| 00:00.18 | luch | and if i type them in really really fast it actualy works |
| 00:03.06 | luch | so i have about one second to type in my extension and thats it |
| 00:23.30 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:25.17 | *** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net) |
| 00:37.02 | andresmujica | done. :) now.. why on earth i don't have the COMPLETECALLER logged.. :/ checking... |
| 00:39.22 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 00:47.06 | *** join/#asterisk davevg (n=davevg-b@75.97.64.33) |
| 00:48.17 | *** join/#asterisk devyll (n=paul@89.36.24.2) |
| 00:49.31 | devyll | are Hangup Hangup() Hungup and Hungup() all correct and do the same thing ? (terminates the connection/channel ) (ex: s,5,Hungup ) ? |
| 00:53.17 | WindowsUser | iono mostp eople use Hangup() |
| 00:56.50 | *** join/#asterisk neoalex (n=chatzill@cpe-72-225-191-51.nyc.res.rr.com) |
| 00:57.31 | neoalex | hello I need some help setting up voice mail to e-mail which was working fine up until a few months ago |
| 00:58.18 | neoalex | the problem I'm having is that now I'm trying to set it up to send using ssmtp |
| 00:58.34 | neoalex | ssmtp works fine when testing from command line |
| 00:58.43 | neoalex | asterisk doesn't send anything |
| 00:58.57 | neoalex | is there any logs I could look in to see if it's even trying? |
| 00:59.07 | neoalex | are* there |
| 01:00.29 | WindowsUser | change mailcmd to be cat > /tmp/file |
| 01:00.44 | WindowsUser | and see if anything makes it there? |
| 01:00.58 | WindowsUser | or try turning up debug really high and see if that help |
| 01:01.25 | neoalex | I tried core set verbose 1000 that didn't help |
| 01:01.37 | neoalex | let me try the dumping the mail like you suggested |
| 01:03.01 | WindowsUser | if the mailcmd sends anything to STDERR it shows up when I go asterisk -d |
| 01:07.16 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 01:07.39 | Lantizia | Hey I've upgraded our phones to the latest version 7 snom firmware but now even tho all the phones are registered no calls can be made... anyone heard about that? |
| 01:08.12 | Lantizia | google gives me less than nothing on this one, we were on 7.1.30 which worked ok, but 7.3.23 is what they're on now |
| 01:19.55 | *** join/#asterisk sacitec (n=sacitec@189.129.249.246) |
| 01:21.23 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 01:22.41 | neoalex | ok this is a dumb question... I have asterisk started with asterisk -f |
| 01:22.46 | neoalex | how do I kill it? |
| 01:22.55 | neoalex | without it restarting |
| 01:24.40 | hardwire | ctrl-c |
| 01:24.46 | hardwire | do I win? |
| 01:25.24 | neoalex | no |
| 01:25.29 | sacitec | hello, i'm testing the beta version of chan_skype with asterisk 1.4.x. I'm following the pre-configured .conf file to set up my existing(and non logged) skype account, that i want to use as test scenario. When i load try to load skype user from CLI, i get this debug(http://pastebin.com/m76f630a5), and never get logged in. Anyone has succedes using chan_skype ? All comments are welcomed =) |
| 01:25.35 | hardwire | neoalex: you want it to die right? |
| 01:25.39 | hardwire | but you don't want it to.. restart? |
| 01:25.47 | neoalex | youp |
| 01:25.52 | hardwire | so you want it dead? |
| 01:25.56 | neoalex | youp |
| 01:25.58 | hardwire | init 0? |
| 01:26.18 | neoalex | thanks, very helpful |
| 01:27.02 | hardwire | well if you can connect to it via asterisk -r |
| 01:27.05 | hardwire | you can run shutdown now |
| 01:27.43 | [TK]D-Fender | <PROTECTED> |
| 01:27.51 | [TK]D-Fender | sacitec: want another? ;) |
| 01:27.59 | neoalex | I can, but no matter how I stop it, it restarts the process |
| 01:28.09 | sacitec | que ? |
| 01:28.10 | hardwire | [TK]D-Fender: luser-base is an RBL right? |
| 01:28.11 | hardwire | :) |
| 01:28.21 | [TK]D-Fender | neoalex: Sounds like you're running safe_asterisk |
| 01:28.32 | sacitec | y lo que quisiste decir es..? |
| 01:28.39 | neoalex | the process only says asterisk -f |
| 01:28.53 | neoalex | I think it's the startup script that starts it when the machine first boots up |
| 01:28.58 | file | sacitec: you can't use your existing account, you have to use one created from the business control panel |
| 01:29.06 | [TK]D-Fender | neoalex: As launched by an init using safe_asterisk no doubt. Look for THAT process |
| 01:29.21 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-80.tricom.net) |
| 01:30.10 | neoalex | no results for ps -ax | grep safe_asterisk |
| 01:30.22 | sacitec | file: you mean the control panel that comes when you first install skype on you GUI system, or it's a special version ok skype ? |
| 01:30.24 | [TK]D-Fender | neoalex: Stop grepping and look at EVERYTHING |
| 01:30.36 | [TK]D-Fender | neoalex: * does not restart itself. Some other script is diong that |
| 01:30.44 | file | sacitec: it's a web based interface on the skype website, I believe this is in the documentation |
| 01:30.59 | neoalex | I know but the only asterisk process I see says asterisk -f |
| 01:31.51 | sacitec | file: thanks a lot ! |
| 01:32.33 | [TK]D-Fender | neoalex: passtebin the entire thing |
| 01:33.38 | neoalex | http://pastebin.com/m4439e75 |
| 01:34.01 | sacitec | file: have you got success using this chan ? |
| 01:34.23 | file | I haven't touched it yet, but others have |
| 01:34.38 | hardwire | :D |
| 01:34.48 | WindowsUser | the skype for asterisk? |
| 01:34.55 | hardwire | I'm going to try it out this weekend. |
| 01:35.02 | hardwire | just to miff [TK]D-Fender |
| 01:35.06 | WindowsUser | works for me on 1.4.26 and svn checkout of 1.6.1 |
| 01:36.12 | WindowsUser | hi dont pm me, i rarely notice them |
| 01:36.15 | [TK]D-Fender | neoalex: Go hunt down your init script and see what its calling |
| 01:37.05 | neoalex | trying to find it now, that's another problem I can't remember who's calling it |
| 01:39.10 | manxpower | Personally, I'd rather not have a way for Skype users to call me. I try to use stuff that uses open standards like SIP. |
| 01:43.37 | jaytee | only thing more pesky than Skype users are those damn Philistines! |
| 01:43.57 | sacitec | WindowsUser: did you get a successfully connection with skype user ? |
| 01:46.39 | WindowsUser | yea |
| 01:51.30 | sacitec | uhmm, any clue why i'm getting my debug and unable to connect ? http://pastebin.com/m76f630a5 |
| 01:51.53 | sacitec | password is correct |
| 01:51.59 | *** part/#asterisk manxpower (n=eric@84.sub-70-222-54.myvzw.com) |
| 01:54.11 | WindowsUser | thats a lot of gibberish |
| 01:55.08 | WindowsUser | what debug level is that? |
| 01:55.32 | WindowsUser | if you do "skype show users" are you logged in or out? |
| 01:57.10 | *** join/#asterisk Alfio (n=Amunoz@190.94.56.23) |
| 01:58.15 | sacitec | the higgest |
| 01:58.19 | sacitec | logged out |
| 01:58.35 | WindowsUser | did you create the account via the BCP on skype.com? |
| 02:00.56 | WindowsUser | skype is being very particular on that point |
