it was definately easy to remember... |
| 06:47.12 | idespinner | not all SIP phones can do letters... |
| 06:47.20 | p3nguin_ | Sure they do. |
| 06:47.38 | idespinner | as an example the 3com's cant... |
| 06:47.40 | fiddur | Today I prefer not firewalling the phone server, looking forward to the offset of directly interconnected SIP calls... |
| 06:47.48 | p3nguin_ | Every single SIP phone I have ever seen can support letters in their secrets. |
| 06:47.56 | fiddur | (well, not firewalling SIP-ports, that is) |
| 06:48.05 | p3nguin_ | If it can't, it's pretty useless. |
| 06:50.14 | p3nguin_ | Wow, I can't even imagine a device not allowing letters in the passwords. |
| 06:51.28 | idespinner | the 3Com VCX phone system and all associated phones are numbers only for extension and password... |
| 06:51.28 | idespinner | the phones of course can be used elsewhere, but they are number only... |
| 06:51.31 | p3nguin_ | What about the device's username? |
| 06:51.39 | idespinner | its the extension... |
| 06:52.04 | p3nguin_ | Why should it be required to be the same? That's a poor design. |
| 06:52.10 | vader-- | how about cisco 7940G phones? |
| 06:52.14 | p3nguin_ | What about it? |
| 06:52.20 | p3nguin_ | s/it/them/ |
| 06:52.22 | vader-- | can they support numbers? |
| 06:52.29 | vader-- | i mean letters |
| 06:52.32 | p3nguin_ | well of course. |
| 06:52.44 | p3nguin_ | They support a bunch of characters. |
| 06:52.52 | p3nguin_ | one moment. |
| 06:53.08 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:53.24 | idespinner | cisco's are good |
| 06:53.34 | p3nguin_ | apg -a1 -n1 -m13 -x26 -MSNCL -E^[]{}\"? -s |
| 06:53.34 | idespinner | full character support |
| 06:53.38 | p3nguin_ | not full. |
| 06:53.51 | idespinner | wierd nat things with them though... |
| 06:53.58 | p3nguin_ | It can't support entering the above-listed characters via phone menu. |
| 06:54.09 | p3nguin_ | ^ [] {} " and ? |
| 06:54.45 | p3nguin_ | You could enter them in config files, though. |
| 06:54.48 | idespinner | is it possible to us [ and ] in asterisk? |
| 06:54.55 | idespinner | using \ escape codes? |
| 06:54.57 | p3nguin_ | probably |
| 06:55.08 | p3nguin_ | I don't see why not. |
| 06:55.16 | vader-- | any of you guys running asterisk in vmware? |
| 06:55.18 | p3nguin_ | secret=jfb[49rcf]f9g |
| 06:55.26 | p3nguin_ | should work just fine. |
| 06:56.09 | p3nguin_ | Just don't expect to be able to key it in from the menu of your 7940/7960. |
| 06:57.18 | idespinner | vader--, ive heard of people running it in large deployments, but i'm sure you know the reprocussions of doing so... no meetme, iax2, dahdi... |
| 06:57.22 | p3nguin_ | That apg command is what I decided on for creating Cisco compatible secrets between 13 and 26 characters. |
| 07:00.27 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 07:05.25 | vader-- | why no meetme? |
| 07:07.28 | vader-- | and iax2? |
| 07:07.35 | vader-- | dahdi is hardware support right? |
| 07:09.07 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:10.57 | smooth_penguin | iax2 should work |
| 07:11.09 | smooth_penguin | or I dont see why it wouldnt |
| 07:11.23 | vader-- | same with meetme |
| 07:11.50 | ChannelZ | hmm.. ok in voicemail, you listen to a message, and hit 3 for advanced options. It says "to send a message, press 5". I hit 5, it asks for an extension, and I type in 200. It plays that person's name, but then in the console I get "leave_voicemail: No entry in voicemail config file for '200'" and then it re-reads the options menu |
| 07:11.55 | ChannelZ | ..huh? |
| 07:14.52 | p3nguin_ | bug |
| 07:15.28 | s2krish | i tried to install dahdi, i did $sudo make all. But got make[1]: Entering directory `/home/itosasia/asterisk/dandi/linux /bin/sh: build_tools/make_version_h: Permission denied |
| 07:15.50 | s2krish | make failed |
| 07:16.52 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 07:18.27 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 07:19.03 | ChannelZ | I think it told you why |
| 07:24.15 | s2krish | make -C linux all |
| 07:24.16 | s2krish | make[1]: Entering directory `/home/itosasia/asterisk/dandi/linux' |
| 07:24.18 | s2krish | /bin/sh: build_tools/make_version_h: Permission denied |
| 07:24.20 | s2krish | make[1]: *** [include/dahdi/version.h] Error 126 |
| 07:24.21 | s2krish | make[1]: Leaving directory `/home/itosasia/asterisk/dandi/linux' |
| 07:24.23 | s2krish | make: *** [all] Error 2 |
| 07:24.29 | s2krish | that was output |
| 07:24.41 | vader-- | idespinner why won't any of those things run in vmware? |
| 07:24.56 | ChannelZ | <s2krish> i tried to install dahdi, i did $sudo make all. But got make[1]: Entering directory `/home/itosasia/asterisk/dandi/linux /bin/sh: build_tools/make_version_h: Permission denied |
| 07:25.01 | ChannelZ | See last two words |
| 07:25.32 | s2krish | wat that mean, sudo should have all permission as this run as root |
| 07:27.09 | ChannelZ | that would be for you to figure out |
| 07:27.49 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rxpgcgowiwbxeyto) |
| 07:28.07 | *** join/#asterisk donatas_ (~donatas@office.kis.lt) |
| 07:28.21 | donatas_ | whats is wrong here, because I hear silence.. http://p.defau.lt/?hHQFnsx5t9SzPTM20MoxhA |
| 07:29.52 | ChannelZ | I dunno, what is wrong? |
| 07:30.20 | s2krish | i just did $sudo chmod 777 -R dandhi, worked |
| 07:31.25 | donatas_ | used codec is g729 |
| 07:31.28 | donatas_ | btw |
| 07:31.53 | ChannelZ | donatas: what is happening, as opposed to what you want to happen? your log doesn't really show anything of interest |
| 07:32.13 | donatas_ | ChannelZ: I hear silence, instead of music. |
| 07:32.18 | ChannelZ | Do you have a g729 license? |
| 07:32.32 | FSB_1 | "because I hear silence" |
| 07:32.39 | FSB_1 | How can one hear silence? :D |
| 07:32.59 | ChannelZ | If a tree falls in a forest, will it kill a fuzzy bunny? |
| 07:33.34 | donatas_ | FSB_1: :) it means, that i hear nothing :) |
| 07:33.46 | FSB_1 | Heh |
| 07:33.50 | ChannelZ | RE: Do you have a g729 license? |
| 07:33.56 | donatas_ | ChannelZ: i have license from asterisk.hosting.lv |
| 07:34.02 | FSB_1 | Because silence is the total absense of sounds. Which means you cannot hear it. :P |
| 07:34.22 | FSB_1 | *absence |
| 07:34.35 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 07:35.15 | ChannelZ | so "g729 show licenses" shows something? |
| 07:35.45 | donatas_ | i can only debug "g729 debug" |
| 07:36.13 | ChannelZ | eh? |
| 07:36.28 | donatas_ | no more options after g729, only debug |
| 07:36.38 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 07:36.47 | ChannelZ | what version of * is this? |
| 07:36.55 | donatas_ | 1.4.21 |
| 07:43.06 | ChannelZ | hmm in 1.4 you should have "show g729 licenses" |
| 07:44.25 | donatas_ | no such option |
| 07:44.30 | ChannelZ | oh wait |
| 07:44.49 | ChannelZ | you're using the hacked g729 (IE you do not have a g729 license) |
| 07:45.04 | donatas_ | yes, i use hacked |
| 07:45.14 | ChannelZ | well there you go |
| 07:45.18 | donatas_ | :) |
| 07:45.27 | donatas_ | okey, i will try to buy one, for testing |
| 07:45.38 | ChannelZ | it's not transcoding or registering that it is available to transcode or who knows what |
| 07:46.46 | donatas_ | if i buy g729 license from digium.com, how quick should i get it ? |
| 07:47.39 | ChannelZ | instantly |
| 07:48.04 | ChannelZ | you basically download a little license tool which generates a hostid for your system, sends it to digium, and they send back a license |
| 07:48.27 | donatas_ | oh, roger :) |
| 07:48.35 | donatas_ | going to buy. |
| 07:51.01 | ChannelZ | Huh. They're on sale even. |
| 07:52.10 | *** join/#asterisk Buklov (~buklov@213.138.71.254) |
| 07:52.47 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:57.55 | *** join/#asterisk Kumbang (~kumbang@167.205.24.69) |
| 08:05.11 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:06.05 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 08:11.55 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:16.57 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:17.07 | *** join/#asterisk r15 (~anuj@122.177.252.148) |
| 08:17.14 | r15 | hi everyone. |
| 08:17.37 | r15 | i am getting error ACL error (permit/deny) while registering a user |
| 08:17.53 | r15 | i tried to reload configs as well as rebooted server. |
| 08:18.32 | r15 | but client is getting 404 error and on cli i can see chan_sip.c .. ACL error (permit/deny) |
| 08:18.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:18.57 | r15 | any idea? |
| 08:20.27 | ChannelZ | missed the question |
| 08:21.12 | r15 | i am unable to register a user through xlite, and on asterisk cli i have ACL error |
| 08:22.40 | r15 | chan_sip.c .11393 failed for 'IP' -ACL error (permit/deny) |
| 08:22.46 | r15 | i am not using any tcp wrappers |
| 08:22.50 | s2krish | r15 you sud configure sip.conf and extension.conf |
| 08:23.18 | r15 | earlier it was working |
| 08:23.19 | s2krish | i just success, |
| 08:23.21 | s2krish | http://www.krishnasunuwar.com.np/2010/02/asterisk-installation-and-configuration-guide/ |
| 08:23.56 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 08:24.32 | ChannelZ | well without seeing your sip.conf all we can say is "you have something configured wrong" |
| 08:26.19 | r15 | ok just a moment |
| 08:27.35 | ChannelZ | and more of the console when the phone tries to register |
| 08:29.07 | r15 | ChannelZ: http://www.pastebin.com/d6c3acbc5 |
| 08:29.40 | ChannelZ | see line 3 |
| 08:29.51 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:30.15 | r15 | yes? [chris] |
| 08:30.28 | ChannelZ | no it says [chirs] |
| 08:32.16 | r15 | ohh |
| 08:32.35 | r15 | earlier it worked, someone might have changed that |
| 08:32.56 | ChannelZ | or earlier you were making the same type-o into your softphone.. ? :) |
| 08:33.34 | r15 | i just modified sip.conf with correction to chris |
| 08:33.42 | r15 | reloaded and it worked |
| 08:34.17 | ChannelZ | Praise jebus! |
| 08:34.18 | r15 | ok Thanks ChannelZ |
| 08:34.21 | ChannelZ | sure |
| 08:34.22 | ChannelZ | I'm off to bed |
| 08:34.24 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 08:34.33 | r15 | thanks s2krish |
| 08:34.35 | ChannelZ | have fun |
| 08:34.58 | r15 | yes Good Night ChannelZ |
| 08:37.57 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:38.00 | *** part/#asterisk r15 (~anuj@122.177.252.148) |
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| 08:49.46 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 08:58.36 | *** part/#asterisk gidas (~linas@94.75.253.244) |
| 09:09.12 | *** join/#asterisk themolester (themoleste@cpe-173-171-165-200.tampabay.res.rr.com) |
| 09:09.53 | themolester | what would cause two way audio, but consistant hangup 20s after call answer |
| 09:10.31 | themolester | also, if I wait more rings, it still cuts off at exactly 20s |
| 09:10.58 | themolester | from the time the handset answers, not from incoming call on asterisk |
| 09:11.50 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 09:24.19 | *** join/#asterisk stefanlsd (~stefanlsd@ubuntu/member/stefanlsd) |
| 09:24.45 | stefanlsd | can someone tell me if i plug a phone or a phone line into - Module 0: Installed -- AUTO FXS/DPO |
| 09:25.52 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 09:26.41 | kaldemar | stefanlsd: a phone to FXS, a line to FXO. |
| 09:27.31 | stefanlsd | kaldemar: thanks! i've been trying to debug a channel unavail and i dunno if the lines are plugged in right :) |
| 09:27.51 | themolester | stefanlsd i find it helps to remember which is which by the s at the end |
| 09:27.54 | themolester | think 'server' |
| 09:28.01 | themolester | not sure if that helps... |
| 09:28.28 | stefanlsd | themolester: kk. thanks. i think 'station' is better.. but yeah, helps |
| 09:28.58 | themolester | i'm more of a networking guru who's just starting to get into phones |
| 09:29.15 | themolester | but, that probably makes more sense :) |
| 09:31.35 | kaldemar | themolester: option L() for app Dial could limit the call. what technology are you using? |
| 09:32.26 | themolester | kaldemar sip hardphone/softphone and sip trunk (flowroute) |
| 09:32.34 | themolester | outgoing calls work perfect |
| 09:33.47 | kaldemar | set verbosity to 10, enable sip debug and pastebin cli output of a call. |
| 09:34.20 | themolester | one sec while I clean up a log |
| 09:42.26 | themolester | http://pastebin.ca/1806565 |
| 09:43.03 | themolester | i have some private ips in there, i had externip set but disabled during troubleshooting (backwards, i know... testing nat reflection) |
| 09:43.35 | *** join/#asterisk alexsea7 (~Sheeju_Al@122.169.210.138) |
| 09:44.35 | stefanlsd | ok, im confused again. channel 1 is generated by dahdi_genconf and is signalling=fxo_ks , dahdi scan shows port=1,FXS (so what is it?) |
| 09:45.26 | kaldemar | stefanlsd: FXS devices use FXO signalling, and vice versa. |
| 09:46.37 | stefanlsd | kaldemar: heh. thats where it gets confusing. so i belive dahdi_scan port 1 is fxs - so i plug a phone into that |
| 09:47.06 | themolester | stefanlsd so you had it right the first time on accident? |
| 09:48.02 | stefanlsd | themolester: umm, maybe :) |
| 09:48.02 | kaldemar | themolester: can't see an obvious reason there. you better ask in #freepbx, i don't know what dialparties.agi does. |
| 09:50.01 | themolester | kaldemar it doesn't look like a nat issue though, right? |
| 09:52.23 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 09:53.07 | kaldemar | themolester: if both parties can hear each other, it shouldn't be. there are RTP timeout parameters in sip.conf to hangup a call though, but that would require no RTP/RTCP activity. |
| 09:54.15 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:54.48 | themolester | what if the two endpoints (trunk and phone) are communicating correctly and asterisk is just confused |
| 09:54.57 | themolester | is there a way to disable the timeouts on asterisk? |
| 09:58.15 | kaldemar | doesn't look like asterisk is confused. |
| 09:59.21 | stefanlsd | dahdi show channels is only showing pseudo default default |
| 09:59.33 | stefanlsd | that means i dont have any channels right? |
| 10:00.47 | tzafrir | stefanlsd, any chance you don't include dahdi-channels.conf ? |
| 10:02.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:02.39 | stefanlsd | tzafrir: in chan-dahdi.conf i do include dahdi-channels.conf |
| 10:02.51 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-icopkszlivxuywom) |
| 10:03.01 | tzafrir | stefanlsd, either #include it, or copy its content |
| 10:03.10 | stefanlsd | may be more freepbx related, i'll give it a go there |
| 10:03.56 | stefanlsd | tzafrir: aah, stupid. i was removing the # thinking it was a comment from the includes |
| 10:09.17 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 10:09.27 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 10:15.55 | themolester | stefanlsd I've done that (or been confused and had to double check) on more than one occasion |
| 10:16.49 | themolester | don't know who the genius was that decided to use the same character for includes that is used as comments in many languages (or vice versa) |
| 10:20.59 | *** join/#asterisk matteo (~matteo@openwrt/developer/matteo) |
| 10:21.47 | *** join/#asterisk erbse (~Andre@voxgate.winet.ch) |
| 10:23.50 | *** join/#asterisk Pastoolio (~null@blowfish.x86.co.za) |
| 10:24.26 | Pastoolio | hey ppl. i am looking for something similar to hudlite or flash operator panel where i can see which extentions etc are connect. it needs to also support iax extensions if possible |
| 10:24.58 | Pastoolio | any suggestions |
| 10:25.28 | tuxx- | best opensource switchboard i know is FOP |
| 10:25.29 | tuxx- | ;p |
| 10:26.21 | Pastoolio | tuxx-: the issue i have is its limited screen space |
| 10:26.33 | Pastoolio | seems like it can only handle 36 buttons |
| 10:26.53 | Pastoolio | has anyone looked at isymphony? |
| 10:27.19 | *** part/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
| 10:28.37 | Gido-E | Pastoolio ? |
| 10:28.42 | Gido-E | you can resize the buttons |
| 10:29.19 | Pastoolio | i have tried, is it done in the styles config file? |
| 10:29.37 | Pastoolio | lemme looked at it again |
| 10:30.05 | *** join/#asterisk abhijitd1973 (~79f2de02@gateway/web/freenode/x-ygtjnlkyrsgwvhna) |
| 10:31.45 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 10:31.53 | *** join/#asterisk qasim (~qasim@119.152.62.229) |
| 10:31.57 | Pastoolio | yay, ok cool |
| 10:31.59 | qasim | hello |
| 10:32.02 | Pastoolio | i have it resized |
| 10:32.05 | Pastoolio | thanks Gido-E |
| 10:32.12 | qasim | i need some help using asterisk real time |
| 10:32.20 | tzafrir | Pastoolio, look into monast |
| 10:32.21 | qasim | can any one help me out? |
| 10:33.03 | tzafrir | qasim, maybe. We'll be able to tell better after you ask some specific question |
| 10:33.24 | *** join/#asterisk m0t3jl (~m0t3jl@ip-40.galance.net) |
| 10:33.50 | m0t3jl | Hi, what can cause a SIP channel to refuse to negotiate T.38? |
| 10:34.48 | abhijitd1973 | Hi, I have Asterisk 1.6 and trying to implement chat functionality using ejabbered 2.1.2 on CentOS 5.3 64 bit. The client is ekiga, Ejabbered seems to be configured, asterisk is able to connect to jabber server, still, when message is sent by the user, Asterisk console says - SIP/2.0 405 Method Not Allowed. Any help? |
| 10:34.49 | *** join/#asterisk joobie (~joobz@CPE-124-179-211-169.lns1.lon.bigpond.net.au) |
| 10:35.02 | qasim | first of all i would like to introduce my self. My name is qasim and i am relatively a new user of asterisk |
| 10:35.04 | stefanlsd | sigh. i have another problem. I have two fxo cards with two different phone lines going into them. the one works great, the other one reports alarm and onhook. If i phone it, i get engaged tone. I then swopped the working line to the other channel, and thats engaged. so it looks like its on the card or config... any idea? |
| 10:35.40 | Pastoolio | tzafrir: will do that thanks |
| 10:35.59 | qasim | the steps i followed for my asterisk realtime was first i installed mysql and then asterisk and then i installed asterisk addons |
