| 00:03.50 | ChannelZ | 3102 |
| 00:04.21 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 00:04.24 | parim | ChannelZ: how about a 2 port fxo? |
| 00:04.57 | ChannelZ | hmm dunno on that one |
| 00:05.08 | parim | ChannelZ: thanks. |
| 00:18.04 | *** join/#asterisk Neo31 (~Neo31@unaffiliated/neo31) |
| 00:19.00 | Neo31 | hello, please I need some help with digium-current repo, i have a centos 5.4 installation and i have problemes using yum. it says : Cannot retrieve repository metadata for repository digium-current |
| 00:19.55 | Neo31 | i used the path located here : http://www.asterisk.org/downloads/yum |
| 00:22.06 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 00:23.22 | exothermc | is 1.6.x backwards compatable with 1.4.x ? |
| 00:25.37 | Neo31 | exothermc it's the dead walk here :p euh sry I mean dead chat :p |
| 00:25.37 | carrar | some things are |
| 00:25.41 | carrar | some aren't |
| 00:25.58 | Neo31 | finally someone answering :) |
| 00:26.00 | exothermc | carrar: What common things aren't? |
| 00:26.16 | carrar | everything that was added in 1.6 and or changed |
| 00:26.21 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 00:26.29 | carrar | common is relative |
| 00:26.56 | TJNII | exothermc: Perhaps a reading of upgrade.txt is in order? |
| 00:27.04 | exothermc | carrar: How would things added not be backwards compatible? I can see things changed, but added makes no sense. |
| 00:27.12 | exothermc | TJNII: ya getting into that now. |
| 00:27.31 | carrar | ADDED |
| 00:27.39 | carrar | You know, was not there before |
| 00:27.44 | Neo31 | does anyone know abt the metadata digium problem i have ? |
| 00:27.46 | carrar | and is now |
| 00:28.08 | carrar | NEW |
| 00:28.49 | exothermc | carrar: which would have no effect of using 1.4 conf files on 1.6, and therefor has no effect on backwards compatibility. |
| 00:28.59 | TJNII | Unless there are bugs. |
| 00:29.10 | TJNII | Which there always will be in software. |
| 00:29.20 | carrar | You're question notes you want to use 1.6 configs on 1.4 |
| 00:29.29 | carrar | I assume just configs |
| 00:30.04 | exothermc | carrar: sorry I guess I wasn't clear. I'm upgrading to 1.6.x from 1.4.x |
| 00:31.08 | carrar | there might be some minor syntex stuff, but like TJNII said read the upgrade stuff |
| 00:31.11 | carrar | just run it |
| 00:31.14 | carrar | see what breaks |
| 00:31.18 | exothermc | not the (I guess, more commonly assumed) path of upgrading from 1.6.x to 1.4.x |
| 00:31.31 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 00:31.38 | exothermc | ya going to give it a go now. |
| 00:31.50 | TJNII | Have fun. Don't try it on a production box. |
| 00:31.50 | carrar | everyones configs are differnt |
| 00:32.07 | carrar | Where is the fun in that TJNII! |
| 00:32.39 | TJNII | Eh, to each his own. |
| 00:32.45 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 00:32.47 | carrar | heheh |
| 00:34.15 | TJNII | Today I got the joy of having someone breathe down my neck to get a box back up while debugging a PCI slot failure. After an hour I realized the FNG tried to drive a screw through the mainboard before realizing he had to turn a assembly around. I prefer to avoid that kind of fun. |
| 00:35.36 | Neo31 | well at least can anyone tell me what is the difference between asterisk-current repo and digium-current repo? may be i could just ignore the digium repo!? |
| 00:35.52 | *** join/#asterisk diegomad (mad@186.81.136.28) |
| 00:37.09 | carrar | install from source |
| 00:37.37 | Neo31 | what's the benefit of installing from source carrar ? |
| 00:37.54 | TJNII | Eh, there is something to be said for using a package manager. |
| 00:38.38 | carrar | assuming you've made the packages yourself |
| 00:39.23 | Neo31 | i don't get it! |
| 00:39.43 | TJNII | Yea, but then you still have to keep track of updates yourself. Too easy to never update if you install from source. |
| 00:39.57 | TJNII | Though I will agree source installs have their place, no point in arguing that. |
| 00:40.34 | TJNII | prefers Gentoo. Best (and worse) of both worlds. |
| 00:41.02 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 00:41.23 | Neo31 | well TJNII in case there is a security update i'll have to update anyway |
| 00:41.48 | Neo31 | so may be updating from the repo would be easier |
| 00:42.00 | *** part/#asterisk dcplaya (~IceChat7@cpe-71-67-109-185.cinci.res.rr.com) |
| 00:44.03 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 00:44.04 | *** mode/#asterisk [+o angler] by ChanServ |
| 00:48.17 | *** part/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 00:48.29 | *** join/#asterisk dotnetted (~dotnetted@75.138.79.27) |
| 00:49.25 | dotnetted | hey all - is there a gui available for asterisk 1.6 configuration? |
| 00:49.36 | dotnetted | (ubuntu) |
| 00:50.05 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 00:53.28 | Neo31 | i used freepbx with asterisk 1.4 it should work with asterisk 1.6 too |
| 00:54.54 | *** join/#asterisk roxdragon (~roxdragon@host24-102-dynamic.247-95-r.retail.telecomitalia.it) |
| 00:55.03 | roxdragon | hi |
| 00:55.06 | roxdragon | help please |
| 00:56.11 | Neo31 | roxdragon start by posting the problem so everyone can see it |
| 00:56.40 | dotnetted | Neo31: thanks |
| 00:57.20 | roxdragon | Neo31, http://rapidshare.com/files/378632772/video.zip.html |
| 00:57.32 | roxdragon | Neo31, dahdi don't work |
| 00:57.44 | roxdragon | see clip |
| 00:58.33 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:58.39 | TJNII | video.zip? Yea, right. Pastebin some debug output/ |
| 00:59.42 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:59.53 | Neo31 | u'r welcome roxdragon |
| 01:00.16 | roxdragon | Neo31, thanks |
| 01:00.17 | Neo31 | ^^ roxdragon post it on megavideo or youtube, it would be easier for them :p (in case you can't explain the problem without that video) |
| 01:00.59 | Neo31 | ur welcome dotnetted ^^ |
| 01:01.09 | Neo31 | sry roxdragon ;p |
| 01:03.01 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 01:03.49 | roxdragon | Neo31, http://www.youtube.com/watch?v=pCj9z-LSZus :( dont'work |
| 01:04.43 | Neo31 | roxdragon i am just a newbie with asterisk, but i'll take a look, may be i can help with somethin, who knows :p |
| 01:05.49 | Neo31 | what is that roxdragon ? |
| 01:05.59 | Neo31 | a linux os booting |
| 01:06.02 | Neo31 | so what ? |
| 01:06.27 | roxdragon | Neo31, the problem is dahdi 0.14 secs |
| 01:06.51 | roxdragon | debian Neo31 |
| 01:07.18 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 01:07.19 | Neo31 | ah, sry i have problems to see the text |
| 01:07.22 | Neo31 | hum |
| 01:08.17 | Neo31 | sry i think i can't help with it anyway |
| 01:08.22 | roxdragon | yes Neo31 :( reinstall dahdi? |
| 01:10.20 | Neo31 | it depends on what it does roxdragon, i told you i am just a newbie with asterisk, i dont even know what dahdi does |
| 01:11.45 | TJNII | Dahdi is (basically) a driver package for voice cards. It also provides dummy timing modules for pure VoIP systems. |
| 01:12.10 | roxdragon | yes |
| 01:12.16 | TJNII | Someone who knows is better can probably come up with better wording. I'm pure VoIP, I only use it for the dummy driver. |
| 01:12.52 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 01:14.51 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
| 01:19.06 | exothermc | is there anything that has changed between 1.4.26 and 1.6.2 that would have caused my extensions.conf to not be read? |
| 01:26.36 | Neo31 | dotnetted i just tested freepbx with asterisk 1.6 and it worked fine :) |
| 01:28.21 | TJNII | Neo31: Don't encourage him. |
| 01:28.36 | TJNII | In two weeks he'll be back complaining that freepbx doesn't work. |
| 01:28.46 | sawgood | just wondering if anyone has an opinion on what should be done to the 'default' logger.conf file on a new Asterisk install |
| 01:28.47 | Neo31 | lol TJNII i was testing it anyway :p |
| 01:28.51 | sawgood | Should I change anything to help? |
| 01:29.44 | TJNII | sawgood: Probably not needed. |
| 01:30.16 | sawgood | Do you send your logging statement to a syslog server instead of a file (or both) |
| 01:30.35 | sawgood | I was thinking since I have quite a few boxes, I would setup a syslog server to 'watch' for WARNINGS or ERRORS |
| 01:31.08 | Neo31 | i think it's time to go, i'll finish testing the server tomorrow :) thanks for help :) |
| 01:31.47 | TJNII | All my syslogs go to files. I don't do anything fancy. |
| 01:32.19 | sawgood | TJNII: Do you use tail or grep to search them at times? |
| 01:32.45 | TJNII | Yes |
| 01:33.23 | TJNII | I also have a logrotate daemon on the server boxes. |
| 01:33.40 | sawgood | how often do you rotate the logs? |
| 01:33.56 | TJNII | To be honest, Debian default. :) |
| 01:34.17 | sawgood | oh, I was thinking logrotate was an Asterisk script |
| 01:34.31 | TJNII | No, no. Completely independent. |
| 01:34.35 | carrar | sawgood, I'd setup a syslog server with syslog-ng and swatch |
| 01:35.26 | WIMPy | But Asterisk rotates automativally when the log gets bigger than 2GiB :-) |
| 01:35.55 | carrar | it only rotates it's own log files, not syslogd |
| 01:36.05 | sawgood | I'll do that tonight when I get back in to my LAB (setup a syslog server) |
| 01:36.11 | p3nguin | Maybe its own... |
| 01:37.06 | sawgood | So, I have one client who tells me he, "cannot dial some specific number" (for example (415) 777-1009) (when I try it via an external softphone from my lab, the call goes through |
| 01:37.19 | sawgood | When he tries it from a Snom 320 phone on the LAN it fails |
| 01:37.40 | carrar | tell em its user error and go back to youtube |
| 01:37.42 | sawgood | all Snom 320 phones fail to the same number (there are 4 or5 Snom 320 phones on the LAN) |
| 01:38.08 | sawgood | carrar: actually, tried that, but they call back reporting more numbers with the same issue |
| 01:38.23 | sawgood | I wasn't sure if the Asterisk logger would catch a failed call attempt or not |
| 01:38.51 | carrar | it would if the issue isn't with the phone's digitmap not accepting the numbers |
| 01:38.53 | sawgood | I wasn't sure how I could 'troubleshoot' this in a post event (without having the Asterisk CLI open) |
| 01:39.43 | sawgood | can one 'dial out' from the Asterisk CLI to test a number (instead of using a softphone) |
| 01:39.45 | sawgood | that would be cool |
| 01:39.50 | carrar | yes |
| 01:39.53 | TJNII | /var/log/asterisk/messages or CLI + screen with a large buffer |
| 01:40.01 | sawgood | I guess you would need a speaker and mic though |
| 01:40.10 | carrar | not to just dial a number |
| 01:40.17 | carrar | for testing |
| 01:40.29 | sawgood | what would the syntax for that be? |
| 01:40.35 | sawgood | I'll try it now on a LAB box I have |
| 01:40.48 | Slugs_ | sawgood, like originate? |
| 01:41.26 | sawgood | no originate command for me (at the CLI) |
| 01:41.53 | WIMPy | channel originate |
| 01:42.48 | Slugs_ | originate proto/channel/phone# application playback /var/lib/asterisk/sounds/sound.wav |
| 01:43.09 | Slugs_ | send somebody a sound! ;) |
| 01:43.22 | sawgood | neat! |
| 01:43.29 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 01:43.41 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:43.41 | sawgood | launch a callback campaign |
| 01:43.48 | Slugs_ | indeed |
| 01:43.55 | TJNII | Ooh, I'll have to remember that when I want voicemail to call me at work. |
| 01:44.02 | TJNII | Looks easier than a call file. |
| 01:44.15 | Slugs_ | it is |
| 01:44.24 | Slugs_ | i think atleast |
| 01:44.51 | Slugs_ | i call my wife w/ a diff song all the time ;) |
| 01:45.05 | Slugs_ | when playback() is done it just hangs up on her |
| 01:45.05 | TJNII | Haha |
| 01:45.13 | sawgood | channel originate "301 302" |
| 01:45.19 | sawgood | would that call from 301 to 302? |
| 01:45.22 | TJNII | She actually listens to the whole thing? |
| 01:45.32 | Slugs_ | not anymore! |
| 01:45.33 | Slugs_ | ;) |
| 01:45.47 | Slugs_ | as soon as she hers 1 line she hangs up |
| 01:45.52 | Slugs_ | sawgood, |
| 01:46.00 | Slugs_ | is this sip? |
| 01:46.07 | sawgood | yes 100% |
| 01:46.56 | Slugs_ | sec |
| 01:48.55 | Slugs_ | i thought you were making an outbound call, i don't know if you can do internally like that.... |
| 01:49.35 | sawgood | I'll try an external call to my cell phone ... |
| 01:49.44 | sawgood | I have a SIP trunk for in/out calls |
| 01:49.47 | sawgood | no FXO |
| 01:50.53 | Slugs_ | that's ifne |
| 01:50.55 | Slugs_ | fine |
| 01:51.06 | exothermc | is extensions.conf still the default place for contexts? |
| 01:51.13 | sawgood | yes |
| 01:51.14 | Slugs_ | yes |
| 01:51.32 | TJNII | There is also extrensions.ael, if you're suffering from dialplan confusion. |
| 01:51.34 | Slugs_ | ./etc/asterisk/.... |
| 01:51.51 | TJNII | s/extrensions.ael/extensions.ael/ |
| 01:52.12 | exothermc | TJNII: Ya I moved out the extensions.ael just to be sure |
| 01:52.24 | Slugs_ | permissions? |
| 01:52.25 | TJNII | Probably wise. |
| 01:52.48 | Slugs_ | is it not being read or something? |
| 01:53.48 | exothermc | hmm good call checking now, but asterisk running as root should have access to all. |
| 01:54.33 | exothermc | ok so 'core set debug atleast 9' doesn't do what I expected it would ( |
| 01:54.40 | Slugs_ | sawgood, originate SIP/302/cellnumber application args |
| 01:55.04 | exothermc | I was hoping for lots of debug, but still only getting Call from 'peername' to extension '12345' rejected because extension not found |
| 01:55.13 | TJNII | Don't forget that Asterisk doesn't load config changes as they happen. You have to reload the affected modules. |
| 01:55.24 | Slugs_ | true |
| 01:55.32 | exothermc | TJNII: just a general "reload" doesn't do it? |
| 01:55.48 | Slugs_ | only if you edit dialplan |
| 01:56.03 | Slugs_ | only dialplan |
| 01:56.18 | sawgood | ty! |
| 01:56.22 | Slugs_ | np |
| 01:56.50 | TJNII | exothermc: Not sure, I've never used a bare "reload" command. Restart would be safer, or dialplan reload. |
| 01:57.16 | TJNII | Eh, reload might be safe, I don't know. |
| 01:57.26 | TJNII | I defer tom someone with more knowledge. |
| 01:57.32 | TJNII | s/tom/to/ |
| 01:58.00 | exothermc | how do I turn up more debug on the console? |
| 01:58.02 | WIMPy | reload reloads much more than just the dialplan, but not everything. |
| 01:58.19 | leifmadsen | everything that is a module |
| 01:58.30 | leifmadsen | and whatever modules have the ability to be reloaded |
| 01:59.27 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 01:59.36 | WIMPy | I can't remember what, but I had effects where a reload didn't work, but a module reload ... did. |
| 01:59.56 | sawgood | Slugs_: can you give me an example of "application args" (the syntax)? |
| 02:01.08 | Slugs_ | <PROTECTED> |
| 02:01.40 | exothermc | ok got it |
| 02:01.48 | Slugs_ | originate SIP/302/cellnumber application playback /var/lib/asterisk/sounds/soundfile.wav |
| 02:01.49 | exothermc | some errrors in my config file |
| 02:02.14 | exothermc | which showed up when trying to load module load pbx_config.so |
| 02:05.22 | *** join/#asterisk Jed__ (~Jed@ool-4574c525.dyn.optonline.net) |
| 02:06.16 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 02:12.24 | ChannelZ | shake shake shake |
| 02:17.58 | exothermc | any idea why this no longer seems to work with 1.6.2: exten => *97,n,Set(VMBOX=${SIPPEER($[ "${CHANNEL}" : "\[^/\]+/(.+)\[-\]\[^-\]+" ]:mailbox)}) ?> |
| 02:26.56 | *** join/#asterisk methodvon (~methodvon@c-69-255-206-41.hsd1.va.comcast.net) |
| 02:29.04 | exothermc | Where is the docs on asterisk applications? |
| 02:29.18 | ChannelZ | core show application xxx |
| 02:30.07 | exothermc | ChannelZ: I probably didn't build it right: Not available |
| 02:30.24 | ChannelZ | what application? |
| 02:30.31 | exothermc | voicemailmain |
| 02:30.51 | ChannelZ | you have app_voicemail loaded? |
| 02:31.02 | exothermc | trying to find a way to allow peers to access their voicemail without a password, and force their mailbox. |
| 02:31.10 | exothermc | ChannelZ: ya it works fine. |
| 02:34.18 | *** join/#asterisk sourcode (~code@ppp-58-8-118-233.revip2.asianet.co.th) |
| 02:34.54 | TJNII | exothermc: How many users? |
| 02:35.09 | exothermc | 250 |
| 02:36.02 | exothermc | we used to just pull the mailbox off the sip peer and stuff it in the channel var with this: exten => *97,n,Set(VMBOX=${SIPPEER($[ "${CHANNEL}" : "\[^/\]+/(.+)\[-\]\[^-\]+" ]:mailbox)}) |
| 02:36.26 | TJNII | Yea, my regex isn't good enough to follow that. |
| 02:36.30 | exothermc | but ${SIPPEER($[ "${CHANNEL}" : "\[^/\]+/(.+)\[-\]\[^-\]+" ]:mailbox)} is coming up blank |
| 02:36.43 | ChannelZ | it doesn't look right, the very first part, \[^ |
| 02:36.55 | exothermc | is there any other way to access to the mailbox sip var? |
| 02:37.01 | TJNII | Just keying off $CALLERID(num) won't work? |
| 02:37.20 | exothermc | TJNII: and let people spoof it? |
| 02:37.40 | ChannelZ | Who cares? |
| 02:37.43 | TJNII | shrugs |
| 02:37.49 | ChannelZ | Does everyone know everyone else's voicemail passwords? |
| 02:38.13 | exothermc | ChannelZ: doubt it they are all forced to set their own password on VMbox setup. |
| 02:38.30 | exothermc | ChannelZ: but I can't stop them from telling other people what their password is. |
| 02:38.45 | TJNII | goes back to trying to slove that jumble of resistors in today's xkcd |
| 02:39.01 | ChannelZ | Can you stop them from running into each other's offices and using their phones if they're intent on breaking into other people's voicemail? |
| 02:39.20 | carrar | exothermc: how about something like exten => *97,1,VoicemailMain(${CALLERID(number)}@contexthere|s) |
| 02:39.32 | carrar | assuming their extension number is the voicemail number |
| 02:39.57 | exothermc | carrar ya assuming I don't want to prevent people from spoofing the number. |
| 02:40.09 | carrar | set some policies |
| 02:40.21 | carrar | then require a password |
| 02:40.24 | carrar | drop the |s |
| 02:40.40 | TJNII | carrar: Suggested already. He's worried about security on his passwordless voicemail system. |
| 02:40.49 | carrar | haha |
| 02:41.54 | exothermc | The users aren't physically located next to each other. |
| 02:41.56 | ChannelZ | You could just set an extra variable for every device in sip.conf etc of each person's mailbox number |
| 02:42.18 | ChannelZ | but... |
| 02:42.25 | carrar | PARANOIDVM=6200 |
| 02:42.35 | exothermc | well in sip.conf peers I have: mailbox=12345 |
| 02:42.36 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 02:42.42 | exothermc | How do I access that? |
| 02:42.52 | ChannelZ | SIPPEER(whatever,mailbox) should show you the mailbox=xxx set in sip.conf |
| 02:43.06 | exothermc | ahh cool |
| 02:43.39 | ChannelZ | that's assuming they're all SIP |
| 02:43.56 | exothermc | which they are. |
| 02:44.06 | carrar | what if they call from the outside to check their voicemail? |
| 02:44.12 | ChannelZ | Fuck them! |
| 02:44.16 | carrar | heh |
| 02:44.21 | *** join/#asterisk jarvis141 (~Adium@84.242.101.194) |
| 02:44.22 | exothermc | carrar: that is what passwords are for. |
| 02:44.30 | ChannelZ | That's what the passwords ARE ALREADY FOR |
| 02:44.45 | carrar | What if I walk over to user's X phone and hit voicemail? |
| 02:44.51 | carrar | NO password!! |
| 02:44.57 | ChannelZ | What happened to "I don't want people spoofing their mailbox".. what good is that if they can just call in an outside line and type in whatever box they want |
| 02:45.04 | exothermc | carrar: How do you know where user' X is located, or where their phone is? |
| 02:45.20 | carrar | Cause I am the devil |
| 02:45.24 | TJNII | Psychic powers. |
| 02:45.46 | *** join/#asterisk blaines (~blaines@64-130-243-194.sparkplugbb.net) |
| 02:45.50 | exothermc | ChannelZ: You don't allow passwordless access unless it is from the auth'd peer. |
| 02:46.00 | jarvis141 | Hi, guys. I'm trying to interconnect Cisco CME and Asterisk, but Asterisk tells me "Registration failed, no matching peer found".. I've probably forgot some configuration :/ Could you help me? |
| 02:46.12 | ChannelZ | I didn't know this had anything to do with passwordless anything |
| 02:46.37 | exothermc | ok how do I pull the peer name out of the current channel. |
| 02:48.08 | ChannelZ | You could pull off SIP/ I think like ${CHANNEL:4} |