| 02:07.05 | *** join/#asterisk BadHAL (n=nn@70-5-70-9.pools.spcsdns.net) |
| 02:07.27 | neoalex | [TK]D-Fender: I finally found it |
| 02:07.40 | neoalex | I have it in /etc/inittab respawn |
| 02:10.30 | [TK]D-Fender | EW |
| 02:10.46 | [TK]D-Fender | neoalex: This is not the kind of thing you spawn in there |
| 02:11.50 | neoalex | yeah I know, it was probably 3 AM 2 years ago and I couldn't get the debian script to run properly, so this was just a dirty suppose to be temporary fix |
| 02:12.28 | neoalex | anyway, I've commented it out and did an init 2 again but it's still respawning |
| 02:13.06 | WindowsUser | read the manpage for init, you need to -HUP it or something |
| 02:13.09 | neoalex | any other way besides restarting the server because I'm sure fsck is hell bent on spending an hour to check my disk |
| 02:13.12 | WindowsUser | kill -HUP init |
| 02:16.05 | neoalex | kill -HUP 2 right? |
| 02:16.08 | neoalex | still respawning |
| 02:17.14 | neoalex | got it... init q |
| 02:20.08 | WindowsUser | horray for man pages |
| 02:20.23 | neoalex | ok, all this was so I can start asterisk -d |
| 02:20.30 | neoalex | but still nothing when I leavea a voicemail |
| 02:20.45 | neoalex | and it did write the voicemail to the /tmp file |
| 02:20.54 | neoalex | well the email I mean |
| 02:22.51 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:23.08 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:24.22 | neoalex | does anyone tried using ssmtp for voicemail to e-mail before? |
| 02:30.13 | WindowsUser | ssmtp < /tmp/file |
| 02:30.35 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-45.tricom.net) |
| 02:30.44 | WindowsUser | well, sendmail -t < /tmp/file i guess |
| 02:31.05 | WindowsUser | damn thing should give some kind of error, or maybe the to address is wrong? |
| 02:33.49 | joako | Anyone has used Asterisk with bluetooth mobiles? What is the best way to get it running from scratch (channel driver, asterisk version, etc)? |
| 02:34.06 | *** join/#asterisk anthm][ (n=anthm@CPE-72-128-94-253.wi.res.rr.com) |
| 02:34.31 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 02:34.33 | WindowsUser | chan_mobile is in asterisk-addons i thought |
| 02:34.49 | WindowsUser | get 1.6.1.1 or 1.6.1 svn if you want to try out skype as well |
| 02:35.45 | WindowsUser | hrm i might have a svn checkout of asterisk-addons as well, odd |
| 02:36.12 | joako | WindowsUser: Ok... I will see about that I haven't tried 1.4 yet. I just recall their being various implementations |
| 02:36.35 | joako | That also reminds me has anyone been able to link the Windows Live phone service to Asterisk? I am told it is SIP |
| 02:36.38 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:36.55 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:37.27 | WindowsUser | i dont know anything about that actually |
| 02:37.35 | WindowsUser | is that part of live messenger? |
| 02:41.13 | grandpapadot | Hey guys, I have a debian 5.02 system running asterisk 1.4.26. When transcoding ulaw->g726 I get garbled audio, transcoding everything else seems to work fine, any suggestions? |
| 02:41.56 | joako | Are you using Sipura or Linksys devices? |
| 02:42.27 | WindowsUser | g726 is DECT |
| 02:42.32 | grandpapadot | asterisk->asterisk |
| 02:42.53 | grandpapadot | seems better with g726aal2 |
| 02:44.04 | joako | Not sure what to say I've had issues with Linksys phones and ATA that do support "G726" I believe there are some settings in codecs.conf I would check that and have you tried any other "compressed" codec e.g. gsm? |
| 02:44.17 | grandpapadot | Yea, gsm works fine.. hrm... |
| 02:44.36 | WindowsUser | you're going from ulaw to g726 in asterisk |
| 02:44.41 | WindowsUser | but where does the call end up? |
| 02:45.02 | grandpapadot | polycom(ulaw)->asterisk->g726->asterisk |
| 02:45.40 | WindowsUser | and the second asterisk plays to chan_oss or something? |
| 02:46.16 | joako | Personally I would use G729 on the polycoms and between asterisk machines.... I don't believe there is another common codec between the two |
| 02:46.57 | grandpapadot | g729 passthrough works fine |
| 02:47.08 | grandpapadot | hrm... |
| 02:49.00 | joako | Although I can't recall recent compatibility issues between dissimilar Asterisk versions, what versions are the 2 machines? |
| 02:49.10 | grandpapadot | 1.4.26 on both .. |
| 02:49.16 | grandpapadot | One is debian 5, the other is debian 4 |
| 02:49.28 | hardwire | hi |
| 02:59.58 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 03:28.02 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 03:38.16 | joako | So chan_mobile only works with CSR chipset bluetooth, not broadcom? |
| 03:39.40 | WindowsUser | i guess? |
| 03:40.04 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-245.tricom.net) |
| 03:40.10 | WindowsUser | Bus 002 Device 002: ID 0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode) |
| 03:40.16 | WindowsUser | I'll believe that |
| 03:40.24 | grandpapadot | Hey guys, I thought ilbc wasn't included in asterisk after 1.2? When I show codecs in 1.4.26, it's there... |
| 03:41.04 | WindowsUser | core show translation |
| 03:41.23 | WindowsUser | prometheus*CLI> core show codecs |
| 03:41.23 | WindowsUser | Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. |
| 03:41.24 | grandpapadot | It's there but with a bunch of '-' |
| 03:41.30 | WindowsUser | the - means its not available |
| 03:41.35 | grandpapadot | ahh |
| 03:42.49 | grandpapadot | Tnx. |
| 03:45.22 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 03:47.18 | joako | WindowsUser: Either way I don't think the mobile I intend to use will work! I was trying to pair it with the loaner car I had and it would pair fine but the car would never consider it connected. |
| 03:47.36 | WindowsUser | what phone? |
| 03:48.25 | WindowsUser | also theres like 2 or more ways for a phone to connect to an audio gateway, theres the dumb headset profile that can answer a call, and a profile that allows dialing |
| 03:48.48 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar) |
| 03:49.34 | joako | Samsung r210... it's a $50 phone for a $30/month unlimited service |
| 03:50.16 | joako | They had the nerve to ask me if I wanted to pay $5/month + $40 deductable for insurance! |
| 03:51.05 | WindowsUser | hahahaha |
| 03:51.22 | WindowsUser | well how long is the contract for the phone? |
| 03:51.32 | WindowsUser | or no contract? ;) |
| 03:51.53 | joako | No contract... I could return the phone to the store and exchange it for another one if it didn't work |
| 03:52.15 | joako | But I would "have" to wait a month since you get the 1st month free when you activate a new line :) |
| 03:52.27 | joako | www.metropcs.com FWIW |
| 03:52.39 | WindowsUser | what country? im in canada |
| 03:52.54 | joako | Country to your north |
| 03:53.09 | WindowsUser | russia? |
| 03:53.18 | WindowsUser | or did you mean south? ;) |
| 03:53.29 | joako | Yes lol |