| 10:36.12 | joobie | hey guys.. im getting this weird behaviour with a sip peer.. i keep getting messages like this "[Feb 23 21:33:49] NOTICE[5120]: chan_sip.c:12723 handle_response_peerpoke: Peer 'pennytel' is now Lagged. (3191ms / 3000ms)" .. "[Feb 23 21:32:04] NOTICE[5120]: chan_sip.c:16223 sip_poke_noanswer: Peer 'pennytel' is now UNREACHABLE! Last qualify: 4215" .. only thing is, if i setup a ping to that host, i get a perfect response.. any |
| 10:36.13 | joobie | <PROTECTED> |
| 10:36.22 | tzafrir | stefanlsd, "alarm" on an FXO line: no line connected |
| 10:36.32 | qasim | i configured asterisk realtime using this link http://hostseries.com/asterisk-realtime-installation-guide/ |
| 10:36.55 | qasim | every one else had it working by using this method |
| 10:37.24 | qasim | but i am getting an error like this [Feb 23 14:00:50] WARNING[19962]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 10:37.25 | qasim | [Feb 23 14:00:50] WARNING[19962]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 10:37.25 | qasim | [Feb 23 14:00:50] NOTICE[19962]: chan_sip.c:21500 handle_request_register: Registration from 'Robert<sip:105@192.168.1.50>' failed for '192.168.1.51' - No matching peer found |
| 10:37.40 | FSB_1 | What about pastebins? |
| 10:37.46 | tzafrir | ~pb |
| 10:37.47 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 10:37.50 | FSB_1 | You're filling up my logs with poop. |
| 10:38.15 | qasim | the main cause that i think is that my asterisk cannot connect to the database |
| 10:38.20 | qasim | can any one help |
| 10:38.21 | qasim | :) |
| 10:38.29 | qasim | i think that my question was too long |
| 10:38.34 | qasim | ill sumit up |
| 10:38.50 | qasim | i just want to connect mysql database with asterisk realtime |
| 10:39.01 | qasim | any one has a good tutorial link or some thing |
| 10:39.05 | qasim | ? |
| 10:39.33 | tzafrir | qasim, right. the relevant message is: found to engine 'mysql', but the engine is not available |
| 10:39.49 | qasim | sorry |
| 10:40.19 | stefanlsd | tzafrir: kk. thanks. i'll check it again. when i swop lines in the two fxo cards i have it works, so i hope its not the module |
| 10:40.42 | qasim | tzafrir can you help me with this issue? |
| 10:40.54 | tzafrir | qasim, I suspect you don't have the module res_config_mysql installed . Not installed, not installed properly, not loaded, whatever |
| 10:41.15 | qasim | yes its res_mysql.conf |
| 10:41.41 | qasim | i copied it to /etc/asterisk and configured it as told in the link i just sent |
| 10:41.47 | qasim | i can also send it again if you like |
| 10:41.48 | tzafrir | qasim, what happens if you run: module unload res_config_mysql.so |
| 10:42.03 | tzafrir | and then: module load res_config_mysql.so |
| 10:42.04 | qasim | yes thats the other thing |
| 10:42.26 | qasim | i read in another tutorial that you have to look for this file but i cudnt find it |
| 10:42.29 | tzafrir | What do you see on the CLI? |
| 10:42.41 | qasim | one sec lemme check |
| 10:42.50 | tzafrir | What version of Asterisk do you use? |
| 10:43.04 | qasim | i am using 1.6 |
| 10:43.14 | qasim | 1.6.2 to be exact |
| 10:43.46 | qasim | yes i got an error trying to unload it |
| 10:43.49 | qasim | ast_unload_resource: Unload failed, 'res_config_mysql.so' could not be found |
| 10:44.11 | *** join/#asterisk eddyo (~edd@dxb-b125537.alshamil.net.ae) |
| 10:44.18 | qasim | how can i install this module? |
| 10:44.29 | qasim | i have allready installed asterisk addons |
| 10:45.28 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 10:46.17 | abhijitd1973 | Hi, I have Asterisk 1.6.0.20 and trying to implement chat functionality using ejabbered 2.1.2 on CentOS 5.3 64 bit. The client is ekiga, Ejabbered seems to be configured, asterisk is able to connect to jabber server, still, when message is sent by the user, Asterisk console says - SIP/2.0 405 Method Not Allowed. Any help? |
| 10:46.49 | kaldemar | qasim: how did you install asterisk-addons? |
| 10:47.04 | qasim | tar -xzf asterisk-addons... |
| 10:47.10 | qasim | then ./configure |
| 10:47.19 | qasim | then make and then make install |
| 10:47.23 | eddyo | hi guys |
| 10:47.25 | eddyo | ztcfg -vv |
| 10:47.40 | eddyo | 4 channels to configure. |
| 10:47.49 | eddyo | how shall i config these :| |
| 10:47.57 | kaldemar | qasim: do you have libmysqlclient-dev installed? |
| 10:48.06 | qasim | kaldemar can i PM you and explain my problem? |
| 10:48.18 | kaldemar | no, we'll do this here. |
| 10:48.25 | qasim | :) ok sorry |
| 10:48.33 | qasim | one sec lemme check |
| 10:48.34 | m0t3jl | Hi, what can cause a SIP channel to refuse to negotiate T.38? |
| 10:48.49 | tuxx- | eddyo: zapgenconf i think :) |
| 10:49.02 | tuxx- | that will make your /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 10:49.14 | eddyo | wrong command |
| 10:49.32 | eddyo | genzaptelconf |
| 10:49.36 | tuxx- | ye, thats the one :) |
| 10:49.36 | eddyo | i even tried this |
| 10:49.38 | eddyo | no luck |
| 10:49.57 | tuxx- | what do you see in /etc/zaptel.conf ? |
| 10:50.11 | qasim | i have installed mysql server and mysql client |
| 10:50.22 | tuxx- | TEARS OF THE DRAGON |
| 10:50.23 | tuxx- | \o/ |
| 10:50.45 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-dbdxkoypqftqufmc) |
| 10:51.41 | kaldemar | qasim: res_config_mysql depends on the development package of mysql client. you need to install it, then re-run configure, make and make install. |
| 10:51.41 | qasim | i think tzafrir was right on my problem |
| 10:51.44 | qasim | i cudnt find res_config_mysql.so |
| 10:52.09 | kaldemar | qasim: your problem is that you don't have the module. it doesn't get installed when you install asterisk-addons unless you do what i told you. |
| 10:52.12 | qasim | as he told me to try to unload and load this module again |
| 10:52.16 | qasim | ok |
| 10:52.20 | tzafrir | eddyo, if you get no errors after that, it means that all was well |
| 10:52.26 | alexsea7 | ERROR[4791] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory, asterisk crashed after this error plase help |
| 10:52.58 | kaldemar | qasim: you can't unload or load it, if you don't have the module. you need to install it first. |
| 10:53.08 | tzafrir | alexsea7, I suspect it is unrelated. It's probably the next module Asterisk tried to load after trying codec_dahdi.so |
| 10:53.37 | qasim | ok i will follow your steps and thanks for the help :) |
| 10:55.15 | alexsea7 | tzafrir: I don;t see any other error in messages or full logs, but asterisk crashed 3 times |
| 10:56.08 | alexsea7 | how do I find out the reason for asterisk crash? |
| 10:58.16 | *** join/#asterisk Tim_Toady (~moi@77.49.236.7.dsl.dyn.forthnet.gr) |
| 10:59.46 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 11:03.01 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 11:07.56 | tzafrir | qasim, in the source directory of asterisk-addons, can you file res_config_mysql.c ? |
| 11:08.07 | tzafrir | Was res_config_mysql.so built from it? |
| 11:08.40 | eddyo | tzafrir |
| 11:08.47 | eddyo | http://pastebin.ca/1806627 |
| 11:08.48 | eddyo | http://pastebin.ca/1806628 |
| 11:08.52 | eddyo | have a look mate |
| 11:08.58 | stefanlsd | tzafrir: 4 WCTDM/4/3 FXSKS (In use) RED(SWEC: MG2) - thats the line plugged in or not... (hardware issue?) |
| 11:10.11 | angryuser | Is it possible to sent a SUBSCRIBE from asterisk to external provider (some third party service) and to monitor the state of that subscribed device ? |
| 11:10.20 | *** join/#asterisk matteo` (~matteo@openwrt/developer/matteo) |
| 11:10.36 | tzafrir | stefanlsd, "RED" - so the card reports it is not plugged |
| 11:10.51 | tzafrir | eddyo, asterisk uses those channels |
| 11:11.08 | eddyo | what am i supposed to do mate? |
| 11:11.17 | tzafrir | no need to further mock at the Zaptel level. Can you state the problem, exactly? |
| 11:11.21 | qasim | one second lemme check |
| 11:11.36 | tzafrir | eddyo, What do you expect to work and doesn't? |
| 11:11.36 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 11:11.50 | eddyo | incoming zap |
| 11:11.56 | eddyo | and outgoing as well |
| 11:12.01 | eddyo | incoming fast hangup |
| 11:12.09 | eddyo | and outgoing all circuits are busy |
| 11:12.20 | eddyo | incoming and outgoing routes are perfectly created |
| 11:12.44 | tzafrir | eddyo, what's the dialplan that you use? |
| 11:12.56 | tzafrir | dialplan show default |
| 11:13.07 | tzafrir | specifically: dialplan show s@default |
| 11:13.26 | qasim | i think it made it |
| 11:13.30 | qasim | [LD] res_config_mysql.o -> res_config_mysql.so |
| 11:13.39 | qasim | this msg was shown when i compiled it |
| 11:13.48 | qasim | should i manually copy it? |
| 11:13.49 | tzafrir | qasim, 'make install' (as root) should install it |
| 11:13.56 | qasim | i did it again |
| 11:14.32 | qasim | kaldemar told me to add libmysqlclient-dev this package and then reinstall addons |
| 11:14.41 | qasim | i did it again but i get the same error message |
| 11:14.41 | eddyo | tzafrir |
| 11:14.42 | eddyo | http://pastebin.ca/1806637 |
| 11:14.53 | eddyo | :| |
| 11:14.58 | angryuser | qasim, you use debian ? |
| 11:15.04 | qasim | yes |
| 11:15.12 | qasim | i am downloading cenos though |
| 11:15.18 | qasim | its about to be finished |
| 11:15.28 | angryuser | its libmysql++-dev |
| 11:15.30 | qasim | and yes i installed it as a rooy |
| 11:15.42 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 11:15.52 | angryuser | its libmysqlclient++-dev |
| 11:15.55 | angryuser | * |
| 11:16.16 | qasim | that package was also available but i will install it and try also |
| 11:16.29 | angryuser | relaunch ./configure script |
| 11:16.40 | angryuser | and check by "make menuselect" |
| 11:16.54 | angryuser | you will see that mysql is selected or not |
| 11:16.54 | qasim | i didnt find any package by this name |
| 11:16.59 | qasim | E: Couldn't find package libmysqlclient++-dev |
| 11:17.00 | tzafrir | qasim, alternatively: aptitude install asterisk-mysql |
| 11:17.01 | qasim | ok lemme check |
| 11:17.12 | angryuser | apt-cache search libmysql |
| 11:17.30 | tzafrir | libmysqlclient15-dev |
| 11:17.52 | qasim | i did menu select |
| 11:17.53 | tzafrir | (which is a build-dep of asterisk-mysql, or rather: asterisk-addons) |
| 11:18.20 | eddyo | tzafrir |
| 11:18.28 | eddyo | HAVE U SEEN what i pasted? |
| 11:18.42 | qasim | and under resource modules i am getting res_config_mysql |
| 11:18.48 | tzafrir | eddyo, yes. If you count on FreePBX, you send them to the wrong context |
| 11:19.09 | angryuser | qasim, all good |
| 11:19.19 | angryuser | qasim, press X |
| 11:19.32 | angryuser | make && make install && make samples (if needed) |
| 11:19.40 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 11:19.54 | qasim | i did |
| 11:19.58 | qasim | :( |
| 11:20.03 | angryuser | qasim, > enjoy ? |
| 11:20.09 | qasim | but it didnt install it in the modules folder |
| 11:20.11 | eddyo | what u mean tzafrir |
| 11:20.14 | qasim | should i manually copy it there |
| 11:20.15 | qasim | ? |
| 11:20.17 | tzafrir | Amazing how much extra work people spend on installing from source when they can just use apt-get |
| 11:20.31 | angryuser | qasim, hm, launch you asterisl |
| 11:20.37 | qasim | :) |
| 11:20.45 | angryuser | qasim, core show module like mysql |
| 11:20.48 | qasim | i am trying your way tzafrir |
| 11:22.00 | angryuser | tzafrir, yes but, that is better for learning |
| 11:22.30 | angryuser | at least first 10 times |
| 11:22.39 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 11:23.22 | qasim | yes for learning point of view its good to compile and install it as i have learned a lot about asterisk this way but both ways are eqaully good if you are a pro like both of you |
| 11:23.23 | qasim | :D |
| 11:24.12 | qasim | yes the file is there now |
| 11:24.53 | qasim | and it is connected to the data base |
| 11:25.26 | qasim | thank you so much tzafrir and angryuser and kaldemar you guys are life saver for me:D |
| 11:25.43 | *** join/#asterisk [netman] (~netman@40.Red-88-17-244.dynamicIP.rima-tde.net) |
| 11:25.55 | qasim | and btw tzafrir's way was the one that worked |
| 11:26.32 | stefanlsd | tzafrir: do io / conflicts happen with these tdm400's? im getting totally erratic results. both lines work, one at a time works. as soon as i put both in, one goes red |
| 11:30.53 | tzafrir | stefanlsd, Is it connected to the PSTN? |
| 11:31.11 | tzafrir | Or to some other device? Maybe bad wiring? |
| 11:32.46 | stefanlsd | tzafrir: straight to phone line.. it is a adsl line with a splitter. maybe the splitter is bad... |
| 11:33.06 | tzafrir | two different lines? |
| 11:39.06 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 11:39.59 | *** join/#asterisk aandrade (~aandrade@189.58.7.253.dynamic.adsl.gvt.net.br) |
| 11:41.54 | stefanlsd | tzafrir: yeah. two different lines. but actually now i suspect a faulty fxo module... unplugged everything. just one line that works, channel 3, fine, plug that into channel 4, red. so it must be the fxo module for channel 4 |
| 11:44.51 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 11:44.56 | lindi- | angryuser: you learn even more if you create your own package |
| 11:45.37 | angryuser | lindi-, yea, about creating package |
| 11:47.22 | asteriskATmarmuD | i want to set up an auto-dialer (don't need GUI, web-interface) - found vicidial dialer (astGUIclient), gnu dialer and AMI - looking for recommendations |
| 11:47.51 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 11:49.14 | angryuser | asteriskATmarmuD, the simpliest auto dialer is from elastix |
| 11:49.20 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 11:49.42 | asteriskATmarmuD | Asterisks AMI seems interesting, but getting automated info on the call status (busy, hung up, answering machine, fax machine etc.) seems complicated or impossible |
| 11:49.50 | asteriskATmarmuD | ok, looking up elastix now |
| 11:49.53 | asteriskATmarmuD | thx |
| 11:50.41 | stefanlsd | tzafrir: sigh. thanks for all your help. will check hardware |
| 11:59.00 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 12:01.03 | garymc | Hi all, is anyone familiar with Polycom phones? My IP330 works ok. When im on the phone and another call comes in my line 2 button flashes. Now the boss has a brand spanking IP650 and when another call comes in while he is on the phone non of his 6 line buttons flash. So he is not noticing other calls coming in. Any reason why this would be ahppening? |
| 12:05.06 | qasim | tzafrir can we install asterisk the same way? if so how can we install sample config files with it? |
| 12:05.09 | qasim | in debian |
| 12:05.28 | tzafrir | qasim, the packages asterisk and asterisk-config |
| 12:05.41 | qasim | actually i am starting with new installation of ubuntu |
| 12:05.44 | qasim | thanks man |
| 12:06.10 | tzafrir | and those are likely to be already installed if asterisk-mysql is installed |
| 12:06.22 | qasim | ok |
| 12:10.13 | asteriskATmarmuD | elastix as dialer possible without GUI/webbrowser? |
| 12:12.33 | zamba | my logs are filling up with warning messages like this: rc_avpair_new: unknown attribute 1490026597 |
| 12:12.37 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 12:12.38 | zamba | what do they mean? |
| 12:13.23 | zamba | looks like it happens everytime a call is routed through the server |
| 12:17.51 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 12:21.34 | tzafrir | asteriskATmarmuD, what do you need? |
| 12:22.00 | garymc | Hi fellas. How do I know if tftp is installed on my ditro of asteriskNOW? |
| 12:23.36 | *** join/#asterisk mohawk (~ross@host217-40-110-153.in-addr.btopenworld.com) |
| 12:24.37 | garymc | well i typed tftp and i think it is there |
| 12:25.43 | garymc | ok I got a folder called tftpboot should i just put my config files in here? |
| 12:25.45 | asteriskATmarmuD | tzafrir: I need a dialer for asterisk, dynamic, fast, no GUI |
| 12:26.06 | tzafrir | asteriskATmarmuD, what phone do you use? |
| 12:26.13 | tzafrir | Where is Asterisk installed? |
| 12:26.31 | asteriskATmarmuD | tzafrir: and an interface to get status on all ongoing calls, another server needs that to take action |
| 12:27.13 | asteriskATmarmuD | tzafrir: debian lenny, we want to call out 200 lines and patch the to inbound/inhouse analog lines |
| 12:27.44 | asteriskATmarmuD | tzafrir: got test-server with asterisk, berofix card (ISDN) and tpm410 (4 analog FXS) |
| 12:28.24 | themolester | what is the correct usage of localnet= with multiple nets... multiple lines, or whitespace deliminated? |
| 12:29.05 | *** part/#asterisk rossh (~ross@host217-40-110-153.in-addr.btopenworld.com) |
| 12:29.28 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 12:29.48 | asteriskATmarmuD | tzafrir: all up and running, just tried to deal with vicidial... |
| 12:30.41 | asteriskATmarmuD | tzafrir: how to get numbers dialed, get status on calls etc. perferably get all info to another server (not asterisk, handling our interviewers) |
| 12:35.03 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:38.48 | *** part/#asterisk alexsea7 (~Sheeju_Al@122.169.210.138) |
| 12:39.50 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:39.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:40.23 | m0t3jl | Hi, what can cause a SIP channel to refuse to negotiate T.38? |
| 12:40.40 | asteriskATmarmuD | ok, let me simply put it this way: is the vicidial approch using meet me rooms to costly, not performant enough? |
| 12:41.05 | qasim | tzafrir: is there any documentation available for asterisk in debian package? |
| 12:41.35 | tzafrir | generally: the package asterisk-doc |
| 12:41.45 | tzafrir | sadly it also includes the huge api docs |
| 12:41.47 | qasim | i mean on the internet? |
| 12:41.55 | tzafrir | (those were removed in later packaging) |
| 12:42.01 | qasim | ok |
| 12:43.14 | coppice | m0t3jl: I expect it doesn't support T.38 |
| 12:47.36 | m0t3jl | coppice, what can I check/look for to find out more about that? Is that a matter of allowed codecs or something? |
| 12:47.37 | coppice | either its not allowed or its not even supported by the equipment |
| 12:47.37 | m0t3jl | coppice, should there be something like allow=t38 ? |
| 12:47.37 | m0t3jl | coppice, I am using t38pt_udptl=yes in the [general] section in sip.conf |
| 12:48.25 | coppice | does the other end support T.38? |