| 02:48.24 | exothermc | <PROTECTED> |
| 02:48.24 | exothermc | <PROTECTED> |
| 02:48.28 | exothermc | that should do it. |
| 02:48.31 | exothermc | hopefully |
| 02:48.40 | exothermc | tries it out. |
| 02:50.37 | fenrus | jarvis141, interconnecting CME <-> Asterisk with sip ? |
| 02:50.50 | jarvis141 | fenrus: yup, correct |
| 02:51.18 | fenrus | jarvis141, did you create a peer in the CME ? |
| 02:51.37 | fenrus | try enable some debugging for sip and registring in the cme |
| 02:52.09 | ChannelZ | ${CHANNEL(peername)} since I guess ${CHANNEL} has the call ID on it |
| 02:52.50 | jarvis141 | fenrus: yup, I've made some, CME tries to register (I would like to use Asterisk as a voicemail), but Asterisk trhrows "no matching peer" |
| 02:53.18 | jarvis141 | so possibly only something missing in sip/extensions/voicemail config :/ |
| 02:53.38 | TJNII | Try just sip |
| 02:53.47 | jarvis141 | type=peer in sip.conf maybe? |
| 02:53.53 | TJNII | Turn on sip debug and find the problem |
| 02:54.07 | fenrus | yea, debugging is neat |
| 03:06.00 | ChannelZ | boobs are neat |
| 03:09.20 | spenguin[work] | http://openbts.sourceforge.net/ |
| 03:09.29 | spenguin[work] | never realised they used asterisk |
| 03:11.27 | jarvis141 | okay, I got debugging enabled, can see packets :) any hints what I've missed in configuration? |
| 03:11.49 | carrar | line 2 |
| 03:12.59 | jarvis141 | REGISTER packet from CME, 404 reply from Asterisk |
| 03:13.51 | ChannelZ | Yes I see the problem right there. |
| 03:14.07 | TJNII | Whatever you do, don't pastebin anything for us to see. |
| 03:14.13 | TJNII | Otherwise we might be able to help. |
| 03:15.03 | exothermc | anyone use BLF and polycoms ? |
| 03:15.54 | jarvis141 | So, what would you like to know, to direct me some way? :) I tried to follow a guide, I'm still little lost in Asterisk config |
| 03:16.40 | TJNII | You know that debug you're getting? Pastebin it. And your sip.conf. |
| 03:16.52 | ChannelZ | Miss Cleo has the night off. |
| 03:18.30 | fenrus | jarvis141, try pastebin some of the relevant config from the CME and the sip config from the asterisk ? |
| 03:18.40 | fenrus | we might be able to help you then :) |
| 03:19.45 | jarvis141 | I've followed this guide, only changed the IPs of CME and * .. http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration |
| 03:20.23 | ChannelZ | Ok so we know you know how to read, but apparently have selective hearing? |
| 03:22.46 | jarvis141 | my config: extensions.conf http://paste.misacek.net/163/ | sip.conf http://paste.misacek.net/164/ | voicemail.conf http://paste.misacek.net/165/ |
| 03:23.45 | jarvis141 | SIP and voicemail only modified, both at the bottom |
| 03:23.57 | jarvis141 | extensions taken from the example |
| 03:25.15 | ChannelZ | I can't hit that site |
| 03:25.29 | TJNII | Pastebin the SIP DEBUG as well. |
| 03:26.45 | jarvis141 | site should be now up, sorry.. going to look for a debug log |
| 03:27.47 | ChannelZ | what is the cisco thing trying to register _as_? |
| 03:28.47 | jarvis141 | SIP debug http://paste.misacek.net/166/ |
| 03:29.05 | jarvis141 | ChannelZ: How do you mean? |
| 03:30.15 | ChannelZ | The username/password it's trying to register with |
| 03:30.37 | *** join/#asterisk sourcode (~code@ppp-58-8-118-233.revip2.asianet.co.th) |
| 03:30.55 | ChannelZ | you have no secrets in your sip.conf and 3 different dynamic peers, so Asterisk can't figure out who is trying to register. |
| 03:32.19 | ChannelZ | it's trying to reg as 5555202 but there is no peer with that name |
| 03:33.19 | jarvis141 | so the authentication is required, or optional only? I haven't set any pass/secret in CME, too |
| 03:34.57 | ChannelZ | it doesn't necessarily have to no but the bigger problem is that Asterisk can't figure out what peer in sip.conf to match because 1. the source address doesn't exist as a sip peer (the stuff [201], [202], etc in sip.conf) and 2. None of the peers have IP addresses so it can't figure it out by that either. |
| 03:35.43 | ChannelZ | which is what it's telling you, "no matching peer found" |
| 03:36.23 | jarvis141 | so should I rename the [20x] to [555520x] and add the IP? |
| 03:36.44 | ChannelZ | You could do one or the other, it depends on the rest of your config. |
| 03:37.06 | ChannelZ | Or change the peer name the CME is using (or is that your DID number or something?) |
| 03:38.47 | *** join/#asterisk geneticx (~geneticx@c-65-34-240-65.hsd1.fl.comcast.net) |
| 03:40.25 | *** join/#asterisk sourcode (~code@ppp-58-8-118-233.revip2.asianet.co.th) |
| 03:43.56 | *** join/#asterisk blaines (~blaines@64-130-243-194.sparkplugbb.net) |
| 03:44.27 | *** join/#asterisk boodu (~antoine@175.158.129.128) |
| 03:50.56 | *** join/#asterisk diegomad (mad@186.81.136.28) |
| 03:53.17 | jarvis141 | aaw, finally registrations seem fine ^^ what does it mean "unmonitored" status in sip show peers? |
| 03:53.18 | *** join/#asterisk Brookss (~ae03770d@gateway/web/freenode/x-izeqqbwaesqfbrjb) |
| 03:54.14 | justing | it means you haven't set them to qualify |
| 03:54.37 | justing | http://www.voip-info.org/wiki/view/Asterisk+sip+qualify |
| 03:55.36 | jarvis141 | many thanks :) pretty fine thing to set |
| 03:55.50 | fenrus | nice |
| 03:56.12 | fenrus | i'm thinking about integrating an asterisk with an CME to learn how to set it up :) |
| 03:57.43 | jarvis141 | I have to do it in my bachelor thesis :/ .. it's like..learn Asterisk in 21 days :)))))) |
| 03:59.52 | justing | whats an CME? |
| 03:59.56 | justing | are you talking about callmanager express? |
| 04:00.05 | jarvis141 | Cisco CallManager Express |
| 04:00.22 | jarvis141 | or, newly a CUCM :)) |
| 04:00.45 | justing | oh yeah, asterisk works okay with cucme |
| 04:00.48 | justing | not perfect, but it works |
| 04:01.32 | jaytee | i used a bunch of sneaky tricks to integrate a Nortel Meridian system with Asterisk although I had two separate voicemail systems. Since it was for a phased migration I didn't bother with trying to integrate CallPilot voicemail and Asterisk's voicemail |
| 04:05.07 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 04:05.28 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 04:07.55 | exothermc | can you hint off multiple peers? |
| 04:09.15 | justing | no |
| 04:09.50 | exothermc | so how do I have exten 1234 show busy when any of 4 peers is busy? |
| 04:10.03 | exothermc | multiple hints? |
| 04:11.26 | justing | you can try hinting on a channel group |
| 04:11.28 | justing | not sure if that works or not. |
| 04:11.43 | justing | i don't know if there's another way, sorry |
| 04:14.06 | manxpower | why don't you check extensions.conf.sample and the Wiki. It would not surprise me if you could do something like exten => 668,hint,SIP/1234&SIP/5678 |
| 04:16.09 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 04:16.32 | exothermc | manxpower: been looking, I'm going to try that though. |
| 04:17.16 | jarvis141 | what do you think about this guide? http://www.pasewaldt.com/cme/cme_index.htm |
| 04:19.43 | fenrus | looks nice |
| 04:22.19 | jarvis141 | I think so :) .. anyway, thanks for your help, guys :) I hope it will finally become working |
| 04:24.52 | *** part/#asterisk jarvis141 (~Adium@84.242.101.194) |
| 04:25.30 | exothermc | manxpower: Ya just tied the & and it only catches the first peer |
| 04:28.10 | manxpower | exothermc: check the wiki and search the mailing list archives |
| 04:28.12 | manxpower | ~mailinglist |
| 04:28.13 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 04:28.15 | manxpower | ~answers |
| 04:28.15 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 04:30.34 | *** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net) |
| 04:31.47 | Kobaz | so, this is weird |
| 04:32.00 | Kobaz | i set up a dahdi t1 card as NI2 |
| 04:32.13 | Kobaz | and i have an external pri->sip gateway set up as ni2 |
| 04:32.32 | Kobaz | and callerid doesn't work right... and adtran tells me to switch the gateway to nortel dms 100 |
| 04:32.39 | Kobaz | and then callerid works again |
| 04:33.08 | Kobaz | so... either the asterisk pri drivr is implemented wrong, or the adtran firmware is wrong |
| 04:33.11 | manxpower | Kobaz: add a Wait(1) at the start of the dialplan. |
| 04:33.26 | Kobaz | i would point at the asterisk pri driver... since i've had other problems |
| 04:33.28 | manxpower | Kobaz: BTW, callerid name or callerid number? |
| 04:33.32 | Kobaz | callerid name |
| 04:34.02 | Kobaz | if you've ever looked a the libpri code, you would be surprised it works at all |
| 04:34.10 | manxpower | *nod* There are two ways to send that info. As part of the setup message and as a facility message after the call setup. put a wait(1) to wait for the facility message. chances are yo can lower that to something like .25 |
| 04:34.30 | manxpower | it's a pretty common thing to need to do |
| 04:35.18 | Kobaz | there's three ways actually |
| 04:35.39 | Kobaz | in the setup, in display, or in umm, what was it |
| 04:35.46 | Kobaz | oh yeah facility |
| 04:35.55 | Kobaz | adtran supports all three |
| 04:35.59 | manxpower | was not familiar with the display message. |
| 04:36.04 | Kobaz | libpri only does display |
| 04:36.17 | manxpower | are you sure about that? |
| 04:36.30 | manxpower | I thought it supported it as part of the setup |
| 04:36.34 | Kobaz | well |
| 04:36.38 | Kobaz | it can read it as part of the setup |
| 04:36.41 | Kobaz | it only sends it out as display |
| 04:36.58 | manxpower | On the other hand, you ARE the one with PRI problems from hell and it's not because you are a n00b. 8-| |
| 04:37.05 | manxpower | Ah, OK. |
| 04:37.06 | Kobaz | haha |
| 04:37.07 | Kobaz | yeah i know |
| 04:37.11 | Kobaz | libpri crashes left and right |
| 04:37.21 | Kobaz | if you're using ni2 on one end, and att custom on the other |
| 04:37.21 | manxpower | I don't know if I've ever actually had to send CID name. |
| 04:37.26 | Kobaz | i realized this recently |
| 04:38.00 | Kobaz | att custom is similar enough to ni2 that call setup and signalling work though |
| 04:38.07 | manxpower | was that most of your "hell" from the PRI problems? |
| 04:38.11 | Kobaz | but the pri txqueues get corrupted |
| 04:38.12 | Kobaz | yeah |
| 04:38.16 | Kobaz | i think so |
| 04:38.37 | Kobaz | well, i did fix a bug in chan_dahdi |
| 04:38.52 | Kobaz | to not sending progress indications on an already answered channel |
| 04:39.18 | manxpower | I'd love to be able to force OOB indications pre-answer. |
| 04:39.19 | Kobaz | i think libpri is just not spec-compliant |
| 04:39.28 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 04:39.44 | manxpower | XO likes to send inband audio messages and and not hang up the call at seemingly random times. |
| 04:39.45 | florz | is there any channel driver in asterisk that's spec-compliant? =:-) |
| 04:40.03 | manxpower | florz: it there were it wouldn't run with half the gear out there |
| 04:40.05 | Kobaz | the local channel ? |
| 04:40.31 | Kobaz | it seems like ni2 just accepts anything |
| 04:40.49 | Kobaz | the adtrans are picky... if it gets a message it doesnt like, it ignores it |
| 04:41.03 | Kobaz | well, ignores it and ignores the call, if it's a call setup |
| 04:41.03 | florz | manxpower: I doubt that - or at least most of the deviations certainly aren't necessary |
| 04:41.16 | florz | Kobaz: hehe :-) |
| 04:42.04 | florz | well, yeah, I thought that maybe iax would be another candidate - for lack of any real spec ;-) |
| 04:42.19 | Kobaz | manxpower: i've had other issues with dahdi though besids this signalling shenanagins... i've have kernel panics with dahdi |
| 04:42.26 | manxpower | florz: Um. IAX is an RFC. |
| 04:42.51 | florz | manxpower: hu? I very much doubt that ... but would you mind telling me the number? |
| 04:43.12 | Kobaz | i think i believe manxpower, he wouldn't make that up |
| 04:43.14 | manxpower | RFC 5456 |
| 04:43.35 | Kobaz | manxpower: hey wait... that's the carrier pidgon control protocol! |
| 04:43.42 | manxpower | well THAT was a useless URL |
| 04:44.20 | manxpower | http://tools.ietf.org/html/rfc5456 |
| 04:44.22 | manxpower | that is better |
| 04:44.47 | manxpower | February 2010 |
| 04:45.02 | manxpower | if you were on the asterisk mailing lists...... |
| 04:45.43 | Kobaz | come on kernel |
| 04:45.46 | Kobaz | compile faster |
| 04:45.50 | Kobaz | i need to take up at 5 |
| 04:45.52 | Kobaz | wake |
| 04:47.20 | manxpower | what TZ are you in, Kobaz? |
| 04:47.21 | florz | I didn't even know they published stuff as RFCs that they at the same time label as "this is not gonna be a standard, ever, be cautious with using it" ;-) |
| 04:47.35 | Kobaz | est |
| 04:47.42 | manxpower | florz: I noticed that too. In any case there is an official spec for it now. |
| 04:48.15 | florz | yeah, and at least looks more comprehensive than the "spec" I read some years ago |
| 04:48.22 | manxpower | even if it's just the official spec of an unofficial protocol |
| 04:48.42 | manxpower | I've not used IAX in at least 5 years |
| 04:49.09 | manxpower | I was burned by it one too many times back in 1.0/1.2 |
| 04:49.27 | Kobaz | yeah, i stopped using iax once i figured out how to get sip going nicely with nats |
| 04:49.36 | Kobaz | iax is laggy with call teardown |
| 04:49.36 | manxpower | Kobaz: not hard to do at all |
| 04:49.42 | Kobaz | that's the thing i dislike the most |
| 04:49.51 | Kobaz | manxpower: yeah it's not hard, i just never played with it |
| 04:50.01 | manxpower | Kobaz: we had random audio dropouts with IAX2 |
| 04:50.12 | manxpower | if everything is SIP, you can, in theory use reinvites |
| 04:50.17 | Kobaz | yeah |
| 04:50.29 | Kobaz | i've never had any real problems with iax |
| 04:50.33 | Kobaz | other than the silly delays with teardown |
| 04:50.46 | Kobaz | you hang up on one side... and the phone keeps ringing for like 20 seconds on the other |
| 04:50.52 | manxpower | 1.4 had massive improvements to IAX2 |
| 04:51.15 | ChannelZ | hmm yeah I've never had that |
| 04:52.25 | Kobaz | i gotta drive from albany to jersey tomorrow |
| 04:52.30 | Kobaz | and then from jersey to nyc |
| 04:52.43 | Kobaz | and then nyc to staten island, and then i get to take a breather for a few days |
| 04:52.48 | manxpower | I have to drive from Huntsville to Steele, AL (2hr drive, leave at 5am) |
| 04:52.50 | Kobaz | and then head back to pa on monday |
| 04:53.00 | manxpower | Kobaz: the company I work for is on staten island |
| 04:53.02 | *** join/#asterisk Kyosh (whoa@96.246.232.222) |
| 04:53.04 | Kobaz | ah |
| 04:53.08 | Kobaz | i thought you worked in georgia |
| 04:53.11 | manxpower | ~manxpower |
| 04:53.12 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 04:53.23 | manxpower | I live in Huntsville, telecommute |
| 04:53.27 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 04:53.35 | Kobaz | ah |
| 04:53.41 | Kobaz | i used to live on staten island |
| 04:53.45 | manxpower | have a cabin on a mountain that I can also telecommute |
| 04:53.48 | Kobaz | they don't call it crappy apple island for nothing |
| 04:54.23 | manxpower | Kobaz: tomorrow we are provisioning something like 150 phones |
| 04:54.31 | manxpower | one location. |
| 04:54.38 | Kobaz | sounds fun |
| 04:54.42 | Kobaz | i get to set up 3 tomorrow |
| 04:55.00 | manxpower | Kobaz: I'm on the software side/server side. I'm not the running around unboxing and plugging in phones |
| 04:55.00 | Kobaz | and a new server at another spot with 4 t1s |
| 04:55.04 | Kobaz | hehe |
| 04:55.11 | manxpower | we have PFYs for that |
| 04:55.13 | Kobaz | well, my company has four people |
| 04:55.57 | Kobaz | me, a sales/marketing/semicoder guy, a db/interface programmer, and another gui programmer |
| 04:56.10 | Kobaz | so it's either me, or the sales guy plugging in phones |
| 04:56.16 | manxpower | you familiar with what a Tag and Locate is? |
| 04:56.28 | ChannelZ | Geese, right? |
| 04:56.35 | Kobaz | Tags florz |
| 04:56.38 | Kobaz | your it |
| 04:56.48 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 04:56.54 | manxpower | telco comes in and tags a new line with the number on the 66-block |
| 04:56.59 | Kobaz | yeah |
| 04:57.08 | Kobaz | mm, never knew there was a name for it |
| 04:57.11 | Kobaz | makes sense |
| 04:57.13 | manxpower | we order it, ILEC installs it, send them back to T&L |
| 04:57.51 | Kyosh | I have asterisk 1.4.21 set up to operate in real time with mysql. I am trying to use the call-limit variable by setting the value to 2, but it is not working. it allows as many calls as possible with no regard to the call-limit variable. is there a known issue with this version? is there a way i can script this in the database? thanks. |
| 04:58.02 | manxpower | we had set up T&L for a line at a customer 4 times, each time they said they tagged and located it. We finally made them come out again and while our tech was there, the ILEC tech could not find the line. turns out they never even patched it up to the customer's suite. |
| 04:58.13 | manxpower | crap like that really pisses me off. |
| 04:58.39 | Kobaz | haha |
| 04:58.41 | Kobaz | dude |
| 04:58.53 | manxpower | customer has been without fax for days |
| 04:59.01 | Kobaz | i've had installations through a VAR to an end customer |
| 04:59.13 | Kobaz | we tell the var... we need a t1, one power outlet, one network jack |
| 04:59.29 | Kobaz | var says, sure... tells us the signalling on the t1 and everything |
| 04:59.46 | Kobaz | we get onsite... no t1... there's 14 fxs lines, no power outlets, and no network jack |
| 05:00.11 | manxpower | Ah! We once ordered a frame relay connection and the telco handed us off an ethernet port. |
| 05:00.24 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-bqtknbqtzgjlwyoj) |
| 05:00.58 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 05:01.13 | Kobaz | hehe |
| 05:01.23 | Micc | is there a way in extensions.conf to check if a macro or extension exists? |
| 05:01.43 | Micc | I thought I saw an application for checking if an extension existed in 1.6 |
| 05:01.45 | manxpower | Micc: 1.6.??, I think so. 1.4 I don't think so |
| 05:02.29 | manxpower | 'night everyone |
| 05:03.10 | Kobaz | Micc: EXISTS |
| 05:03.12 | Kobaz | nifht |
| 05:03.13 | Kobaz | night |
| 05:04.55 | Micc | oh, it must be a function |
| 05:05.01 | Micc | I didn't see that in core show applications |
| 05:05.40 | Kobaz | it's core show functions |
| 05:05.43 | Kobaz | it's a function |
| 05:05.57 | p3nguin | I'd rather just run dialplan show and see if it's there. |
| 05:09.44 | Kyosh | Is ther a known issue with Call-Limit running in Asterisk Real-Time ? |
| 05:09.50 | Kobaz | probably |
| 05:09.58 | Kobaz | realtime is kind of picky |
| 05:10.09 | Kyosh | hmm |
| 05:10.11 | Kobaz | if you want stuff stored in a db... use static realtime via extconfig |
| 05:10.21 | Kyosh | thats what i am doing |
| 05:10.27 | Kobaz | then it should work fine |
| 05:10.36 | Kyosh | sadly its not |
| 05:10.47 | Kobaz | select from your table and make sure it's outputting the data right |
| 05:10.52 | Kobaz | do sip show peer <foo> |
| 05:10.55 | Micc | do I have to do ${VALID_EXTEN(context,exten)} or can I do it without the ${} ? |
| 05:11.01 | Kobaz | you'll see the call limit settings if they got set right |
| 05:11.34 | Kobaz | or maybe not |
| 05:11.40 | Kobaz | hmm i guess call limit isnt in the peer output |
| 05:11.42 | Kyosh | ok well, its not static cause sip show peer <anything> shows nothing |
| 05:12.18 | Kobaz | run your query directly in the db |
| 05:12.22 | Kobaz | make sure itns returning the right data |
| 05:12.32 | Kyosh | run what query? |
| 05:13.11 | Kobaz | the query you're using for extconfig |
| 05:13.27 | Kyosh | but i am not using a query in extconfig |
| 05:13.29 | Kobaz | select from the table/view whatever |
| 05:13.33 | Kobaz | and make sure it's right |
| 05:13.38 | Kyosh | sipusers => mysql,dbast,sipfriends |
| 05:13.39 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 05:13.43 | Kobaz | oh |
| 05:13.47 | Kobaz | don't do it that way |
| 05:13.58 | Kyosh | thats the only way that works |
| 05:14.01 | Kobaz | [settings] |
| 05:14.12 | Kobaz | sip.conf => odbc,pbx,sip_conf |
| 05:14.15 | Kobaz | or whatever |
| 05:14.22 | Kyosh | right |
| 05:14.33 | Kobaz | use sip.conf... not sipusers/sippeers |