| 03:59.37 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 04:07.51 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 04:11.58 | *** join/#asterisk joako_ (n=joako@opensuse/member/joak0) |
| 04:12.42 | *** join/#asterisk BeeBuu (n=beebuu@219.135.40.94) |
| 04:14.05 | BeeBuu | what's the command that will play my busy message before record a busy voicemail,voicemail(111@default,b) doesn't work,please help |
| 04:15.51 | WindowsUser | what does it do now? |
| 04:16.18 | *** join/#asterisk joako_ (n=joako@opensuse/member/joak0) |
| 04:16.30 | BeeBuu | it can't play busy message before record |
| 04:16.45 | BeeBuu | and i had make the busy message |
| 04:17.03 | grandpapadot | VoiceMail(b111@default) |
| 04:17.47 | BeeBuu | grandpapadot: i had tried tooï½ï½ |
| 04:18.03 | grandpapadot | That's it. Did you record your busy message? |
| 04:18.19 | [TK]D-Fender | BeeBuu: "core show application voicemail". and show us that you made the recording and the failed attempt to enter the box playing the message |
| 04:20.11 | BeeBuu | [TK]D-Fender: voicemail(111@default|b)? |
| 04:20.26 | [TK]D-Fender | BeeBuu: "," not "|" |
| 04:20.44 | [TK]D-Fender | BeeBuu: and show us your failure |
| 04:20.54 | [TK]D-Fender | eealong with a dump of the VM folder. |
| 04:21.17 | BeeBuu | [TK]D-Fender: thanks.let me try |
| 04:22.24 | BeeBuu | -- Executing [111@sipuser:1] VoiceMail("SIP/1001-081de3f0", "111@default|b") in new stack |
| 04:22.25 | BeeBuu | -- <SIP/1001-081de3f0> Playing '/var/spool/asterisk/voicemail/default/111/temp' (language 'en') |
| 04:23.10 | BeeBuu | [TK]D-Fender: can you see "111@default|b"? but asterisk still playing temp |
| 04:23.32 | BeeBuu | what's problem now? |
| 04:23.53 | grandpapadot | Hey TK, do you know the effective bandwidth usage of asterisk's GSM codec? I know the codec calls for like a 13k bitrate, but I was trying to find the IP overhead. |
| 04:23.54 | [TK]D-Fender | BeeBuu: it will ALWAYS play the temp over any other message <- |
| 04:24.21 | [TK]D-Fender | BeeBuu: that made so you can override yourr normal message while on vacation and not have to re-record your orgianals upon your return |
| 04:24.45 | BeeBuu | [TK]D-Fender: so what can i do next? |
| 04:24.48 | [TK]D-Fender | grandpapadot: 20kbps ~ |
| 04:24.53 | grandpapadot | Thanks. |
| 04:24.56 | [TK]D-Fender | BeeBuu: . DUH... REMOVE IT |
| 04:25.05 | BeeBuu | remove the temp message? |
| 04:25.08 | [TK]D-Fender | YES |
| 04:27.19 | BeeBuu | [TK]D-Fender: Done.it's OK. It mean the temp over other message! no document say this~~~ |
| 04:27.26 | BeeBuu | [TK]D-Fender: thanks. |
| 04:30.08 | grandpapadot | Hey! 1.4.26's moh=files now resumes where you left off, no more madplay, yah!!!! |
| 04:31.28 | *** join/#asterisk joako_ (n=joako@opensuse/member/joak0) |
| 04:36.30 | WindowsUser | hrm |
| 04:36.36 | WindowsUser | does the spa3102 have on hold music? |
| 04:37.25 | [TK]D-Fender | WindowsUser: No, why would it? |
| 04:37.36 | [TK]D-Fender | WindowsUser: Since when does a phone generate its own music? |
| 04:37.55 | WindowsUser | i dunno |
| 04:38.22 | jblack | The spa3102 is a phone? |
| 04:39.08 | [TK]D-Fender | ATA |
| 04:39.10 | [TK]D-Fender | Close enough |
| 04:39.34 | jblack | Ok, if it's anything like the SPA-8K, then it can do music on hold from a stream |
| 04:40.05 | jaytee | true |
| 04:40.12 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 04:41.16 | jblack | [TK]D-Fender: Seems like a reasonable question to me... |
| 04:41.16 | [TK]D-Fender | BRB |
| 04:41.32 | [TK]D-Fender | jblack: Again, what would the phone device generate it? |
| 04:42.17 | jblack | gives [TK]D-Fender a sad, but oddly comforting, look |
| 04:42.17 | [TK]D-Fender | jblack: a PBX yes, but why a phone? SIP phones & ATA's connect to more central system that are in charge of such things |
| 04:42.17 | grandpapadot | g711->gsm hardley has any cpu overhead, g711->ilbc is like 33% (on a 1.6GHz Atom N270) |
| 04:42.29 | jblack | WindowsUser: Check for streaming. It may be able to take music from a stream, and play that. Think "icecast" like stuff. |
| 04:42.37 | [TK]D-Fender | grandpapadot: Low bits are not free bits :) |
| 04:42.43 | grandpapadot | ;) |
| 04:43.01 | jblack | Those devices are a lot more than phones, tk. They're practically a mini-pbx. |
| 04:43.07 | joako_ | jblack: SPA8000 is a gateway only, no? You still need some PBX between that and your other endpoints. Asterisk is a nice opensource PBX that supports MOH from streams :) |
| 04:44.00 | [TK]D-Fender | joako_: He knows these things already.. |
| 04:44.05 | jblack | Nope. Get a pile of sip accounts from various providers, and hook each one up to a different phone, with various customized ringtones, call waiting, three way calling, etc. |
| 04:44.18 | WindowsUser | well i have a spa3102 and when i press flash on a call theres some moh action going on |
| 04:44.25 | jblack | It's not _nearly_ as functional as *. |
| 04:44.29 | WindowsUser | and im not sure how to tell if its asterisk or the unit |
| 04:44.30 | [TK]D-Fender | remembers jblack's start here, along with the curious fellow who recommended the SPA-8000 to him... |
| 04:44.38 | jblack | That was you. :) |
| 04:44.54 | [TK]D-Fender | jblack: OMG! \o/ |
| 04:45.04 | jblack | I've had it for... maybe two years now? |
| 04:45.17 | [TK]D-Fender | jblack: Easily. How's the She-bitch doing? |
| 04:45.46 | jblack | It's still a pita. fiddle in the web interface for too long, and it needs a cold start |
| 04:46.00 | [TK]D-Fender | jblack: I was talking about the Ex ;) |
| 04:46.00 | jblack | My bet is they cheaped out on ram |
| 04:46.15 | joako_ | jblack: Better than grandstream... fiddle with it and you need to toss it out |
| 04:46.30 | grandpapadot | grandscream sucks in every possible way |
| 04:46.47 | [TK]D-Fender | BRB... |
| 04:46.52 | jblack | Let's not play childish "let's call something by a similiar sounding derogatory name". That's so... |
| 04:47.11 | grandpapadot | grandsucks |
| 04:47.16 | grandpapadot | grandgarbage |
| 04:47.24 | grandpapadot | grandcheapassbrokeshit |
| 04:47.49 | *** join/#asterisk akant2 (n=chatzill@70-59-167-73.omah.qwest.net) |
| 04:48.04 | jblack | Dude, you done insulting yourself? |
| 04:48.19 | grandpapadot | grandjblackfanboy |
| 04:48.22 | jblack | Could have just said "old, fat and forgetful" |
| 04:48.27 | grandpapadot | lol |
| 04:48.42 | jblack | I'm not a granstream fan. Never had any of their stuff |
| 04:49.09 | jblack | I bet you picked on your brother's last name when you were a kid. :) |
| 04:49.35 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 04:49.58 | WindowsUser | fender i guess the SPA3102 knows to ask asterisk to put a call on hold? |
| 04:50.18 | [TK]D-Fender | WindowsUser: Naturally |
| 04:54.38 | WindowsUser | that is so cool |
| 04:56.41 | [TK]D-Fender | WindowsUser: ATA is just like any other SIP phone, you just plug the handset onto it |
| 04:59.20 | *** join/#asterisk kihote (n=chatzill@118.69.66.118) |