| 12:49.34 | m0t3jl | coppice, are we talking about the actual device on the other side or the SIP provider? |
| 12:52.55 | *** join/#asterisk rossh (~ross@host217-40-110-153.in-addr.btopenworld.com) |
| 12:56.10 | m0t3jl | coppice, because the SIP provider assured me even though faxing over VoIP is not that well supported he supports it (but only using the alaw codec). |
| 12:56.39 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 12:57.05 | *** join/#asterisk matteo (~matteo@openwrt/developer/matteo) |
| 12:58.16 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 12:58.50 | coppice | if he only supports alaw, then obviously T.38 won't work |
| 12:59.47 | *** join/#asterisk H4U (~AdamH@79-121-154-141.eurotel.managedbroadband.co.uk) |
| 13:01.09 | H4U | Recomend any UK VOIP providers that are well estabished in UK? |
| 13:01.32 | H4U | ^that work well with Asterisk/Switchvox |
| 13:03.01 | m0t3jl | coppice, oh... And is it possible to use alaw to transfer faxes? ;) |
| 13:05.16 | coppice | the reliability of using alaw over the internet for faxing is highly variable |
| 13:06.05 | qasim | thanks tzafrir i installed a fresh copy of ubuntu and asterisk with asterisk-mysql and all looks good :) |
| 13:06.32 | qasim | now i just have to configure according to my task |
| 13:07.08 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 13:07.11 | shamelessn00b | Hi all |
| 13:07.18 | shamelessn00b | anyone used zanzibar with asterisk? |
| 13:09.08 | *** join/#asterisk klashniv (~klashniv@41.191.76.68) |
| 13:12.12 | m0t3jl | coppice, okay, but what would I have to do in order to get it working? ;) |
| 13:15.05 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-mtsywhkbrtrgihxe) |
| 13:17.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:18.23 | *** join/#asterisk fink (~guest@32.165.152.86) |
| 13:23.42 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:29.47 | garymc | Can anyone help me with my Tftp stuff. Ive got a Polycome IP550 im trying to get to use the Tftp to boot of. Ive created a Macaddress.cfg an phone204.cfg and a sip.cfg |
| 13:30.15 | garymc | my polycom is saying error loading macaddress.cfg |
| 13:31.10 | *** join/#asterisk nickaugust (~anonymous@207-224-59-105.hlrn.qwest.net) |
| 13:31.15 | garymc | macaddress being 004F21FD553 |
| 13:31.26 | garymc | not that means anything |
| 13:31.28 | garymc | :S |
| 13:31.42 | FSB_1 | Yes it does. |
| 13:31.50 | FSB_1 | It will be hacked by noon. |
| 13:31.55 | garymc | not that that means anything to you ( i meant) :P |
| 13:32.19 | garymc | just aswell i put the wrong mac number in lo9l |
| 13:32.21 | garymc | lol |
| 13:32.27 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 13:33.09 | tzafrir | garymc, tcpdump is your friend |
| 13:33.21 | garymc | ive never heard of tcp dump |
| 13:33.22 | tzafrir | as well as the logs of the tftpd |
| 13:33.40 | garymc | I havent got a tftp log. Ive just looked in /var/log |
| 13:33.44 | tzafrir | a very intuitive name, considering tftp is UDP |
| 13:33.57 | garymc | sorry tftp |
| 13:34.18 | tzafrir | garymc, tcpdump is a simple packet sniffer |
| 13:34.24 | garymc | ok |
| 13:34.25 | FSB_1 | No |
| 13:34.26 | FSB_1 | Dumper |
| 13:34.31 | tzafrir | tcpdump -n 'udp port 69' |
| 13:35.18 | SuPrSluG | garymc: don't use capital letters in <mac>.cfg |
| 13:35.19 | tzafrir | or better: |
| 13:35.20 | tzafrir | tcpdump -v -n 'udp port 69' |
| 13:36.16 | garymc | that doesnt work |
| 13:36.54 | garymc | [TK]D-Fender : Where can i get some decent example tftp cfg files for my polycom phones? |
| 13:37.44 | BCS-Satori | garymc: polycoms site when you download firmware has config files in the .zip |
| 13:38.11 | SuPrSluG | garymc: it comes with an example -> 000000000000.cfg |
| 13:38.12 | *** join/#asterisk captiancrash (~jonathan@12.71.218.232) |
| 13:38.14 | FSB_1 | garymc: Are you a novice with Linux? |
| 13:38.24 | garymc | FSB_1 yes |
| 13:38.38 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 13:38.44 | FSB_1 | Don't you think you should be familiar with Linux and troubleshooting on linux before trying to configure telephony services? |
| 13:38.54 | SuPrSluG | garymc: if you use capital letters in the mac it won't work |
| 13:39.01 | *** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com) |
| 13:39.06 | garymc | Right im looking at the firmware and theres lots of differnt versions. How do I know what Boot version my phone is running as it is a second hand ip550 |
| 13:39.17 | garymc | ok |
| 13:39.26 | tzafrir | FSB_1, that's an odd requirement. What Linux has to do with telephony? |
| 13:39.52 | FSB_1 | tzafrir: At least he would know if he reaced the tftpserver and whatnots,. |
| 13:40.07 | FSB_1 | With the troubleshooting tools linux distributions often provide. |
| 13:40.21 | FSB_1 | Since he installs it ontop of a linux distribution. |
| 13:40.27 | shamelessn00b | hey, anyone used zanzibar to integrate asterisk with spjinx4\?? |
| 13:41.04 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 13:41.08 | FSB_1 | tzafrir: Am I thinking about this stupidly? |
| 13:41.33 | FSB_1 | IMO to many people do stuff without even having the basics nailed down. |
| 13:41.50 | FSB_1 | (Regardless of what, and yes, me too at times.) |
| 13:42.24 | [TK]D-Fender | garymc: Go look in your phones menus |
| 13:43.08 | [TK]D-Fender | FSB_1: No, he's running off a cooked GUI install ISO and has been running on luck. |
| 13:43.31 | [TK]D-Fender | FSB_1: WARNING! : chan_headlesschicken.so is already loaded! |
| 13:43.39 | shamelessn00b | lol |
| 13:43.43 | FSB_1 | ;D |
| 13:43.47 | garymc | [TK]D-Fender : Iam very lucky |
| 13:43.52 | garymc | always |
| 13:43.54 | garymc | :) |
| 13:44.30 | [TK]D-Fender | garymc: http://farm4.static.flickr.com/3019/2685319969_21f126ce34_o.jpg |
| 13:44.40 | BCS-Satori | Could someone tell me what I am doing wrong, I am trying to configure if an extension on a phone dials itself it checks its own voice-mail. This is what I have so far but doesn't work "exten = _X.,1,GotoIf($["${CALLERID(num)}" = "${CALLERID(num)}"]?VoiceMailMain(${CALLERID(num)}))" |
| 13:44.42 | garymc | do i even want to look at that? |
| 13:45.26 | [TK]D-Fender | BCS-Satori: You are comparing the Caller ID to... THE CALLER ID. FRFs of course its always gonna be the same as ITSELF |
| 13:45.32 | [TK]D-Fender | FFS* |
| 13:45.46 | garymc | [TK]D-Fender that was made up by an unlucky person or someone who got lucky once then unlucky the rest of their life |
| 13:45.47 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 13:46.26 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:48.06 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:48.59 | jaytee | that was a funny demotivate poster |
| 13:49.20 | beek | mornin' [TK]D-Fender, jaytee. |
| 13:50.10 | jaytee | morning beek |
| 13:50.50 | [TK]D-Fender | jaytee: I've got a large collection :) |
| 13:52.08 | ManxPower-work | BCS-Satori: CALLERID(num) will ALWAYS match CALLERID(num) |
| 13:52.16 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 13:52.27 | ManxPower-work | Maybe you want $["${CALLERID(num)}" = "${EXTEN}"] |
| 13:53.33 | BCS-Satori | ManxPower-work: ya I just changed it to that; Is it correct to use "_X." When I attempt to dial it, calls are using stdexten not [default] where this is placed |
| 13:55.16 | [TK]D-Fender | BCS-Satori: Its your dialplan... since when should we assume what any of your extens, macros, or contexts look like? |
| 13:55.30 | [TK]D-Fender | BCS-Satori: or that "stdexten" is meant to mean anything specific at all? |
| 13:56.07 | [TK]D-Fender | BCS-Satori: "exten = _X.,1,GotoIf($["${CALLERID(num)}" = "${CALLERID(num)}"]?VoiceMailMain(${CALLERID(num)}))" <- this isn't from a macro anyway |
| 13:58.34 | ManxPower-work | [TK]D-Fender: looks like he's doing stuff without understanding it. |
| 14:00.31 | ManxPower-work | Many people use _X. I call those "lazy people" |
| 14:01.04 | [TK]D-Fender | ManxPower-work: thats another matter, and one I wouldn't independently slam someone for before seeing the big picture |
| 14:01.37 | [TK]D-Fender | ManxPower-work: Especially when there's so much more other stuff :) |
| 14:02.08 | BCS-Satori | ManxPower-work: that is why I asked if that is correct or a better way then using "_X." |
| 14:02.45 | ManxPower-work | BCS-Satori: use something more specific like _4XXX |
| 14:03.26 | ManxPower-work | how does the system know you are dialing an extension or are dialing an outside number? |
| 14:04.03 | *** join/#asterisk ttwhy (~tekkno@p4FECF0E3.dip.t-dialin.net) |
| 14:06.29 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 14:06.45 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net) |
| 14:09.10 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 14:09.19 | Katty | shivers |
| 14:09.40 | Katty | WHY MUST YOU BE SO COLD WEATHER |
| 14:09.44 | Katty | WHAT HAVE I DONE TO YOU |
| 14:11.02 | beek | mornin' Katty |
| 14:11.09 | Katty | huddles with beek |
| 14:11.33 | *** join/#asterisk Cuz (~plastik@mail.gradeatechs.com) |
| 14:13.32 | garymc | My IP550 says couldnt connect to boot server |
| 14:13.39 | garymc | any ideas? |
| 14:14.23 | garymc | how would i know if my tftp is working? |
| 14:14.28 | [TK]D-Fender | garymc: Try being more vague |
| 14:14.41 | [TK]D-Fender | garymc: Get a tftp client and try it by hand yourself |
| 14:16.38 | *** join/#asterisk d-k-t-2 (~D@112.202.232.46) |
| 14:18.53 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 14:19.05 | elliot98 | gives a hearty wave to all |
| 14:19.32 | *** join/#asterisk high-freq (~hfreq@99.188.122.87) |
| 14:19.42 | beek | [TK]D-Fender: You're in rare form today! |
| 14:20.17 | [TK]D-Fender | beek: Not too different from the norm... |
| 14:20.47 | beek | Ahh, but the pithy comments are much quicker today. |
| 14:21.09 | [TK]D-Fender | beek: Multi-tasking is better tuned ;) |
| 14:23.17 | *** join/#asterisk matteo` (~matteo@openwrt/developer/matteo) |
| 14:26.08 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 14:26.19 | Skeeter- | anyone having trouble downloading from asterisk? |
| 14:26.20 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:29.25 | Skeeter- | just came back |
| 14:30.45 | TheDavidFactor | Skeeter-, yep I was seeing the same thing, seems to be working fine now though |
| 14:31.56 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:31.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:32.04 | Skeeter- | TheDavidFactor, good to hear i wasnt the only one |
| 14:33.42 | *** join/#asterisk skymeyer (~skymeyer@91.183.54.9) |
| 14:35.49 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 14:38.27 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:306b:7d5f:a077:acde) |
| 14:38.30 | cusco | hi |
| 14:39.07 | cusco | sip client --> local asterisk -- DIAL --> remote asterisk --> some queue... |
| 14:39.27 | cusco | does the dialplan in the localasterisk need to Answer() before the dial= |
| 14:39.27 | cusco | ? |
| 14:39.36 | cusco | before dialing to remote asterisk? |
| 14:39.43 | *** join/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 14:40.09 | [TK]D-Fender | cusco: No |
| 14:40.52 | cusco | [TK]D-Fender: that is how I have it, then while I am queuing in remote asterisk, I can't hear any on-hold sound |
| 14:41.12 | cusco | should I use Playback() or something before the Dial() ? |
| 14:41.21 | [TK]D-Fender | cusco: Unrelated |
| 14:41.25 | *** join/#asterisk qasim (~qasim@115.186.29.216) |
| 14:42.36 | cusco | thing is, if I Answer() first I can hear the queueing sound from remote asterisk... |
| 14:43.06 | [TK]D-Fender | cusco: Should only need to do so on the remote side, and you should be preventing reinvites. |
| 14:43.07 | cusco | but if I don't Answe() and her no sound, does it take less Bandwidth? |
| 14:43.20 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 14:43.25 | cusco | (sorry about mistyping) |
| 14:44.26 | cusco | if I playback() some sound locally until call is answered, does the on-hold sound from remote asterisk travel to local asterisk? |
| 14:44.27 | [TK]D-Fender | cusco: No. |
| 14:44.40 | cusco | oh :/ |
| 14:45.09 | [TK]D-Fender | [09:44]<cusco>if I playback() some sound locally until call is answered, does the on-hold sound from remote asterisk travel to local asterisk? <-- I wasn't saying no to this |
| 14:45.16 | [TK]D-Fender | [09:43]<cusco>but if I don't Answe() and her no sound, does it take less Bandwidth? <- no |
| 14:45.19 | cusco | yes I unserstood |
| 14:45.35 | cusco | the second questiong would therefore have "yes" as a reply... right? |
| 14:46.02 | cusco | sound keeps travelign even tho I don't hear it, thus not saving any BW |
| 14:47.50 | [TK]D-Fender | cusco: pastebin an actual call for us to examine |
| 14:50.35 | qasim | tzafrir do we have to install mysql-server exclusively or we are good to go when we install asterisk-mysql? |
| 14:52.04 | tzafrir | qasim, do you want to run a mysql server on that system? |
| 14:52.04 | tzafrir | if so: install mysql-server? |
| 14:52.04 | qasim | yes i want my database in that same server |
| 14:52.04 | tzafrir | if so: install mysql-server |
| 14:52.04 | qasim | ye si did install it |
| 14:52.05 | qasim | and it is now giving me error |
| 14:52.10 | qasim | MySQL RealTime: Invalid database specified: asterisk |
| 14:52.20 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:52.30 | qasim | i have database called asterisk and relative tables in it |
| 14:52.55 | donatas_ | what is the problem ? http://asterisk.pastebin.com/JrRDvAMK |
| 14:53.03 | donatas_ | i cant check license |
| 14:53.10 | adnc | sometimes i have in my CLI this message: -- Got SIP response 400 "Bad Request" back from 192.168.193.201 |
| 14:53.18 | adnc | is this something i should worry about? |
| 14:53.23 | donatas_ | no |
| 14:53.24 | qasim | http://hostseries.com/asterisk-realtime-installation-guide/ |
| 14:53.41 | qasim | this is the link where you can find database tables for asterisk |
| 14:53.53 | [TK]D-Fender | adnc: Depends what it is a RESPONSE TO |
| 14:54.37 | adnc | [TK]D-Fender, eveything works fine as far as i can see, would you recommend investing more research into it? |
| 14:55.23 | adnc | 201 is a nokia sip client |
| 14:56.12 | donatas_ | who are using g729 codec? |
| 14:56.27 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 14:57.04 | qasim | can anyone tell me how to access asterisk documentation files? |
| 14:57.06 | [TK]D-Fender | adnc: That still answers nothing |
| 14:58.10 | adnc | [TK]D-Fender, i don't know for what it is a response to |
| 14:58.15 | adnc | i have it randomly |
| 14:58.26 | [TK]D-Fender | adnc: Well go look then. |
| 14:58.35 | adnc | and i don't know how to find out for what it was a response to |
| 14:59.31 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 14:59.52 | donatas_ | wtf |
| 14:59.52 | donatas_ | asterisk*CLI> g729 show licenses |
| 14:59.53 | donatas_ | 0/0 encoders/decoders of 0 licensed channels are currently in use |
| 15:00.25 | donatas_ | as i run # ./benchg729, i got Unable to check for valid G.729 licenses. |
| 15:02.05 | *** join/#asterisk heliosj (~jeff@i216-58-41-253.cybersurf.com) |
| 15:02.22 | *** join/#asterisk collink (~collink@adsl-065-082-196-004.sip.asm.bellsouth.net) |
| 15:03.16 | heliosj | I know this is a fairly vague question, but has anyone had an issues with Eyebeam dropping calls after 30 or 90 seconds (exactly) after a recent upgrade? Both 1.4 and 1.6.2.x? |
| 15:05.01 | *** join/#asterisk WinZ (~winz@82.146.61.218) |
| 15:05.52 | WinZ | guys, is it possible to forward incoming fax call from asterisk to an external number (jFax) and successfully receive fax? |
| 15:06.15 | voipmonk | heliosj: might be wireshark time |
| 15:07.00 | Katty | so i think i'm going to get rid of the bird bath camera |
| 15:07.05 | Katty | and setup FerretCam instead. |
| 15:08.18 | *** join/#asterisk giesen (giesen@dirtypackets.net) |
| 15:08.41 | *** join/#asterisk MAbbas (~abbas4u@203.215.177.194) |
| 15:09.22 | giesen | I'm trying to troubleshoot an MOH problem in 1.4, in the console the MOH starts and immediately stops |
| 15:09.25 | giesen | <PROTECTED> |
| 15:09.28 | giesen | -- Stopped music on hold on SIP/5544-08be4250 |
| 15:09.37 | giesen | but I'm unable to determine what's causing it |
| 15:09.48 | giesen | are there any debug commands available for moh |
| 15:09.52 | heliosj | voipmonk: Sure, that sounds like effort. I was hoping someone might have a magic answer. ;) |
| 15:09.52 | giesen | I dont seem to be able to find any |
| 15:10.00 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 15:10.13 | MAbbas | Hi All, whats the load limit for asterisk .. I am using ver 1.4.20.1 |
| 15:10.38 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 15:11.11 | qasim | i am relatively new but i think that it depends on your hardware not the software |
| 15:11.47 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:11.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:13.28 | Katty | hi Deeewayne |
| 15:13.53 | Deeewayne | morning Katty :-) |
| 15:14.23 | Katty | bmoraca_work: ping |
| 15:14.38 | qasim | hello when i try to load res_config_mysql.so module it gives me this error |
| 15:14.42 | qasim | MySQL realtime: no requirements setting found, using 'warn' as default. |
| 15:14.44 | *** join/#asterisk matteo` (~matteo@openwrt/developer/matteo) |
| 15:14.47 | qasim | can any one help please |
| 15:15.12 | qasim | tzafrir? it gives me this error when i try to reload that module |
| 15:15.21 | qasim | MySQL realtime: no requirements setting found, using 'warn' as default. |
| 15:15.35 | *** part/#asterisk matteo` (~matteo@openwrt/developer/matteo) |
| 15:16.20 | *** join/#asterisk clintc (~clintc@n128-227-126-206.xlate.ufl.edu) |
| 15:17.27 | [TK]D-Fender | giesen: Look at your MoH class definition. Look where it says its getting MoH from. is there anything there? |
| 15:17.57 | giesen | yes, it's definitely there |
| 15:18.30 | giesen | it's pulling music from a streaming source |
| 15:18.32 | giesen | using a script |
| 15:18.41 | giesen | when I run the script on its down, it's definitely getting output |
| 15:18.45 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.19.170) |
| 15:18.46 | giesen | s/down/own/ |
| 15:20.35 | *** join/#asterisk basty (~basty@2001:4cd8:1:0:21c:b3ff:fec2:ec18) |