| 05:14.34 | Kyosh | essentially the same thing i have, but no queries are defined |
| 05:14.51 | Kobaz | no, it's not essentially the same thing |
| 05:14.56 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 05:15.02 | Kobaz | sip.conf behaves differently than sipusers |
| 05:15.57 | Kyosh | so if i change the sipusers => mysql,dbast,sipfriends and sippeers => mysql,dbast,sipfriends to sip.conf => mysql,dbast,sipfriends it will understand all the parameters in the db? |
| 05:16.12 | Kobaz | probably not |
| 05:16.23 | Kobaz | you need to follow the syntax for defining config files in the db |
| 05:16.30 | Kobaz | it's in the wiki |
| 05:16.42 | Kyosh | what wiki? |
| 05:16.47 | Kobaz | voip-wiki |
| 05:16.53 | Kyosh | already did, thats how i got this |
| 05:17.03 | Kyosh | you mean voip-info wiki right? |
| 05:17.05 | Kobaz | er |
| 05:17.06 | Kobaz | yeah |
| 05:17.09 | Kyosh | yup |
| 05:17.09 | Kobaz | one of those |
| 05:17.19 | Kobaz | it's late... i wanna goto bed... but i'm still working on some stuff |
| 05:17.20 | Kobaz | heh |
| 05:17.56 | Kyosh | k thanks for the heads up |
| 05:18.23 | Kobaz | np |
| 05:19.56 | Kobaz | nap time |
| 05:36.01 | *** join/#asterisk infobot (ibot@rikers.org) |
| 05:36.01 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 05:36.05 | Micc | I have pandentic=yes in sip.conf |
| 05:37.43 | Micc | asterisk has been running for weeks. |
| 05:41.13 | justing | with aastra phones? |
| 05:45.16 | Micc | justing, yes. 6730 and 31i's |
| 05:45.43 | Micc | firmware 2.5.2.1010 |
| 05:47.41 | justing | have you always had pendantic=yes in the cnof? |
| 05:47.42 | justing | conf* |
| 05:49.21 | Micc | no, I added that a few weeks back when I first saw this eror. |
| 05:49.45 | justing | interesting |
| 05:51.28 | *** join/#asterisk infobot (ibot@rikers.org) |
| 05:51.28 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 05:53.39 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 06:04.31 | *** join/#asterisk ibrahim-laptop0 (~ibrahim@195.175.50.222) |
| 06:04.52 | ibrahim-laptop0 | Hello, |
| 06:05.03 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:06.25 | ibrahim-laptop0 | Is there a way to set minimum length of DTMF duration? For example, ignore "0 ms duration" of DTMF? |
| 06:11.05 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 06:13.58 | ChannelZ | Well there is relaxdtmf |
| 06:14.40 | ChannelZ | other than that I imagine setting a specific duration would be hacking the source |
| 06:18.26 | ChannelZ | however it looks like, in 1.6.1 anyway, min is 80ms.. this would be for inband dtmf I guess |
| 06:19.30 | ibrahim-laptop0 | yes I have found a setting on source code and changed it, but it is related minimum duration of dtmf and If dtmf duration is less than that duration asterisk generating regular dtmf itself |
| 06:21.18 | ibrahim-laptop0 | I need to ignore "0 ms duration" of DTMFs |
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| 06:40.40 | ChannelZ | hmm my system can't seem to detect much below 50ms |
| 06:41.42 | *** join/#asterisk J4zen (~j4zen@95-36-106-201.dsl.alice.nl) |
| 06:41.46 | J4zen | Goodmorning guys |
| 06:41.53 | ChannelZ | Cheerios |
| 06:42.56 | J4zen | I have a rather challenging scenario, im curious if anyone has some bright ideas to solve it. |
| 06:43.33 | J4zen | I'm asked to interconnect 40 offices worldwide via Asterisk, each office has an internet connection and has 4~6 analogue telephone lines which they are currently using. When the client makes a call from any of his offices to another office, he wants the call to go out via internet. Other calls go via the analogue lines (ISDN) on that location as they normally do. I would apreciate some ideas on how to handle this in the most cost-efficient way wh |
| 06:43.57 | J4zen | The logical solution would be to place one asterisk machine in each office with a 4 port BRI interface, when calling locally calls go out via BRI. If another office is called, it goes out via internet. Problem is that this is quite an expensive solution since you're running 40 asterisk machines where maybe 5 could normally handle this kind of traffic. |
| 06:44.27 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 06:44.28 | J4zen | Another scenario (which scares me) is to setup one centrale asterisk machine which is used by all 40 offices, then use ATA's or some kind of remote BRI card(is there such a thing?). This way i would only need the remote interface at each location rather then an asterisk box. What scares me is stability. |
| 06:44.38 | J4zen | How would you guys approach this situation, im very curious. |
| 06:46.12 | Belgarath | J4zen: you could put satelite asterisk in remote place |
| 06:46.31 | florz | the latter probably wouldn't be able to make use of any local telephone lines |
| 06:46.52 | J4zen | Belgarath: Satelite asterisk? |
| 06:47.06 | Belgarath | J4zen: asterisk in each location and one centerl one |
| 06:47.34 | Belgarath | local calls go through local asterisk and if it goes to another branch it goes through central one |
| 06:47.45 | Belgarath | if you put only one central box |
| 06:47.58 | Belgarath | any communication disruption brings whole company down |
| 06:48.28 | ChannelZ | I would think with * in each location in order to drive the local BRIs, you'd just route your "internal" calls from point-to-point as necessary. |
| 06:48.51 | ChannelZ | Each setup would therefore be independent of one another so if one dies, it doesn't affect anyone else |
| 06:49.03 | Belgarath | exactly |
| 06:49.23 | ChannelZ | So you contradicted yourself by suggesting "a central one" |
| 06:49.27 | J4zen | That's true, but you could create a fail-over for the centrale asterisk machine in stead for stability |
| 06:49.48 | J4zen | I agree though, a * box at each location would be the most logical solution, but also the most expensive one |
| 06:50.16 | ChannelZ | Yes but short of offloading the BRI to an ITSP, you've got to have some way to connect to the phone system in each location and that's where most of the expense is |
| 06:50.26 | florz | essentially it all hinges on whether you can switch to VoIP for your PSTN connection in all/most locations |
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| 06:51.21 | J4zen | that really depends, the 40 offices are actually more than 40 office. It's 40 countries with multiple offices |
| 06:51.52 | ChannelZ | And you need an analog card at each place to run the handsets, another expense (but still probably cheaper than replacing all the phones?) |
| 06:51.57 | J4zen | since VoIP isn't as widely spread as i'd like it to be, and we don't want to partner up with local trunk providers, we'd probably have to terminate the PSTN ourselves at their office |
| 06:51.58 | florz | if you can, it could be reasonable to set up some "central" server with failover or live redundancy, possibly distributed across the globe |
| 06:52.52 | florz | if you can not, there isn't really much you can do about the cost of interfacing to the PSTN in each location, or at least in each country |
| 06:54.14 | J4zen | That's what i was afraid of, theyre looking for a cost-saving solution attained by placing inter-office calls via asterisk. |
| 06:55.12 | J4zen | I suppose a local asterisk box with a BRI interface is the only way to go |
| 06:55.14 | ChannelZ | Probably better off using softphones on computers for that :) |
| 06:55.43 | florz | J4zen: you could switch to VoIP phones, of course, then you only need BRI to the PSTN side |
| 06:55.46 | ChannelZ | Because it won't be cheap putting a computer, BRI interface, and ATA at each location |
| 06:55.52 | ChannelZ | Cheap is of course always a relative term |
| 06:56.04 | J4zen | it's not a headset-type of company really, its industrial, so softphones probably wouldn't fit their needs. |
| 06:56.24 | ChannelZ | How many phones, max, at any given location? |
| 06:56.33 | *** join/#asterisk sulex (~sulex@95.236.116.143) |
| 06:56.41 | J4zen | i'd say about 15 to 20 max |
| 06:57.48 | *** join/#asterisk sulex (~sulex@95.236.116.143) |
| 06:58.04 | ChannelZ | ok so that's a ~$1000 card just for the ATAs |
| 06:58.05 | J4zen | small * server: 300$, BRI interface 4 port: +/- 500$, no clue about ATA interface cards for analogue phones |
| 06:58.16 | J4zen | i see |
| 06:58.24 | J4zen | so about 2k usd per location |
| 06:58.57 | florz | J4zen: then it may actually be a reasonable solution to switch to VoIP phones (probably offer more functionality and will have to be bought at some point anyhow?) and use some cheap BRI interface for the 1 to 2 BRI lines per location |
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| 07:00.01 | J4zen | it all comes down to the number of inter-office calls they make on a yearly basis i suppose, i currently have no idea how long the ROI on such an implementation would be |
| 07:00.03 | florz | J4zen: maybe you actually can limit yourself to just one line per location, and route all outgoing calls through the internet? |
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| 07:00.04 | ChannelZ | The cost for buying all new VoIP phones is probably similar, but more flexible in the long run |
| 07:00.46 | florz | J4zen: (with BRI as a fallback) |
| 07:01.04 | ChannelZ | But then there's infrastructure cost of making sure there's ethernet run everywhere... |
| 07:01.22 | florz | J4zen: but, yeah, it obviously makes sense only if they make some reasonable amount of inter-office calls :-) |
| 07:01.46 | J4zen | ChannelZ: indeed, plus a ton of extra man-hours implementing the thing |
| 07:01.59 | ChannelZ | well there will be a lot of that regardless :) |
| 07:02.34 | J4zen | well hooking up the current analogue phones to an ATA interface is a lot less expensive then having to redo their ethernet infrastructure :P |
| 07:03.01 | florz | sure that that will be true in the long run? |
| 07:03.04 | ChannelZ | Maybe. You get to deal with lots of configuration, tuning the lines, echo... |
| 07:03.19 | J4zen | There is that, yes. |
| 07:03.55 | J4zen | Personally i would greatly prefer switching to VoIP and have them see it as a long-term investment which is bound to happen in the near future anywya |
| 07:03.58 | J4zen | anyway* |
| 07:04.05 | J4zen | VoIP phones that is. |
| 07:04.57 | ChannelZ | I don't see a "cheap" fix to do what they want to do.. but would an investment now pay off for them in the long run? As you say it really depends on how much money is being spent making long distance calls between offices now |
| 07:05.44 | J4zen | Another thing came to mind, if we do keep analogue phones rather than SIP phones. It would become absolutely unmanageable, we don't have an office in each and every country and traveling expensive would go through the roof. |
| 07:06.08 | J4zen | where SIP phones could easely be managed remotely |
| 07:06.30 | florz | well, with analog phones there simply is nothing to be managed? =:-) |
| 07:06.53 | J4zen | well, they do break eventually at which point they'll call us =P |
| 07:07.19 | florz | but you can not fix physical defects of SIP phones remotely, either ;-) |
| 07:07.27 | J4zen | hush |
| 07:07.29 | J4zen | ;) |
| 07:07.34 | ChannelZ | subcontract :) |
| 07:07.59 | J4zen | yeah, luckely they do have their own IT-staff in each country so they can take quite a bit of management of our shoulders |
| 07:10.11 | J4zen | well, i apreciate the input ChannelZ,florz and Belgarath. I'll think about it and see what we can come up with. if you happen to have any further suggestions, feel free to PM me :) i'd apreciate it |
| 07:10.37 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:10.38 | florz | obviously, it all depends on the money that could be saved - but I think that if anything a solution that essentially switches completely to VoIP (phones and ITSPs) would be the best long-term solution for the money, plus some interfacing with the PSTN for incoming numbers and in places where good ITSPs are not available, at least for now |
| 07:11.33 | ChannelZ | Good luck |
| 07:11.34 | J4zen | since it's globally i think we'd really need to look at it office-by-office, because the situations/bandwith/acceptance of VoIP are massivly different world-wide |
| 07:11.50 | J4zen | thanks ChannelZ |
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| 07:11.57 | florz | if you do that, you might even be able to additionally save some bucks for BRI lines and stuff |
| 07:12.28 | J4zen | yeah, maybe have some ITSP's partnered up if they are well-known/organised. |
| 07:13.13 | florz | J4zen: well, for outbound it' relatively easy - after all, you can in principle use US ITSPs for calling french numbers from germany ... |
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| 07:13.58 | J4zen | yeah, inbound is the tricky part. also we don't want to increase their local cost-per-call |
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| 07:14.32 | ChannelZ | well lots of ITSPs have termination in many different places |
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| 07:15.02 | ChannelZ | it's not like every office needs their own account through a different ITSP local to them. |
| 07:15.02 | J4zen | can you name one you would recommend? |
| 07:15.15 | florz | J4zen: yeah, but potentially reducing the number of lines may save you more than the additional expenses for international calls to their local neighbourhood ;-) |
| 07:15.19 | ChannelZ | Not really, since I'm guessing this is mostly european? |
| 07:15.26 | J4zen | no globally really |
| 07:15.43 | J4zen | europe, USA, china, russia, asia and more |
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| 07:17.01 | florz | J4zen: and after all, calling most EU countries, say, from a German ITSP, is (a) potentially cheaper than the local ex-monopoly and (b) not necessarily more expensive than using an ITSP in the respective destination country |
| 07:17.54 | J4zen | That's true, which is why it's probably best to really check office-by-office or eat least country-by-country |
| 07:18.05 | florz | yeah |
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| 07:19.21 | ChannelZ | Yeah the US is sort of easy, there are several ITSPs that operate in dang near every state |
| 07:19.53 | florz | but the point is that you don't necessarily need 40 ITSPs for that - and it may even be cheaper to make a contract with some ITSP for all the minutes you terminate from some group of countries |
| 07:20.26 | florz | that way possibly getting cheaper rates than when you buy from the local telcos individually |
| 07:20.57 | florz | erm, s/terminate/originate/, rather |
| 07:22.29 | J4zen | i suppose it'll be a tedious task to figure out which ITPS to contract |
| 07:23.13 | florz | well, depends |
| 07:23.29 | florz | on the one hand you can ask people for recommendations |
| 07:24.02 | florz | on the other hand, for outbound it's relatively easy to switch around a bit and test them out |
| 07:25.26 | J4zen | How about outbound caller ID's, can operators send out an Outbound ID of a inbound line not registered to them? |
| 07:25.37 | J4zen | well, ofcourse they 'can'. But are they allowed to? |
| 07:25.56 | florz | technically, yes, whether they are allowed depends on the local jurisdiction, obviously ;-) |
| 07:26.23 | florz | but I guess if you can prove that those numbers are actually yours, it should be possible in many places |
| 07:26.50 | J4zen | yes that would make sence, but they can be stupidly stubborn |
| 07:27.12 | florz | well, that depends on the number of minutes you want to buy, I guess ;-) |
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| 07:27.35 | J4zen | heh |
| 07:30.39 | florz | (and for at least two ITSPs in .de I know that they at least did offer that in the past as a regular part of the product ;-) |
| 07:30.44 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.226.94) |
| 07:31.21 | J4zen | i'll keep that in mind, they do have offices in germany |
| 07:35.35 | florz | who would have thought that ;-) |
| 07:35.41 | J4zen | while we're on the topic, they do have a lot of offices in Europe. Finding a suitable european provider would be a good starting point. What can you recommend? |
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| 07:45.43 | florz | well, given that I am somewhat involved with one, I won't say too much on that topic - but I guess it's safe to say that sipgate is the most prominent one in .de, as maybe a starting point |
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| 07:47.52 | florz | the http://www.ip-phone-forum.de/ probably has quite a bit of information on the topic, too, in case you are able to decipher it ;-) |
| 07:52.10 | J4zen | thanks, i'm dutch myself so i should be able to decipher it ;) |
| 07:56.02 | florz | that may make it close to easy ;-) |
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| 08:26.22 | carrar | w00t |
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| 08:27.45 | shamelessn00b | HURR DURR |
| 08:27.54 | shamelessn00b | Finally resolved the issue |
| 08:28.00 | shamelessn00b | of channel states |
| 08:28.02 | shamelessn00b | !!!!!!!!! |
| 08:28.20 | shamelessn00b | problem was being caused by asterisk DB and application logic |
| 08:29.42 | c0rnoTa | issue num plz |
| 08:36.07 | shamelessn00b | I dont post on the lists |
| 08:36.13 | shamelessn00b | just discuss it over here |
| 08:36.19 | shamelessn00b | on the channel |
| 08:36.35 | shamelessn00b | issue was that I was getting channel state up and down at the same time |
| 08:36.49 | shamelessn00b | on my asterisk box |
| 08:37.09 | shamelessn00b | cdrs were getting all messy etc, and asterisk kept on crashing after a while |
| 08:37.16 | shamelessn00b | in peak hours |
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| 08:45.40 | carrar | Whats the problem with that! |
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| 09:08.38 | fiddur | Hi. I have a problem with AGI command "STREAM FILE". I send: STREAM FILE tt-monkeys "123" the agi debug says it's streaming the file, and the cli says "Playing 'tt-monkeys' (escape_digits=0123) (sample_offset 0)", but there is no sound on the phone. Using EXEC PLAYBACK... with the same file works well. This is from trunk 2010-02-08 (r245385). |
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| 09:19.23 | fiddur | doh... seems playback automatically answers the channel, which I missed on agi. |
| 09:25.51 | *** join/#asterisk sujith_h (~sujith@219.64.78.55) |
| 09:27.43 | sujith_h | Hi all, I am using asterisk 1.4. I am able to register the 2 extensions using sip softphones. But I am not able to call. Hence I had a look at the log file. It says "NOTICE[9983] manager.c: 127.0.0.1 failed to authenticate as 'admin'" |
| 09:27.49 | sujith_h | can someone help me |
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| 09:34.28 | J4zen | you've changed your password in manager.conf? |
| 09:38.59 | fiddur | The softphones shouldn't be trying to login as managers... |
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| 09:44.59 | Polysics | hello |
| 09:45.07 | Polysics | does NoOp generate an AMI event, please? |
| 09:50.48 | fiddur | I hope not :P |
| 09:53.20 | Polysics | hmm, i wonder how i can fire something from AGI taht AMI picks up |
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| 09:53.33 | Polysics | fiddur, and you're right, i sort of brain farted there :-) |
| 09:53.58 | sujith_h | fiddur: actually I have added password in manager.conf |
| 09:54.32 | fiddur | Polysics: Tried "UserEvent" application? |
| 09:54.39 | sujith_h | J4zen: There wasn't password mentioned in manager.conf, I added |
| 09:55.17 | fiddur | sujith_h: If your problem is that sip-clients can't register, to be able to dial and receive calls, manager.conf has nothing to do with it... |
| 09:58.36 | sujith_h | fiddur: I am using ekiga. For example I have extensions 2222 and 1111 for test. Ekiga shows 2 extensions as registered. When I execute "sip show users" in the CLI, it shows me the 2 users. But the problem is when I dial, then no response comes :( |
| 09:59.05 | sujith_h | So I were wondering is this an issue with the message that I saw in the log file: "NOTICE[9983] manager.c: 127.0.0.1 failed to authenticate as 'admin'" |
| 10:00.50 | Belgarath | no |
| 10:01.07 | Belgarath | run asterisk -vvvvr and try again |