| 05:04.12 | WindowsUser | yea I'm slowly learning about SIP and asterisk |
| 05:13.16 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 05:15.27 | thehar | grandstream does in fact suck. |
| 05:16.03 | thehar | we use hundreds of 286's and not even grandstream can fix our bugs |
| 05:16.11 | thehar | altho they are bugs related to the sonus switch we use |
| 05:23.43 | [TK]D-Fender | thehar: So what you're saying is you are incapable of buying a quality product? |
| 05:23.44 | [TK]D-Fender | :p |
| 05:28.05 | thehar | heh |
| 05:28.27 | thehar | we are 78% done replacing them with pap2ts which aren't the best thing to replace them with but the cost benefit is much better |
| 05:30.43 | WindowsUser | what'd be the best thing to replace them with? |
| 05:32.21 | neoalex | I still can't get voicemail to e-mail to work, I've replaced ssmtp with msmtp and that works from console fine too, however mailcmd never seems to get called since there's nothing in the log |
| 05:32.48 | neoalex | it's exactly the problem described here: https://issues.asterisk.org/view.php?id=15199 |
| 05:33.04 | thehar | they are a fine ATA |
| 05:37.06 | WindowsUser | neoalex: what user is asterisk running as? |
| 05:41.25 | neoalex | root |
| 05:43.02 | WindowsUser | hrm it must be something simple |
| 05:47.57 | neoalex | could this cause it: Prefixing the mailbox with an option is deprecated ('su8600') |
| 05:48.02 | [TK]D-Fender | neoalex: Someone else's problem doesn't help us. It went in and sat idle for months after 2 days of activity |
| 05:48.42 | [TK]D-Fender | neoalex: And yes, not following instructions when they app tells you to your face should be a shoot-on-sight-offense :p |
| 05:48.58 | [TK]D-Fender | reaches for his ClueBbat (tm) |
| 05:49.13 | WindowsUser | neoalex: but is it putting the voicemail into like /var/spool/asterisk/voicemail/context/8600/INBOX |
| 05:50.37 | neoalex | Yes it does, and not only that, it also writes the email if I use: mailcmd=cat > /tmp/vmtest |
| 05:51.29 | WindowsUser | and then sendmail -t < /tmp/vmtest works? |
| 05:51.41 | neoalex | actually it's putting the voicemail in /var/spool/asterisk/voicemail/default/8600/unavail |
| 05:54.17 | neoalex | trying now |
| 05:56.55 | neoalex | it doesn't send the email no, but I think I see the issue |
| 05:57.10 | neoalex | there's an _ in front of the recepient address in the log for msmtp |
| 05:57.48 | neoalex | I have no _ in front of the email address in voicemail.conf |
| 05:58.06 | neoalex | so I'm have no idea why asterisk is adding that |
| 06:04.31 | *** join/#asterisk darksmurf (n=asdf@166.135.30.88) |
| 06:05.52 | WindowsUser | is the _ in front in the /tmp/vmtest? |
| 06:06.50 | darksmurf | ack! problems. Setup: Asterisk 1.4, on DSL behind NAT with an IPMAP so all traffic seems to come from a specific public IP. Asterisk thinks it has a 192.x.x.x IP. I am trying to use externip to tell it the true pub IP, but when I 'sip debug' the peer I still see the 192.x.x.x IP listed in the INVITE. Is this correct? I am getting a 403 Forbbiden error when trying to connect to a SIP Trunk. (Velocity Networks if anyone has any suggestions about that...) |
| 06:08.03 | neoalex | nope, found the problem, in vmtest there was a space |
| 06:08.10 | darksmurf | I should mention that I think the reason for the 403 is because my provider will not try to connect to a private IP. They want to see my public IP in the invite. Sound plausable? |
| 06:08.17 | neoalex | in voicemail.conf I had a space after the comma |
| 06:08.21 | neoalex | that was the issue |
| 06:08.22 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 06:08.25 | WindowsUser | horray |
| 06:08.28 | neoalex | finally works now |
| 06:08.34 | neoalex | thank you for all your help |
| 06:09.03 | [TK]D-Fender | darksmurf: PASTEBIN your configs and the SIP DEBUG of a failed call |
| 06:09.05 | [TK]D-Fender | ~pb |
| 06:09.06 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 06:09.18 | darksmurf | fender will do |
| 06:12.13 | darksmurf | failed call: http://pastebin.com/m73a4fba8 |
| 06:14.14 | darksmurf | config is done via trixbox, what parts do you need? I have been able to use a friend's asterisk box work as a SIP trunk. The only thing I have done is add nat=yes and externip=99.x.x.x to the general section of sip.conf (rather, trixbox's really convoluted sip.conf replacement using seemingly hundreds of #includes.) |
| 06:15.24 | darksmurf | using extern IP I would expect to not see ANY references to the 192.x.x.x IP. Maybe I am wrong. |
| 06:15.44 | [TK]D-Fender | darksmurf: sip.conf and everything it INCLUDES |
| 06:16.01 | [TK]D-Fender | darksmurf: And you have done it wrong if the internal IP shows up |
| 06:16.25 | darksmurf | thats what I thought. stand by, this is going to take a bit. |
| 06:17.11 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 06:24.46 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 06:28.33 | darksmurf | http://pastebin.com/m3c6c7382 |
| 06:28.35 | darksmurf | configs |
| 06:28.59 | darksmurf | I did omit some .confs that were #included because they were empty |
| 06:29.39 | drmessano | Youre not supposed to edit sip.conf in trixbox |
| 06:30.19 | [TK]D-Fender | DarkYou left out "localnet" <- * has no idea what is LOCAL so when would it decide what isn't? |
| 06:30.26 | [TK]D-Fender | darksmurf: You left out "localnet" <- * has no idea what is LOCAL so when would it decide what isn't? |
| 06:30.40 | darksmurf | drmessano, I didn't, I edited the file sip.conf says to edit, sip_general_custom.conf |
| 06:30.42 | drmessano | yup |
| 06:30.52 | drmessano | ok |
| 06:30.59 | darksmurf | localnet would be 192.168.1.1/255.0.0.0 ? |
| 06:31.06 | darksmurf | err..duh |
| 06:31.14 | [TK]D-Fender | darksmurf: I highly doubt that netmask |
| 06:31.26 | darksmurf | yeah... |
| 06:31.30 | drmessano | I completely doubt the netmask |
| 06:31.40 | drmessano | and the IP |
| 06:31.43 | drmessano | wrong and wrong |
| 06:31.55 | drmessano | 192.168.1.0/255.255.255.0 perhaps |
| 06:32.18 | darksmurf | that looks better |
| 06:32.34 | drmessano | or "correct" |
| 06:35.14 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 06:36.46 | darksmurf | the 'contact' line in the invite is correct now, thanks. Now I am getting a 404 error. I think that may be because I am not sending a full 10 (11?) digit phone number. |
| 06:40.14 | WindowsUser | probably |
| 06:41.01 | WindowsUser | I know flowroute requires 11 digit dialing for north america |
| 06:41.29 | WindowsUser | you can append digits via the dialplan if you need to btw |
| 06:42.27 | [TK]D-Fender | WindowsUser: ....TRIXBOX |
| 06:43.04 | [TK]D-Fender | darksmurf: As of now... |
| 06:43.06 | [TK]D-Fender | ~trixbox |
| 06:43.07 | infobot | extra, extra, read all about it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 06:43.37 | darksmurf | GOT IT. Thanks everyone. Now it is time to play with the dial rules. |
| 06:44.29 | *** join/#asterisk shinao1 (n=shinao1@41.219.196.40) |