| 15:20.41 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 15:20.43 | basty | Hi |
| 15:21.27 | qasim | can anyone help me with this error.. "MySQL RealTime: Invalid database specified: asterisk" |
| 15:21.52 | qasim | i am using asterisk realtime but it seems that i cant connect to my database |
| 15:21.55 | qasim | can anyone help |
| 15:21.57 | qasim | ?? |
| 15:22.01 | basty | I have updated asterisk 1.2 to asterisk 1.4 - everything worked well. But now I noticed something weirdo. As soon as I try to call somebody external and the external puts me into "hold" - I hear my OWN musiconhold instead of his... |
| 15:22.25 | Katty | so i'm thinking about taking down camera number two, and putting it in front of the ferret cage |
| 15:23.00 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 15:23.36 | qasim | tzafrir can you help me with this? |
| 15:24.05 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 15:24.49 | *** part/#asterisk jayesh (~jay@122.172.124.160) |
| 15:24.59 | tzafrir | qasim, is that database available from any other mysql client? Are you familiar with mysql? |
| 15:25.15 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 15:25.27 | qasim | yes i am using phpmyadmin |
| 15:25.34 | qasim | and it is available there |
| 15:26.12 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:26.32 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:26.47 | *** join/#asterisk tonywho (~cibnetsac@200.121.247.79) |
| 15:26.58 | tonywho | hello |
| 15:28.08 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 15:28.29 | qasim | tzafrir: WHen i try to unload and then load the module it give me an error "MySQL realtime: no requirements setting found, using 'warn' as default." |
| 15:28.41 | qasim | may be there is some small settings that i am missing? |
| 15:29.17 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 15:29.22 | tonywho | need someone to login my server |
| 15:29.39 | ttwhy | Hi, how do i increase the sound quilty? whats the best codec? |
| 15:29.50 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:29.54 | tzafrir | is not really familiar with realtime |
| 15:30.13 | tzafrir | anybody here is? |
| 15:30.39 | tonywho | im here |
| 15:30.52 | qasim | are you familliar with asterisk realtime? |
| 15:31.00 | qasim | tonywho... |
| 15:31.05 | tonywho | im a begginer |
| 15:31.22 | tonywho | i have my asteisk runuing now |
| 15:31.36 | tonywho | asterisk + a2billing |
| 15:31.41 | coppice | ttwhy: what are you trying to increase from? |
| 15:31.47 | tonywho | but still have questions |
| 15:35.01 | [TK]D-Fender | tonywho: Maybe you should get around to asking them... |
| 15:35.12 | tonywho | hi |
| 15:35.29 | qasim | any one familiar with realtime? |
| 15:37.08 | [TK]D-Fender | qasim: You haven't shown us anything. |
| 15:39.39 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:39.48 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 15:40.41 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 15:40.47 | Katty | tonywho: why are you sending me private messages |
| 15:41.41 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:41.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:43.40 | [TK]D-Fender | Katty: A dangersous mix of lack of cummunications skills, desperation, and lack of good graces? |
| 15:44.15 | *** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net) |
| 15:45.55 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 15:47.36 | tonywho | i just said hi |
| 15:48.19 | qasim | [TK]D-Fender: when i try to register a peer i get this error MySQL RealTime: Invalid database specified: asterisk |
| 15:48.33 | [TK]D-Fender | qasim: Still don't see anything... |
| 15:49.51 | qasim | ok here's the thing |
| 15:49.57 | qasim | i am using asterisk realtime |
| 15:50.18 | qasim | i have configured my system i.e the config files as well as the databases |
| 15:50.26 | Katty | tonywho: you just said hi.... to me in private.... to someone you don't even know? |
| 15:50.38 | Katty | tonywho: heh |
| 15:50.47 | qasim | but what i think is that it is unable to connect to the database |
| 15:51.06 | qasim | i think that i am missing some setting in the conf files |
| 15:51.18 | qasim | http://hostseries.com/asterisk-realtime-installation-guide/ |
| 15:51.29 | qasim | this is the link from where i am configuring my system |
| 15:51.30 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 15:51.32 | Assid | heya |
| 15:51.39 | ttwhy | coppice, its a ulaw codec. I'am trying to get a dial in connection which links chan_capi to SIP ... but till now, the quality is awful. I think its because of the codec (but iam quite new to asterisk, so maybe it depends on more than just the codec ;) ) |
| 15:51.49 | qasim | do you need more information? |
| 15:52.07 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:52.43 | coppice | ttwhy: your problem is not the codec. ulaw is what the PSTN uses |
| 15:53.19 | coppice | although if you are using chan_capi chances are you are in a country where you should really be using alaw |
| 15:53.36 | [TK]D-Fender | qasim: Where do I see YOUR configs? Where do i see your proving you can conenct to it with those credentials via a direct client? |
| 15:53.41 | [TK]D-Fender | qasim: PASTEBIN is your friend. |
| 15:53.44 | [TK]D-Fender | ~pb |
| 15:53.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:54.28 | Assid | so i have a question.. if i have dial and i sent a parameter for music on hold, and mention the context.. does that mean once the person picks up the call and puts it on hold again, they will continue to get the predefined context |
| 15:54.36 | Assid | or will they get the default context |
| 15:54.48 | tonywho | chill out |
| 15:55.29 | qasim | <[TK]D-Fender> do you want me to send the config file? |
| 15:55.52 | Assid | or if i set using SetMusicOnHold |
| 15:55.54 | [TK]D-Fender | qasim: If you expect any help you'll pastebin all of the kinds of information I have suggested. |
| 15:56.33 | qasim | ok |
| 15:56.43 | qasim | what do you want me to do? |
| 15:56.56 | [TK]D-Fender | [10:53]<[TK]D-Fender>qasim: Where do I see YOUR configs? Where do i see your proving you can conenct to it with those credentials via a direct client? <- WAKE UP |
| 15:58.34 | elliot98 | things it's great to be back |
| 15:58.41 | qasim | http://pastebin.com/t0XexZYc |
| 15:59.09 | elliot98 | wonders if Katty's squirels are still being well fed |
| 15:59.13 | qasim | this is in the file res_mysql.conf |
| 16:00.25 | [TK]D-Fender | wasAnd the rest? |
| 16:00.30 | [TK]D-Fender | qasim: And the rest? |
| 16:00.32 | Katty | elliot98: indeed. in fact i've got to refill the food bin for them this weekend |
| 16:00.41 | qasim | ok one sec |
| 16:01.11 | qasim | can i send you the link from which i am following? |
| 16:01.21 | Katty | elliot98: i'm about to set up FerretCam |
| 16:01.33 | Katty | elliot98: hopefully i'll have some time to do it tonight or this weekend if not tonight |
| 16:02.17 | elliot98 | looks out into the distance with a content smile |
| 16:02.23 | elliot98 | thumbs up Katty |
| 16:02.37 | qasim | http://pastebin.com/bqBkfG5k |
| 16:02.48 | qasim | this is in the file extconfig.conf |
| 16:03.09 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 16:03.23 | [TK]D-Fender | qasim: Keep going... |
| 16:03.43 | qasim | http://pastebin.com/Jb2tRXYy |
| 16:03.58 | qasim | this is in the file extentions.conf |
| 16:04.02 | [TK]D-Fender | .... |
| 16:04.14 | qasim | and thats preety much it |
| 16:04.31 | qasim | these were the only settings that i found in the tutorial |
| 16:04.43 | qasim | now is there anything i am missing? |
| 16:05.09 | [TK]D-Fender | qasim: Where do i see you connecting to it LOCALLY like I asked? Where do i see proof that the PID file is where it says it is and that the DB server is running? |
| 16:05.53 | qasim | i am using phpmyadmin and i will paste its information just a second |
| 16:06.22 | [TK]D-Fender | qasim: NO, not PHPmyadmin. |
| 16:06.25 | [TK]D-Fender | CLI CLIENT! |
| 16:06.40 | qasim | then?? |
| 16:06.43 | [TK]D-Fender | qasim: and show me that its running and the sock file is where tis supopsed to be. |
| 16:07.07 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 16:07.58 | qasim | socket /var/run/mysqld/mysqld.soc |
| 16:08.03 | qasim | socket /var/run/mysqld/mysqld.sock |
| 16:08.19 | qasim | this is what i am getting from phpmyadmin variables page |
| 16:08.24 | qasim | and yes the file is there |
| 16:08.26 | qasim | i checked |
| 16:08.39 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 16:09.33 | qasim | pid file /var/run/mysqld/mysqld.pid |
| 16:09.42 | qasim | this is the pid of mysql server |
| 16:09.46 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:10.07 | qasim | does this help? |
| 16:12.57 | qasim | <[TK]D-Fender> do you need some more information?? |
| 16:13.25 | [TK]D-Fender | qasim: Show me that you can connect at cli using the credentials you have shown |
| 16:17.07 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:17.39 | qasim | http://pastebin.com/mXbJAYGb |
| 16:17.41 | qasim | there you go |
| 16:17.50 | *** join/#asterisk lbarth (~lbarth@62.216.165.71) |
| 16:18.06 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 16:18.25 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 16:19.10 | qasim | or better yet |
| 16:19.11 | qasim | http://pastebin.com/dismTTev |
| 16:19.25 | qasim | this shows that i am connected to my database "asterisk" |
| 16:20.04 | drfreeze | Hello |
| 16:20.22 | *** part/#asterisk stefanlsd (~stefanlsd@ubuntu/member/stefanlsd) |
| 16:20.25 | drfreeze | Anyone know if it is possible to change the ring type of a parked call when it rings back |
| 16:20.47 | [TK]D-Fender | qasim: Show me your tables. |
| 16:21.14 | [TK]D-Fender | drfreeze: what ring type? |
| 16:22.12 | drfreeze | [TK]D-Fender: the sound of the ring. Options are 1-12 on a polycom phone. |
| 16:22.21 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 16:22.22 | ManxPower-work | drfreeze: Yes, but don't ask me how I did it. It was several years ago. Basically when the call times out, make sure it timesout to an extension where you change the Alert info, then send the call on to the original device. |
| 16:22.27 | [TK]D-Fender | drfreeze: .... Maybe you should look at what is being called... |
| 16:22.49 | drfreeze | I usually control the ring with a SIPAddHeader call |
| 16:23.06 | qasim | http://pastebin.com/C5dkbWgm |
| 16:23.14 | qasim | this is my table for sip_buddies |
| 16:23.15 | *** join/#asterisk basty (~basty@2001:4cd8:1:0:21c:b3ff:fec2:ec18) |
| 16:23.18 | basty | hi again |
| 16:23.55 | qasim | http://pastebin.com/9EDSYgTW |
| 16:24.07 | qasim | this is table for extentions |
| 16:24.12 | basty | there must be something weirdo with musiconhold on asterisk 1.4.29 - as soon as I try to call an external (not asterisk) pbx and this one set me on hold - I am listening to my own musiconhold music instead of his. Anyone could confirm that ? |
| 16:24.39 | qasim | http://pastebin.com/ERH8FhAb |
| 16:24.50 | qasim | this is for voice messages |
| 16:24.51 | ChannelZ | basty: Yours is probably better anyway |
| 16:24.51 | [TK]D-Fender | drfreeze: And what is * doing? |
| 16:25.16 | basty | ChannelZ: true...but actually i want to listen to the external.. ;-) |
| 16:25.26 | [TK]D-Fender | qasim: dbhost = 127.0.0.1 <- comment this line out and restart * |
| 16:25.35 | *** join/#asterisk Erestar (~jim@c-98-236-90-228.hsd1.wv.comcast.net) |
| 16:25.38 | qasim | ok one second |
| 16:26.14 | Erestar | Can anyone give me the right combination of spaces and quotes to make this do what I want it to? GotoIf(${SYSTEMSTATUS}=SUCCESS?success:fail) |
| 16:26.23 | drfreeze | ManxPower-work: I'm looking at features.conf and don't see how to direct a timed-out parked call to an ext. |
| 16:26.28 | Erestar | Where ${SYSTEMSTATUS} will be either SUCCESS or APPERROR |
| 16:26.52 | ManxPower-work | drfreeze: It's not documented and what little documentation there is, is wrong. You'll have to debug it by watching the dialplan |
| 16:26.57 | Erestar | The above GotoIF express is always evaluating to true no matter what I try |
| 16:27.04 | ChannelZ | basty: well it's hard to say without seeing some console and figuring out what might be going on; Either the remote system is sending a strange invite to 127.0.0.1 or a LAN IP that happens to match your own, or who knows... |
| 16:27.04 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 16:27.09 | guax | ~book |
| 16:27.10 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:27.15 | *** join/#asterisk heit0050 (~heit0050@mail2.heitkeconsulting.com) |
| 16:27.23 | ChannelZ | runs off to work |
| 16:27.30 | drfreeze | ManxPower-work: Did you controll it with context? |
| 16:27.37 | ManxPower-work | GotoIf($[${SYSTEMSTATUS}=SUCCESS]?success:fail) There are literally thousands of examples of this on the web |
| 16:27.45 | [TK]D-Fender | Erestar: Would help if you put an EXPRESSION in there. |
| 16:27.51 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 16:27.57 | [TK]D-Fender | Erestar: go read the CHANNELVARIABLES doc again |
| 16:27.58 | basty | ChannelZ: with a sip debug I could figure it our, right ? ;-) |
| 16:28.01 | ManxPower-work | drfreeze: IIRC, yes. |
| 16:28.08 | drfreeze | cool, thanks |
| 16:28.17 | qasim | load_mysql_config: MySQL RealTime: No database host found, using localhost via socket |
| 16:28.19 | ManxPower-work | drfreeze: I did this in like 1.2.12 |
| 16:28.22 | qasim | this is the error i got |
| 16:28.47 | [TK]D-Fender | qasim: that's only a warning so far. See if it works |
| 16:29.34 | elliot98 | I understand why asterisk would open UDP sockets...but why is it opening up a bunch of Unix sockets? |
| 16:29.35 | Erestar | [TK]D-Fender, ${SYSTEMSTATUS}=SUCCESS isn't an expression? ;) |
| 16:29.46 | [TK]D-Fender | Erestar: No, it isn't |
| 16:29.47 | ManxPower-work | Erestar: NO IT IS NOT! |
| 16:29.49 | qasim | no i tried to register sip it gives the same error |
| 16:30.02 | [TK]D-Fender | qasim: Show me.. |
| 16:30.09 | ManxPower-work | Erestar: That is why you MUST read the book. |
| 16:30.32 | [TK]D-Fender | Erestar: and that doc in your tarball I told you to refer to |
| 16:30.46 | heit0050 | I recently turned on monitoring on my queue with Mix-monitor. The audio is in one file, but I hear one party only for the first half, and then the second party only for the second half. Does anyone know if this is the expected behavior? |
| 16:30.53 | ManxPower-work | Correct. channelvariables.tex. If you can't read .tex files -- well go complain to Digium |
| 16:31.16 | qasim | http://pastebin.com/0X2R8HtR |
| 16:31.27 | fifer | Anyone know if there is a diference between the standard Aastra 6755i and the 8x8 version? |
| 16:31.32 | qasim | this is what i get when i try to connect through eyebeam |
| 16:31.35 | tzafrir | isn't a HTML version of it available on-line somewhere? |
| 16:31.44 | tzafrir | Or at least a PDF version? |
| 16:31.51 | ManxPower-work | tzafrir: I'm sure there is, but I don't know where. |
| 16:32.14 | ManxPower-work | I've not seen an HTML version in the tarball |
| 16:32.20 | Erestar | ManxPower-work, Ok, I'm going. I've been using voip-info.org for most of my references though |
| 16:32.37 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:32.46 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:32.59 | ManxPower-work | Erestar: That should be your last resort. Much of the information is very outdated and some is even downright wrong. Where EXACTLY did you see that ${SYSTEMSTATUS}=SUCCESS is an expression? |
| 16:33.15 | elliot98 | is there a quick reference that lists the proper compile order for asterisk.tar.gz, asterisk-addons, dahdi, libpri, oh323, spands...? |
| 16:33.47 | ManxPower-work | elliot98: zaptel/dahdi, libpri, spandsp, asterisk, asterisk-addons |
| 16:33.59 | qasim | http://pastebin.com/0X2R8HtR |
| 16:34.07 | Erestar | ManxPower-work, I didn't. I extrapolated it based on an example. Now I see that I need to do $[expression] to make it actually evaluate |
| 16:34.08 | Erestar | Thanks |
| 16:34.10 | [TK]D-Fender | qasim: Please pastbin all the configs again in 1 pastebin. |
| 16:34.17 | qasim | ok |
| 16:34.27 | [TK]D-Fender | qwas and include the filenames in question before the contents |
| 16:35.20 | elliot98 | ManxPower-work: what about the H323 module? |
| 16:35.37 | ManxPower-work | elliot98: the module is either part of Asterisk or part of Asterisk addons |
| 16:36.05 | Erestar | ManxPower-work, [TK]D-Fender Thank you both |
| 16:36.08 | elliot98 | alrighty! thanks |
| 16:36.08 | ManxPower-work | elliot98: and there is no "the" h323 module. There are at least 4 H323 modules. |
| 16:37.02 | elliot98 | ManxPower-work: I know, never really understood why H323 never standardized |
| 16:37.46 | qasim | http://pastebin.com/rKWE0XQQ here it is |
| 16:37.49 | Katty | decides to setup ferretcam over lunch today |
| 16:37.59 | Katty | Ferret Viewing ETA 1.5hrs! |
| 16:38.01 | ManxPower-work | elliot98: Mostly licensing issues. |
| 16:38.09 | qasim | <[TK]D-Fender>: http://pastebin.com/rKWE0XQQ |
| 16:38.31 | Katty | also, i'm going ot sign up for some vocational classes on pre-vet medicine |
| 16:38.36 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 16:38.39 | Katty | they sound fun |
| 16:39.01 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 16:39.31 | Katty | hi superbeef |
| 16:40.14 | [TK]D-Fender | qasim: dbhost = 127.0.0.1 <- I told you to comment this out. why isn't it commented out? |
| 16:40.23 | superbeef | yo |
| 16:40.59 | qasim | i commented it and it didnt work then i again uncommented it |
| 16:41.07 | superbeef | Do guys use any nifty tools for parsing the asterisk full log? |
| 16:41.10 | superbeef | you guys |
| 16:41.12 | superbeef | (gals) |
| 16:41.12 | qasim | http://pastebin.com/EzqzXanF |
| 16:41.24 | elliot98 | I see |
| 16:41.27 | qasim | this is the things i get when i restart asterisk |
| 16:42.03 | Katty | superbeef: i use asterisk-stat |
| 16:42.11 | Katty | superbeef: it's a php based query thing |
| 16:42.18 | Katty | superbeef: nothin particularly fancy |
| 16:42.28 | qasim | i will comment it again if you like? |
| 16:42.31 | Katty | superbeef: but it allows the people around here to look at the call records easily |