| 10:01.13 | Belgarath | you shoudl see more debug |
| 10:01.26 | Belgarath | when making a call |
| 10:01.43 | fiddur | sujith_h: I understand that you wonder, but the manager-message must be from somewhere else. |
| 10:04.46 | sujith_h | fiddur: ok |
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| 10:07.32 | sujith_h | Belgarath: http://pastebin.com/JRfYP9r8 This the message that I got when I tried to call |
| 10:08.54 | Belgarath | looking |
| 10:09.26 | Belgarath | seems you get a busy signal |
| 10:09.41 | Belgarath | do you have one ekiga with two acounts |
| 10:09.49 | Belgarath | and you try to call one from the other ?> |
| 10:10.00 | Belgarath | or is it two different ekigas ? |
| 10:10.13 | sujith_h | Belgarath: it is 2 different ekigas |
| 10:10.38 | Belgarath | SIP/haynes-081635d0 is circuit-busy |
| 10:10.38 | sujith_h | Belgarath: And I am trying to call from one ekiga to other |
| 10:10.52 | Belgarath | is hynes the one you are calling from or to ? |
| 10:11.28 | Belgarath | you seem to get constant reload or sth by the look of the log anyway ... |
| 10:12.10 | Belgarath | can you call asterisk echo or voicemail form both phones ? |
| 10:12.18 | sujith_h | Belgarath: I am cally haynes |
| 10:12.27 | sujith_h | s/cally/calling |
| 10:13.11 | Belgarath | can you give me oiutput of sip show peers ? |
| 10:13.19 | sujith_h | Belgarath: sure :) |
| 10:14.03 | sujith_h | Belgarath: |
| 10:14.05 | sujith_h | Belgarath: http://pastebin.com/dAkBKZxq |
| 10:14.43 | Belgarath | can you call voicemail form both ? |
| 10:15.40 | sujith_h | Belgarath: Need little time to figure out voicemail in ekiga |
| 10:16.01 | Belgarath | call voicemail number |
| 10:16.12 | Belgarath | i am not talking about ekiga voicemail just asteriks one |
| 10:16.25 | Belgarath | *97 is the default in most guis |
| 10:28.29 | sujith_h | Belgarath: My extensions_additional.conf says http://pastebin.com/wyBrimS0 |
| 10:31.00 | Dovid | how do I reload asterisk on 1.6.2.6 ? |
| 10:33.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:33.46 | Belgarath | Dovid: config reload name_of config |
| 10:33.54 | Belgarath | od core restart now |
| 10:34.07 | Belgarath | there might be a better way |
| 10:34.08 | Belgarath | :) |
| 10:34.33 | Belgarath | sujith_h: well |
| 10:34.42 | Belgarath | you do not have direct routes for this numbers |
| 10:34.50 | Belgarath | it goes directly to voicemail |
| 10:35.09 | Belgarath | remove exten 1111 and 2222 |
| 10:35.13 | Belgarath | and try again |
| 10:36.15 | sujith_h | Belgarath: removing the last 2 lines of numbers from the extensions_additional.conf right? |
| 10:40.35 | sujith_h | Belgarath: When I removed the exten 1111 and 2222, how will I test the call? |
| 10:44.42 | Belgarath | they would still work |
| 10:44.49 | Belgarath | they do not have to be in extensions.conf |
| 10:45.02 | tzafrir | Chainsaw, hi |
| 10:45.19 | tzafrir | Dovid, officially: 'module reload' |
| 10:45.31 | Chainsaw | ltns tzafrir |
| 10:45.47 | sujith_h | Belgarath: If i remove the extensions from the extensions_additional.conf, and I dial 1111, which is haynes, the connection doesn't go. It says user not found :( |
| 10:45.48 | tzafrir | However, there should be an alias for 'reload' in the default cli_aliases.conf |
| 10:46.09 | Belgarath | oh yes ... |
| 10:46.17 | Belgarath | show me your users.conf |
| 10:46.42 | Belgarath | give them numbers instear of names and u are set |
| 10:46.54 | sujith_h | Belgarath: sorry |
| 10:47.28 | Belgarath | in [] instead of having haynes put 1111 |
| 10:49.17 | michael-i | I'm having trouble getting flash-hook transfers to work in 1.6.1. Do I only need transfer=yes on my analog extensions or additional parameters (assuming the wink, etc times are ok) |
| 10:54.27 | sujith_h | Belgarath: You mean like this, http://pastebin.com/X0Ek5rzV ? |
| 10:57.16 | tzafrir | michael-i, threewaycalling = yes ;? |
| 10:57.19 | Belgarath | sujith_h: no |
| 10:57.28 | Belgarath | change it in user(S).conf |
| 11:01.29 | sujith_h | Belgarath: I haven't added anything in users.conf |
| 11:01.44 | sujith_h | Belgarath: I think because of that routing issue came up :( |
| 11:02.24 | Belgarath | well you can add them only in sip if you wish but users.conf is better if you may switch to iax one day |
| 11:02.42 | Belgarath | also it allows you to separate users from providers |
| 11:04.19 | sujith_h | Belgarath: I had added it in the users.conf as http://pastebin.com/NJpgc6bC |
| 11:04.27 | sujith_h | Belgarath: I hope this is ok right? |
| 11:05.21 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 11:09.38 | Belgarath | should be ok |
| 11:09.53 | Belgarath | and remove bits form sip conf and extensiosn.conf now |
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| 11:33.48 | sujith_h | Belgarath: removing what part of exetensions,conf and sip.conf? My sip_additional.conf is http://pastebin.com/Bq91ZnJK and the extensions_additional.conf is http://pastebin.com/XnnQx0sM |
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| 11:35.19 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 11:39.23 | Dovid | Mantis seems to be down :( |
| 11:43.23 | Belgarath | sujith_h: anything that refers 1111 or 2222 |
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| 11:45.21 | kaldemar | Dovid: seems up to me |
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| 11:48.38 | sujith_h | Belgarath: when I remove the reference of 1111 and 2222 then I am getting messages like this in the asterisk -vvvvr http://pastebin.com/EXgLgSGz |
| 11:48.52 | sujith_h | Belgarath: That is it is taking me to the voicemail |
| 11:48.56 | *** join/#asterisk sourcode (~code@ppp-58-8-118-233.revip2.asianet.co.th) |
| 11:50.55 | Belgarath | have you reloaded asterisk after adding the users.conf ? |
| 11:51.10 | Belgarath | if yes show me you |
| 11:51.18 | Belgarath | sip show peers again |
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| 11:56.01 | sujith_h | Belgarath: http://pastebin.com/FBdhH6Xv |
| 11:56.37 | Belgarath | yeah |
| 11:56.44 | Belgarath | you didn;t change the entries in ekigas |
| 11:56.47 | Belgarath | did you ? |
| 11:56.52 | Belgarath | to use number as a user |
| 11:57.53 | sujith_h | Belgarath: nope i didn't changed anything in ekiga. Let me do that :( |
| 11:59.35 | sujith_h | Belgarath: In ekiga if I say auth user as name only then it allows me to register. If i give number it says cannot register :( |
| 12:06.16 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:07.29 | Belgarath | have you restart/reload asterisk after changes to users.conf ? |
| 12:07.36 | sujith_h | Belgarath: yah |
| 12:07.49 | Belgarath | show me your users.conf extensions.conf and sip.conf |
| 12:08.14 | sujith_h | sure |
| 12:08.43 | Belgarath | I would be going for lunch now |
| 12:08.49 | Belgarath | and be back in an hour |
| 12:08.53 | sujith_h | Belgarath: ok |
| 12:09.16 | sujith_h | Belgarath: users.conf http://pastebin.com/Tf9gRSgR |
| 12:11.28 | sujith_h | Belgarath: sip_additional.conf http://pastebin.com/YZhVKByN |
| 12:12.35 | sujith_h | Belgarath: extensions_additional.conf http://pastebin.com/1G4yf7kn |
| 12:12.38 | Belgarath | put host=dynamic |
| 12:12.43 | Belgarath | in each section of users.conf |
| 12:12.50 | sujith_h | Belgarath: ok |
| 12:13.09 | Belgarath | remove ahynes and sujith from sip.conf |
| 12:13.33 | Belgarath | remove entries from global in extensions |
| 12:13.48 | Belgarath | and restart |
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| 12:14.55 | sujith_h | Belgarath: so in the sip.conf, I should be having [1111] instead of [haynes] right? |
| 12:15.19 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:15.58 | sujith_h | Belgarath: or you are pointing towards the username section? |
| 12:19.40 | Lyma | Bom dia! Good morning! |
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| 12:31.08 | batphone | can anyone recommend a fairly robust, near real-time, cdr analysis tool for wholesale traffic? |
| 12:31.27 | batphone | id like to monitor ASR and ACD on a per carrier and per direction basis. |
| 12:31.35 | batphone | cause codes, trigger alerts on them, etc. |
| 12:31.47 | batphone | graphs? |
| 12:32.16 | batphone | anyone use ntop for it? found a pdf about it. |
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| 12:51.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:00.38 | spiceycurry | morning |
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| 13:16.06 | FreezeS | is there a way I can find out who hanged up a call ? |
| 13:16.12 | FreezeS | in 1.4 |
| 13:18.03 | kaldemar | in the dialplan? |
| 13:19.50 | FreezeS | yes, that would be ideal |
| 13:19.56 | FreezeS | if not, is there any other way ? |
| 13:20.08 | kaldemar | use option g for app dial |
| 13:20.13 | FreezeS | (except analyzing the traces) |
| 13:21.24 | FreezeS | so, if I use g, how can I know which party hung up ? |
| 13:22.38 | kaldemar | if you use g and the dialplan execution goes to a priority that follows the dial line, you do something to indicate that is was the destination channel that hung up. |
| 13:22.38 | spiceycurry | I am having problems compiling Asterisk 1.6.x on 64 bit Centos.. what would the best resource be to find the most common solutions? |
| 13:22.55 | Chainsaw | spiceycurry: You could stick the compile error on pastebin and give us a link. |
| 13:23.00 | Chainsaw | spiceycurry: In most cases we're able to figure it out. |
| 13:23.22 | spiceycurry | ok, sounds good, I will get the information and post it up. |
| 13:23.24 | spiceycurry | thanks |
| 13:24.53 | FreezeS | kaldemar: so if it jumps to h it means the source channel that hung up ? |
| 13:25.43 | [TK]D-Fender | FreezeS: No, the dialed device hung up |
| 13:25.45 | *** join/#asterisk Peste_Bubonica (~eduardo@189.63.202.100) |
| 13:25.58 | [TK]D-Fender | FreezeS: Sorry.. reverse that |
| 13:26.38 | Peste_Bubonica | Hi all |
| 13:27.03 | FreezeS | so, if the call goes to the next priority => the dialed party hung up, if it goes to h => dialing party hung up |
| 13:27.04 | kaldemar | FreezeS: it will jump to h extension in any case, if it exists. use the priority after Dial to let the h extension know who hung up and put the result in your choice of place in the h extension. |
| 13:27.15 | FreezeS | ahaaa |
| 13:27.54 | FreezeS | thanks :) |
| 13:27.56 | kaldemar | FreezeS: unless i remember that wrong. try it out yourself, you'll see how it works with a little playing. |
| 13:28.08 | FreezeS | I will test |
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| 13:38.55 | spiceycurry | I am having compile errors for Asterisk 1.6, the OS is CentOS-64 v5. Here is a post bin clip: http://pastebin.com/ErEFRMeP |
| 13:39.24 | spiceycurry | Kernal is: 2.6.18-164.15.1.el5 |
| 13:39.45 | spiceycurry | *kernel |
| 13:41.04 | spiceycurry | Here is the paste bin of the compile errors (mostly chan_ooh323.c errors) http://pastebin.com/ErEFRMeP. |
| 13:41.07 | WIMPy | Did you put up the whole source? |
| 13:41.32 | spiceycurry | just above the problems started |
| 13:41.34 | leifmadsen | don't compile chan_ooh323 :) |
| 13:41.54 | leifmadsen | what version of Asterisk? |
| 13:42.00 | leifmadsen | (1.6 is totally ambiguous) |
| 13:42.08 | spiceycurry | is the chan_ooh323 failing because of the 64bit? |
| 13:42.12 | spiceycurry | getting version no0w |
| 13:42.18 | leifmadsen | I have no idea what it's failing |
| 13:42.24 | leifmadsen | s/what/why |
| 13:42.37 | WIMPy | Not Asterisk, Asterisk-Addons |
| 13:42.37 | spiceycurry | asterisk-1.6.2.6 |
| 13:42.48 | spiceycurry | asterisk-addons-1.6.2.1 |
| 13:43.05 | WIMPy | Looks more like no Asterisk found. |
| 13:44.29 | spiceycurry | here is what I did... http://pastebin.com/spNbpDaX |
| 13:44.35 | *** join/#asterisk devoid (yiffstar66@unaffiliated/devemo) |
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| 13:46.11 | Katty | hi |
| 13:46.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 13:46.20 | elzid | hi - eagi question guys - audio stream is provided on fd 3 but what if you have multiple calls on different extensions running eagi simultaneously? Are the audio streams segregated? |
| 13:46.24 | spiceycurry | I tried without ooh323 |
| 13:46.30 | Katty | my asterisk does not work at all how to fix pls |
| 13:47.50 | spiceycurry | Does this look successful, I disabled format323 and ooh323.. http://pastebin.com/ybzv70pS |
| 13:48.01 | elzid | eagi accesses stream at fd 3, but surely that is for the eagi being executed, what if multiple eagis are running? |
| 13:49.05 | WIMPy | elzid: Somehow you answered the question yourself: It's multiple processes. |
| 13:49.46 | WIMPy | spiceycurry: Line 336 says so. |
| 13:50.22 | spiceycurry | ok perfect. Now, I suppose I will start the next step of installing zaptel |
| 13:50.28 | spiceycurry | thanks for the help |
| 13:50.30 | spiceycurry | :) |
| 13:50.33 | elzid | <WIMPy>: but if I wanted to cat /dev/fd/3 via eagi, there is only one fd 3 device right? How can multiple eagi calls read audio streams from one fd 3 device? |
| 13:50.44 | elzid | <WIMPy>: am I missing something fundamental? |
| 13:50.54 | *** join/#asterisk dajhorn (~dajhorn@206.16.96.160) |
| 13:50.56 | WIMPy | Maybe :-) |
| 13:51.05 | WIMPy | FDs are local to processes. |
| 13:51.44 | WIMPy | And if you think about it, it wouldn't work if they weren't. |
| 13:51.44 | *** join/#asterisk smooth_penguin (~smoove@59.95.53.85) |
| 13:51.44 | WIMPy | or couldn't |
| 13:52.24 | elzid | <WIMPy>: of course! I'm sorry for the silly question - thanks! |
| 13:52.53 | spiceycurry | What is chan_ooh32? |
| 13:53.03 | spiceycurry | Is that for the "Fax for Asterisk" product? |
| 13:53.06 | leifmadsen | if you don't know, you don't need it |
| 13:53.08 | leifmadsen | no |
| 13:53.09 | spiceycurry | for the channels? |
| 13:53.13 | leifmadsen | it's for H.323 |
| 13:53.21 | leifmadsen | disable it |
| 13:53.23 | leifmadsen | you don't need it |
| 13:53.42 | spiceycurry | ok, cool. I suppose I will have a problem anyhow when I go to install the channels, as they do not have a 64-bit version as of yet for "Fax for Asterisl |
| 13:53.45 | leifmadsen | faxing stuff is res_fax and res_fax_spandsp |
| 13:53.54 | spiceycurry | ok cool |
| 13:54.13 | spiceycurry | If I plan on using the "Fax for Asterisk" product, I should really be using 32 bit OS anyhow right? |
| 13:54.42 | coppice | why would you want to use fax for asterisk? |
| 13:55.03 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 13:55.18 | spiceycurry | We want to have a pool of 100-200 fax's |
| 13:55.23 | jaytee | hi Katty |
| 13:55.49 | Katty | hi (= |
| 13:55.56 | coppice | spiceycurry: use one of the freely available options, and save :-) |
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| 13:56.18 | Katty | jaytee: let me get some stuff together for you |
| 13:56.25 | spiceycurry | coppice: what options are available? |
| 13:56.29 | jaytee | Katty, cool thanks |
| 13:56.39 | spiceycurry | I think: Hylafax, Asterfax |
| 13:56.55 | coppice | avoid asterfax |
| 13:57.12 | spiceycurry | The idea is to have 5000+ numbers forwarded to a pool of 200 or so pooled fax's |
| 13:57.17 | coppice | asterisk + iaxmodem + hylafax is very robust. good for high volumes |
| 13:57.20 | spiceycurry | (for incoming and outgoing) |
| 13:57.27 | moos3 | hylafax is way easier to get up and going then asterfax |
| 13:57.31 | spiceycurry | ok, I will take a look |
| 13:57.49 | coppice | res_fax_spandsp will do several hundred concurrent channels on a quad core |
| 13:58.12 | coppice | and you don't have to deal with Mr Slimy who is behind AsterFAX |
| 13:58.40 | spiceycurry | haha cool |
| 13:59.23 | spiceycurry | Where could I find a good resource site for the Asterisk/Hyla/IAX ? |
| 14:00.15 | coppice | the iaxmodem project page at sf.net tells you how you set it up. questions to the hylafax mailing list |
| 14:03.39 | spiceycurry | sweet |
| 14:03.44 | spiceycurry | thanks coppice |
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| 14:11.21 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 14:12.46 | shamelessn00b | Hi I have a quick question, I have a context with extensions s,1,2 and 3, I have a Background app call in extension s, and for that Background application call, I dont want asterisk to jump to extension 3 if the user presses 3 on the playback of the prompt |
| 14:13.15 | shamelessn00b | is there a way to restrict the Background application and make it respond to only a specified set of digits |
| 14:13.17 | shamelessn00b | ? |
| 14:14.22 | shamelessn00b | [TK]D-Fender: I resolved the issue of that channel status I was discussing with you some time back :) |
| 14:17.57 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 14:18.13 | rare1980_ | hey all,, i have question regarding chanspy.. |
| 14:19.02 | rare1980_ | when i dial chanspy number 555. i can listen to an IVR.. and IVR just say the last station number which is on call. |
| 14:19.33 | sulex | shamelessn00b: first thing it comes to mind is to have at the first priority of ext 3(extension=>3,1 a gotoif that will make a jump to extension=>3,3(having background at extension=3,2 ) if the issued dtmf is 3 |
| 14:19.55 | rare1980_ | lets say i have 3 agents on call Chanspy IVR only say the last agent ID who is on call. on this example IVR says agent ID 3 |
| 14:19.59 | rare1980_ | always |
| 14:20.02 | [TK]D-Fender | shamelessn00b: No. |
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| 14:20.19 | rare1980_ | how can i listen to any agent to whom i want to listen |
| 14:20.20 | rare1980_ | ? |
| 14:20.23 | rare1980_ | any help plz |
| 14:20.29 | Peste_Bubonica | Hi all... im having a strange issue, with asterisk 1.6.2.x... I have a call in this scenario: sip phone -> local asterisk -> remote asterisk (voicemail, meetme)... Always when I use any application that uses Playbacks, the diaplan stalls. Even If I create a extension, using the Playback command, or SayDigits, SayNumer, the dialplan stalls before some playback statements... |
| 14:20.38 | shamelessn00b | sulex: thanks |
| 14:20.55 | Peste_Bubonica | with the exactly same configturation files, but, with asterisk 1.6.1, all things works really nice.. |
| 14:21.22 | Peste_Bubonica | im using iax between servers, but, connecting the two boxes via SIP, the probelm persists... |
| 14:21.37 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 14:21.50 | rare1980_ | ??? any help on this |
| 14:22.06 | Peste_Bubonica | If I connect a phone directly to the remote server, the problem does no occours... im thinking that this problem occours only on a bridged call.. |
| 14:23.05 | Peste_Bubonica | Ive increased the debug and verbose to 999, but nothing relevant is showed... the dialplan simple stalls, and I need to manual hangup, and try again, and again.. |
| 14:23.33 | Peste_Bubonica | this is a small log report, of two calls, to a extension that say some numbers, and some records...: http://pastebin.com/Xe9RS4t5 |
| 14:23.52 | sulex | rare1980_: give a look at monitor app |
| 14:24.03 | sulex | rare1980_: and play with it |
| 14:24.16 | [TK]D-Fender | rare1980_: "core show application chanspy" |
| 14:24.35 | sulex | oh yeah apologies, thanks [TK]D-Fender |
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| 14:31.13 | rare1980_ | ko thanks |
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| 14:32.26 | wcselby | o/ |
| 14:32.47 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 14:38.11 | *** join/#asterisk tumaix (~tomaz@187.65.196.165) |
| 14:38.52 | tumaix | ppl , I'm getting a cdr_sqlite3_custom.c:229: error: conflicting types for ‘sqlite3_log’ while trying to build asterisk 1.6.1.18 on the Arch Linux with uname -a as Linux tomaz-desktop 2.6.33-ARCH #1 SMP PREEMPT Sun Apr 4 10:27:30 CEST 2010 x86_64 AMD Phenom(tm) II X4 955 Processor AuthenticAMD GNU/Linux |
| 14:39.28 | tumaix | I did a search and see that this is indeed an error on the asterisk 1.6.2.7 , but not in 1.6.1.18 |
| 14:39.35 | tumaix | can anybody confirm that too? |
| 14:40.07 | *** join/#asterisk AndreBrasil (~andre@187.65.196.165) |
| 14:40.58 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:42.08 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 14:42.09 | Polysics | hello |
| 14:42.20 | Polysics | anyone knows why my DIALSTATUS is not returning correctly? |