| 06:44.29 | *** join/#asterisk peterosd (n=Peter@c83-255-66-116.bredband.comhem.se) |
| 06:45.08 | darksmurf | ah, sorry about that. I shall head over there from now on. |
| 06:45.43 | darksmurf | Still, thanks for the help all the same. |
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| 07:35.09 | Micc | I can't use some things because it says I don't have a timing device. Is there any way to get around that without buying special hardware? |
| 07:35.34 | Micc | I have the dahdi_dummy driver installed. |
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| 08:41.59 | kron4eg | what is the difference between 1.4 and 1.6 branches? |
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| 09:33.53 | svm_invictvs | Hiya |
| 09:34.29 | svm_invictvs | So I'm setting up some hosted PBX service w/ asterisk. I have a SIP client channel set up, and I'm looking to set up like 4 extensions for individual offices. |
| 09:34.39 | svm_invictvs | Should I enable or disable "allowguest"? |
| 09:52.05 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 09:56.58 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 10:04.30 | *** join/#asterisk phryk (n=phryk@yggdrasil.phryk.net) |
| 10:04.40 | phryk | Mornin' |
| 10:05.39 | phryk | I'm moving into my own place soon, and I thought about setting up an asterisk daemon on my miniserver, what exactly do i need? Anything besides a working linux and internet access? |
| 10:07.50 | kron4eg | TDM card maybe? |
| 10:08.25 | phryk | what is that? |
| 10:09.43 | kron4eg | card to connect your asterisk server with PSTN |
| 10:10.13 | phryk | pstn is the "normal" telephone net? |
| 10:10.55 | kron4eg | yep |
| 10:11.13 | kron4eg | line from your local telco |
| 10:11.27 | phryk | If I have a provider that grants me a voip flatrate, can i somehow go through that? |
| 10:11.49 | kron4eg | sure |
| 10:11.50 | phryk | I mean if I get that for free, why should I pay for plain telephone services :) |
| 10:11.56 | kron4eg | pstn jusn an option |
| 10:12.18 | kron4eg | phryk, no reason to pay :) |
| 10:12.21 | phryk | So with voip i don't need a tdm card but still can phone to the normal telephone net |
| 10:12.35 | phryk | kron4eg: I wouldn't have to pay, even if i would go through the ptsn? |
| 10:14.24 | kron4eg | you have to pay your local telco if you using it's line, but, since you not planning to use it, you don't have to warry about it |
| 10:15.03 | phryk | okay |
| 10:17.23 | phryk | I think I'll construct some weird kind of hands-free speaking system :D |
| 10:20.50 | svm_invictvs | You could just get a SIP provider, too |
| 10:23.41 | kron4eg | OMG |
| 10:23.48 | svm_invictvs | ? |
| 10:23.49 | kron4eg | just discovered Adhearsion |
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| 10:30.12 | phryk | svm_invictvs: If my internet provider provides voip, i should have sip access, right? |
| 10:36.27 | *** part/#asterisk kron4eg (n=kron@212.56.196.230) |
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| 10:41.09 | box2 | sip is not for the feint of heart |
| 10:41.54 | phryk | why not? |
| 10:42.08 | phryk | it's similiar to http and smtp, so it can't be that hard to understand... |
| 10:47.17 | *** join/#asterisk mechbangirc (n=mechbang@static-host119-73-6-139.link.net.pk) |
| 10:48.35 | mechbangirc | is there anyway i can define global variables in a separte file and then use them in my dialplan? |
| 10:49.13 | *** join/#asterisk Lantizia (n=cyor@212.57.229.111) |
| 10:49.49 | Lantizia | Lo, since upgrading my snoms to the latest version 7 - they still register but have status UNREACHABLE... any tips on what I should check for? |
| 10:52.09 | box2 | phryk: sip is the leading cause of stroke, diabetes, and road rage |
| 10:52.48 | box2 | you may grow to understand it, but the DANGERS INVOLVED |
| 10:52.51 | box2 | shudders |
| 10:53.44 | phryk | I have people in my im contactlists, who sent me chain mail. |
| 10:54.03 | phryk | I haven't turned mad, yellow and stinky yet. |
| 10:54.07 | phryk | I am fucking invincible. |
| 10:56.07 | box2 | heh |
| 10:56.37 | box2 | i will sing a nice tune for your eulogy |
| 10:58.27 | phryk | I know a shaman. He will revive me. |
| 11:00.29 | box2 | shamans are crafty with their legal documents, be sure not to sign anything which leads to you being revived as a zombie slave |
| 11:00.41 | box2 | that shit happens a lot more than one would think |
| 11:01.13 | box2 | or at least have a roomate co-sign for you, as collateral |
| 11:01.36 | phryk | I already did that. |
| 11:02.09 | phryk | He is my slave. I once killed him and revived him. He is not able to do harm to me. |
| 11:06.39 | box2 | that's awesome |
| 11:06.43 | box2 | i wish i could do that |
| 11:07.36 | phryk | Trust me, that was damned hard work... |
| 11:08.08 | phryk | Killing a Voodoo-Shaman isn't as simple as it sounds... |
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| 11:43.56 | *** join/#asterisk yahh (n=root@122.169.75.81) |
| 11:44.09 | yahh | Hi guys |
| 11:45.26 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-245.tricom.net) |
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| 11:46.31 | yahh | I want to use asteriskwin32 on my windows system |
| 11:46.35 | yahh | with the FXO card |
| 11:46.57 | yahh | Anyone have experiance with it? |
| 11:48.49 | Alfio | yahh asterisk for windows its not recomended |
| 11:51.37 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-245.tricom.net) |
| 11:53.18 | yahh | Alfio: why..? |
| 11:53.44 | yahh | i know it is made for linux |
| 11:53.51 | yahh | and made work with windows |
| 11:54.02 | yahh | thats why it is better to use in linux |
| 11:54.35 | yahh | but is there any specific issue......then tell me plz... |
| 11:54.49 | box2 | not to sound like a fanboy or anything, but windows just isn't fun to use |
| 11:56.34 | Psychobilly | AsteriskWin32 0.66b build from Asterisk 1.2.26.2 |
| 11:56.48 | *** join/#asterisk Gnarfy (n=mirc@213.144.244.115) |
| 11:57.17 | *** join/#asterisk Gnarfy (n=mirc@213.144.244.115) |
| 11:57.21 | Psychobilly | yahh its just outaded and judging by the look of the web page it must also be totaly unmaintained |
| 11:58.08 | Psychobilly | last update was on jan 2008 |
| 11:58.16 | yahh | hmm |
| 11:58.37 | yahh | that version must be working |
| 11:59.00 | Psychobilly | sure :P |
| 11:59.31 | yahh | it is not updated, but upto 1.2.26.2 is should be ok |
| 11:59.56 | Psychobilly | go with linux thats the advice u will get here |
| 12:00.42 | yahh | plz give me guidance, i wanted to use it in windows |
| 12:01.13 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 12:02.17 | Psychobilly | i dont think anyone here has ever used this |
| 12:03.08 | Psychobilly | from my little experience with * running it on windows sounds just wrong, especially such and old unmaintained version |
| 12:08.05 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 12:08.22 | *** part/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 12:17.46 | errotan | yahh even if asterisk would work in windows the FXO card definietly won't |