| 16:42.52 | Katty | superbeef: but we obviously dump into a database...it doesn't parse a log file |
| 16:42.55 | qasim | should i try installing mysql on other pc? |
| 16:42.58 | qasim | will that work? |
| 16:43.18 | [TK]D-Fender | qasim: "module load res_mysql.so" |
| 16:43.21 | [TK]D-Fender | qasim: from * CLI |
| 16:43.34 | superbeef | Katty: hm... so there's nothing out there that can kind of follow all the debug, warning, etc, Ref #'s that float in the log |
| 16:43.57 | Katty | superbeef: well it's all php |
| 16:44.02 | Katty | superbeef: just rewrite the query |
| 16:44.22 | superbeef | Katty: but that just looks at CDR right? |
| 16:46.08 | donatas | who are using g729 ?? |
| 16:46.35 | qasim | <[TK]D-Fender> my module name is "res_config_mysql.so" not "res_config_mysql.so" |
| 16:46.44 | qasim | i will reload it and then tell it to you |
| 16:47.07 | qasim | [Feb 23 21:46:56] NOTICE[5887]: config.c:1968 ast_config_engine_register: Registered Config Engine mysql |
| 16:47.43 | [TK]D-Fender | qasim: Now try reloading the modules that failed earlier |
| 16:47.57 | [TK]D-Fender | qasim: Could be a pre-load order problem. |
| 16:48.38 | Katty | superbeef: it looks at whatever you put in the database |
| 16:49.58 | ManxPower-work | donatas: I use G729 |
| 16:50.01 | qasim | i unloaded and loaded it many times now |
| 16:50.06 | qasim | but no luck |
| 16:50.41 | ManxPower-work | qasim: You do NOT know enough to tell [TK]D-Fender that something is not going to work. Stop second guessing the one guy left on that channel that is willing to help you. |
| 16:50.44 | donatas | ManxPower-work: have you ever got such error opening benchg729 file ? Unable to check for valid G.729 licenses. |
| 16:50.59 | ManxPower-work | donatas: no. You must have screwed up the intall |
| 16:51.01 | ManxPower-work | install |
| 16:51.07 | donatas | I got email from digium, with KEY |
| 16:51.15 | donatas | do you mean register? |
| 16:51.28 | idespinner | donatas, did you do ./register ? |
| 16:51.28 | ManxPower-work | donatas: did you install the codec? Did you register the codec? |
| 16:51.44 | donatas | i registered the codec |
| 16:51.58 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:52.01 | donatas | and loaded, but i see this: 0/0 encoders/decoders of 0 licensed channels are currently in use |
| 16:52.01 | ManxPower-work | donatas: then there should be a g729 command in the CLI |
| 16:52.07 | donatas | yes, there is |
| 16:52.19 | ManxPower-work | donatas: then follow the instructions CAREFULLY and install it again. |
| 16:53.03 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 16:53.16 | donatas | Please enter your Key-ID: this should be a key from email ? |
| 16:53.18 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:53.30 | ManxPower-work | donatas: WHAT DO THE INSTRUCTIONS SAY? |
| 16:53.39 | *** join/#asterisk d-k-t (~D@112.202.232.46) |
| 16:53.52 | donatas | register to generate a valic license. |
| 16:53.55 | donatas | valid |
| 16:54.08 | ManxPower-work | DID YOU READ THIS: http://downloads.digium.com/pub/telephony/codec_g729/README |
| 16:54.19 | donatas | yes |
| 16:54.32 | ManxPower-work | donatas: contact Digium support. |
| 16:54.46 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:55.04 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:55.39 | qasim | http://pastebin.com/DdSRjJqM |
| 16:55.46 | qasim | this is what i got from mysql |
| 16:55.58 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:57.04 | [TK]D-Fender | qasim: I asked you to try reloading the OTHER modules manually following this new successful base load success |
| 16:57.25 | ManxPower-work | [TK]D-Fender: Just give up. He's never going to get it working |
| 16:58.02 | donatas | ManxPower-work: i got this, by registering: http://p.defau.lt/?43yQNYt3EVXWIwuXWGMdTg |
| 16:58.23 | ManxPower-work | He's going to know so little about his system that he's going to leave it wide open to hackers. |
| 16:58.29 | donatas | and i don't see directory named /var/lib/asterisk/license |
| 16:58.56 | ManxPower-work | donatas: The G729 is a commercial product from Digium. If you want support for that product, contact Digium. |
| 16:59.04 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 16:59.39 | qasim | manxpower-work this is how you learn no one is perfect in the first run any ways thank you all for your help and support i will try to figure this out on my own |
| 17:00.02 | ManxPower-work | qasim: you keep ignoring what [TK]D-Fender is telling you. You cannot do that and expect to be successful. |
| 17:00.04 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 17:00.32 | [TK]D-Fender | goes off to lunch |
| 17:00.49 | ManxPower-work | Telecommunications is complicated, confusing, has a seep learning curve and REQUIRES an attention to detail or you will have major security issues. |
| 17:00.58 | ManxPower-work | s/seep/steep |
| 17:01.26 | bmoraca_work | but security issues are soooo much fun! |
| 17:01.45 | bmoraca_work | i love it when people bounce international calls through my softswitch! |
| 17:01.48 | ManxPower-work | At my last job we usually got 1 -2 panicked calls per week from someone that just got a $10,000 phone bill because they did not secure their PBX -- mainly because they didn't even know how to |
| 17:02.09 | bmoraca_work | i love it when people bounce international calls through my softswitch! |
| 17:02.11 | bmoraca_work | er |
| 17:02.14 | bmoraca_work | stupid up arrow |
| 17:02.16 | ManxPower-work | Oh look! A GUI! Now I don't have to worry about technical stuff! |
| 17:02.36 | qasim | i know but how are you gonna go on security side unless you have a server running |
| 17:02.39 | qasim | ?? |
| 17:02.43 | tonywho | can someone help me with security of my box? |
| 17:02.52 | qasim | first you need a server to adress security issues |
| 17:02.56 | qasim | and that is how you learn |
| 17:02.59 | bmoraca_work | tonywho: are you having a specific problem? |
| 17:03.02 | ManxPower-work | qasim: How are you going to get the server running if you don't follow the simpliest instructions from [TK]D-Fender |
| 17:03.08 | qasim | i know |
| 17:03.09 | tonywho | i set up my server |
| 17:03.21 | tonywho | can you tell me security holes |
| 17:03.28 | tonywho | if i give you |
| 17:03.32 | tonywho | the address |
| 17:03.33 | qasim | i just started learning it few days back and i know i am not a smart guy like you and i was trying to follow them |
| 17:03.46 | qasim | that is y i came to this forum.. i.e to learn |
| 17:04.29 | bmoraca_work | tonywho: any place that you can get to your server is a potential hole. use your edge firewall or iptables to close them up so that only the networks that need access have access. example: port 80 for administration (don't leave it open to people who don't need administrative access) |
| 17:04.49 | ManxPower-work | qasim: We don't mind if you don't know something, what really pisses most of us off is the user ignoring what we tell them, doing things without telling us (addind back in options you were told to remove). |
| 17:05.21 | qasim | i did that and told him that it didnt work... |
| 17:05.49 | ManxPower-work | qasim: But he had a very good reason for leaving it out and you invalided everything after that. |
| 17:05.51 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 17:06.09 | ManxPower-work | you were trying to second guess the person that was helping you. |
| 17:06.30 | tonywho | but if i need customer to access it, i need port 80 open |
| 17:06.39 | qasim | i never did that... i was doing exactly as he told me to do... |
| 17:06.41 | tonywho | to acces UIcustomer |
| 17:06.44 | qasim | any ways |
| 17:06.47 | ManxPower-work | tonywho: Why would they need access to port 80. Asterisk doesn't even have a GUI. |
| 17:07.14 | bmoraca_work | tonywho: why would you have a customer accessing your own server on port 80? |
| 17:07.28 | tonywho | client to see balances |
| 17:07.52 | ManxPower-work | tonywho: Something you wrote yourself? |
| 17:08.12 | bmoraca_work | tonywho: that should not be hosted on your PBX. that should be on a separate server. |
| 17:08.14 | tonywho | what¨? |
| 17:08.40 | ManxPower-work | tonywho: Asterisk does not have a web server. Therefore if you are running a web service on your PBX you must have installed or wrote it yourself? |
| 17:08.50 | ManxPower-work | Are you sure you're not using a GUI for Asterisk? |
| 17:09.37 | tonywho | i habe elastix |
| 17:09.39 | bmoraca_work | probably running a2billing or something |
| 17:09.42 | tonywho | all in the same server |
| 17:09.48 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net) |
| 17:09.49 | ManxPower-work | ~elastix |
| 17:09.50 | infobot | methinks elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 17:10.03 | ManxPower-work | You're not even using regular Asterisk |
| 17:10.27 | ManxPower-work | bmoraca_work: someone using FreePBX asking how to secure their server. That's FUNNY. |
| 17:11.09 | bmoraca_work | it can be done...but i doubt a novice could do it. he could pay someone to do it, i suppose |
| 17:11.56 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:14.33 | *** join/#asterisk keyp (~keyp@66.184.128.98) |
| 17:17.24 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 17:21.13 | *** join/#asterisk Victor_Yure_ (~victor@unaffiliated/victoryure/x-837844) |
| 17:22.33 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:22.53 | ariel_ | hello everyone |
| 17:25.18 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 17:25.30 | *** part/#asterisk tonywho (~cibnetsac@200.121.247.79) |
| 17:28.20 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 17:29.27 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:33.10 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:34.40 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 17:37.08 | *** part/#asterisk Assid (~assid@unaffiliated/assid) |
| 17:39.46 | ruben23 | hi guys how do i setup voicemail for incoming calls, not being answered.. |
| 17:40.07 | [TK]D-Fender | ruben23: call Voicemail() in your dialplan |
| 17:40.18 | [TK]D-Fender | rubits a dialplan app like any other. |
| 17:40.23 | [TK]D-Fender | ruben23: its a dialplan app like any other. |
| 17:40.38 | *** join/#asterisk Geminizer (~me@cpe-76-180-27-4.buffalo.res.rr.com) |
| 17:41.53 | idespinner | ruben23, there is a macro that does it all for you mostly i think its [macro-std-extension] |
| 17:42.22 | Geminizer | Hello all. Would anyone know the cause of this type of message: Spawn extension (...) exited non-zero ? All I have is a Playback(file) followed by SayDigits(...), and the latter ends up returning non-zero |
| 17:43.19 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:43.23 | idespinner | do you have waitexten or anything else prior to that? |
| 17:43.47 | [TK]D-Fender | idespinner: Who says he has a macro like that? |
| 17:44.06 | [TK]D-Fender | Geminizer: Would help if you SHOWED us your failed call and dialplan. |
| 17:44.48 | idespinner | never said he had it, just that there is one in existence that can be found fairly easily.... |
| 17:45.45 | Geminizer | http://pastebin.com/FetXhQEc |
| 17:46.31 | [TK]D-Fender | Geminizer: And the failed call? |
| 17:47.20 | *** join/#asterisk lenne_dk (~leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 17:47.58 | lenne_dk | Hi. Are there countermeasures for brute-force attacks to log in to asterisk? |
| 17:48.25 | p3nguin_ | Sure. |
| 17:48.26 | [TK]D-Fender | lenne_dk: fail2ban <- |
| 17:48.33 | p3nguin_ | fail2ban can take care of it. |
| 17:49.16 | Geminizer | http://pastebin.com/AZnMgwHZ |
| 17:49.33 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:49.38 | lenne_dk | Okie. Some romanian tried first all extensions of 1 to 4 digits, then 11267 passwords on the extensions that worked. |
| 17:50.13 | lenne_dk | then I made a blackhole route for 89.165.131.103 |
| 17:52.03 | lenne_dk | One should think asterisk should just answer "username or password incorrect", so the intruder didn't know if an extension was right. |
| 17:52.33 | p3nguin_ | ruben23: http://pastebin.com/tRXvqxef |
| 17:52.46 | [TK]D-Fender | lenne_dk: there is a generl sip option to respond 401 on everything instead of 404, |
| 17:52.49 | p3nguin_ | lenne_dk: You don't login with extensions. |
| 17:53.36 | p3nguin_ | alwaysauthreject = yes |
| 17:53.44 | [TK]D-Fender | Geminizer: Looks like it dies after the 1st digit. Enable SIP DEBUG to see who quit, and why |
| 17:54.27 | ruben23 | p3nguin_:thanks. |
| 17:54.34 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:55.10 | p3nguin_ | ruben23: If you don't want to use followme, take out the two lines which contain "followme" in them. |
| 17:56.07 | lenne_dk | Somehow he first tested all extensions to see which existed, then he tried to register. |
| 17:56.18 | p3nguin_ | No, he tried usernames. |
| 17:56.45 | lenne_dk | Words :-) |
| 17:57.00 | p3nguin_ | Yeah, the wrong words. |
| 17:57.07 | lenne_dk | But I'll set alwaysauthreject = yes. Thanx |
| 17:57.38 | p3nguin_ | If the usernames just so happen to be the same as the extension numbers, that's fine, but they aren't trying to login with "extensions," since that's not how things work. |
| 17:58.04 | ManxPower-work | Most people eventually realize that setting the SIP userid to be the same as the extension is a mistake. |
| 17:58.15 | lenne_dk | In which conf shoud I put alwaysauthreject = yes? sip.conf? |
| 17:58.17 | p3nguin_ | Phones are not extensions, and you don't login with extension information. |
| 17:58.27 | ManxPower-work | lenne_dk: and allowguest=no |
| 18:00.11 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 18:00.17 | lenne_dk | Well, it didn't complain when I put it in sip.conf, so it must be right ;-) |
| 18:01.03 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 18:01.04 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 18:02.25 | Katty | ferretcam is up and running! |
| 18:02.39 | Naikrovek | you'll need a zookeeper license soon |
| 18:02.53 | Katty | http://www.ustream.tv/channel/the-nut-house-bird-bath <- Ferrets |
| 18:03.15 | TheDavidFactor | Does anyone use Fax For Asterisk? I'm trying to set up a test box with the one free channel you can get. I think I've got it working, but when I try to send a fax to the * box my fax software says "Timeout waiting for reinvite to fax" I'll pastebin config, anyone know what I need to do to fix this? |
| 18:03.18 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 18:05.07 | Geminizer | does "X-Asterisk-HangupCause: Normal Clearing" mean anything? |
| 18:05.09 | ManxPower-work | lenne_dk: Asterisk does not generally complain about invalid options in config files. This is how it was designed |
| 18:05.19 | ManxPower-work | Geminizer: That means the call ended normally |
| 18:05.44 | ManxPower-work | TheDavidFactor: turn off reinvites |
| 18:07.40 | TheDavidFactor | ManxPower-work, will do |
| 18:07.52 | TheDavidFactor | Here's the pastebin: http://pastebin.com/NzGRiCrZ |
| 18:13.11 | idespinner | Geminizer, to me, it looks like the remote party hung up. Thats just what I can gather from what you send. Although i'm sure that your certain this is not the case |
| 18:13.31 | leifmadsen | TheDavidFactor: which version? |
| 18:13.36 | leifmadsen | (of asterisk) |
| 18:14.14 | TheDavidFactor | asterisk 1.6.2.4 |
| 18:14.20 | TheDavidFactor | 64bit |
| 18:15.42 | leifmadsen | TheDavidFactor: huh, actually, try 1.6.2.3-rc2 (if this is a development box). It'll have more fixes beyond what 1.6.2.4 has, which includes from T.38 changes I think |
| 18:16.04 | leifmadsen | in fact, at this point you might even try 1.6.2 branch to see if it is a configuration issue or an existing (fixed) bug |
| 18:17.10 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.29 | TheDavidFactor | leifmadsen, thanks |
| 18:21.28 | coppice | how can you use FAX for Asterisk with a 64 bit build of Asterisk? |
| 18:23.32 | p3nguin_ | coppice: http://downloads.digium.com/pub/telephony/fax/res_fax/asterisk-1.4/x86-64/ |
| 18:24.05 | p3nguin_ | or http://downloads.digium.com/pub/telephony/fax/res_fax/asterisk-1.6.2.0/x86-64/ |
| 18:24.13 | coppice | interesting. the web page still says its not available |
| 18:24.15 | p3nguin_ | or whatever other branch you use. |
| 18:25.20 | coppice | I wonder if they'll ever get V.34 FAX out the door |
| 18:27.13 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 18:29.41 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:30.53 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 18:31.12 | benngard | regarding fax and asterisk, i claim that trunk version works very vell, i have like 20 faxes around in sweden that use a soulition like: fax - spa2102 - asterisk - avaya - pstn, all hardcoded to g711 alaw, but as soon as i discover a fax i switch to t38, did start to implemnet it for about a month ago and no complains so far |
| 18:32.35 | coppice | the SPA2102 and SPA3102 have a *very* quirky T.38 implementation, so just be happy it works for you |
| 18:32.52 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 18:35.06 | benngard | what is so quirky with them? the hard part was to get it to work over the ooh323 channel driver, but may (alexandr) fixed that with some very small help from me |
| 18:35.12 | *** join/#asterisk avajadi (~avajadi@94.126.224.225) |
| 18:35.24 | Katty | i need help deciding a name for the ferret cam |
| 18:35.33 | Katty | i was thinking maybe Fuzzybutt Ferret Flat |
| 18:35.58 | rubberneck | is there a way to turn off the color formatting of the CLI? |
| 18:36.07 | coppice | well, if you find a box that doesn't behave well try changing the FAX to ATA lead for one a few metres longer. its that quirky |
| 18:38.18 | p3nguin_ | rubberneck: I think you have to change your init script to not colorize the console at startup. |
| 18:38.41 | keyp | Is there a way for a user to send a message from the voicemail system to more than one recipient? Not a static voicemail group, but a dynamic list entered from the phone? |
| 18:38.51 | Skeeter- | anyone ever done videoconferencing?? |
| 18:39.12 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:39.15 | rubberneck | p3nguin_: Thx Ill take a look. |
| 18:39.23 | ManxPower-work | keyp: no |
| 18:39.30 | ManxPower-work | keyp: you should do that in your MTA |