| 14:42.30 | Polysics | i am using SIP on realtime, qualify=yes is up |
| 14:42.38 | wcselby | Polysics - it doesn't like you |
| 14:42.42 | [TK]D-Fender | Polysics: DETAILS |
| 14:42.58 | wcselby | or, you could show us some logs, cli, debugs, etc |
| 14:43.20 | Polysics | after a Dial in an AGI script, the DIALSTATUS variable is simply empty |
| 14:43.38 | [TK]D-Fender | Polysics: Where's the backup? |
| 14:44.03 | *** join/#asterisk sourcode (~code@ppp-58-8-118-233.revip2.asianet.co.th) |
| 14:44.11 | wcselby | Polysics - we need to see logs so we can see what it's doing |
| 14:44.12 | Polysics | [TK]D-Fender, backup? |
| 14:44.21 | [TK]D-Fender | Polysics: ... |
| 14:44.28 | Polysics | i guess i need to turn aGI debug on and run it again, one second |
| 14:45.11 | tumaix | anybody care about my build problem? |
| 14:45.28 | wcselby | tumaix - i can not confirm it, so I didn't respond to it |
| 14:45.30 | wcselby | sorry |
| 14:45.53 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 14:46.13 | superbeef | does this syntax look valid? PauseQueueMember(|Local/${BVACODE}@from-internal/n) |
| 14:46.24 | tumaix | wcselby: thanks ;) |
| 14:46.34 | [TK]D-Fender | superbeef: What version? |
| 14:46.41 | superbeef | 1.4.30 |
| 14:46.41 | Naikrovek | W H O A |
| 14:46.42 | Naikrovek | http://cgi.ebay.com/J4904A-ProCurve-2848-Layer-3-10-100-1000-Gigabit-Switch-/270559905838?cmd=ViewItem&pt=COMP_EN_Hubs&hash=item3efea0882e |
| 14:46.53 | *** join/#asterisk Kyosh (whoa@96.246.232.222) |
| 14:46.56 | [TK]D-Fender | superbeef: How do I know what's in that var? Where is your ATTEMPT? Considering actually TRYING it at some point? |
| 14:47.42 | [TK]D-Fender | superbeef: what do the instructions say? |
| 14:48.31 | superbeef | [TK]D-Fender: http://pastebin.com/FT9b6spG |
| 14:49.10 | superbeef | It's legacy code that was ported from 1.2 systems.. basically user dials 55** to pause themselves in the queue |
| 14:50.07 | [TK]D-Fender | superbeef: Read the instructions. Try it. You tell us |
| 14:50.58 | superbeef | gotcha |
| 14:51.32 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:54.39 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:55.23 | spiceycurry | How does iaxmodem handle T.38? |
| 14:55.42 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 14:55.45 | DelphiWorld | hi |
| 14:55.49 | DelphiWorld | using dahdi_scan |
| 14:55.54 | DelphiWorld | what alarms=REC mean? |
| 14:56.02 | spiceycurry | Rather, How does spandsp/iaxmodem handle T.38? |
| 14:56.36 | wcselby | spiceycurry - spanDSP answers the fax call and receives the fax in t.38 |
| 14:57.18 | wcselby | spiceycurry - hmmm....I may be confusing technologies actually |
| 14:58.04 | wcselby | i use iaxmodem to bridge fax calls coming in from my t1 over to my hylafax server |
| 14:58.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:58.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:58.52 | wcselby | this client doesn't do t.38 here. the other client i have that does uses fax for asterisk to receive the fax directly into asterisk, using the ReceiveFax() command |
| 14:59.26 | wcselby | so hmmm, sorry I guess I'm typing out my train of thought, when I should be just doing it all in my head. too early still. |
| 15:01.31 | DelphiWorld | leifmadsen: ! |
| 15:03.32 | *** join/#asterisk edge (~edge@Ganon.EdgeServ.Com) |
| 15:04.30 | edge | I'm super interested in learning Asterisk. Are there any good documents or manuals ? |
| 15:04.38 | leifmadsen | ~docs |
| 15:04.39 | infobot | docs is probably for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 15:04.44 | leifmadsen | ~book |
| 15:04.45 | 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 |
| 15:06.03 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:06.04 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:07.51 | superbeef | [TK]D-Fender: fixed it.. for some reason one box used from-queue instead of from internal... anyway I was mostly curious if variable syntax has changed dramatically.. i had been burned on that once before... but I get your point |
| 15:09.27 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 15:09.31 | *** part/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 15:09.46 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:10.27 | *** join/#asterisk MmixX (mmixx@61.14.191.143) |
| 15:11.10 | ariel_ | Morning |
| 15:13.09 | *** join/#asterisk dajhorn_ (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 15:13.51 | Katty | hello my asterisk does not work at all how to fix pls |
| 15:14.26 | ChannelZ | plug in harder ok |
| 15:14.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:14.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:15.05 | Slugs_ | katty, turn on ur electricity |
| 15:15.40 | superbeef | Katty: i like to let my hard drive fill up and watch what happens |
| 15:16.19 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 15:16.23 | DelphiWorld | in pri config |
| 15:16.27 | DelphiWorld | crc is only 4? |
| 15:16.31 | DelphiWorld | no other value? |
| 15:16.44 | Katty | ya'll are funny |
| 15:17.02 | tzafrir | DelphiWorld, "crc4" is the name of an option |
| 15:17.12 | DelphiWorld | tzafrir: and the value? |
| 15:17.22 | tzafrir | if it appears, that value is set |
| 15:17.23 | smooth_penguin | hey Katty |
| 15:17.38 | DelphiWorld | tzafrir: but what is the value? |
| 15:17.59 | DelphiWorld | tzafrir: ah |
| 15:18.05 | DelphiWorld | tzafrir: got you |
| 15:18.18 | tzafrir | if enabled, CRC4 checksum is used to check the data on the line. For E1 you should normally have it enabled |
| 15:18.54 | DelphiWorld | tzafrir: algeria don't use CRC |
| 15:19.07 | tzafrir | so don't set it |
| 15:20.31 | ariel_ | funny all my setup for E1's are without CRC4 |
| 15:22.06 | coppice | an ISDN E1 without CRC4 is dumb, but most non-ISDN E1s don't use CRC4 and this seldom give problems |
| 15:24.15 | keith4 | is a telco asks me "what type of PRI channel" i need for asterisk... what should I tell them? |
| 15:24.17 | *** part/#asterisk tumaix (~tomaz@187.65.196.165) |
| 15:24.50 | keith4 | he mumbled something about "n2", but his accent was thick and it was difficult to understand what he actually wanted |
| 15:25.21 | *** join/#asterisk roe (~roe___@207-172-35-242.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) |
| 15:26.20 | malcolmd | keith4: probably trying to convey "national 2" when he said that. national 2 is fine. |
| 15:26.22 | *** join/#asterisk pentanol (~pentanol@77.35.52.134) |
| 15:27.30 | keith4 | malcolmd: alright, thanks. do you know where i can read about the different options? |
| 15:27.49 | ariel_ | Also make sure that they tell you how your inbound numbers are sent, with the last 4 or 10 digits, I always request them to set them up for 10 digits |
| 15:28.14 | malcolmd | ^ +1 to what ariel said; you want all 10 digits if you can get them |
| 15:28.33 | keith4 | this particular customer has DIDs scattered all over different exchanges, so last 4 is probably not even an option |
| 15:29.00 | keith4 | but that's very good to know, thanks |
| 15:29.40 | Kyosh | I have Asterisk 1.4.21 set up to operate in RealTime with MySQL. I am trying to use the call-limit variable by setting the value to 2, but it is not working. It allows as many calls as possible with no regard to the call-limit variable. If anyone is familiar with this issue, I would appreciate any assistance. Also, is there a way I can script this in the dialplan? Thanks. |
| 15:29.41 | malcolmd | keith4: for t1 your options on the asterisk side are national isdn 1, national isdn 2 (NI2), nortel dms100, at&t 4ess, lucent 5ess. |
| 15:30.29 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 15:30.46 | keith4 | malcolmd: nice. that's enough specific keywords to find more info. thanks |
| 15:30.57 | ariel_ | Kyosh: from testing call-limit only works on type=peer |
| 15:31.30 | Kyosh | i havent had luck with that either |
| 15:31.37 | *** part/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 15:32.25 | *** join/#asterisk TimeRider (~steve@82.132.248.88) |
| 15:37.06 | spiceycurry | Is there any point using DAHDI over Zapatel? |
| 15:37.23 | leifmadsen | it's newer and is maintained? |
| 15:37.41 | spiceycurry | Zap is not maintained any longer? |
| 15:37.45 | leifmadsen | (I originally read that as if you were saying Voice over IP and was like, "wtf... that makes no sense") |
| 15:37.48 | leifmadsen | it is not |
| 15:37.55 | ariel_ | other then I can't really tell how to correctly speak it, it works fine. |
| 15:37.55 | leifmadsen | DAHDI is the replacement |
| 15:38.03 | leifmadsen | DAW-DI |
| 15:38.05 | spiceycurry | ok |
| 15:38.09 | leifmadsen | DAW-DEE rather |
| 15:38.17 | leifmadsen | I refuse to say DADDY |
| 15:38.20 | ariel_ | dady |
| 15:38.21 | spiceycurry | LOL |
| 15:38.26 | ariel_ | daady |
| 15:38.30 | spiceycurry | I was looking for my DADHI earlier |
| 15:38.31 | ariel_ | arhg da ha di |
| 15:38.51 | spiceycurry | where can I find my DADHI? |
| 15:38.59 | wcselby | i've always said DAH-DEE |
| 15:39.01 | leifmadsen | DEE-AH-DI |
| 15:39.30 | leifmadsen | DEE-AW-DEE |
| 15:39.31 | leifmadsen | heh |
| 15:39.36 | leifmadsen | I like DEE-AW-DEE |
| 15:39.38 | ariel_ | you see |
| 15:39.43 | leifmadsen | kinda like audi :) |
| 15:39.58 | spiceycurry | It seems that Zapatel won't even work with 1.6.2... so gonna have to do the dahdi |
| 15:40.09 | wcselby | leifmadsen - yeah that's how I say it |
| 15:40.09 | spiceycurry | as per " Zaptel-to-DAHDI.txt" |
| 15:40.17 | wcselby | audi with a d in front |
| 15:40.19 | spiceycurry | I say "Daddy" |
| 15:40.34 | ariel_ | so who's your Daddy |
| 15:40.38 | spiceycurry | lol |
| 15:40.45 | wcselby | but do you say "eeks" ? |
| 15:41.17 | leifmadsen | yes, I say eeks |
| 15:41.34 | wcselby | i know you do leifmadsen :P |
| 15:41.36 | spiceycurry | You can find out who your daddy is with the command " DAHDIScan " |
| 15:43.48 | spiceycurry | Asterisk died with code 1. |
| 15:43.48 | spiceycurry | Automatically restarting Asterisk. |
| 15:43.48 | spiceycurry | Asterisk ended with exit status 1 |
| 15:44.04 | spiceycurry | I tried to start asterisk /etc/init.d/asterisk start |
| 15:44.11 | spiceycurry | where should I start looking for problems? |
| 15:45.12 | leifmadsen | kill safe_asterisk and run 'asterisk -c' to find out why asterisk is failing on startup |
| 15:45.17 | spiceycurry | ok |
| 15:45.35 | spiceycurry | that was easy |
| 15:45.36 | spiceycurry | No 'modules.conf' found, no modules will be loaded. |
| 15:45.36 | spiceycurry | Unable to open AMI configuration manager.conf, or configuration is invalid. Asterisk management interface (AMI) disabled. |
| 15:45.36 | spiceycurry | Can't find indications config file indications.conf. |
| 15:45.48 | bmoraca_work | that's somewhat important |
| 15:45.59 | leifmadsen | someone didn't install any configuration files :) |
| 15:46.12 | spiceycurry | haha, that would be me... ;) |
| 15:46.21 | spiceycurry | so I suppose I can do a make config |
| 15:46.40 | spiceycurry | or wait |
| 15:46.51 | spiceycurry | how do I install those config files? |
| 15:47.32 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:48.14 | bmoraca_work | in your source directory, configs/indications.conf.sample should start you out |
| 15:48.21 | bmoraca_work | or potentially, you could do "make samples" |
| 15:48.32 | bmoraca_work | you'll just need to clean them up before you actually push your system to production |
| 15:48.41 | *** join/#asterisk pentanol (~pentanol@77-35-7-125.pppoe.primorye.net.ru) |
| 15:48.53 | spiceycurry | ah ok |
| 15:48.56 | spiceycurry | cool, thanks |
| 15:49.17 | *** join/#asterisk davido1 (~davido1@p54B0A71B.dip0.t-ipconnect.de) |
| 15:49.26 | davido1 | Hi guys |
| 15:49.59 | davido1 | I've a small problem... I just installed asterisk 1.6.2.5 and the Queue() app is killing me |
| 15:51.28 | davido1 | In short, the problem is that when I call the queue and hang up --that is, without waiting for an agent to answer my call--, my call stays there, the phones keep ringing and, if I try to pick up the call, Asterisk dies... in the rain |
| 15:51.44 | davido1 | :s |
| 15:54.01 | leifmadsen | any reason you're not using the latest version? :) |
| 15:54.15 | leifmadsen | regardless, more information is required |
| 15:54.51 | davido1 | leifmadsen: err.. I installed the version in the Ubuntu repos |
| 15:54.58 | leifmadsen | ya... don't do that ;) |
| 15:55.00 | leifmadsen | install from source |
| 15:55.19 | spiceycurry | Hmmmmmmm I take it that " WARNING WARNING WARNING" messages are not so hot... |
| 15:55.27 | leifmadsen | depends what they are |
| 15:55.32 | leifmadsen | warnings are not errors |
| 15:55.33 | Polysics | ok, basically when i do a call, if the caller hangs up, either before or after the called party answers, the DIALSTATUS variable is empty |
| 15:55.36 | wcselby | davido1 - could you show us the CLI logs from this happening (reproduce it with cli verbosity 10 and copy paste it to a pastebin) |
| 15:55.39 | spiceycurry | Your Asterisk modules directory, located at /usr/lib/asterisk/modules |
| 15:55.41 | davido1 | leifmadsen: yeah well... that's not my decision to make |
| 15:55.44 | Polysics | which makes it impossible to tell if the call went OK or not |
| 15:56.20 | wcselby | Polysics - we need to see the logs that you are seeing so we can help troubleshoot |
| 15:56.25 | wcselby | Polysics - use pastebin |
| 15:56.27 | wcselby | ~pb |
| 15:56.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:57.25 | spiceycurry | Ok... here is the messages I just got http://pastebin.com/tXdA5kEx |
| 15:57.32 | spiceycurry | how can I resolve that? |
| 15:57.53 | wcselby | spiceycurry - if you're concerned about it, remove those modules from that path |
| 15:58.10 | wcselby | but otherwise, that usually means that it compiled okay and you can keep moving forward, even with those modules |
| 15:58.20 | spiceycurry | sweet. ty |
| 15:58.23 | Polysics | nvm, i thin k i have Adhearsion to blame |
| 15:58.34 | spiceycurry | I have config files dammit!! :D |
| 15:58.36 | wcselby | Polysics - glad we could help |
| 15:58.53 | spiceycurry | And the lovely: "Asterisk Ready." |
| 15:59.08 | davido1 | wcselby: http://pastebin.com/TtTERBa8 |
| 15:59.21 | spiceycurry | Shit, how do I get out of asterisk -c --- quit doesn't work, or exit |
| 15:59.25 | spiceycurry | ctrl-c? |
| 15:59.56 | leifmadsen | core stop now |
| 16:00.04 | leifmadsen | it'll stop the asterisk process |
| 16:00.11 | spiceycurry | sweet, ty |
| 16:00.18 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:02.45 | wcselby | does asterisk support md5 authentication for sip providers? |
| 16:03.05 | leifmadsen | md5secret |
| 16:03.15 | leifmadsen | if that's what you mean |
| 16:03.36 | wcselby | nevermind, found the config examples I needed for this provider. |
| 16:03.44 | wcselby | thanks though |
| 16:03.54 | davido1 | Great, I actually found that asterisk segfaults when that happens ¬¬ |
| 16:06.48 | *** join/#asterisk lhz (~shrekz@c-b9aa72d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 16:07.06 | wcselby | davido1 - what version of asterisk, how is the call incoming? as in, what technology? DAHDI, SIP, IAX, etc? |
| 16:08.02 | davido1 | wcselby: Asterisk 1.6.2.5-0ubuntu1, call is SIP <-> SIP |
| 16:08.41 | davido1 | wcselby: Funny thing is, if I call the queue and hang up, Asterisk doesn't notice that I hung up and just goes on... |
| 16:09.20 | wcselby | davido1 - yeah, I see that. it's not detecting the hangup for some reason |
| 16:09.45 | spiceycurry | my dahdi Tool is showing two unconfigured spans... card 0, Span 1, and card 0, span 2... how do I configure the card? |
| 16:09.52 | davido1 | wcselby: yup... so stwange |
| 16:10.37 | spiceycurry | the alarms are unconfigured |
| 16:12.06 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:12.51 | wcselby | davido1 - can you setup a test system with a source 1.6.2.6 install to see if you experience the same issue? |
| 16:13.19 | wcselby | davido1 - otherwise, you may want to open a new bug on the issue tracker (http://issues.asterisk.org) |
| 16:13.33 | wcselby | especially if it's segfaulting |
| 16:14.23 | davido1 | wcselby: yup, I guess I can :) Thanks mate |
| 16:14.53 | wcselby | i do see a couple chan_sip.c updates in 1.6.2.6 |
| 16:16.54 | spiceycurry | how can I test my DAHDI configuration? |
| 16:18.25 | Qwell | make a call |
| 16:19.18 | batphone | trying to get this asterisk cd to burn right |
| 16:19.28 | batphone | i built a little distro that auto installs |
| 16:19.41 | batphone | but i get io errors and csum problems, missing files, when the installer rips |
| 16:19.49 | batphone | the sha1 hash is good |
| 16:20.14 | batphone | wierd. tried different brands of disks, burnt in different os's and drives, and booted in different computers... no dice anywhere |
| 16:20.34 | batphone | so i guess my off topic question is: how do i take the sha1 hash from the iso image burned to a cdrom? |
| 16:20.54 | batphone | reversing the mkisofs might do that but i'd think theres no way it would match because of the way mkisofs works. |
| 16:21.12 | batphone | any ideas guys? |
| 16:21.23 | batphone | i have lots of boxen to deploy soon ;/ |
| 16:23.11 | batphone | man you voip guys are supposed to stick together |
| 16:23.22 | batphone | i got people breathing down my neck right now and i feel retarded |
| 16:23.30 | batphone | i can author a linux distro but i can burn a damned boot cd? |
| 16:23.46 | batphone | s/can/can't/2 |
| 16:23.59 | [TK]D-Fender | batphone: Sorry, this is #asterisk , not #howtobuildanoscd |
| 16:24.08 | batphone | well its a asterisk cd anyways |
| 16:24.11 | batphone | so ;P |
| 16:24.47 | batphone | got a gui that sales and customers use to build config files on the web, then an iso is written with all that stuff ready to go |
| 16:25.03 | batphone | they get a download link and a cdrom in the mail later if they want |
| 16:25.10 | wcselby | where do asterisk dump files go? |
| 16:25.14 | Naikrovek | wish i knew how to help ya, but i've never done what you're doing |
| 16:25.17 | batphone | so this is good thing. i am not making lots of money, just a living and helping people. |
| 16:25.45 | batphone | cool cool well thanks for some feedback. i tried #linux but they are all trying to mount /dev/sda |
| 16:25.50 | batphone | or something ;) |
| 16:26.22 | wcselby | ahhh, /tmp |
| 16:27.37 | *** join/#asterisk rdircio (~admin@189.242.22.188) |
| 16:27.56 | davido1 | thanks guys... ciao |
| 16:27.57 | *** part/#asterisk davido1 (~davido1@p54B0A71B.dip0.t-ipconnect.de) |
| 16:28.29 | batphone | have you guys used netsapiens? |
| 16:28.37 | batphone | i have an interop issue with it and asterisk sometimes |
| 16:28.52 | batphone | where the From URI won't match the IP of the outgoing interfaace on the * box |
| 16:28.53 | wcselby | never heard of it |
| 16:29.06 | batphone | its a big mysql driven switch/lcr |
| 16:30.12 | batphone | pretty cool stuff actually. some serious throughput though. i have 1700 calls and 100 cps up on one dual proc box with 8 gigs of ram. |
| 16:30.33 | batphone | thats wholesale though. i connect all my customers to asterisk boxes at their office which connect back to my lcr |
| 16:30.45 | *** join/#asterisk rdircio (~admin@189.242.22.188) |
| 16:30.46 | batphone | i wish you all could work with me. i need help so bad. |
| 16:31.20 | paulc | looks around for the rittalin... |
| 16:31.37 | paulc | I'll work with lots of people, for the right price ;-) |
| 16:31.55 | paulc | (rittalin for co-workers - they're a bit ADD this morning..) |
| 16:31.57 | spiceycurry | @paulc - concerta is better... lasts all day |
| 16:32.06 | wcselby | batphone - it doesn't appear to be an asterisk issue, but a cd burning issue? |
| 16:32.23 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 16:32.37 | wcselby | did you make an iso? try remaking the iso? |
| 16:33.27 | batphone | well the iso has worked in the past and nothing has changed there in the process |
| 16:33.36 | batphone | its an automatic sort of thing done with perl and what not |
| 16:33.56 | batphone | config your phones and stuff in the ui, and ext.conf and all that gets written. /home/polycom/ is populated. etc. |
| 16:34.10 | batphone | i know its not really * related but thought id ask |