| 12:18.39 | yahh | but digium card should supported to asterisk-win |
| 12:19.39 | errotan | why do you want to run it on windows anyway ? if you have some programs that are made for windows you should start windows in a virtual environment (like virtualbox) |
| 12:20.41 | yahh | my friend is going to use that and he is not knowing Linux |
| 12:21.04 | yahh | He just knowing windows |
| 12:23.11 | Psychobilly | even better, u will install it and he will never touch it |
| 12:23.17 | Psychobilly | it will be safe this way |
| 12:26.59 | jaytee | digium cards aren't supported on AsteriskWin32. the driver mods are only available for linux, not Windows |
| 12:27.54 | jaytee | according to some sources AsteriskWin32 was started as an April Fool's joke that got out of hand and now won't die |
| 12:28.03 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 12:29.03 | yahh | are you sure...digium FXO cards is not suppored? |
| 12:29.08 | Psychobilly | lol |
| 12:30.38 | jaytee | yahh, unless someone has taken zaptel code and ported it to the Windows driver spec then no. |
| 12:31.26 | yahh | okay... |
| 12:31.56 | Psychobilly | yahh maybe u can try some pbx software for windows, like yate or freeswitch |
| 12:32.06 | Alfio | or 3cx |
| 12:32.12 | Psychobilly | dont know how good they are though |
| 12:34.05 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 12:38.28 | *** join/#asterisk seth911 (n=Seth@C-59-100-13-206.for.connect.net.au) |
| 12:39.19 | seth911 | i have a phone line plugged into my modem i am trying to make calls from it, which works, but no sound is being sent or recieved |
| 12:39.33 | seth911 | would anyone know the solution or problem? |
| 12:40.45 | yahh | thank you Psychobilly |
| 12:40.54 | yahh | thank you all |
| 12:41.47 | yahh | anyone have experiance with freeswitch or yate for windows? |
| 12:41.47 | errotan | freeswitch works on windows well |
| 12:41.51 | jaytee | yahh, try asking in #freeswitch or #yate |
| 12:42.54 | yahh | errotan: with FXO card |
| 12:43.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:43.15 | errotan | FXO card ? |
| 12:43.47 | yahh | A card for PSTN line |
| 12:44.24 | errotan | i know what is an FXO card but on windows you can't find any working card |
| 12:45.07 | yahh | any specific reason? |
| 12:45.09 | errotan | you need to use voip gateways if you want to work with windows |
| 12:45.39 | errotan | reason: they are made for linux apps :) |
| 12:46.13 | yahh | no single card with drivers for windows? |
| 12:48.37 | seth911 | anyone? |
| 12:48.45 | errotan | i never heard of any, there are not so many pci fxo cards on the market |
| 12:49.18 | jaytee | there's two particular reasons why the vast majority of pci FXO or FXS cards don't have drivers for Windows. One is the cost of developer tools to write device drivers using the Windows DDK and the other is LATENCY. |
| 12:51.28 | jaytee | commercial phones systems from companies like Nortel or Avaya don't use Windows as an OS. Nortel uses a customized kernel based on SRV5 Unix. stripped down to be minimal and run with very low latency. |
| 12:52.52 | yahh | i see |
| 12:53.12 | jaytee | Windows system developers have never learned to develop minimal stripped down versions of their OS. Some people think they even strive for bloat in order to drive people to upgrade memory or get a faster CPU so their investments in companies like Intel and Micron will reap extra revenue. |
| 12:53.59 | yahh | :) |
| 12:54.17 | jaytee | even linux with a GUI like Gnome running on it tends to introduce too many interrupts (latency) which makes the earlier cards from Digium and Sangom have problems due to missed interrupts. |
| 12:56.38 | *** join/#asterisk Katty (n=Katty@mail.copi-rite.com) |
| 12:56.58 | Katty | hi! :> |
| 12:57.24 | jaytee | morning Katty *hugs* |
| 12:57.38 | Katty | herroes |
| 12:58.19 | Katty | Do you know how to change nicknames using nickserv? I can't seem to use /nick with jmirc |
| 12:58.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:59.44 | jaytee | nope, I've always just used /nick with Xchat and even with mIRC on Windows I'd use /nick |
| 13:00.06 | Katty | hmmmmmk |
| 13:00.14 | jaytee | never used jmirc |
| 13:00.50 | hesco | what application will wait for multiple digits? |
| 13:00.57 | jaytee | I prefer Xchat since I can use any perl or python script plugin |
| 13:01.35 | Katty | Well I prefer irssi, but its not blackberry friendly |
| 13:02.21 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 13:02.24 | Katty | Hesco read()? |
| 13:02.25 | jaytee | hesco, WaitExten |
| 13:02.45 | *** join/#asterisk kron4eg (n=kron@212.56.211.50) |
| 13:03.15 | Katty | waitexten is good |
| 13:03.19 | hesco | thanks, will try read() then. WaitExten did not quite work for me for some reason |
| 13:03.36 | hesco | and it is not an extension I am waiting for but an account number |
| 13:03.37 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 13:03.45 | Katty | You have to tell waitexten how long to wait |
| 13:04.06 | Katty | and then what to do next, obviously |
| 13:04.22 | Katty | Ahhh, k |
| 13:05.33 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar) |
| 13:06.05 | jaytee | hesco, if you're waiting for the caller to input an account number then use Read(). Read() will allow you to assign what the caller enters to a variable that you can then handle in your dialplan any way you choose. |
| 13:06.49 | Katty | I'd recommend rerouting all calls to jaytees zoo |
| 13:07.09 | Katty | We can put a phone in with the monkies |
| 13:07.13 | hesco | that ought to take the load off our overworked staff, eh? |
| 13:07.22 | hesco | thanks jaytee and katty |
| 13:07.34 | jaytee | sets up a macro to route all incoming calls to Katty's blackberry |
| 13:07.37 | hesco | reading the show application read now |
| 13:07.49 | Katty | )= |
| 13:08.55 | Katty | No calls today! Me and mom are out yard saleing |
| 13:09.30 | jaytee | always buy the hummels. good investment strategy :-) |
| 13:09.58 | Katty | Hummels? |
| 13:10.42 | jaytee | http://www.hummelsatadiscount.com/ |
| 13:11.11 | Katty | Brb |
| 13:12.01 | *** join/#asterisk Katty (n=Katty@mail.copi-rite.com) |
| 13:12.42 | Katty | Hard to switch apps on this thing |
| 13:26.30 | hesco | My NoOp is printing to the *CLI: (NoOp) Options: (The Account Number is: $ACCOUNTNUMBER . . . |
| 13:26.53 | *** join/#asterisk [netman] (n=netman@107.Red-88-8-164.dynamicIP.rima-tde.net) |
| 13:26.59 | hesco | how do I get it to interpolate that variable before printing it out? |
| 13:27.30 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 13:28.13 | jaytee | NoOp(The account number is ${ACCOUNTNUMBER}) |
| 13:29.33 | jaytee | hesco, one of these days when you're really really bored you might take a look at the channelvariables.txt file in the tarballs. full of good information |