| 18:39.53 | ManxPower-work | Skeeter-: millions of people |
| 18:40.01 | p3nguin_ | The email server takes care of Asterisk's voicemail, now? |
| 18:40.24 | ManxPower-work | p3nguin_: I was thinking he meant "voicemail notifications" |
| 18:40.53 | p3nguin_ | I think he wants to be able to leave voicemail for a group of people, but not have a pre-configured voicemail group. |
| 18:40.54 | keyp | ManxPower-work, nope. p3nguin_ is correct. |
| 18:41.02 | keyp | exactly |
| 18:41.04 | Skeeter- | anyone down here ever done videoconferencing?? |
| 18:42.42 | ManxPower-work | p3nguin_: also since he's talking about voicemail groups, he might be using a GUI |
| 18:44.17 | keyp | I am using FreePBX, yes. |
| 18:44.27 | ManxPower-work | ~freePBX |
| 18:44.27 | 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, or fpbx |
| 18:44.54 | ManxPower-work | One of the few pieces of Asterisk FreePBX doesn't totally screw up is the voicemail stuff. |
| 18:45.00 | p3nguin_ | But since he wants to NOT configure voicemail groups, the FreePBX aspect of it seems irrelevant. |
| 18:45.23 | keyp | FreePBX doesn't seem to have much of anything to do with the parts I want to change. |
| 18:45.39 | ManxPower-work | keyp: You should expect that with FreePBX |
| 18:47.39 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 18:47.55 | p3nguin_ | To reiterate, the question was how to send voicemail to a group (from within the voicemail system) without configuring voicemail groups. |
| 18:48.18 | ManxPower-work | And to reiterate, I said I didn't think it could be done |
| 18:49.01 | p3nguin_ | Maybe you meant to say it, but I don't see where you said it. |
| 18:49.49 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 18:50.45 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:51.03 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 18:52.19 | Katty | infobot: crittercam |
| 18:52.20 | infobot | from memory, crittercam is Katty's live broadcast of The Nut House @ http://ustre.am/8H5d and The Fuzzy Ferret Flat @ http://ustre.am/bEBU |
| 18:52.20 | Katty | infobot: crittercam? |
| 18:52.21 | infobot | somebody said crittercam was Katty's live broadcast of The Nut House @ http://ustre.am/8H5d and The Fuzzy Ferret Flat @ http://ustre.am/bEBU |
| 18:52.21 | Katty | infobot: forget crittercam |
| 18:52.21 | infobot | Katty: i forgot crittercam |
| 18:52.22 | Katty | infobot: are you alive? |
| 18:52.22 | infobot | The dead cannot live |
| 18:52.23 | ManxPower-work | ~alive |
| 18:52.23 | infobot | alive is, like, someone who still lives |
| 18:52.23 | Katty | okay who broke infobot |
| 18:52.24 | coppice | ferrets are just working class mink |
| 18:52.25 | Katty | oh |
| 18:52.26 | Katty | my |
| 18:52.43 | ManxPower-work | There's nothing wrong with playing with the bot, just do it in private and wash your hands after. |
| 18:53.12 | Katty | infobot: crittercam is Katty's live broadcast of The Nut House @ http://ustre.am/8H5d and The Fuzzy Ferret Flat @ http://ustre.am/bEBU |
| 18:53.13 | infobot | okay, Katty |
| 18:53.13 | ManxPower-work | coppice: I've never seen a ferret do any work ever |
| 18:53.30 | Katty | i must be awfully laggy |
| 18:53.50 | coppice | ManxPower-work: sounds a lot like many working class people |
| 18:54.13 | seanbright | oh dear |
| 18:54.21 | seanbright | that room must smell wonderful |
| 18:54.47 | Katty | seanbright: actually it does. |
| 18:54.52 | Katty | seanbright: and requires daily cleaning |
| 18:55.30 | Katty | seanbright: even their bedding is swapped around every 3 or 4 days |
| 18:56.01 | ManxPower-work | hugs his low maintenance cat. |
| 18:56.07 | Katty | ManxPower-work: :P |
| 18:56.13 | Katty | ManxPower-work: i love cats too. |
| 18:56.19 | ManxPower-work | I'd hug the electric litter box, but...ewwwww. |
| 18:56.20 | Katty | ManxPower-work: i've never met a cat that i didn't get along with. |
| 18:56.33 | coppice | ManxPower-work: actually, ferrets are quite industrious when then find something motivating. have you seen one make a kill? |
| 18:56.45 | ManxPower-work | Katty: you love pretty much anything warmblooded, don't you? |
| 18:56.51 | Katty | ManxPower-work: pretty mch |
| 18:56.59 | Katty | ManxPower-work: i'd spoil a mouse. |
| 18:57.04 | ManxPower-work | coppice: all I've ever seen them do is lay around like a sock full of sand. |
| 18:57.19 | Katty | ManxPower-work: and probably stop by schnucks once a week to purchase 1/4lb of some divine cheese |
| 18:57.37 | Katty | ManxPower-work: but not just warm blooded |
| 18:57.44 | Katty | ManxPower-work: i'm rather fond of snakes and reptiles too |
| 18:57.44 | p3nguin_ | Lorraine cheese! |
| 18:58.08 | Katty | ManxPower-work: spiders....not so much |
| 18:58.25 | Katty | bugs freak me out )= |
| 18:58.49 | ManxPower-work | I kill wasps and ants, the rest I try to not harm. |
| 18:59.03 | Katty | ants don't really count as bugs |
| 18:59.09 | Katty | they're not creepy |
| 18:59.13 | Katty | wasps....definately creepy |
| 18:59.30 | ManxPower-work | Has anyone used accudatatech.com for CNAM services. |
| 18:59.43 | coppice | I think ants are pretty creepy when they swarm |
| 18:59.48 | ManxPower-work | Katty: Wasps can sting over and over. Honey bees, at least, will die if they sting you. |
| 19:00.02 | Naikrovek | i found an MF'ing cockroach last night. |
| 19:00.04 | Katty | bumbles bees don't |
| 19:00.07 | Naikrovek | instant girl mode for me |
| 19:00.12 | ManxPower-work | Katty: I don't kill those. |
| 19:00.21 | Naikrovek | screamin', kickin over things in an attempt to kill the monster, etc |
| 19:00.24 | Katty | i was stung in the back by a bumble bee when i was a kid |
| 19:00.28 | Naikrovek | doesn't do well around cockroaches |
| 19:00.30 | p3nguin_ | I hate cockroaches. |
| 19:00.30 | coppice | ants only sting you once, but they always have plenty of friends to follow on :-) |
| 19:00.32 | Katty | it was nesting is a bird house |
| 19:00.43 | Katty | coppice: ants? sting? |
| 19:00.48 | Katty | coppice: i've never been stung by an ant |
| 19:00.50 | Naikrovek | bite |
| 19:00.52 | ManxPower-work | Katty: fireants bite |
| 19:00.52 | Katty | or bitten |
| 19:00.58 | Katty | fireants? |
| 19:00.58 | p3nguin_ | They're the only existence I know of that can survive being microwaved for several minutes. |
| 19:01.06 | Naikrovek | fireants bite AND inject venom |
| 19:01.08 | Naikrovek | snakes |
| 19:01.10 | Katty | creepy |
| 19:01.15 | Katty | i don't think i've seen any fireants around here |
| 19:01.18 | ManxPower-work | Katty: yes, all over the south USA. They can kill a child |
| 19:01.24 | Katty | dang. |
| 19:01.27 | coppice | Katty: many ants can inflict pain worse than a bee |
| 19:01.33 | Naikrovek | if the child doesn't know to run i would think |
| 19:01.39 | ManxPower-work | I get bit by fire ants all the time |
| 19:01.47 | Katty | p3nguin_: have you seen any fireants in missouri? |
| 19:01.48 | [TK]D-Fender | http://en.wikipedia.org/wiki/Bullet_Ants |
| 19:01.50 | [TK]D-Fender | WORse |
| 19:01.56 | p3nguin_ | katty: nope |
| 19:01.58 | Katty | phew |
| 19:01.59 | coppice | if you piss off an ant's nest they can do some serious harm |
| 19:02.07 | Katty | coppice: oh i'd never wreck an ant nest |
| 19:02.09 | p3nguin_ | katty: They could exist, though. |
| 19:02.11 | Katty | coppice: that's just impolite |
| 19:02.28 | ManxPower-work | I'll do anything I can to kill a fire ant mound. |
| 19:02.29 | Katty | coppice: mostly i don't get grumpy with them on the counter either |
| 19:02.37 | coppice | you might just lay in the wrong spot one day |
| 19:02.40 | Katty | coppice: usually i just clean it up and they'll go away |
| 19:02.48 | p3nguin_ | We do have cowkillers around here, though. Though they aren't really ants, they sure look like ants and many people call them ants. |
| 19:02.50 | Katty | coppice: yeah i might, you never no |
| 19:02.55 | Katty | s/no/know/ |
| 19:03.08 | Katty | googles |
| 19:03.25 | ManxPower-work | I usually spread fire any killer stuff around my cabin a couple of times a year. |
| 19:03.36 | Katty | p3nguin_: so THAT"S what those are |
| 19:03.43 | Katty | p3nguin_: yes i've seen man of those around here |
| 19:04.00 | p3nguin_ | katty: You do NOT want to get stung by one. You'll hurt for weeks. |
| 19:04.02 | Katty | p3nguin_: i usually run screaming the other way :P |
| 19:04.17 | Katty | and those garden spiders |
| 19:04.21 | Katty | the big black and yellow ones |
| 19:04.26 | Katty | round the side of the house |
| 19:04.39 | Katty | alksjdfowiahlgkjsdlfkjasdlf |
| 19:04.41 | Katty | CREEPY |
| 19:05.10 | p3nguin_ | http://lancaster.unl.edu/pest/resources/cowkiller.shtml |
| 19:05.22 | Katty | p3nguin_: i had a dream one night that i'd gone to flordia for vacation, and took pippin |
| 19:05.36 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:05.40 | Katty | p3nguin_: and while we were on the beach a group of creepy crawly something or others got ahold of him |
| 19:05.43 | Katty | p3nguin_: and killed him :< |
| 19:06.41 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 19:06.53 | p3nguin_ | http://www.semiantics.com/images/100_0702.jpg |
| 19:07.08 | themolester | i had a dream one night that I was in florida on vacation, then I realized I lived here |
| 19:07.14 | Katty | themolester: i hate you |
| 19:07.16 | Katty | themolester: :P |
| 19:07.19 | themolester | :) |
| 19:07.39 | themolester | about 5 minutes from the beach too :) |
| 19:07.45 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:08.15 | themolester | did i make everybody jelous and not want to help? |
| 19:09.06 | themolester | cause, if not... i've got this weird 20 second hangup problem with two way audio on incoming calls only |
| 19:09.15 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 19:09.20 | Nugget | jealous? of florida? sheesh. |
| 19:09.47 | Nugget | sure, it's warm, but everything is covered in a 6" deep blanket of bugs and lizards and you're surrounded by old people and tourists. |
| 19:10.03 | themolester | depends on the city |
| 19:10.53 | themolester | though, they are a bit higher concentration accross the board... the old people that is |
| 19:10.56 | themolester | :) |
| 19:11.21 | Nugget | a cockroach once ate one of my flip flops in panama beach. |
| 19:11.25 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 19:11.39 | bmoraca_work | omg, cisco's website is abominable |
| 19:11.52 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 19:12.12 | ManxPower-work | themolester: that sounds like a reinvite issue |
| 19:12.13 | Nugget | yes it is |
| 19:12.17 | themolester | Nugget lol |
| 19:12.24 | Katty | telnet |
| 19:12.27 | Katty | :< |
| 19:12.29 | Katty | not first :< |
| 19:13.18 | themolester | ManxPower-work want to take a look? http://pastebin.ca/1807165 |
| 19:13.23 | Nugget | huggles Katty anyway |
| 19:13.52 | Katty | http://www.youtube.com/watch?v=f0Y-SvS9kwo |
| 19:13.54 | Katty | hugs Nugget |
| 19:14.12 | Nugget | Goats are cool |
| 19:14.22 | Katty | very cool |
| 19:14.28 | themolester | i have tried canreinvite=no and also created a codec mismatch to keep things from reinvite |
| 19:14.39 | Katty | i intend to get a pair if i ever move outside the city limits and have room |
| 19:16.27 | p3nguin_ | themolester: Did you check to see if the channel was in fact state: Up before reaching the time when it hangs up? |
| 19:16.36 | Katty | i'm sure riddick would love to herd a couple goats |
| 19:17.15 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.10 | themolester | p3nguin_ sip show channels? |
| 19:18.33 | ManxPower-work | ~freePBX |
| 19:18.34 | 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, or fpbx |
| 19:18.45 | p3nguin_ | themolester: core show channels verbose |
| 19:18.50 | ManxPower-work | themolester: No, I'm not interesting looking at FreePBX |
| 19:19.04 | p3nguin_ | actually, it will show it without verbose, too. |
| 19:20.09 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:21.44 | p3nguin_ | themolester: If you know when the call begins, wait 12-15 seconds and run core show channels verbose and check to see how things look. It could help get you to a solution. |
| 19:24.14 | themolester | p3nguin_ http://pastebin.ca/1807182 that is core show channels verbose |
| 19:24.35 | p3nguin_ | You'll have to not use pastebin.ca if you want me to be able to view it. |
| 19:24.51 | themolester | oh, what site do you prefer? |
| 19:24.55 | p3nguin_ | still cannot figure out the obsession with pastebin.ca over pastebin.com |
| 19:25.06 | themolester | the .com site was slow for me on more than one occasion |
| 19:25.17 | themolester | so, you prefer .com? |
| 19:25.33 | p3nguin_ | I cannot figure out why pastebin.ca doesn't display for me unless I use the IP address. |
| 19:25.44 | p3nguin_ | and I hate having to go lookup the IP address for pastes. |
| 19:25.44 | Katty | p3nguin_: it's one less letter to type? |
| 19:25.46 | Katty | p3nguin_: idk |
| 19:25.52 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 19:26.11 | Katty | p3nguin_: maybe it's just Cooler(tm) |
| 19:26.27 | p3nguin_ | You would think it's a DNS issue, but since I can lookup the IP address we know it's not a DNS issue. |
| 19:27.10 | themolester | http://pastebin.com/dJWpYeM3 |
| 19:27.19 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:27.32 | themolester | very odd, because thats exactly what i thought |
| 19:27.34 | themolester | lol |
| 19:27.54 | p3nguin_ | It's like non of my browsers can resolve it. |
| 19:27.57 | p3nguin_ | none |
| 19:28.05 | ManxPower-work | it always works for me |
| 19:28.21 | ManxPower-work | Do you have a proxy? Maybe CSS or script blocker? |
| 19:28.21 | themolester | heh... that was my second thought ... i bet you get these suggestions every time you mention it, right? |
| 19:28.27 | p3nguin_ | dig, host, and nslookup can resolve it just fine, though. It's not a huge inconvenience. |
| 19:29.18 | Katty | sure it is |
| 19:29.18 | themolester | host? is it a linux desktop? |
| 19:29.31 | p3nguin_ | Yes, Linux desktop. |
| 19:29.55 | p3nguin_ | I wonder if it actually is a DNS problem and I haven't dug deep enough to uncover it. |
| 19:30.22 | ManxPower-work | ping by name. dig, host, nslookup many times will resolve even if you have a problem |
| 19:30.54 | p3nguin_ | 64 bytes from pastebin.ca (208.68.18.97): icmp_seq=1 ttl=46 time=81.2 ms |
| 19:31.03 | ManxPower-work | doubt it's a DNS issue then |
| 19:31.04 | themolester | web browsers were pretty pitiful on linux when i tried linuxdesktop last time... opera being the only thing really usable at the time... firefox is much better than the old mozilla though |
| 19:31.27 | p3nguin_ | I haven't ran Windows on the desktop for nearly 10 years. |
| 19:31.47 | ManxPower-work | I recently switched back to a linux desktop |
| 19:32.19 | [TK]D-Fender | I switched about a year and half ago. |
| 19:33.01 | themolester | stop it already, your going to make me go and set up a linux desktop rig again |
| 19:33.03 | themolester | :) |
| 19:33.06 | rubberneck | I feel so gimped when I have to use windows for something. |
| 19:33.21 | ManxPower-work | I'm forced to use Windows for work, but that's all nicely locked in a VM |
| 19:33.25 | [TK]D-Fender | rubberneck: that's OK... they have GIMP for Windows as well ;) |
| 19:33.58 | rubberneck | [TK]D-Fender: yeah the gimp is cool. Takes some getting used to though. |
| 19:34.17 | avajadi | Hi, channel! |
| 19:34.18 | *** join/#asterisk smooth_penguin (~smoove@59.95.3.220) |
| 19:34.24 | Katty | don't you hate it when one of your ears just starts ringing for no particular reason |
| 19:34.29 | Katty | ^_- |
| 19:34.40 | Katty | hello mister operator |
| 19:34.46 | rubberneck | Katty: what? |
| 19:34.51 | avajadi | Anyone familiar with fax for asterisk on 1.6 |
| 19:35.05 | Katty | rubberneck: you've never had that before? |
| 19:35.13 | rubberneck | Katty: WHAT? |
| 19:35.17 | smooth_penguin | hey Katty, they say it happens when one of the cells is dying |
| 19:35.19 | Katty | rubberneck: one of your ears gets a high pitched tone for a few seconds |
| 19:35.19 | smooth_penguin | :P |
| 19:35.30 | rubberneck | Katty: what? |
| 19:35.35 | Katty | i'll take that as a no |
| 19:35.40 | Katty | smooth_penguin: ah |
| 19:35.44 | Katty | s/ah/ha/ |
| 19:36.00 | rubberneck | Katty: Youll have to speak up, cant hear you. got a ringing in my ears. |
| 19:36.03 | themolester | p3nguin_ so, does that pastebin.com link make any sense to you? both channels say they are Up |
| 19:36.06 | p3nguin_ | katty: That happened to me the other day. I had my hand cupped near my ear because I was trying to hear something quiet from another room... it started ringing. I thought it was a pretty stupid coincidence. |
| 19:36.07 | Katty | rubberneck: ;P |
| 19:36.26 | Katty | p3nguin_: yeah they go away pretty quick, at least for me |
| 19:36.28 | [TK]D-Fender | Katty: tinnitus <- |
| 19:36.33 | Katty | no, i have tinnitus |
| 19:36.38 | Katty | well i mean it is |
| 19:36.40 | Katty | technically |
| 19:36.42 | Katty | a ringing in the ears |
| 19:36.46 | Katty | but it's not a constant ringing like i have |
| 19:36.51 | smooth_penguin | wuts tinnitus |
| 19:36.53 | Katty | it's...higher...ringing |
| 19:36.57 | Katty | very briefly |
| 19:37.10 | smooth_penguin | eww |
| 19:37.14 | smooth_penguin | wax build up |
| 19:37.21 | Katty | i don't think that has anything to do with it |
| 19:37.25 | rubberneck | i remember that annoying CRT sound |
| 19:37.27 | [TK]D-Fender | Katty: Phantom Cell-phone Syndrome? :p |
| 19:37.32 | p3nguin_ | themolester: No, my original suspicion was that one channel was remaining in state Ringing, which will timeout shortly. Since they are both Up, I think more debuggin will be required. |
| 19:37.47 | Katty | smooth_penguin: http://www.youtube.com/watch?v=OE5fIoveLoM |
| 19:38.04 | p3nguin_ | Yay, the pastebin.ca link JUST loaded! |
| 19:38.14 | p3nguin_ | Usually it results in a timeout. |
| 19:38.24 | Katty | smooth_penguin: mine wasn't caused by extreme audio |
| 19:38.33 | Katty | smooth_penguin: mine was caused by a drug |
| 19:38.44 | smooth_penguin | :< |
| 19:38.45 | Katty | smooth_penguin: but it's healing, slowly, and the tinnitus is going away (= |