| 16:34.17 | batphone | any idea what # i should be asking in? |
| 16:35.05 | wcselby | not a clue, i only come here for the #asterisk |
| 16:35.11 | wcselby | and then only some times |
| 16:46.10 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 16:47.55 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:49.34 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.229.215.dsl.dyn.forthnet.gr) |
| 16:53.12 | pabelanger | how big is Huntsville, AL? |
| 16:53.54 | p3nguin | 'bout the size of my fist. |
| 16:55.13 | bmoraca_work | pabelanger: on my map, it shows about 1mm in diameter |
| 16:55.15 | jaytee | population is about 180K people |
| 16:55.35 | jaytee | smaller than most US cities but large than alot in terms of geographic size. |
| 16:55.41 | malcolmd | metro's about 395K per wikipedia |
| 16:56.48 | coppice | I am amused when travelling around the US by the tiny hamlets with City in their name |
| 16:57.05 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:58.49 | p3nguin | http://www.pcworld.com/businesscenter/article/194319/caller_id_spoofing_ban_is_bad_for_business.html |
| 16:59.18 | malcolmd | coppice: i enjoy driving through the sticks and coming across signs that say "police jurisdiction" with no other visible sign of civilization |
| 17:02.45 | citywok | lol p3nguin pay your bills and you wont have to worry about them spoofing your mother! |
| 17:03.30 | citywok | i dislike that on DID's you can't change the name, since most providers override the name with the actual lookup for the number you sent |
| 17:04.09 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.229.215.dsl.dyn.forthnet.gr) |
| 17:06.52 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:07.22 | wcselby | [Apr 22 10:16:03] DEBUG[15967] sched.c: Request to schedule in the past?!?! |
| 17:08.48 | p3nguin | Ah, the things we put up with to stream MoH. |
| 17:09.13 | citywok | mine works pretty well coming from shoutcast/aol |
| 17:09.28 | p3nguin | same here |
| 17:10.05 | raden_work | p3nguin, funny thing is there was a truth in caller id like 4 years ago its been illegal for quiete some time |
| 17:11.23 | citywok | does anybody know what L3's minimum commit is? |
| 17:11.29 | wcselby | don't think that's an MOH issue |
| 17:11.58 | citywok | wcselby: i would assume you probably tried a call script in the outgoing folder or whatever, and it was set with an invalid date |
| 17:12.27 | wcselby | i have no idea what happened, I'm going through the logs of a client's trixbox that crashed earlier. i think around the same time their internet crapped out |
| 17:13.07 | p3nguin | I'm going to assume they stream moh and the buffer ran out during the internet outage. |
| 17:13.18 | wcselby | no stream |
| 17:13.27 | wcselby | crappy freeplay that comes with asterisk |
| 17:14.11 | *** join/#asterisk iluminator101 (~iluminato@ool-ad02048d.dyn.optonline.net) |
| 17:14.49 | iluminator101 | hey so i install asterisk from source, but now i am having trouble with install freepbx, i am running centos 5.4 64bit |
| 17:15.10 | wcselby | you mean you don't like the freepbx installation instructions on their site? |
| 17:15.21 | Qwell | iluminator101: #freepbx |
| 17:15.31 | wcselby | remembers the last time I had to install freepbx from source......and shudders |
| 17:16.36 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:18.20 | *** join/#asterisk rene- (~rene@200.79.233.92.cable.dyn.cableonline.com.mx) |
| 17:20.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:25.25 | *** join/#asterisk rare1980_ (~rare1980@115.186.23.49) |
| 17:27.40 | bmoraca_work | anyone have a good international test phone number that terminates to an auto attendant? |
| 17:28.39 | rare1980_ | hey experts, i have a question.. i am using a predictive dialer called stratasoft. this dialer use intel dialogic card to establish its agent audio connection... now what i want... i want to establish its agent audio connection using asterisk SIP. i have defined sip account in sip.conf in asterisk. and i put that sip account info in the dialer. but i m not getting register. |
| 17:28.50 | rare1980_ | <PROTECTED> |
| 17:29.28 | rare1980_ | soo please guide me if stratasoft agent audio connection can be established from asterisk sip?? |
| 17:30.37 | rare1980_ | plz any help or ideas on this??? |
| 17:31.42 | [TK]D-Fender | rare1980_: Odds are nobody else uses them, or have even herd of them. If that software speaks SIP, chances are you can use it with * |
| 17:32.01 | [TK]D-Fender | bmoraca_work: ibm.com <- contacts |
| 17:33.34 | rare1980_ | [TK]D-Fender: soo if that software can get communicate with sip on part UDP 5060 and 10000 to 20000 so then can we established its audio connection??? |
| 17:35.07 | rare1980_ | ??? |
| 17:37.59 | bmoraca_work | [TK]D-Fender: yeah, i ended up calling the US Embassy in England |
| 17:38.54 | ariel_ | bmoraca_work: When I do testing for international numbers I do a google lookup for big name hotels |
| 17:39.25 | ariel_ | Most Marriots have an IVR |
| 17:39.37 | bmoraca_work | that's a good idea, too |
| 17:41.36 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 17:42.47 | highvoltz | Is there any way to get a FXO card to work with vmware esx / asterisk ? |
| 17:43.41 | bmoraca_work | highvoltz: no |
| 17:43.49 | leifmadsen | highvoltz: only the host operating system can access the hardware |
| 17:43.57 | leifmadsen | in the case of ESXi -- the answer is no |
| 17:44.22 | leifmadsen | the host OS needs to be Linux (or FreeBSD if you use the freebsd-dahdi drivers) |
| 17:44.23 | lanning | plus timing is critical, and VMs just don't cut it. |
| 17:44.36 | bmoraca_work | highvoltz: you'll need to get use a SIP-to-PSTN gateway, such as an SPA3102 |
| 17:44.39 | bmoraca_work | lanning: not true :) |
| 17:44.49 | leifmadsen | VMs work fine for a few calls -- you can tweak them to be more responsive, but hardware is out of the question |
| 17:45.24 | citywok | i run msft office communications server w/ audio for 25-30 calls at a time without any issues, in Hyper-V |
| 17:45.29 | *** join/#asterisk punkgode (~punkgode@72.249.66.65) |
| 17:45.39 | highvoltz | bmoraca_work, Ill check it out thanks! |
| 17:45.45 | leifmadsen | that's because MS ComServ is amazing |
| 17:45.49 | leifmadsen | it'll make you coffee too |
| 17:46.03 | leifmadsen | <sarcasm /> |
| 17:46.21 | citywok | i really like * + OCS, i am working on getting my R2 OCS setup going so i can go that way instead of using meetme. |
| 17:46.27 | [TK]D-Fender | rare1980_: why not? SIP is SIP. Go try something and look at what is happening |
| 17:46.38 | citywok | everybody uses livemeeting + meetme, being able to do it all natively in OCS R2 will be super slick |
| 17:47.06 | citywok | it'll make you a very complex coffee pot, but you have to grow the beans first, then roast and grind them. oh, and there is no manual. |
| 17:47.30 | lanning | bmoraca_work: it's hard enough keeping the timer interrupt correct. |
| 17:49.10 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:50.54 | bmoraca_work | lanning: i run multple asterisk VMs in ESXi without issue |
| 17:51.02 | simond | does dahdi now have a way to see line voltage on analog lines? |
| 17:51.21 | lanning | bmoraca_work: against hardware? or just passing SIP/RTP along? |
| 17:51.35 | bmoraca_work | lanning: they're hosted PBXes |
| 17:52.16 | bmoraca_work | lanning: so they run the gamut of utilization |
| 17:52.17 | lanning | bmoraca_work: I am talking about hardware access |
| 17:52.17 | bmoraca_work | lanning: there is no hardware access from within a VM |
| 17:52.18 | bmoraca_work | lanning: i have an external media gateway |
| 17:52.18 | citywok | lanning: we don't use analog anything, all sip |
| 17:52.42 | lanning | then VMs work fine. I was talking about the request of using an FXO device from within a VM. |
| 17:52.48 | *** join/#asterisk rene- (~rene@200.79.233.92.cable.dyn.cableonline.com.mx) |
| 17:52.56 | citywok | sure, you just need an external gateway like bmoraca_work said |
| 17:53.01 | lanning | when did the subject change? |
| 17:53.20 | bmoraca_work | lanning: you made a blanket statements about VMs which was incorrect. |
| 17:53.35 | citywok | bmoraca_work: did you know that ATI drivers suck? |
| 17:53.35 | rene- | Hello, it seems that reinvite or media bypass can not be achieved in SIP when dtmfmode is RFC2833, is that right? most SIP ITSPs do not support SIP INFO |
| 17:53.49 | bmoraca_work | citywok: for the most part, though they're better than they used to be |
| 17:53.52 | citywok | oh wait, that was in 1995. lol |
| 17:53.55 | lanning | sorry, I didn't not qualify about HW access. As, I thought that was what we were talking about. |
| 17:54.19 | citywok | i think all graphics drivers suck if you install all the crappy software that uses 300mb of ram. |
| 17:54.33 | citywok | same for printers and scanners |
| 17:54.53 | Qwell | citywok: my keyboard has software that goes with it |
| 17:54.57 | Qwell | KEYBOARD |
| 17:55.06 | citywok | yea, same. so did my mouse! |
| 17:55.12 | citywok | F u logitech. F u |
| 17:55.18 | citywok | (msft keyboard tho) |
| 17:55.20 | *** join/#asterisk Z_God (~julius@wlan228189.mobiel.utwente.nl) |
| 17:55.36 | Qwell | citywok: rebranded logitech imo |
| 17:55.37 | citywok | next thing you know the hard drive will come with drivers to install... lol |
| 17:55.48 | Qwell | they come with software.. |
| 17:55.49 | bmoraca_work | citywok: they used to |
| 17:56.00 | wcselby | i've had people ask for monitor drivers before |
| 17:56.10 | Qwell | monitors had drivers in win9x |
| 17:56.12 | bmoraca_work | citywok: remember the old days where you needed a fake bios to run hard drives larger than 10gb? |
| 17:56.13 | rene- | yeah and then computers will came with software, what will they think of next |
| 17:56.22 | *** part/#asterisk rene- (~rene@200.79.233.92.cable.dyn.cableonline.com.mx) |
| 17:56.23 | citywok | bmoraca_work: lol, jumper on the HD |
| 17:56.40 | citywok | haha, i had an acer and a viewsonic monitor come with drivers last month... |
| 17:56.50 | bmoraca_work | i'll always remember WD's EZBIOS in my nightmares |
| 17:57.21 | coppice | all monitors come with "drivers", but they really just colour tables |
| 18:00.57 | Kyosh | I have Asterisk 1.4.21 set up to operate in RealTime with MySQL. I am trying to use the call-limit variable by setting the value to 2, but it is not working. It allows as many calls as possible with no regard to the call-limit variable. If anyone is familiar with this issue, I would appreciate any assistance. Also, is there a way I can script this in the dialplan? Thanks. |
| 18:01.16 | [TK]D-Fender | ~itsplist-us |
| 18:01.17 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:02.58 | *** join/#asterisk QaDeS (~mklaus@p54A1AD95.dip0.t-ipconnect.de) |
| 18:05.30 | *** join/#asterisk Krazylegz (~~Vikram@li37-238.members.linode.com) |
| 18:05.39 | Krazylegz | Hello. |
| 18:08.06 | ChannelZ | ohell. |
| 18:08.55 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 18:08.55 | *** join/#asterisk iCEBrkr (~icebrkr@72.251.206.106) |
| 18:08.55 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 18:08.55 | *** join/#asterisk FreezeS (~KVIrc@82.208.157.125) |
| 18:08.55 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:08.55 | *** join/#asterisk P1ersson (~P1ersson@213-64-217-60-no50.tbcn.telia.com) |
| 18:08.55 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 18:08.56 | *** join/#asterisk Belgarath (~Belgarath@banda.pl) |
| 18:08.56 | *** join/#asterisk zamba (marius@flage.org) |
| 18:08.56 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 18:08.56 | *** join/#asterisk DarkNet (~FreeNoden@courriel-quebec.com) |
| 18:08.56 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:08.56 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 18:08.56 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 18:08.56 | *** join/#asterisk McLazarus (~McLazarus@dogpile.mcallister.ws) |
| 18:09.07 | *** join/#asterisk citrus2 (~citrus2@mail.serviceobjects.com) |
| 18:09.07 | ChannelZ | puts his pants back on |
| 18:09.38 | citrus2 | i just installed asterisk 1.6.2 from source when i run asterisk -r i don't see extensions.conf being loaded can anyone tell me why/ |
| 18:11.12 | *** join/#asterisk karega (~anoans@205.168.84.98) |
| 18:11.15 | karega | Hello |
| 18:11.30 | karega | Does anyone have a link for a simple SIP session? |
| 18:11.35 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:11.59 | karega | I am attempting to interact with an SIP listener through telner |
| 18:12.02 | karega | telnet |
| 18:12.34 | bmoraca_work | karega: SIP is generally UDP, telnet is TCP. not going to work. |
| 18:13.05 | leifmadsen | unless you've enabled SIP TCP signalling on 1.6.2 (or whatever the first version was to have it) |
| 18:13.19 | coppice | according to the SIP specs TCP support is compulsory |
| 18:13.36 | leifmadsen | specs are for breaking |
| 18:13.36 | karega | ok |
| 18:13.53 | karega | so my SIP supports TCP |
| 18:14.07 | karega | how do I interact with an SIP listender? |
| 18:14.15 | *** join/#asterisk Alagar (~Administr@122.164.32.135) |
| 18:14.18 | karega | such as using the MESSAGE method |
| 18:15.16 | [TK]D-Fender | coppice: RFC compliance = No comment :) |
| 18:15.30 | [TK]D-Fender | ~siprfc |
| 18:15.31 | infobot | i guess siprfc is http://www.faqs.org/rfcs/rfc3261.html |
| 18:15.35 | [TK]D-Fender | ^^^^^^^ |
| 18:15.42 | [TK]D-Fender | karega: Happy browsing |
| 18:15.44 | keith4 | i'm trying to use a nagios plugin check_sip to make sure asterisk is functional. the readme just says "Full SIP uri, eg sip:uri@example.com". I can't figure out what that corresponds to in asterisk, so i just keep getting 404s. help? |
| 18:16.38 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 18:17.06 | *** join/#asterisk SaiSoma (~Anth8708@client105.jdcc.edu) |
| 18:17.20 | idespinner | keith4, sounds like something in your extensions.conf in your 'default' context |
| 18:17.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.19 | keith4 | idespinner: that's what i was hoping, but sip debug shows it looking in [banned] |
| 18:18.25 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 18:18.26 | karega | [TK]D-Fender, that document doesn't cover the MESSAGE methog |
| 18:18.29 | karega | MESSAGE method |
| 18:18.44 | keith4 | ugh. some idiot set the default sip context to banned |
| 18:18.49 | ruben23 | hi how do i do with one way audio voice for calls on asterisk,i have NAt |
| 18:19.21 | ariel_ | ping hardwire you around? |
| 18:21.27 | [TK]D-Fender | karega: Go find a better SIP book then |
| 18:22.06 | *** join/#asterisk ryanlin_ (~ryanlin@you.better.not.nulroute.us) |
| 18:22.25 | ruben23 | how do i config asterisk to do with NAT environment.. |
| 18:22.29 | [TK]D-Fender | ~sipnat |
| 18:22.30 | 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 |
| 18:22.30 | ryanlin_ | i am trying to intergrate cisco callmanager express with asterisk |
| 18:22.39 | ryanlin_ | i configured the peering entry in sip.conf |
| 18:22.59 | ryanlin_ | and the voicemail entry in extensions.conf |
| 18:23.05 | ryanlin_ | that didn't work, any ideas? |
| 18:23.25 | [TK]D-Fender | ryanlin_: "didn't work". Sorry, could you be a little more vague? |
| 18:23.30 | [TK]D-Fender | :) |
| 18:23.33 | ryanlin_ | lol sure |
| 18:23.39 | ryanlin_ | one sec..let me do some debug |
| 18:23.48 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 18:24.47 | keith4 | slap it around a bit, that might help |
| 18:25.29 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.229.215.dsl.dyn.forthnet.gr) |
| 18:25.32 | karega | damn I want to use SIP so I don't have to write my own session initiated protocol but the SIP documentation out there sucks |
| 18:26.44 | [TK]D-Fender | karega: I fail to see you even listing what you want to do... |
| 18:26.57 | citrus2 | anyone know why asterisk 1.6.2 wouldn't look for my extensions.conf file? |
| 18:26.59 | citrus2 | to parse |
| 18:27.19 | justing | did you rm -rf * on your config directory? |
| 18:27.25 | ryanlin_ | [Apr 22 11:32:35] WARNING[3791]: pbx.c:8849 pbx_builtin_background: ast_streamfile failed on SIP/172.16.11.250-00000001 for E |
| 18:27.36 | citrus2 | no its their its just not parsing it when i do a 'reload' |
| 18:27.40 | ryanlin_ | that's the error that i receive every time when i make the call |
| 18:27.50 | justing | citrus2, but did you delete any OTHER config files? |
| 18:27.51 | [TK]D-Fender | ryanlin_: PASETBIN is your friend |
| 18:27.54 | ryanlin_ | isee |
| 18:27.55 | [TK]D-Fender | ~pb |
| 18:27.55 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:28.07 | *** join/#asterisk Tim_Toady (~moi@77.49.29.230.dsl.dyn.forthnet.gr) |
| 18:28.34 | [TK]D-Fender | citrus2: Check your permissions. check yoru modules.conf . |
| 18:29.25 | ryanlin_ | http://inetpro.org/pastebin/3084 |
| 18:30.04 | iluminator101 | what a cheap and reliable sip provider that you use guys have used and can recommend |
| 18:30.06 | citrus2 | permisisons are fine here is my modules.conf |
| 18:30.08 | citrus2 | http://pastebin.com/YhCnWQqr |
| 18:30.20 | ryanlin_ | that's the output from 'sip set debug on' |
| 18:30.42 | [TK]D-Fender | ryanlin_: Interesting. Try with a stock * recording |
| 18:31.04 | ryanlin_ | hmm..that works |
| 18:31.11 | [TK]D-Fender | ryanlin_: thn your file is bad |
| 18:31.16 | ryanlin_ | i have fiddled around the permission |
| 18:31.30 | ryanlin_ | here is the thing |
| 18:31.37 | ryanlin_ | the file works in another asterisk system |
| 18:31.46 | ryanlin_ | however, that asterisk system is quite old |
| 18:31.54 | [TK]D-Fender | ryanlin_: the file is bad. SOMETHING is wrong with it. either permissions, format, etc |
| 18:31.59 | ryanlin_ | hmm |
| 18:32.00 | ryanlin_ | ok |
| 18:32.38 | ryanlin_ | in terms of the sip exchange, does it look ok? |
| 18:32.53 | ryanlin_ | cisco callmanager express <====> asterisk |
| 18:34.23 | [TK]D-Fender | rynCall processes. You get audio with a "not bad" recording, right? |
| 18:34.26 | [TK]D-Fender | ryanlin_: You tell me. |
| 18:34.35 | *** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi) |
| 18:34.51 | [TK]D-Fender | ryanlin_: Mind you it didn't match a peer.. but that didn't sto the call from progressing as unauthed |
| 18:34.55 | ryanlin_ | no..the call just hangs up by itself |
| 18:35.11 | ryanlin_ | isee |
| 18:35.21 | ryanlin_ | i will check into the audio file |
| 18:36.24 | hardwire | ariel_: si |
| 18:36.29 | hardwire | ariel_: sort of |
| 18:36.39 | hardwire | <- headache |
| 18:36.41 | citrus2 | does my modules.conf look fine >http://pastebin.com/YhCnWQqr |
| 18:36.51 | ariel_ | sorry to hear that hardwire |
| 18:36.59 | hardwire | I'll live to tell the tale. |
| 18:37.02 | ariel_ | let me know when your able to work then |
| 18:37.32 | hardwire | today probably not great.. working on a prefix reduction system to take 800k prefixes and make it a smaller yet just as informative list. |
| 18:37.43 | hardwire | I'm down to 100k :) |
| 18:37.54 | ariel_ | wow |
| 18:37.59 | ariel_ | ok tomorrow then |
| 18:38.01 | hardwire | ok |
| 18:38.06 | ryanlin_ | hardwire: wow..on a asterisk? |
| 18:38.09 | ariel_ | I have a digium board for you... |
| 18:38.16 | ryanlin_ | i could never wrap my head around the voip |
| 18:38.19 | hardwire | btw I set up drbd and heartbeat for some other stuffs. so I'm fresh on the topic now |
| 18:38.31 | hardwire | ryanlin_: the voip is powerful. |
| 18:38.41 | ariel_ | hardwire: nice |
| 18:38.45 | hardwire | ryanlin_: 800k prefixes to match against.. not really in extensions.conf.. for LCR |
| 18:38.46 | ariel_ | hope you feel better |
| 18:38.54 | hardwire | yup.. me too |
| 18:39.21 | ryanlin_ | isee |
| 18:40.36 | bmoraca_work | one of my providers has 4400 individual rates for international calling...eventually, i'll get more and do some LCR, but for now, my international needs are very low |
| 18:45.46 | paulc | Part of the problem is granularity.. like the UK - they said 01xx and 02xx = landline, 07xx = mobile.. should be easy, right? noooo cos different mobile carriers have different interconnect rates, so you end up with a bazillion 07xxxxx routes with a handful of different price points |
| 18:48.18 | bmoraca_work | yep |
| 18:49.29 | hardwire | yeh |