| 13:30.25 | hesco | should have mentioned, its being called from an agi script |
| 13:31.18 | jaytee | sorry, AGI not spoken here |
| 13:33.13 | hesco | I found: doc/api/html/chanvars_8c.html, is that what you are talking about? |
| 13:33.42 | jaytee | nope |
| 13:34.22 | jaytee | what version are you running? |
| 13:36.32 | hesco | 1.6.0.3-rc1 |
| 13:36.49 | hesco | (NoOp) Options: (The Account Number is: ${ACCOUNTNUMBER}) |
| 13:37.03 | grandpapadot | Hey guys, VoiceMailMain has a parameter "p" that is suppsed to "Consider the mailbox parameter as a prefix to the mailbox that |
| 13:37.03 | grandpapadot | <PROTECTED> |
| 13:37.08 | grandpapadot | shit, sorry |
| 13:37.09 | hesco | that is the result whether or not I escape the { and } |
| 13:37.58 | grandpapadot | Hey guys, VoiceMailMain has a parameter "p" that is suppsed to "Consider the mailbox parameter as a prefix to the mailbox that is entered by the caller." However, it appears in 1.4.26 that parameter is active no matter what. Anybody else experience this? |
| 13:48.58 | *** join/#asterisk dexteruk (n=dexteruk@hst-4-6.cisbg.com) |
| 13:49.17 | dexteruk | Does anyone know where i can get some help with a2billing? |
| 13:52.50 | hesco | jaytee: found: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+channelvariables.txt |
| 13:53.04 | hesco | and references in google to this file missing from 1.6.0.9 |
| 13:53.30 | jaytee | hesco, yep and in 1.6 it's in /doc/tex in the tarball and it's a .tex file not a text file. |
| 13:55.29 | *** join/#asterisk Katty (n=Katty@mail.copi-rite.com) |
| 13:56.41 | Katty | I found a leppard print rug and two stuffed dinos for riddick so far :> |
| 13:57.05 | *** join/#asterisk pointer (n=pointer@aj.catt.com) |
| 13:58.20 | grandpapadot | Ok, when compiling on the Intel Atom N270, the i586 flags need to be set for the transcoder to work properly. |
| 14:00.50 | jaytee | wonders how many concurrent calls a Intel Atom can handle |
| 14:01.16 | pointer | hrm. I've been busy for a while and just realized that my outbound provider is gone (nufone). Any recommendations for another provider that will let me set outbound CID (forwarding calls from another sip provider to my mobile)? |
| 14:02.08 | grandpapadot | I'm actually testing that, I'm able to sustain around 80 g729->g729 and about 120 g711-g711 and about 60 g711->gsm and about 10 (lol) g711-g722 ... 1.6GHz, 1GB |
| 14:02.21 | pointer | and by a while, it looks like I mean months :-\ |
| 14:03.09 | grandpapadot | With the exception of gsm, anything transcoded on the Atom N270 just kills it. |
| 14:03.50 | grandpapadot | But we're using it for a endpoint aggregator so transcoding isn't necessary. |
| 14:03.55 | *** join/#asterisk devyll (n=paul@89.36.24.2) |
| 14:05.58 | devyll | any ideea why navigating in an IVR is really crappy from mobile phone ? some times (lot of times) when I press a key it doesn't do nothing ... i have to wait untill the end of the Background(audio) to be able to choose an option from the ivr however this is happening randomly .. sometims it works with the mobile phone too .... from a local phone is working just fine .. am I missing something ? |
| 14:07.41 | grandpapadot | All, I'm looking for a way to evaluate a string and see if it's a number. I can do it by letting MATH() fail, but I was looking for a cleaner (non-hacker) way ... |
| 14:07.45 | grandpapadot | In 1.4 dp |
| 14:08.13 | Katty | Pointer, we use bandwidth.com |
| 14:08.31 | pointer | Katty: thanks! |
| 14:08.36 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 14:09.32 | pointer | Katty: do they have a pay as you go or do you use the prepaid/monthly plans? |
| 14:19.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 15:09.54 | Lantizia | Anyone had issues with snoms behind a NAT and asterisk since firmware 7.1.33? |
| 15:17.13 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 15:21.23 | *** join/#asterisk many (n=many@dslb-188-098-011-250.pools.arcor-ip.net) |
| 15:23.46 | hesco | jaytee: thanks, found it. |
| 15:24.23 | hesco | how was it again I convert .tex to .pdf? Was it dvips file.tex; ps2pdf file.dvi ??? |
| 15:24.54 | hesco | I seem to think there was a third step along the way, but I can't for the life of me remember what that might have been. |
| 15:29.55 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 15:34.18 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 15:53.31 | *** join/#asterisk SkramX (i=mark@phalse.2600.COM) |
| 15:53.54 | SkramX | anyone know the voicexml syntax to have the system say "first", "second", "third" for 1,2,3? |
| 16:05.29 | *** join/#asterisk many (n=many@dslb-188-098-011-250.pools.arcor-ip.net) |
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| 16:11.23 | *** join/#asterisk VladTheImpaled (n=kodegear@wsip-68-14-246-71.ph.ph.cox.net) |
| 16:13.56 | VladTheImpaled | Need some advice on dial plan strategy. I have one ITSP but multiple DIDs (4). I need to control incoming calls. I've put the full details of the problem at http://pastebin.com/m1e1eaea1 |
| 16:14.34 | VladTheImpaled | This is with Asterisk 1.4.26 |
| 16:16.48 | WindowsUser | goto() a different context |
| 16:17.02 | Psychobilly | VladTheImpaled make seperate contexts for each line |
| 16:17.55 | VladTheImpaled | Psychobilly: Thanks. Yes, that is what I thought. But my problem is that how do I handle that in sip.conf? The ITSP is telling me to have one sip.conf for their service, which ties to one DID |
| 16:18.06 | WindowsUser | goto() a different context <--- |
| 16:18.12 | Psychobilly | what WindowsUser said |
| 16:18.35 | WindowsUser | exten => 5555551212,1,Goto(wakeupcall,797,1) |
| 16:19.05 | VladTheImpaled | Aahhh.. I think I see. So basically the incoming sip.conf is singular, and I do some sort of IF statement on $EXTEN to work out where the call was supposed to go? |
| 16:19.29 | WindowsUser | if you want |
| 16:19.46 | WindowsUser | but I'd goto right at the start |
| 16:20.42 | WindowsUser | exten => 2125551111,1,Goto(CompanyA,${EXTENSION},1) |
| 16:20.48 | WindowsUser | exten => 2125551112,1,Goto(CompanyB,${EXTENSION},1) |
| 16:20.58 | WindowsUser | and then write up CompanyA and B contexts |
| 16:21.15 | VladTheImpaled | that's exactly what I was looking for |
| 16:21.32 | VladTheImpaled | Thanks so much for this. I'm still trying to get my head around it all, but gradually its sinking in |
| 16:22.50 | [TK]D-Fender | VladTheImpaled: exten => 2125551111,1,Goto(CompanyA,s,1) <- should run IVR's off non numeric extens so they can't be dialed recursively. Thats what "s" is for |
| 16:23.14 | [TK]D-Fender | VladTheImpaled: So no need to make variable in your Goto. then just make your "s" exten in [CompanyA] |
| 16:24.41 | VladTheImpaled | [TK]D-Fender: So if I use s for incoming DID reference, I just reference s as the exten in the other contexts? |
| 16:26.39 | [TK]D-Fender | VladTheImpaled: No, in your case your ITSP dials an extension matching the DID that arrived to them. You match on this, and YOU jump to separate places to process as you wish inside your dialplan |