| 19:39.00 | Katty | smooth_penguin: i can only hear it in a very quiet room now |
| 19:39.21 | p3nguin_ | I guess if I wait 10 minutes, I can view people's pastes. |
| 19:39.23 | smooth_penguin | kk, well in some movie they say its because of dying ear cells |
| 19:39.28 | smooth_penguin | ear drum* |
| 19:39.39 | smooth_penguin | or wait that was in 'Children of men' |
| 19:39.50 | smooth_penguin | I think that ringing was because of bomb explosions |
| 19:39.58 | ManxPower-work | p3nguin_: that sounds a lot like an MTU issue. |
| 19:40.10 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 19:40.26 | p3nguin_ | You'd think it would happen on more than just one site I try to use... and also when viewing the site by IP address. |
| 19:40.31 | Katty | smooth_penguin: tinnitus can be a result of hearing loss |
| 19:40.40 | smooth_penguin | hrm |
| 19:40.45 | Katty | smooth_penguin: or perminent ear damage |
| 19:40.56 | *** join/#asterisk DGMurdockIII (~dgmurdock@208-70-41-206.bb.hrtc.net) |
| 19:40.57 | Katty | smooth_penguin: but mine is just a drug side effect |
| 19:41.01 | adnc | if i do a reload in the CLI, are the calls interrupted in this moment? |
| 19:41.11 | Katty | adnc: no |
| 19:41.16 | adnc | cool |
| 19:41.17 | p3nguin_ | You shouldn't do a "reload" on the CLI, anyway. |
| 19:41.24 | adnc | Katty, thanks |
| 19:41.26 | DGMurdockIII | anyone now anyting about having vonage connect to a linux firewall |
| 19:41.29 | adnc | p3nguin_, ohhh |
| 19:41.31 | p3nguin_ | Reload what you need to reload, or restart. |
| 19:41.41 | Katty | ^- reload dialplan |
| 19:41.42 | Katty | etc. |
| 19:41.44 | p3nguin_ | sip reload, for example. |
| 19:41.52 | Katty | dialplan reload maybe |
| 19:41.52 | adnc | p3nguin_, i did changes on the extensions.conf |
| 19:41.53 | Katty | i forget |
| 19:41.53 | p3nguin_ | dialplan reload, for another example. |
| 19:41.55 | ManxPower-work | DGMurdockIII: your question makes no sense |
| 19:42.02 | adnc | ohh, than dialplan reload |
| 19:42.07 | p3nguin_ | extension.conf changes, dialplan reload. |
| 19:42.13 | DGMurdockIII | and when i start talking on a call it gose ok then after a while the call cualty gose down hill |
| 19:42.14 | smooth_penguin | DGMurdockIII, why would vonage want to connect to the linux fw |
| 19:42.18 | Katty | adnc: a specific reload is a better habit to get into |
| 19:42.22 | smooth_penguin | you mean through it |
| 19:42.26 | DGMurdockIII | no that what i do |
| 19:42.26 | Katty | adnc: but if you just reload, the world will not come to an end |
| 19:42.31 | ManxPower-work | DGMurdockIII: Which version of Asterisk are you using? |
| 19:42.33 | p3nguin_ | or will it? |
| 19:42.39 | Katty | bonks p3nguin_ |
| 19:42.43 | adnc | Katty, nice to know, i'll use that way |
| 19:42.50 | DGMurdockIII | not using Asterisk |
| 19:42.58 | Katty | then why are you in here, DGMurdockIII? |
| 19:43.00 | adnc | Katty, thanks |
| 19:43.02 | smooth_penguin | lol |
| 19:43.03 | Katty | for the conversation? |
| 19:43.05 | DGMurdockIII | just thougt you guys might now somthing about my problem |
| 19:43.11 | Katty | ahhh, i see |
| 19:43.14 | Katty | well i do that all the time |
| 19:43.17 | Katty | i ask for recipes. |
| 19:43.24 | DGMurdockIII | becse you guys now about voip |
| 19:43.24 | Katty | and all sorts of non asterisk related stuff |
| 19:43.37 | Katty | bmoraca_work: speaking of recipes |
| 19:43.48 | Katty | bmoraca_work: i sure hope you get me that peach shrimp recipe before this weekend or i'm going to call your wife :P |
| 19:43.56 | bmoraca_work | lol |
| 19:44.01 | smooth_penguin | DGMurdockIII, well if you suspect the firewall drop it |
| 19:44.05 | smooth_penguin | and check |
| 19:44.15 | smooth_penguin | probably some for of rate limiting |
| 19:44.17 | DGMurdockIII | butcan use use astrix to get free phone calls |
| 19:44.31 | smooth_penguin | which allows burst and then starts limiting |
| 19:44.32 | bmoraca_work | i'll try to remember it |
| 19:44.49 | DGMurdockIII | ok |
| 19:44.57 | DGMurdockIII | i'll tell my friend |
| 19:45.05 | DGMurdockIII | to check |
| 19:45.08 | DGMurdockIII | but |
| 19:46.11 | DGMurdockIII | is there a good way to set up a voip phone system to use like ooma |
| 19:46.29 | DGMurdockIII | but with out having to pay for the system |
| 19:46.37 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 19:46.47 | DGMurdockIII | and is it possable to transfer you phone number over |
| 19:46.54 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:46.57 | ManxPower-work | This is just too painful |
| 19:47.06 | [TK]D-Fender | DGMurdockIII: * is a TOOL. It does not make service free <- |
| 19:47.24 | *** join/#asterisk SchleimKeim (~skull17@88.84.7.169) |
| 19:47.30 | p3nguin_ | Some companies will provide services for free, but that has nothing to do with Asterisk. |
| 19:47.59 | bmoraca_work | DGMurdockIII: ooma is not going to allow you to connect to their service without one of their devices. they're not going to provide you with the information you'll need. |
| 19:48.13 | *** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 19:48.49 | DGMurdockIII | you kinda get what i want to do |
| 19:49.05 | SchleimKeim | hello everybody :) |
| 19:49.10 | Katty | SchleimKeim: herroes |
| 19:49.15 | Katty | SchleimKeim: did you bring us cookies |
| 19:49.16 | DGMurdockIII | i want to do somthing like ooma but with out having to use there serves or hardward |
| 19:49.44 | bmoraca_work | DGMurdockIII: not gonna happen. no such thing as a free lunch. |
| 19:50.07 | Katty | DGMurdockIII: well i'm not sure what ooma is, but i have setup an asterisk server with two phones at my house. i still had to pay for all the hardware, and I still pay for the incoming/outgoing phone calls. |
| 19:50.29 | Katty | DGMurdockIII: it's highly unlikely you're going to find anything "free" |
| 19:50.33 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 19:50.51 | Katty | DGMurdockIII: it is a highly competative market, so you can probably find a few products at very good price. |
| 19:50.52 | DGMurdockIII | can you get callerid for free |
| 19:51.09 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 19:51.12 | [TK]D-Fender | DGMurdockIII: CallerID on **WHAT**? |
| 19:51.15 | Katty | DGMurdockIII: Skype seems to be reasonably competative and you can buy skype specific handsets. |
| 19:51.25 | DGMurdockIII | i hate skype |
| 19:51.32 | [TK]D-Fender | My Polycom phones send CallerID for free... now that I bought the phones :p |
| 19:51.37 | Katty | lol |
| 19:52.02 | DGMurdockIII | are there any prebuilt asterisk devices |
| 19:52.27 | themolester | DGMurdockIII one option is magicjack, I was reading just yesterday that after you purchase the device, many people were pulling sip usernames/passwords out of software dumps |
| 19:52.33 | [TK]D-Fender | DGMurdockIII: www.digium.com |
| 19:52.44 | Katty | DGMurdockIII: i believe digium sells an asterisk appliance. |
| 19:52.45 | themolester | that wouldn't be "free" service, but really really close |
| 19:52.55 | [TK]D-Fender | themolester: And just as quick they keep booting everybody off. |
| 19:52.56 | smooth_penguin | switchvox |
| 19:53.03 | themolester | however, i wouldn't expect it to be around next year |
| 19:53.05 | themolester | :) |
| 19:53.43 | Katty | pats her little asterisk server. |
| 19:53.55 | themolester | [TK]D-Fender good thing to know... i kept the page up and was going to read up more on it tomorrow after i solve my more pressing issues |
| 19:53.59 | themolester | :) |
| 19:54.06 | SchleimKeim | curses his little asterisk server cause he doesnt understand it *g* |
| 19:54.35 | bmoraca_work | i wonder what's going to happen to all their customers' phone numbers when ooma and magicjack go out of business |
| 19:54.50 | Katty | i bet they'll stop working ;) |
| 19:54.54 | bmoraca_work | though i suppose someone else will just come along and buy up the customer base so it'll be a non-issue |
| 19:54.55 | Katty | or someone will buy it out |
| 19:54.57 | Katty | like AT&T |
| 19:55.12 | bmoraca_work | i wouldn't anticipate AT&T buying it out |
| 19:55.12 | p3nguin_ | They've been around for enough years already that I don't expect they'll just go out. |
| 19:55.30 | Katty | p3nguin_: probably right |
| 19:56.01 | p3nguin_ | Though I have no idea how they have money to operate, since they have such low rates. |
| 19:56.19 | Katty | they sell crack |
| 19:56.41 | Katty | that's my theory anyway :P |
| 19:57.01 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 19:58.29 | leifmadsen | I have no theories |
| 19:58.33 | leifmadsen | theoretically |
| 19:58.42 | *** join/#asterisk fatgoose (~obelix@modemcable069.121-57-74.mc.videotron.ca) |
| 19:58.44 | fatgoose | hi |
| 19:58.46 | Katty | that's because you're a 9 |
| 19:58.55 | leifmadsen | I'm totally the 6 |
| 19:58.57 | Katty | brownie points if you get the reference |
| 19:59.04 | Katty | NO BROWNIE POINTS FOR YOU |
| 19:59.11 | themolester | ashton kutcher probably went to at&t and told them to give ooma a rate of like a quarter of a tenth of a cent per minute or he would let loose his million twittards on them |
| 19:59.21 | leifmadsen | Katty: Daniel Tammet? |
| 19:59.52 | Katty | leifmadsen: ^_- |
| 19:59.53 | Katty | leifmadsen: no |
| 19:59.57 | leifmadsen | you're wrong |
| 20:00.01 | leifmadsen | I'm right |
| 20:00.04 | leifmadsen | that's just how it works |
| 20:00.14 | leifmadsen | hides quickly |
| 20:00.38 | Katty | leifmadsen: http://www.imdb.com/title/tt0810988/ <- Rent, Download, Watch. |
| 20:00.40 | Katty | leifmadsen: Immediately. |
| 20:00.54 | leifmadsen | sorry, I'm working :) |
| 20:01.07 | Katty | leifmadsen: k,after work then |
| 20:01.18 | Katty | OH HEY |
| 20:01.19 | Katty | new release day |
| 20:01.42 | seanbright | ohhh |
| 20:01.52 | seanbright | johnny depp and helena bonham carter in a tim burton movie!? |
| 20:01.58 | seanbright | what will they think of next?? |
| 20:01.58 | bmoraca_work | got me a HTC Touch Pro 2 last night...pretty decent damn phone! |
| 20:02.07 | Katty | seanbright: alice in wonderland? |
| 20:02.10 | seanbright | aye |
| 20:02.12 | fatgoose | I've a problem with call transfer, when I got call from the external (sip trunk) to an SPA942 phone, then from that phone I transfer to another SPA942 phone the caller (the one from the sip trunk) does not hear the callee (monster cutting voice) |
| 20:02.13 | Katty | :> |
| 20:02.29 | fatgoose | But, when I transfer from a grandstream, everything is ok |
| 20:02.38 | *** join/#asterisk Alagar (~Administr@122.164.38.173) |
| 20:02.42 | vader-- | any of you guys running asterisk in vmware? |
| 20:02.47 | seanbright | or sweeney todd |
| 20:02.53 | Katty | vader--: i think some are running it on xen |
| 20:02.59 | Katty | vader--: or whoever bought out xen |
| 20:03.06 | seanbright | or corpse bride |
| 20:03.09 | Katty | vader--: but i am not running it on vmware. |
| 20:03.22 | [TK]D-Fender | bmoraca_work: WinMo.... meh |
| 20:03.24 | Katty | seanbright: i don't know how helena bonham is. |
| 20:03.26 | bmoraca_work | vader--: i'm running on VMware |
| 20:03.34 | *** part/#asterisk lindi- (~lindi@130.233.157.226) |
| 20:03.38 | seanbright | Katty: the woman from fight club |
| 20:03.40 | Katty | seanbright: but alice in wonderland is the only tim burton film with depp in it i could think of that's coming out soon |
| 20:03.41 | vader-- | im trying to find someone who is/has run asterisk in vmware with around 20 conncurrent calls |
| 20:03.43 | seanbright | tim burton's wife |
| 20:03.46 | bmoraca_work | [TK]D-Fender: it runs touchflow 3d, which is very much not like winmo |
| 20:03.46 | KavanS | helena bonham carter |
| 20:03.48 | KavanS | ^^ google her |
| 20:03.49 | Katty | seanbright: oh |
| 20:03.53 | Katty | seanbright: she's in all his movies |
| 20:03.56 | Katty | seanbright: isn't she? |
| 20:04.10 | [TK]D-Fender | bmoraca_work: I do like my HTC Touch w/ unlim data + GPS via Google maps... |
| 20:04.15 | seanbright | the recent ones, yes. |
| 20:04.37 | [TK]D-Fender | bmoraca_work: But I'd rather have one of the newer Android models with a much bigger hi-res screen... |
| 20:04.38 | Katty | http://www.themovieinsider.com/dvd-releases/february/2010/#day23 |
| 20:04.50 | Katty | cirque du freak wasnt' very good |
| 20:04.50 | bmoraca_work | [TK]D-Fender: yeah, i used to have a Touch (Sprint Mogul)...then i got to upgrade to a Touch Pro...then last night i got the TP2 |
| 20:04.53 | Katty | just FYI |
| 20:05.19 | bmoraca_work | [TK]D-Fender: unfortunately, i don't really have a choice. i'd have liked a Nexus One, but i got this guy for free. |
| 20:06.03 | themolester | bmoraca_work i want to get one of those TP2s I still have an 8600, but you should try out spb mobile shell |
| 20:06.06 | bmoraca_work | vader--: i don't have 20 simultaneous calls, but I have had up to 9 g729 calls on one of my boxes. |
| 20:06.17 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:06.30 | Katty | i don't think we've had 20 simultaneous calls ever either |
| 20:06.33 | bmoraca_work | themolester: i used to have a PPC-6600, lol |
| 20:06.35 | [TK]D-Fender | bmoraca_work: Up here we just got the Milestone (HSPA Droid), and there are more like it coming... |
| 20:07.09 | bmoraca_work | nice |
| 20:07.14 | [TK]D-Fender | bmoraca_work: My contract comes due this November at which point I might make a shift |
| 20:07.25 | themolester | i would have had a touch pro, and a touch pro 2, but verizon bought alltel, and I don't want to switch from my grandfathered plan |
| 20:07.41 | Katty | ryan carries the touch pro 2 |
| 20:07.47 | Katty | he put some custom bin on it too |
| 20:07.50 | [TK]D-Fender | bmoraca_work: I'd need to get unlim data however and enough of the perks to jsutify it. Itsw hard to beat my Touch's 10$ unlim data.... |
| 20:07.52 | Katty | it's pretty schnazzy |
| 20:08.11 | Katty | work provides my blackberry, so i can't complain (= it's free |
| 20:08.21 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 20:08.25 | [TK]D-Fender | bmoraca_work: And I discovered the gps about 2 months ago.. that's 2 years of use i didn't know i could have had out of it :) |
| 20:08.37 | themolester | the custom bins are great, and half the reason I wouldn't want to switch to android |
| 20:08.42 | bmoraca_work | [TK]D-Fender: definitely. my plan's ~$90/mo with the phone as modem...but work pays, so i can't complain |
| 20:09.05 | themolester | android took too much apple business model, and not enough let everybody do whatever they want business model |
| 20:10.19 | themolester | the stores are nice,but not worth the concessions of loading whatever you want, or risk jailbreaking and potentially bricking your phone (either during break, or because of break+future upgrade) |
| 20:10.23 | bmoraca_work | vader--: were you ever able to get your hands on a TA924? |
| 20:10.59 | [TK]D-Fender | bmoraca_work: Nexus one is pretty nice... though the Snapdragon chips are about to hit 1.5ghz and the power is getting kinda sick. We seem to be in another generational bump phase so I'm nervous about switching behind the curve |
| 20:11.04 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 20:11.33 | rubberneck | Where can I set the DSCP code point that is used for SIP and RTP packets originating from my * box? |
| 20:11.56 | bmoraca_work | well, as long as i'm working where i am, i'm stuck with sprint and sprint doesn't ahve the Nexus One yet, so QQ for me. |
| 20:12.07 | bmoraca_work | i also need a phone that supports ActiveSync |
| 20:13.07 | bmoraca_work | it does limit me a little, but i can live with a winmo phone to be able to get email, contacts, and calendar synced anywhere and everywhere |
| 20:13.40 | Naikrovek | i had a call with microsoft yesterday and they told me that their office communications server can be a full pbx, and they're going to start marketing it as such this fall |
| 20:13.46 | Naikrovek | maybe then you'll get your activesync phone |
| 20:14.10 | bmoraca_work | i was speaking about a cell phone |
| 20:14.14 | Naikrovek | oh |
| 20:14.16 | Naikrovek | nevermind then |
| 20:14.18 | idespinner | rubberneck, a quick google shows you use IPtables... http://www.freepbx.org/forum/setting-dscp-tos-bits-on-your-voice-packets |
| 20:14.34 | bmoraca_work | although, i have had requests from users to get their outlook contacts in their phone's directories |
| 20:15.01 | rubberneck | idespinner: hmmm, a quick grep of configs shows it can be done in sip.conf, but thanks anyhow. |
| 20:15.04 | bmoraca_work | i've been curious to try out Microsoft's ResponsePoint, but I can't justify the cost of the system just to try it out |
| 20:17.44 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:19.12 | idespinner | bmoraca_work, are you referring to outlook contacts in a cellphone or sip phone? |
| 20:19.26 | bmoraca_work | in a SIP phone |
| 20:19.53 | *** join/#asterisk smooth_penguin (~smoove@59.95.3.220) |
| 20:20.08 | idespinner | have you seen polycoms productivity suite(if they are polycoms) its close... |
| 20:20.41 | bmoraca_work | i believe that can only do LDAP, not Microsoft Exchange directories |
| 20:20.51 | idespinner | yes, your correct |
| 20:21.02 | idespinner | its almost there though... |
| 20:21.04 | vader-- | bmoraca adtran is supposely sending me a test unit |
| 20:21.12 | bmoraca_work | cool |
| 20:21.27 | vader-- | idespinner you said lastnight i won't be able to get meetme, iax2, and dadhi working on asterisk in a vmware enviroment? |
| 20:21.38 | vader-- | i can see dadhi but why iax2 and meetme? |
| 20:21.41 | idespinner | vader--, in general yes... |
| 20:21.44 | idespinner | timing |
| 20:21.55 | idespinner | you know dahdi_dummy... |
| 20:22.06 | idespinner | i hear that may be resolved soon. |
| 20:22.09 | bmoraca_work | dahdi_dummy works fine in VMware |
| 20:22.28 | bmoraca_work | meetme works as well, though it's definitely not as robust |
| 20:22.49 | vader-- | what do you mean by not as a robust? |
| 20:23.03 | bmoraca_work | and iax2 relies on dahdi when in trunkmode for timing and meetme needs a dahdi timing source |
| 20:23.10 | bmoraca_work | vader--: you won't be able to get as many people in the call |
| 20:23.50 | idespinner | bmoraca_work, i thought dahdi_dummy was unreliable under most virtual servers |