| 18:49.50 | hardwire | I'm reducing the rates based on unique costs as well as grouped prefixes. |
| 18:49.59 | citywok | intermittently my phones lose their registration |
| 18:50.00 | hardwire | it's very interesting what I've done so far.. I'm rather impressed with myself. |
| 18:50.04 | citywok | how can i figure out what is going on? |
| 18:50.05 | hardwire | :P |
| 18:50.09 | citywok | (20-50 phones at one time) |
| 18:50.11 | hardwire | citywok: kick them. |
| 18:50.14 | hardwire | citywok: call the ISP? |
| 18:50.15 | hardwire | :P |
| 18:50.34 | citywok | yea believe me i've been lighting up qwest like there's no tomorrow in the lsat few months |
| 18:50.45 | citywok | if they knew how to configure a router life would be AWESOME |
| 18:51.16 | citywok | some fucking idiot put the LAN & DMZ subnets on the same vlan. bridged my entire datacenter. removing the bridge poisened the ARP tables in the core router and took everything down. |
| 18:51.30 | hardwire | citywok: that's unfortunate |
| 18:51.42 | hardwire | lots of ISP platforms that offer "metro ethernet" foul things up that way. |
| 18:51.45 | citywok | yea, that's just one of qwest's 10 mistakes since february |
| 18:51.55 | hardwire | to be honest.. it's their billing platform that is doing most of the damage |
| 18:52.10 | citywok | qwest's billing is actually the best of all the telecoms i've worked with |
| 18:52.24 | citywok | not that that is saying much, or setting a high bar. |
| 18:52.45 | hardwire | sure.. but it's the auto-provisioning that gleans data from the billing system and sets up automatic switch/router configuration that is the problem. |
| 18:52.56 | hardwire | at least it is up here with a few telcos |
| 18:53.02 | citywok | oh, qwest doesn't auto-provision the devices |
| 18:53.09 | citywok | they have some dumbass write and then copy/paste the config in to the router |
| 18:53.13 | hardwire | :P |
| 18:53.17 | citywok | (no, they don't check to make sure that it actually works, they just ship it out) |
| 18:53.19 | hardwire | diff slaves |
| 18:53.33 | hardwire | bbl.. need to bang brain against something sharp |
| 18:53.46 | citywok | out of all of my offices, the most stable connection is the one in the phillipines 7,000 miles away |
| 18:55.42 | bmoraca_work | i accidentally forgot to disable the NAT helper for SIP on a cisco ios router once |
| 18:55.50 | bmoraca_work | that was ugly |
| 18:56.04 | citywok | did anything work? lol |
| 18:56.16 | citywok | getting SIP to work behind a sonicwall defintely took some arm wrestling. |
| 18:56.17 | bmoraca_work | sure did...while i was at the office... |
| 18:56.27 | bmoraca_work | 15 minutes after i left, though, i got a call and nothing was working |
| 18:56.39 | bmoraca_work | i tell my customers we don't support SonicWalls |
| 18:56.48 | citywok | good idea. |
| 18:56.56 | citywok | the 500 checkboxes you have ot get in the right order are annoying |
| 18:56.59 | citywok | bbl lunch |
| 18:57.23 | hardwire | bmoraca_work: sonicwall supports ipsec thankfully.. so you could at least work that angle. |
| 18:57.26 | hardwire | but it's a pita |
| 18:57.38 | bmoraca_work | yeah |
| 18:57.43 | hardwire | I'm setting up something similar for a client |
| 18:57.57 | hardwire | and verizon uses ipsec for what we use them for |
| 18:58.20 | bmoraca_work | i have a couple clients set up with VPNs to their hosted PBX due to NAT issues with the number of clients they have |
| 18:59.08 | hardwire | yeh |
| 18:59.15 | hardwire | it's handy having that facility available |
| 18:59.27 | hardwire | verizon was bugging me that I'm not using cisco ipsec.. but it worked out. |
| 18:59.48 | hardwire | racoon + linux 2.6 + setkey = win |
| 19:05.51 | *** join/#asterisk miamiseb (~deigo@208.76.35.132) |
| 19:06.23 | bmoraca_work | i just have them VPN to a PIX and I set it up such a way that they only have access to their system |
| 19:06.41 | bmoraca_work | it's a PIX 525...it supports like 10000 IPSec peers |
| 19:07.13 | ariel_ | pix = shit |
| 19:07.27 | bmoraca_work | disagree :) |
| 19:08.28 | ariel_ | just remember we all have our opinion just like ass wholes. |
| 19:11.00 | bmoraca_work | opinions are like assholes...they all stink except mine? |
| 19:12.22 | ariel_ | lol |
| 19:12.54 | ariel_ | ASA, oh we are moving off them as well here. |
| 19:13.38 | bmoraca_work | moving to what? |
| 19:13.57 | ariel_ | it starts with Juni..... |
| 19:14.10 | bmoraca_work | ahhh |
| 19:14.45 | ariel_ | But since most of our wlan (corp) loves Cisco's they have not been told yet that we are moving off them. |
| 19:15.44 | hardwire | I just use linux for most of my routers |
| 19:15.59 | hardwire | and it does whatever the hell I tell it to do |
| 19:16.06 | ariel_ | your the man |
| 19:16.10 | hardwire | and it doesn't bitch and moan.. or have sales reps. |
| 19:16.11 | hardwire | haha |
| 19:16.12 | ariel_ | but I work for a big corp |
| 19:16.23 | ariel_ | sad |
| 19:16.37 | hardwire | can pass apx 8gbit through a hefty linux system |
| 19:16.48 | hardwire | doing so in Texas atm for a client. |
| 19:17.26 | hardwire | and it fails over to another system .. both systems cost less than a single cisco/juni |
| 19:17.41 | citrus2 | i have a couple different asterisk install, is there a way i can check to see what custom sounds folder location it is using? they are different |
| 19:17.45 | hardwire | dunno about power consumption however.. but the flexability made the chioce worthwhile for them. |
| 19:17.51 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.03 | hardwire | citywok: maybe 'core show settings' ? |
| 19:18.31 | ariel_ | just read the asterisk.conf |
| 19:22.35 | bmoraca_work | i don't like linux-based routers |
| 19:22.37 | bmoraca_work | never have |
| 19:22.42 | bmoraca_work | probably never will |
| 19:24.26 | hardwire | they complete me. |
| 19:24.32 | *** join/#asterisk dr_ (~duckz@86.107.84.186) |
| 19:25.02 | hardwire | specifically when I need to use them as both a router and something else |
| 19:25.06 | hardwire | say.. a DNS server.. :) |
| 19:25.11 | hardwire | or asterisk. |
| 19:25.20 | hardwire | oooooooor.. a director |
| 19:33.56 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:34.25 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 19:37.15 | *** join/#asterisk QaDeS_ (~mklaus@p54A1A26E.dip0.t-ipconnect.de) |
| 19:39.37 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 19:40.06 | *** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 19:47.19 | *** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 19:47.51 | iluminator101 | what itsp do you guys is the best for a low monthly rate almost unlimted international and domestic rates |
| 19:49.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:59.52 | iluminator101 | whats easiest pbx to setup for a newbie |
| 20:01.22 | iluminator101 | ie asterisknow, trixbox, pbx in a flash etc |
| 20:01.58 | btsteve | iluminator101: also look at freepbx you and use asterisk or FreeSwitch for the engine |
| 20:21.17 | Naikrovek | iluminator101: and everyone wants cheaper phone service. they're all about the same price, really. and remember, cheapest price will provide pretty close to the worst service |
| 20:33.26 | *** join/#asterisk TimeRider (~steve@5ac3182f.bb.sky.com) |
| 20:37.43 | keith4 | ~itsp |
| 20:37.44 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:38.05 | keith4 | iluminator101: US? |
| 20:39.19 | iluminator101 | yes |
| 20:39.25 | iluminator101 | and international to |
| 20:40.00 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:41.37 | iluminator101 | so what do you guys suggest |
| 20:44.08 | *** join/#asterisk SaiSoma (~Anth8708@client105.jdcc.edu) |
| 20:44.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:45.55 | *** join/#asterisk rdircio (~admin@189.242.22.188) |
| 20:50.36 | *** join/#asterisk SaiSoma (~Anth8708@client105.jdcc.edu) |
| 20:54.09 | *** join/#asterisk SaiSoma (~saisoma@client72.jdcc.edu) |
| 20:55.38 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 20:56.33 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:01.08 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 21:03.06 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:03.52 | pabelanger | ~itsplist-ca |
| 21:03.52 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 21:04.15 | Micc | Will asterisk 1.8 include SRTP? |
| 21:04.42 | pabelanger | Micc: that is the plan |
| 21:05.00 | pabelanger | https://issues.asterisk.org/roadmap_page.php |
| 21:05.06 | Micc | That would be great. |
| 21:05.20 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 21:05.28 | Micc | Are there any ITSP's selling SRTP sip trunks? |
| 21:06.03 | Micc | I'm wondering what we should charge our customers for SRTP. |
| 21:06.18 | Micc | Or if it should just be included. |
| 21:07.00 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:14.38 | Jumpie | hey guys |
| 21:14.47 | Jumpie | anybody have any experience with siemens gigaset? |
| 21:14.47 | hardwire | hey horses |
| 21:14.51 | hardwire | snickers |
| 21:14.54 | Jumpie | like the a 580 |
| 21:15.15 | hardwire | I believe my friend jermudgeon does |
| 21:15.25 | hardwire | pm him |
| 21:22.03 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:23.45 | *** join/#asterisk madler1 (~madler@c-76-98-125-204.hsd1.nj.comcast.net) |
| 21:25.04 | madler1 | hello. I am having an issue configuring a T38 Trunk |
| 21:25.27 | madler1 | I am getting an error that says Unsupported SDP media type in offer: image 4389 udptl t38 |
| 21:25.48 | madler1 | the complete error from CLI is here |
| 21:25.52 | madler1 | http://pastebin.com/0DyP9Pxy |
| 21:29.40 | chazzam | madler1: I don't see any sip debug in that |
| 21:29.43 | *** part/#asterisk Krazylegz (~~Vikram@li37-238.members.linode.com) |
| 21:29.54 | madler1 | one sec |
| 21:30.03 | madler1 | let me get a new trace on it |
| 21:31.53 | *** join/#asterisk DelphiWorld (~Miranda@41.104.76.131) |
| 21:34.04 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 21:38.21 | madler1 | Hi chazzam: This snip should be a bit better http://pastebin.com/1N9Ggyvq |
| 21:38.34 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 21:42.20 | DelphiWorld | hi |
| 21:42.24 | DelphiWorld | what is dahdi_speed? |
| 21:42.29 | DelphiWorld | is giving me Count: 635757 |
| 21:46.16 | Jumpie | im curious how those siemans giga sets work? |
| 21:46.19 | Jumpie | it has an ata looking contorl box |
| 21:46.23 | Jumpie | how does it handle 6 remotes/sip accounts? |
| 21:46.34 | Jumpie | how does asterisk know how to terminate those 6? |
| 21:46.41 | Jumpie | if that box is only connected with one ethernet port |
| 21:46.43 | *** part/#asterisk DelphiWorld (~Miranda@41.104.76.131) |
| 21:46.54 | Jumpie | each remote doesn't have an ethernet port |
| 21:46.58 | Jumpie | so its some kinda wireless |
| 21:47.07 | WIMPy | Err??? |
| 21:47.31 | WIMPy | I don't know about yours, but my Asterisk box is also only connected via one ethernet port. |
| 21:48.28 | Jumpie | nono |
| 21:48.29 | Jumpie | thast not what im saying |
| 21:48.30 | Jumpie | lol |
| 21:48.38 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 21:48.40 | Jumpie | http://www.voipsupply.com/siemens-gigaset-a580-ip |
| 21:48.42 | Jumpie | looking to get that |
| 21:48.51 | Jumpie | it says it supports 6 handsets and 6 sip accounts |
| 21:49.02 | Jumpie | each remote handset does not have its own ethernet port..it connects to this ATA looking wireless gateway |
| 21:49.05 | Jumpie | wireless only |
| 21:49.11 | Jumpie | the gateway connects to your network |
| 21:49.20 | citywok | dude, stop hitting enter after every 3rd word |
| 21:49.23 | Jumpie | so on asterisk, how are you setting up each of the up to 6 remote handsets individual extensions? |
| 21:49.26 | citywok | write a sentence |
| 21:49.31 | Jumpie | i write in spurts |
| 21:49.44 | Jumpie | so..i think there is some kinda web interface on this gateway that helps facilitate this |
| 21:49.58 | russellb | ~enter |
| 21:49.59 | infobot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 21:50.02 | russellb | heh |
| 21:50.03 | WIMPy | Like every SIP phone does? |
| 21:50.20 | Jumpie | hah...emdash, haven't heard that in awhile |
| 21:50.21 | citywok | ty russellb |
| 21:50.41 | WIMPy | However the Siemens SIP-DECT bases are shit. I've got such a thing. |
| 21:50.42 | Jumpie | i type faster than i think sometimes..if that next thing i want to say hasn't hit my brain yet..its not going on that line |
| 21:50.51 | Jumpie | yea..well those are more expensive |
| 21:50.57 | Jumpie | hehe |
| 21:51.08 | citywok | Jumpie: i hope you aren't in 3rd grade any more. maybe use that free-time thinking to proof read what you just wrote. |
| 21:51.28 | Jumpie | citywok why is it your goal to talk down to me? you think i'm some trolling grade school kid? |
| 21:51.37 | Jumpie | i dont irc like i type essays |
| 21:52.24 | Jumpie | i dont care enough about irc or you to make this college thesis material, now instead of being a spelling/grammar/english nazi, i was just looking for some basic help |
| 21:52.40 | citywok | it's fuckign annoying to read 3 words per line, much like it's annoying to READ WHEN PEOPLE CAN'T FIND THEIR CAPS LOCK KEY |
| 21:53.01 | Jumpie | citywok you spent more time bitching about my extra use of enter, than what it is you bitched at |
| 21:53.06 | Jumpie | but anyway.. |
| 21:53.08 | Jumpie | thanks |
| 21:53.15 | citywok | my goal was to make the world a better place :) |
| 21:53.45 | citywok | i've got 15 seconds to spare educating a lost soul, but alas it has fallen on deaf ears and blind eyes. so carry on with your questions that probably won't be answered. |
| 21:54.24 | WIMPy | Was there a question? |
| 21:54.43 | citywok | i dont know, and i'm not about to scroll up 500 lines to start reading it ;) |
| 21:55.58 | Jumpie | i'm not a lost soul lol..i was actually an english major, i apologize about my use of enter, but it's not like im some terrible english speaking |
| 21:56.10 | Jumpie | i just dont' care on irc but ill use less enters for your benefit |
| 21:56.23 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:56.25 | *** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 21:56.39 | iluminator101 | how do i configure asterisk to work with skype |
| 21:56.53 | WIMPy | Telling people, you don't care, is probably not a good start. |
| 21:57.57 | iluminator101 | hey guys |
| 21:58.06 | Jumpie | no..saying i dont care ENOUGH to format my words as if i was in college :) |
| 21:58.34 | citywok | a 3red grader knows how to use a period i think. |
| 21:58.40 | citywok | 3rd. lol. |
| 21:59.37 | Nugget | heh |
| 22:00.08 | florz | hey you fucking |
| 22:00.09 | florz | morons |
| 22:00.13 | florz | now help |
| 22:00.17 | exothermc | How do you subscribe to the status of more than one channel? |
| 22:00.18 | florz | me!1 |
| 22:00.37 | citywok | hint sip/1501&sip/1502 ? |
| 22:00.48 | florz | I don't care |
| 22:00.50 | florz | about you! |
| 22:00.57 | fenrus | Lol |
| 22:01.03 | florz | fenrus: help me! |
| 22:01.06 | fenrus | that sounds like a good way of getting help florz. |
| 22:01.14 | florz | doesn't it? =:-) |
| 22:01.14 | iluminator101 | how do i configure asterisk to work with skype |
| 22:01.27 | Jumpie | lol |
| 22:01.30 | fenrus | florz, did you explain the problem earlier?, i'll backlog if you did.. |
| 22:01.36 | exothermc | citywok: it only gives that status of the first one and ignores the follwoing. |
| 22:02.11 | florz | fenrus: In case you didn't notice, I don't have a problem at all, just demonstrating how to motivate people to help you ;-) |
| 22:02.44 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 22:02.57 | citywok | hmmm, i think i got kicked for doing that. lol |
| 22:03.07 | fenrus | florz, ;) |
| 22:03.47 | florz | fenrus: Erm, I guess you were just playing along? Sometimes it's just too realistic ... =:-) |
| 22:03.52 | exothermc | citywok: Not sure if you saw my response, but that doesn't work. |
| 22:04.02 | exothermc | citywok: and no it didn't kick you. |
| 22:04.15 | citywok | i use this: exten => 1006,hint,SIP/1006&SIP/10061 |
| 22:04.19 | [TK]D-Fender | iluminator101: http://www.digium.com/en/products/software/skypeforasterisk.php |
| 22:04.32 | citywok | though to be honest i'm not sure i ever paid attention to whether or not it worked for the second sip peer |
| 22:04.34 | fenrus | florz, sometimes i just try to get people to realize that they need to explain their problems instead of yelling for help. |
| 22:04.40 | exothermc | citywok: and it actually works for SIP/10061 not just SIP/1006? |
| 22:04.42 | iluminator101 | i am not going buy that |
| 22:04.48 | iluminator101 | there is a way to do it for free |
| 22:05.16 | exothermc | iluminator101: freeswitch I think does. |
| 22:06.05 | exothermc | iluminator101: although your time probably isn't very valuable if would pick that route over paying for it. |
| 22:06.41 | exothermc | [TK]D-Fender: Anyway to subscribe to the status of more than one peer/channel? |
| 22:07.18 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:09.11 | *** join/#asterisk megalomano (~klonstein@38.104.248.2) |
| 22:09.19 | megalomano | hi people |
| 22:09.48 | *** join/#asterisk boodu (~antoine@175.158.129.128) |
| 22:09.56 | boodu | hello |
| 22:10.20 | exothermc | is anyone here a moderator on the users mailing list? |
| 22:10.43 | madler1 | Hi. Just wanted to re-post my T38 issue. The issue I am having is this one Unsupported SDP media type in offer: image 4389 udptl t38 |
| 22:10.56 | madler1 | my full debug is here |
| 22:10.57 | madler1 | http://pastebin.com/1N9Ggyvq |
| 22:11.08 | [TK]D-Fender | iluminator101: Not in Asterisk. Everything else is third party, and I don't know of any free |
| 22:11.16 | madler1 | anyone familiar with the issue? |
| 22:11.26 | megalomano | i need some help , in my asterisk debug cli , appears , "Forbidden" from '"asterisk" , and inmediatly show : SIP/klon-08452280 is circuit-busy |
| 22:11.42 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:11.56 | [TK]D-Fender | megalomano: Bad auth |
| 22:11.58 | citywok | exothermc: yes, the & works |
| 22:12.10 | boodu | do you now if there are template of asteriskNow or elastix or other to openVZ anywhere |
| 22:12.17 | exothermc | citywok: what, ok now I need to figure out why it didn't for me. |
| 22:12.19 | citywok | exothermc: i just went and tested it out. attendant console bound to 1593, hint is 1593&15931, and i made a call from 15931 and it shows 1593 in use |
| 22:13.25 | hardwire | sigh. |
| 22:13.32 | hardwire | it's April.. yo.. why the snow? |
| 22:13.35 | hardwire | srsly. |
| 22:13.41 | hardwire | quit fluffing up my pretty trees. |
| 22:14.39 | megalomano | [TK]-Fender: this account refers to an endpoint , the auth fails by ? |
| 22:14.53 | [TK]D-Fender | megalomano: Go look at a complete call |
| 22:15.36 | exothermc | citywok: looks like that isn't going to work for me |
| 22:15.43 | exothermc | I'm running 1.6.2.x |
| 22:15.52 | exothermc | I'm running 1.6.2.6 |
| 22:15.59 | exothermc | very odd |
| 22:16.14 | citywok | 1.6.1.18 |
| 22:16.23 | citywok | though when i wrote all that code it was probably 1.6.1.5 |
| 22:16.45 | iluminator101 | what do they mean by allows one channel |
| 22:16.55 | boodu | there are no pre-created template with asterisk to openVZ ? |
| 22:16.58 | citywok | one concurrent call limit? |
| 22:17.15 | [TK]D-Fender | iluminator101: 1 channel = 1 call |
| 22:17.17 | citywok | wait, i think that is "concurrency limit reached" |
| 22:17.26 | exothermc | citywok: so just to confirm you have exten => 1593,hint,SIP/1593&SIP/15931 and when 15931 notifies it sends to the 1593 subscription? |
| 22:17.35 | citywok | yes |
| 22:17.42 | citywok | and the attendant console is watching 1593 |
| 22:18.10 | citywok | which is an Aastra 6757i CT with the 60 line thing |
| 22:18.13 | exothermc | hmm I wonder if there is a difference in the way polycom does it. |
| 22:18.33 | exothermc | well probably not since I don't even see the notification leaving asterisk. |
| 22:18.57 | citywok | do you have your phoen bound to subscribecontext in sip.conf? |
| 22:19.13 | citywok | i have subscribecontext = autogenerated-subscribecontext |
| 22:19.18 | exothermc | citywok: no |