| 20:24.00 | vader-- | i wonder if i can use the TA924 as a timing source? |
| 20:24.04 | bmoraca_work | idespinner: never had a problem with it under VMware |
| 20:24.12 | bmoraca_work | vader--: not if your asterisk box is virtualized |
| 20:24.15 | idespinner | is that 1.4 or 1.6? |
| 20:24.21 | bmoraca_work | 1.4 |
| 20:24.58 | geneticx_wrk | Hello, I have a T1 line that is not being used since we moved to another ISP for our internet needs. I would like to use this line for asterisk, can I do so? or I need a voice T1 line ? |
| 20:25.16 | bmoraca_work | geneticx_wrk: you need a voice T1 |
| 20:25.28 | bmoraca_work | either CAS or CCS (PRI) |
| 20:26.36 | geneticx_wrk | bmoraca_work: Ok. do you know if it's easy to upgrade to this sort of line without signing another contract? |
| 20:27.18 | bmoraca_work | geneticx_wrk: you have to talk to your telco about that. they're the only ones who will have that information |
| 20:27.55 | geneticx_wrk | bmoraca_work: k. thank you. |
| 20:36.48 | Katty | scowls at sugarcrm |
| 20:37.40 | Katty | don't make me rip through your php guts! |
| 20:39.59 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 20:41.57 | Deeewayne | rollerblades past Katty |
| 20:42.17 | Katty | performs blade-by hugging upon Defraz |
| 20:42.18 | Katty | oh |
| 20:42.22 | Katty | performs blade-by hugging upon Deeewayne |
| 20:42.30 | Deeewayne | thanks! |
| 20:43.45 | Katty | i think i am going to have to perform surgery on sugarcrm |
| 20:44.35 | Deeewayne | Katty, sadness |
| 20:45.00 | Katty | it's okay |
| 20:47.42 | *** join/#asterisk fink (~guest@static-162-84-93-164.fred.east.verizon.net) |
| 20:47.44 | *** join/#asterisk Alagar (~Administr@122.164.38.173) |
| 20:48.25 | *** join/#asterisk `paul (~kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 20:49.28 | `paul | if i record voicelogs using dialer asterisk records fine but if i manually dial from the phone i get these 2kb logs which are basically blank. could it be a problem with phone settings? codec stuffs? |
| 20:51.20 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 20:53.16 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 20:55.31 | *** join/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 20:56.15 | QbY | does anyone know if there ia guide for querying TNS/Verisign for CNAM? They want either a subscribe or notify for each inbound call |
| 20:59.32 | bmoraca_work | just tried meetme on a virtual asterisk box over VMWare with two callers and there was no problem. |
| 21:02.30 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 21:03.32 | [TK]D-Fender | Facebook finally suggested Mark Spencer as a friend :) |
| 21:04.07 | *** part/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 21:06.37 | Nugget | heh |
| 21:06.53 | Nugget | he'll friend you back but you have to fax your facebook release form to digium first. |
| 21:06.58 | Naikrovek | hehe |
| 21:07.16 | Naikrovek | i don't think i'm friends with anyone in this channel on facebook |
| 21:07.21 | Naikrovek | don't know really |
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| 21:26.53 | rene- | hey guys, quick question, can CDR durations be preserved in the case of IAX/SIP reinvite? |
| 21:28.08 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 22:07.10 | *** join/#asterisk infobot (ibot@rikers.org) |
| 22:07.10 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.4, 1.6.1.16, 1.6.0.24 (2010/02/18), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 22:07.20 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 22:10.51 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 22:11.40 | adnc | someone today suggested me to use dialplan reload instead of a total reload. is there a command for loading the indications.conf. indication reload didnt work |
| 22:15.45 | carrar | module reload res_indications.so ? |
| 22:22.31 | voipmonk | yeah you can unload it and load it again |
| 22:26.41 | *** part/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 22:34.46 | *** join/#asterisk infobot (ibot@rikers.org) |
| 22:34.46 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.4, 1.6.1.16, 1.6.0.24 (2010/02/18), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 22:37.07 | vader-- | which asterisk GUI's do you guys rate as the top? |
| 22:37.15 | vader-- | i love working with asteris'k |
| 22:37.25 | vader-- | asterisk's dialplan through conf files and all |
| 22:37.42 | vader-- | but my coworkers aren't as fluent in that and it would be great to get this in a gui where they could point and click |
| 22:37.55 | vader-- | that way i can give simple tasks to them to do versus me doing them |
| 22:38.07 | t_j | vader--: freepbx is a good solution |
| 22:38.40 | vader-- | thats the one that seems to be getting the most votes |
| 22:39.20 | t_j | it what we use, the helpdesk ops seem ok with it and you can acl your uses so they can only access what they need |
| 22:39.23 | vader-- | i guess it won't manage the config files for my cisco phones? |
| 22:39.48 | t_j | no but there is a provisioning module we use that should not be to hard to make work with cisco |
| 22:40.00 | t_j | It alread supports polycom and granstream |
| 22:40.27 | cidu | aye, our CSR's have felt most comfortable with the elsatix interface, but im sure somebody else will point out a slew of reasons not to use it...even though they like elastix more, we still use unmodded free pbx so the csrs can do what they need |
| 22:40.51 | cidu | without the oddities of elastix |
| 22:40.57 | vader-- | you guys are happy with freepbx? |
| 22:41.09 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:41.31 | t_j | i am, infact I am porting it to FreeBSD I am so happy with the linux deployment so that we can use it in a more generic fasion on the rest of our infrastructure |
| 22:43.22 | vader-- | how about pbxware? |
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| 22:48.12 | p3nguin_ | cidu: Isn't the "elastix interface" in fact FreePBX? |
| 22:48.24 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:48.59 | joako | How do I troubleshoot "DAHDI_SPANCONFIG failed on span 1: Invalid argument (22)" |
| 22:49.02 | bmoraca_work | as far as controlling asterisk, yes, it's just freepbx. |
| 22:49.02 | *** join/#asterisk wpbrown (~wpbrown@wh-gtw-0001.woolfharris.com) |
| 22:49.04 | joako | Invalid argument *WHERE* |
| 22:49.27 | t_j | p3nguin_: looks very similar |
| 22:49.42 | vader-- | is pbxware free? |
| 22:49.52 | p3nguin_ | t_j: I'm pretty sure it IS FreePBX. |
| 22:49.53 | bmoraca_work | never used pbxware or looked at it at all |
| 22:50.28 | wpbrown | I have a simple simple question. When I have a user call in the phone system reports the average hold time. She only reports "1" vs "1 min" do you think a sound file is corrupted in /var/lib/asterisk/sounds/en? |
| 22:50.38 | bmoraca_work | elastix uses freepbx to control asterisk, yes. it also has a bunch of other GUIs ranging from useless to moderately useful |
| 22:51.02 | p3nguin_ | wpbrown: "core set verbose 10" and watch what sound files are trying to play. |
| 22:53.11 | wpbrown | that is cool! |
| 22:53.23 | wpbrown | i got the 30 second marker the one min one will be next. |
| 22:54.52 | joako | Does Asterisk document its own error messages? |
| 22:54.59 | p3nguin_ | Sometimes. |
| 22:55.33 | wpbrown | P3nguin here was the error: [Feb 23 16:54:47] WARNING[29328]: file.c:936 ast_streamfile: Unable to open queue-minute (format 0x44 (ulaw|slin)): No such file or directory |
| 22:56.00 | p3nguin_ | wpbrown: Now you know what the problem is and how to fix it. |
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| 22:56.42 | wpbrown | It says it doesn't exist in any format. Yet they are there. So am I safe to assume they are corrupted files? |
| 22:57.06 | p3nguin_ | Show me that the file exists. |
| 22:57.07 | bmoraca_work | wpbrown: it says it doesn't exist as a ulaw or slin format. |
| 22:57.42 | wpbrown | [Feb 23 16:54:47] WARNING[29328]: file.c:635 ast_openstream_full: File queue-minute does not exist in any format |
| 22:57.51 | wpbrown | then.. |
| 22:57.55 | wpbrown | [Feb 23 16:54:47] WARNING[29328]: file.c:936 ast_streamfile: Unable to open queue-minute (format 0x44 (ulaw|slin)): No such file or directory |
| 22:57.57 | p3nguin_ | Show me that the files exist. |
| 22:58.05 | wpbrown | k let me look. |
| 22:58.22 | p3nguin_ | pastebin.com if you have lots of lines. |
| 22:58.52 | wpbrown | the files should exist in /var/lib/asterisk/sounds/en right? |
| 22:59.36 | p3nguin_ | No idea, since I do not have queue-minute.anyformat on my system. |
| 23:00.22 | wpbrown | I have it in that dir in 7 different formats |
| 23:00.36 | wpbrown | no i dont! |
| 23:00.37 | p3nguin_ | I'm still waiting for you to show me. |
| 23:00.46 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 23:00.46 | wpbrown | that is queue-minutes |
| 23:00.52 | p3nguin_ | That's what I have. |
| 23:01.03 | p3nguin_ | queue-minutes with an S |
| 23:01.06 | carrar | I have green eggs and ham sam I am |
| 23:01.12 | wpbrown | correct |
| 23:02.06 | wpbrown | on phpmyadmin is calls for queue_minutes |
| 23:02.22 | wpbrown | yet the pbx is looking for the other |
| 23:03.21 | p3nguin_ | What is your "queue-minutes =" set to in queues.conf? |
| 23:04.34 | wpbrown | queue-minutes = queue-minutes |
| 23:04.55 | p3nguin_ | Okay, so it isn't trying to use that setting for the "1 minute" announcement, then. |
| 23:05.22 | p3nguin_ | Mine seems to only go as low as 2 minutes, and then says less than 2 minutes when it's less. |
| 23:05.39 | p3nguin_ | So I understand my lack of the queue-minute file(s). |
| 23:05.52 | wpbrown | I don't have it either |
| 23:06.01 | wpbrown | I have minute.* |
| 23:06.11 | wpbrown | but not queue-minute.* |
| 23:06.29 | p3nguin_ | Could this be a bug? |
| 23:07.01 | wpbrown | I like the way yours is set up less than 2 mins is cool |
| 23:07.08 | wpbrown | but mine also reports seconds |
| 23:07.27 | wpbrown | I take about 800 calls a day on this thing.. |
| 23:08.27 | *** part/#asterisk DGMurdockIII (~dgmurdock@208-70-41-206.bb.hrtc.net) |
| 23:10.29 | wpbrown | I don't know if it is a bug or the fact that a friend of mine convinced me that it would be cool to bypass editing some of the files in /etc/asterisk and using mysql and phpmyadmin to directly edit some of the files.. |
| 23:10.31 | bmoraca_work | queues.conf i believe lets you specify those announcements. |
| 23:10.40 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 23:10.48 | wpbrown | To tell you the truth I liked editing the files better in /etc/asterisk |
| 23:10.57 | wpbrown | because that is how I used to do it yrs ago |
| 23:11.14 | wpbrown | I don't understand mysql/phpmyadmin very well |
| 23:11.59 | p3nguin_ | I don't see a setting for the "minute" sound in queues.conf. |
| 23:12.09 | bmoraca_work | queue-minutes is a configurable parameter in 1.6 |
| 23:12.18 | bmoraca_work | line 307 of queues.conf.sample |
| 23:12.23 | p3nguin_ | Yeah, minutes, with an S. |
| 23:12.29 | t_j | why would "core show uptime" not print anything? |
| 23:12.31 | bmoraca_work | right, but it's CONFIGURABLE |
| 23:12.56 | bmoraca_work | i could set "queue-minutes = tt-monkies" if i wanted to |
| 23:13.03 | wpbrown | bmoraca did you see the error asterisk was giving me? |
| 23:13.07 | p3nguin_ | It wouldn't make sense to change it to say minute, because the sound file does not exist, and even if it did, you wouldn't want to say "5 minute" |
| 23:13.31 | bmoraca_work | p3nguin_: you don't need to convince me. wpbrown screwed up his configuration |
| 23:13.56 | bmoraca_work | wpbrown: yes, and you have your queues.conf misconfigured. you're missing an "s" on the file you specify for the "queue-minutes" prompt. go fix it. |
| 23:14.01 | bmoraca_work | or rename the file |
| 23:14.03 | p3nguin_ | I don't even know where the "queue-minute" sound SHOULD be played. |
| 23:14.17 | p3nguin_ | (1704.33) <wpbrown> queue-minutes = queue-minutes |
| 23:14.42 | bmoraca_work | p3nguin_: it shouldn't, and there's no such thing. asterisk doesn't distinquish between time = 1 and time > 1 |
| 23:14.45 | p3nguin_ | It's apparently something else causing the behavior. |
| 23:15.09 | bmoraca_work | no, remember he's said that he's pulling configs from a database. any file that he's pastebined for you is irrelevant. |
| 23:15.13 | p3nguin_ | wpbrown: What happens when the time is 2 or more minutes? |
| 23:15.29 | wpbrown | let me test it and see I think it works |
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| 23:16.24 | wpbrown | Even though I didn't configure this particular box it is my baby to fix now. My programmer buddy fell off the planet. |
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| 23:17.54 | bmoraca_work | wpbrown: you need to figure out where the configs are coming from (conf file or mysql) and fix it there. there is only one announcement for this, and it doesn't distinguish between singular and plural units. you're telling it to play the wrong file. it's plain and simple. |
| 23:18.26 | wpbrown | The conf file is within mysql |
| 23:18.42 | wpbrown | I am trying to figure out where the file is to edit at the moment |
| 23:18.44 | p3nguin_ | bmoraca_work: Are you saying that the announcement would say "one minutes" or "two minute"? |
| 23:18.53 | wpbrown | at this point 1 minutes is better than "1" |
| 23:18.55 | wpbrown | lmao |
| 23:19.01 | p3nguin_ | agreed |
| 23:19.04 | bmoraca_work | p3nguin_: the announcement will say "1 minutes" or "2 minutes" |
| 23:19.26 | bmoraca_work | it doesn't distinguish between plural or singular...as evidenced by the fact that there is no singular audio file |
| 23:19.28 | p3nguin_ | I'm assuming it is nearly impossible to detect it, though. |
| 23:19.28 | wpbrown | i am just going to cp the file to what the database is asking for |
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| 23:20.01 | bmoraca_work | wpbrown: changing the file is going to do nothing if it's pulling from a database. why don't you dump that portion of the database and pastebin it? |
| 23:20.05 | wpbrown | let me test this thing up to 2 minutes and see what it does.. |
| 23:20.24 | wpbrown | oh so the sound files are in the database too? |
| 23:20.28 | bmoraca_work | wpbrown: the config parameter in question is a per-queue setting. |
| 23:20.40 | bmoraca_work | wpbrown: no, but the config parameter that tells asterisk which sound file to play is in the database |
| 23:20.47 | wpbrown | k |
| 23:21.16 | wpbrown | she said 2 minutes |
| 23:21.24 | wpbrown | so it works on 2 not one |
| 23:21.26 | *** join/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 23:21.30 | p3nguin_ | so the plot thickens |
| 23:21.53 | bmoraca_work | wpbrown: i don't care what she said. if you don't get me the configs i asked for, i'm not going to help you further. |
| 23:21.53 | QbY | is it possible in a Dial(DAHDI/x) to have it pick the next available channel??? |
| 23:22.17 | bmoraca_work | QbY: yes, you want dahdi groups |
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| 23:24.19 | wpbrown | just export from the database? |
| 23:24.31 | bmoraca_work | yes |
| 23:24.52 | wpbrown | what is easier csv file usually? |
| 23:25.19 | bmoraca_work | doesn't matter |
| 23:26.07 | avajadi | Is there anybody here who can help me figure out what "(FAX_FAILURE_TRAINING), error: 'INVLD_DCS'" means? |
| 23:36.26 | avajadi | Anybody? |
| 23:37.13 | cidu | p3nguin_, the elastix interface is a fairly modified freepbx front end, there is a button to launch unmodified freepbx, iut just has some integration that seems to make life easier for people that dont actually like computers, and a bunch of other oddities, dunno, i kinda feel like most people would define a difference, even though techniocally under the hood its pretty much the same |
| 23:37.18 | wpbrown | Is it cool to post the pastebin.com url in this channel? |
| 23:37.44 | p3nguin_ | wpbrown: Yes. That is how we prefer it. |
| 23:37.53 | cidu | soo, anybody know how i would pass extra variables to the externnotify command in addition to the 3 it passes automatically to whatever command is defined in externnotify = command? |
| 23:37.55 | wpbrown | bmoraca_work: http://pastebin.com/V4c0kS1j |
| 23:38.12 | wpbrown | Now, this may not be what you are looking for. I am not a mysql guru |
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| 23:38.29 | wpbrown | but i exported the file that had the info in phpmyadmin |
| 23:39.01 | bmoraca_work | interesting. you didn't include column headings, but that should be the way it should be |
| 23:39.35 | wpbrown | should I include them? |
| 23:39.42 | bmoraca_work | wpbrown: at this point, no |
| 23:39.56 | bmoraca_work | wpbrown: just make a copy of the queue-minutes file and rename it queue-minute and be done with it |
| 23:40.03 | wpbrown | hehe |
| 23:40.23 | wpbrown | let me get to work on it. brb |
| 23:40.43 | bmoraca_work | cidu: the elastix interface did not modify freepbx in the slightest. they simply embed it into their own interface. it is, however, identical. |
| 23:41.12 | cidu | ohh, thanks for the clarification, was just trying to be helpfull :) |
| 23:41.45 | bmoraca_work | no problem. some of the other interfaces that elastix has built in to their interface are somewhat useful |
| 23:41.54 | bmoraca_work | but there is also a lot of other crap |
| 23:42.05 | cidu | dunno, i rarely look at the guis |
| 23:42.48 | cidu | just our csrs has said they prefered the elastix interface (i think it was just prettier to them) |
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| 23:46.28 | wpbrown | Okay. Care to know what I did? |
| 23:47.28 | wpbrown | I went into /var/lib/asterisk/sounds/en changed then from the command line cp minute.wav queue-minute.wav |
| 23:48.18 | wpbrown | now she reports "1 min" singular and "2 mins" plural |
| 23:50.10 | bmoraca_work | sounds good. |
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| 23:51.36 | Deeewayne | shit, its 50 minutes past beer o'clock |
| 23:52.07 | bmoraca_work | it's not beer o'clock for another hour |
| 23:52.08 | wpbrown | Bmoraca and p3nguin thanks for your input. |
| 23:52.20 | wpbrown | I appreciate you helping me |
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