| 22:20.13 | citywok | exothermc: read http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 22:20.19 | citywok | look at the stuff under Standard Priorities |
| 22:21.26 | exothermc | citywok: ya these are all under the default context so I'm not sure that applies |
| 22:21.55 | citywok | yea i couldn't tell you since i only do it for the main reception console. |
| 22:22.33 | exothermc | citywok: ya same use case here as well. |
| 22:22.56 | citywok | oh, then just do what i did |
| 22:23.02 | citywok | subscribecontext on the reception extension |
| 22:23.07 | citywok | create a context with all the hints |
| 22:23.13 | *** join/#asterisk Yudaisrael1984 (~Yuda@80.179.161.1.static.012.net.il) |
| 22:23.47 | exothermc | citywok: but how does that solve the problem I have where the first channel is working but the second is not? |
| 22:24.05 | citywok | it works in my case, maybe it will work in yours? lol |
| 22:24.10 | megalomano | <PROTECTED> |
| 22:24.10 | megalomano | <PROTECTED> |
| 22:24.12 | citywok | idk, food for thought. |
| 22:24.23 | megalomano | its the error |
| 22:24.24 | exothermc | citywok: ya worth a try I guess. |
| 22:24.31 | megalomano | in my asterisk debug |
| 22:25.09 | *** join/#asterisk nucleo (nucleo@fedora/nucleo) |
| 22:26.20 | beefpastry | having trouble coming up with the right dtmf recipe...anybody have any experience/advice to share? |
| 22:26.36 | ChannelZ | More salt. |
| 22:26.52 | beefpastry | (Asterisk 1.6 + Polycoms) |
| 22:27.01 | p3nguin | megalomano: That doesn't look like sip debug to me. |
| 22:27.33 | ChannelZ | beef: Something wrong with rfc2833? |
| 22:29.13 | Yudaisrael1984 | hi guys quick question in the sip.conf file how can i add under a trunk a range of authenticated ip addresses in example permit 102.120.120.0/24 |
| 22:29.28 | exothermc | citywok: ya no luck. Just really weird. |
| 22:29.43 | citywok | yea i'm not sure then. i use vanilla 1.6.1.18 and it works for me |
| 22:29.54 | citywok | try going down to 1.6.1 and see if it works. maybe it's a bug regression |
| 22:30.28 | p3nguin | yudaisrael1984: deny=0.0.0.0/0.0.0.0 and permit=102.120.120.0/255.255.255.0 |
| 22:30.38 | exothermc | citywok: 1.6.2 you mean? |
| 22:31.02 | beefpastry | I've tried a little bit of everything...I can get it somewhat working with polycoms on inband (voicemail works fine, remote IVRs work ~75% of the time) (trunks set to rfc2833). Right now, rfc2833 on polycoms works great on remote IVRs but can't get in to voicemail. |
| 22:31.03 | citywok | you said you use 1.6.2. i use 1.6.1.18 and it works for me. try my build and see if it's a config issue, or a regression. |
| 22:31.21 | [TK]D-Fender | megalomano: the other side is refusing you |
| 22:31.28 | exothermc | citywok: oh ya that is an idea. |
| 22:31.40 | megalomano | p3nguin: this looks in the cli , the complete debug is too large |
| 22:32.07 | p3nguin | megalomano: I guess you'll need to figure out how to increase your scroll buffer and copy the whole thing. |
| 22:32.28 | p3nguin | megalomano: Everyone else has to do this, so don't feel like this is a special case. |
| 22:32.50 | beefpastry | I'm extremely close to ditching asterisknow...this and fax issues are getting on my nerves (other than that, everything works great!) |
| 22:32.58 | megalomano | [TK]D-Fender: right , thanks |
| 22:33.27 | citywok | i'm extremely close to ditching the Digium AA50 pile of flaming dog poop |
| 22:34.30 | chazzam | citywok: you should contact digium support for the aa50 |
| 22:34.59 | citywok | chazzam: their solution was "we're actively working on the bug that causes your device to blow up randomly" -- since november. |
| 22:35.29 | citywok | you don't need that CF card to be read/write so people can leave voicemails or do anything. read only mode should be fine. just reboot it! |
| 22:35.30 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 22:36.29 | idespinner | AA50 has plenty of problems as is |
| 22:36.52 | citywok | if you know asterisk for the most part you can make it do whatever you want |
| 22:37.20 | citywok | what drives me nuts is the callerid not working right all the time, and the CF lockup that happens every week or two |
| 22:37.29 | idespinner | CF has given a few people I know plenty of problems |
| 22:37.36 | idespinner | which brand CF do you have? |
| 22:37.48 | citywok | but since i didn't want to spend hours giving them more debug info then i did they weren't interested in fixing the bug. |
| 22:37.59 | citywok | whatever came with it |
| 22:38.12 | idespinner | your best bet is some sorta nighlty reboot script |
| 22:38.25 | citywok | yea except that causes dynamic queue members to be removed |
| 22:38.28 | citywok | so then i have to fix that |
| 22:38.52 | chazzam | add a script for it and make it run on start up in rc? |
| 22:39.01 | citywok | how about they get a new driver for the stupid CF card? |
| 22:39.11 | citywok | "oh we are working on feature enhancements so we won't release a new build until we are done with that" |
| 22:39.22 | citywok | oh, that makes sense. everybody has a crashing device because you are working on feature enhancements. wtf?!? |
| 22:39.48 | citywok | not to mention digium support takes 4 days to reply, and it's always in the middle of the night so you can't reply back with answers to questions until the next day. |
| 22:40.04 | citywok | and people pay for this support? wow |
| 22:40.06 | chazzam | you could ask to work with the day shift |
| 22:40.16 | citywok | if you have the low level support you only get email |
| 22:40.19 | chazzam | or night shift, depending on your timezone |
| 22:40.19 | citywok | and the day shift doesn't do email |
| 22:40.35 | citywok | i bitched enough they let me talk to a human |
| 22:40.42 | idespinner | citywok, live support on the AA50 wont get you much further... i know |
| 22:40.44 | citywok | who was very helpful and helped me out immensely with questions i had |
| 22:41.02 | citywok | but the answer "we're adding features so no new firmware until that is done" just pisses me off |
| 22:41.12 | idespinner | its all hardware issues |
| 22:41.20 | citywok | we're sorry your car drives out of control, we're busy redesigning the stereo and until that is done we wont release a fix. |
| 22:41.25 | exothermc | citywok: you have your subscribecontext in general or in the peer ? |
| 22:41.31 | citywok | in the peer |
| 22:41.37 | citywok | i dont want 200 peers all subscribing for no reason |
| 22:42.36 | exothermc | citywok: well it doesn't make them subscribe just allows them too. |
| 22:42.52 | citywok | i guess that makes sense. lol |
| 22:43.29 | exothermc | citywok: so you have a context that only has hints, or does it have more? |
| 22:43.39 | citywok | only hints |
| 22:43.53 | citywok | it's auto-generated by PHP and contains every extension in the company |
| 22:44.12 | exothermc | citywok: ya odd that completely breaks it for me. I'm guessing I'm doing something wrong. |
| 22:45.34 | bmoraca_work | is there any chance that DTMF not being detected by one number when every other number tried works is an issue on the PBX end? |
| 22:46.12 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 22:52.00 | hardwire | bmoraca_work: depends on if the one number goes to a similar CO than another that works |
| 22:52.35 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 22:52.53 | bmoraca_work | hardwire: well, the support 800 number for the company works, but DTMF is not transfered when that company dials in |
| 22:54.08 | hardwire | whats the scenario.. 800 DID calling into a PBX? |
| 22:54.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:54.25 | bmoraca_work | sort of |
| 22:54.29 | hardwire | do tell |
| 22:54.58 | bmoraca_work | it's a toll-free number that records the callers name and dials a list of numbers in order and asks if they want to accept a call...that call doesn't detect the DTMF |
| 22:55.20 | bmoraca_work | however, when I call the support number for the company that provides that service, DTMF IS passed |
| 22:55.48 | hardwire | so an external service that handles an 800 number dials your PBX at some point |
| 22:56.07 | bmoraca_work | yep |
| 22:56.14 | hardwire | and you have several of those? |
| 22:56.24 | bmoraca_work | nope, just this one |
| 22:56.43 | hardwire | and the 800 number calls several extensions on your PBX? |
| 22:57.10 | bmoraca_work | yes, and none of them can respond |
| 22:57.25 | hardwire | respond being pass DTMF to the caller? |
| 22:57.28 | bmoraca_work | yes |
| 22:57.47 | hardwire | have you tested to see if DTMF is being transmitted by the PBX? |
| 22:57.50 | bmoraca_work | it's not a * code, so it's not features.conf getting in the way...it's just "press 1 to accept" |
| 22:57.51 | bmoraca_work | it |
| 22:57.51 | hardwire | by/from |
| 22:57.53 | bmoraca_work | is |
| 22:58.25 | hardwire | so * passes DTMF back to whatever channel the 800 number uses to reach *.. tested and verified |
| 22:58.50 | bmoraca_work | it's RFC2833 start to finish, works on every number except these inbound calls |
| 22:59.18 | hardwire | so the 800 number provider calls several extensions on your PBX via SIP? |
| 22:59.28 | bmoraca_work | yes |
| 22:59.32 | hardwire | thats strange |
| 22:59.40 | bmoraca_work | well, it dials one DID and I have it ring multiple peers |
| 22:59.49 | hardwire | why doesn't the 800 number come into asterisk first and then asterisk does the queue management. |
| 22:59.51 | bmoraca_work | peer=extension right now (stfu p3nguin) |
| 23:00.01 | hardwire | ok.. I see |
| 23:00.17 | bmoraca_work | this is a thirdparty service that the customer has been using before i started providing them SIP trunking |
| 23:00.24 | hardwire | I don't know if app_queue blocks DTMF |
| 23:00.29 | hardwire | if you're using it |
| 23:00.33 | bmoraca_work | no queues |
| 23:00.45 | bmoraca_work | just multiple SIP extensions in the Dial |
| 23:00.57 | hardwire | ah |
| 23:01.02 | hardwire | whats your dial string? |
| 23:01.38 | bmoraca_work | Dial(SIP/20&SIP/30|5|tr) |
| 23:01.54 | hardwire | add T to the flags |
| 23:02.15 | hardwire | I don't know if that would help |
| 23:02.24 | bmoraca_work | T? why would i want to allow the inbound caller to transfer calls? |
| 23:02.25 | hardwire | but it would keep asterisk in the media path |
| 23:02.32 | hardwire | try it out :P |
| 23:02.36 | bmoraca_work | any dial flags keeps asterisk in the media path |
| 23:02.40 | hardwire | ah ok |
| 23:02.59 | bmoraca_work | canreinvite=no does, too, and that's configured as well |
| 23:03.06 | hardwire | ah ok |
| 23:03.12 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:03.26 | hardwire | so you're seeing out of band DTMF go back to the provider? |
| 23:03.40 | bmoraca_work | yes |
| 23:03.50 | hardwire | then yeh.. call them |
| 23:04.01 | bmoraca_work | (i'm the provider) |
| 23:04.06 | bmoraca_work | not the toll-free provider |
| 23:04.09 | bmoraca_work | but the SIP provider |
| 23:04.09 | hardwire | did you try calling yourself? |
| 23:04.22 | hardwire | I'm talking about the TF provider |
| 23:04.27 | bmoraca_work | as I said, DTMF works in every other situation |
| 23:04.29 | bmoraca_work | ahh, yeah |
| 23:04.30 | hardwire | if you see DTMF leaving and going to their network... then yeh. |
| 23:04.34 | bmoraca_work | i've told the customer to call them |
| 23:04.55 | hardwire | well you said DTMF works in all other situations.. are there other situations with this specific configuration with that specific provider? |
| 23:05.05 | bmoraca_work | nope |
| 23:05.08 | hardwire | ah ok |
| 23:05.08 | bmoraca_work | just this one |
| 23:05.18 | hardwire | yeh.. did you say they didn't allow DTMF? |
| 23:05.40 | bmoraca_work | didn't allow? they have to allow it. it's just not being detected by their system for some reason |
| 23:07.12 | hardwire | strange |
| 23:07.17 | bmoraca_work | my customer's telling me "fixit fixit fixit" and i have to tell them "i can't because it's not my problem" |
| 23:07.26 | hardwire | so the client had the TF provider point to your switch instead of their old one right? |
| 23:07.51 | hardwire | I'd temporarily switch to ULAW with the TF provider and use inband. |
| 23:08.00 | hardwire | it's really hard to block inband :) |
| 23:08.05 | bmoraca_work | call's coming in over the PSTN |
| 23:08.08 | bmoraca_work | a PRI |
| 23:08.18 | hardwire | erm. |
| 23:08.30 | hardwire | wot? |
| 23:08.30 | bmoraca_work | toll-free provider forwards to their DID |
| 23:08.35 | hardwire | ahha' |
| 23:08.43 | hardwire | well suck an egg.. I dunno |
| 23:08.49 | hardwire | I'd have to play |
| 23:08.50 | bmoraca_work | that's my feeling |
| 23:09.06 | bmoraca_work | i played for a while today |
| 23:09.09 | hardwire | it seems weird that if you can see DTMF to/from the * box and to/from the PRI then you're golden. |
| 23:09.13 | bmoraca_work | and i can't figure the shit out |
| 23:09.22 | hardwire | turn on PRI span debugging |
| 23:09.25 | hardwire | I think it shows DTMF |
| 23:09.35 | hardwire | somebody correct me if I am a mistaken man. |
| 23:11.24 | bmoraca_work | not at q931 level |
| 23:11.40 | bmoraca_work | i didn't debug q921, though |
| 23:12.49 | bmoraca_work | additionally, the entire sip media path is ulaw |
| 23:13.01 | bmoraca_work | i should try switching to inband, but that's gonna suck |
| 23:13.42 | hardwire | erm.. if you're not sending anything over the PRI then wth. |
| 23:14.02 | hardwire | checks something |
| 23:14.21 | bmoraca_work | there's two SIP hops from the PRI to the customer's PBX and then one more hop to their phone |
| 23:14.29 | WIMPy | There's no DTMF on PRI. |
| 23:14.53 | WIMPy | Or make that only onband or dialling onformation. |
| 23:15.19 | hardwire | WIMPy: was hoping span debug would reveal DTMF transmission |
| 23:15.34 | hardwire | I tested it.. it doesn't |
| 23:16.12 | hardwire | bmoraca_work: you could always record the call and see if you can see DTMF :) |
| 23:16.29 | WIMPy | When you do out of band DTMF on SIP, you have to generate inband DTMF on Asterisk. |
| 23:16.31 | bmoraca_work | i don't think i can record calls on my AS5400 |
| 23:16.46 | bmoraca_work | my media gateway is an AS5400 |
| 23:16.59 | bmoraca_work | asterisk never directly interacts with the PRI. |
| 23:18.20 | *** join/#asterisk jsjc (~chatzilla@219-90-227-77.ip.adam.com.au) |
| 23:18.22 | WIMPy | ok, s/Asterisk/gateway/ |
| 23:18.36 | bmoraca_work | right |
| 23:18.46 | jsjc | I just reinstalled asterisk and.. looks like chan dahdi is not loading because in CLI> do not have dahdi channel why this could happen? |
| 23:19.11 | bmoraca_work | jsjc: your dahdi config is screwed up |
| 23:19.20 | *** join/#asterisk punkgode (~punkgode@72.249.66.65) |
| 23:19.25 | bmoraca_work | jsjc: what's the output of "dahdi_cfg -vvvvv" |
| 23:20.47 | *** part/#asterisk madler1 (~madler@c-76-98-125-204.hsd1.nj.comcast.net) |
| 23:20.50 | WIMPy | jsjc: Try to manually load chan_dahdi and see, what it tells you. |
| 23:21.15 | jsjc | Command 'module load chan_dahdi.so ' failed. |
| 23:22.13 | chazzam | do you have verbosity turned up at all? |
| 23:23.23 | chazzam | if you have the full log enabled, you can problably get a better explanation there |
| 23:28.29 | *** join/#asterisk Znuff (ibm86@89.45.83.253) |
| 23:30.01 | Znuff | Hi. Excuse me if this is the wrong channel to ask this, but I seriously depleted all my google resources so far in asking this question: Does anyone know of a VoIP provider that would give you more than 30 concurent lines and eventually allowing you to hide your caller id, plus, you can buy credit online? |
| 23:32.07 | fenrus | Znuff, in what part of the world do you reside? |
| 23:32.17 | Znuff | eastern europe, romania |
| 23:32.33 | *** join/#asterisk slinksh0t (~slinksh0t@c-76-108-31-194.hsd1.fl.comcast.net) |
| 23:33.42 | *** join/#asterisk trebork2010 (~trebork20@75-146-220-166-Philadelphia.hfc.comcastbusiness.net) |
| 23:34.24 | fenrus | Znuff, http://www.phonecardmall.com/call/romania.html |
| 23:34.27 | fenrus | perhaps |
| 23:34.45 | Znuff | Oh, I'm not interested in calling romanian phone numbers :) |
| 23:34.50 | Znuff | But thanks for the link |
| 23:36.13 | Znuff | I pretty much want a sip trunk |
| 23:36.38 | trebork2010 | hello all sorry to have to ask but I can't find help elsewhere , I am trying to setup hotdesking as described here http://www.orderlyq.com/asteriskqueues.html |
| 23:37.00 | trebork2010 | I can't seem to get the login and logout extensions to register at all |
| 23:37.03 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 23:38.01 | fenrus | add some debugging |
| 23:38.35 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 23:40.34 | TJNII | Okay, I just want to make sure I'm not completely off base here. Let's say your job for the day is to inspect and, if needed, repair a circuit board assembly. On multiple occasions you've been told that defect Y is the number one failure of this particular board. Would you check for defect Y in your inspection? |
| 23:40.59 | Znuff | Uhm, yes |
| 23:41.08 | Znuff | First thing I'd check |
| 23:41.14 | TJNII | Okay, just checking. |
| 23:41.17 | fenrus | =) |
| 23:41.24 | Znuff | yup, you're sane |
| 23:41.26 | TJNII | That concept seems to be eluding one of my coworkers... |
| 23:41.33 | fenrus | it would be the logical way to proceed. |
| 23:41.37 | Znuff | Cow, workers |
| 23:41.51 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 23:41.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.14 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 23:42.24 | cosmicwombat | anyone interested in helping me with DAHDI ? |
| 23:43.13 | TJNII | Eh, not really. But if you pastebin some debug you might get some help. |
| 23:43.14 | fenrus | TJNII, set up a fault-trace routine |
| 23:43.19 | WIMPy | cosmicwombat: Unlikely, unless you tell us whatyour problem is. |
| 23:43.36 | TJNII | fenrus: You mean a software test? |
| 23:43.51 | fenrus | TJNII, well, that is a start.. |
| 23:44.12 | TJNII | fenrus: It is intermittant electronically, but easy to spot visually. |
| 23:44.13 | fenrus | TJNII, but some "check-this-first" stuff. |
| 23:44.39 | fenrus | like "measure this, and if the voltage is less than ... it needs to be rebuilt" |
| 23:45.15 | TJNII | Yea, this particular failure really doesn't lend itself to that. It is a mechanical problem on the board. |
| 23:45.25 | cosmicwombat | My DAHDI issue is that I am running Asterisk 1.4.30 and DAHDI 2.3.0 and when I change the system.conf to reflect my needs... dahdi stopped loading. It is running, but not in Asterisk |
| 23:45.29 | fenrus | oh, that's a shame |
| 23:45.34 | joobie | for those interested |
| 23:45.40 | TJNII | If it is bad enough it will break the circuit, though most times it will work but is not shippable. |
| 23:45.47 | joobie | just bumped into a perl module that interfaces with the asterisk AMI |
| 23:45.58 | cosmicwombat | I will pastebin the system.conf |
| 23:46.10 | fenrus | TJNII, what kind of board is it? :) |
| 23:46.21 | joobie | seems to be very straight forward to use |
| 23:46.27 | TJNII | fenrus: I'd really love to brag, but I signed a NDA. |
| 23:46.31 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 23:46.39 | fenrus | TJNII, naw, that's too bad. :/ |
| 23:49.11 | cosmicwombat | My system.conf looks like http://pastebin.com/srBdBMMR |
| 23:51.31 | trebork2010 | anybody know where I caould hire somebody to help me get through this setup |
| 23:52.38 | miamiseb | Gnight all! |
| 23:54.31 | WIMPy | cosmicwombat: Apart from timing I don't spot anything obvious. What does dahdi_cfg -v say? |
| 23:54.51 | WIMPy | trebork2010: Maybe you should specify a location. |
| 23:55.04 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 23:55.42 | devmod | hello, where would I go to get started witha local phone number? |
| 23:56.27 | TJNII | ~itsplist-us |
| 23:56.27 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 23:56.42 | TJNII | Or an ATA... |
| 23:56.57 | TJNII | Though I think a SIP account would be easiest. |
| 23:57.43 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 23:58.36 | cosmicwombat | WIMPy: Just sec, I may have figure something out... |
| 23:59.00 | Jumpie | what dose it mean if a handset is 'hdsp ready' ? |
| 23:59.07 | Jumpie | requires some kinda extra something added? |