| 16:26.57 | [TK]D-Fender | ~stdextens |
| 16:26.58 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 16:30.14 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 16:31.46 | [TK]D-Fender | VladTheImpaled: Contexts are import security separations that control what can be done when a device points to one, or when you jump into one. In a multi-level IVR you'd have each sub-menu in its own, etc. |
| 16:32.39 | [TK]D-Fender | VladTheImpaled: For many purposes you'd put the extens that would dial phones, etc often in 1 context, and INCLUDE it in IVR's, and in "collector" contexts that also include contexts that have the extens that let you dial out, etc |
| 16:33.03 | VladTheImpaled | [TK]D-Fender: Yes, that is what I thought. I can definitely see how it separates the calls into their appropriate areas. |
| 16:35.23 | VladTheImpaled | [TK]D-Fender: However for some reason my incoming calls are not being handled and put to their appropriate context. I've pasted what I've done at: http://pastebin.com/m621b5612 |
| 16:35.45 | VladTheImpaled | am I missing something in the 'incoming' context for this? |
| 16:36.03 | [TK]D-Fender | VladTheImpaled: Yeah, thats good for each DID that you can ID that way. Now the real work occurs in those other contexts |
| 16:37.43 | VladTheImpaled | [TK]D-Fender: right, but its not working. I'm getting an error of: [Aug 1 09:30:13] WARNING[30507] pbx.c: Channel 'SIP/sip.broadvoice.com-089cc4c8 |
| 16:37.43 | VladTheImpaled | ' sent into invalid extension 's' in context 'edgeneering', but no invalid handl |
| 16:37.44 | VladTheImpaled | er |
| 16:38.37 | VladTheImpaled | In the context that receives the call, I'm just doing stuff like this: exten => s,1,Answer(100) |
| 16:39.12 | VladTheImpaled | Do I have to be specific to the DID in the 'child' context that it is being sent into? |
| 16:45.26 | VladTheImpaled | hang on, fixed it |
| 16:45.42 | VladTheImpaled | my bad.... I had some legacy mess in there from past attempts but its working now |
| 16:45.48 | VladTheImpaled | Thank you all for this!!!!!! |
| 16:46.53 | [TK]D-Fender | VladTheImpaled: You're welcome, and good to hear |
| 16:50.15 | [TK]D-Fender | is off to enjoy the sun... while it lasts... |
| 16:50.17 | [TK]D-Fender | BBL |
| 16:55.28 | *** join/#asterisk Lantizia (n=cyor@212.57.229.111) |
| 16:56.41 | Lantizia | Hey, I'm having a weird issue where if my snom phone is on 7.1.33 version or higher... it registers... but on the asterisk CLI you see it retransmitting 4 times and then "really destroying SIP dialog" over and over |
| 16:57.11 | Lantizia | any phone on this version or higher can't make or receive calls - but they are registered... the pbx is not behind a nat but the phone is ... but it's never been a problem till now |
| 16:58.09 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 16:59.43 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
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| 17:01.07 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 17:01.31 | grandpapadot | Hey guys is there a way to "see" the generic jitter buffer in action? |
| 17:02.16 | carrar | If you're Neo |
| 17:03.12 | grandpapadot | Or at least verify it's actually doing something, lol |
| 17:07.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:07.36 | *** part/#asterisk SkramX (i=mark@phalse.2600.COM) |
| 17:08.29 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 17:13.04 | devyll | any reason for which trying to choose an option from an ivr ex: 3,1,Goto(blabla) does not always work when calling from mobile phones ? this problem occurs only with mobile phones ... |
| 17:13.08 | devyll | ? |
| 17:14.00 | devyll | and when it does ... it seems i still can choose option number 3 but only after the Background file is finished playing or .. only after a certain amount of time from when the play started |
| 17:14.11 | devyll | any help on diagnosing this problem ? don't know where to start ... |
| 17:15.50 | carrar | devyll, it's a mobile, they don't always have great audio |
| 17:16.05 | carrar | or they don't press the DTMF long enough |
| 17:16.24 | grandpapadot | Will jbenabled=yes not show up in sip show peer output? |
| 17:16.44 | carrar | devyll, use background instead of playback |
| 17:19.11 | grandpapadot | Is there any way to verify a jitter buffer is working in 1.4.26? |
| 17:20.48 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 17:29.41 | Lantizia | OK I've got a system running 1.4.19.1 and I want to upgrade... which file (is it that makeelect thing?) do I keep so that when I recompile from new source it does the same config as the last time? |
| 17:34.55 | WindowsUser | try makeopts |
| 17:36.05 | *** join/#asterisk voxter (n=voxter@190.10.79.146) |
| 17:39.21 | svm_invictvs | phryk: Yeah. If not SIP they're providing *some* type of VoIP. The thing is, some ISPs don't like you hooking up a PBX to their channel. |
| 17:40.05 | svm_invictvs | phryk: Likely it's SIP. But, once again, they may somehow block you using asterisk because they want to charge for hosted PBX |
| 17:40.18 | florz | well, and some isps don't like you actually using their bandwidth ... so what? =:-) |
| 17:40.56 | florz | that someone wants to take advantage of you doesn't really oblige you to much, does it? |
| 17:43.41 | grandpapadot | lol |
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| 17:49.40 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 17:50.13 | hesco | I'm working on an AGI script which includes these two lines: |
| 17:50.16 | hesco | print "EXEC Read ACCOUNTNUMBER \"\", 6, \"\", 1, 4 \n"; |
| 17:50.25 | hesco | print "EXEC NoOp \"The Account Number is: \${ACCOUNTNUMBER} \" \n"; |
| 17:50.43 | hesco | written in perl, in case that was not obvious. |
| 17:51.45 | hesco | however, the *CLI reports: |
| 17:51.48 | hesco | -- User entered '33333' |
| 17:52.19 | hesco | -- AGI Script Executing Application: (NoOp) Options: (The Account Number is: ${ACCOUNTNUMBER} ) |
| 17:53.00 | hesco | what am I missing here that is preventing the proper interpolation of my user defined variable. |
| 17:53.11 | hesco | ??? |
| 17:53.32 | Qwell | \ |
| 18:03.15 | *** join/#asterisk manxpower (n=eric@113.sub-70-222-138.myvzw.com) |
| 18:04.07 | hesco | ~paste |
| 18:04.07 | infobot | extra, extra, read all about it, paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 18:07.01 | hesco | I have an AGI question, pasted at: http://bin.cakephp.org/view/1745668161 Can anyone here please offer some guidance? |
| 18:08.20 | WindowsUser | why are you reading a number into a variable asterisk side and then trying to get it? |
| 18:08.32 | WindowsUser | try READ instead of EXEC READ |
| 18:08.46 | hesco | ok, trying that now |
| 18:09.17 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 18:10.29 | hesco | print "READ ACCOUNTNUMBER \"\", 6, \"\", 1, 4 \n"; |
| 18:10.40 | hesco | did not even pause to have me enter the number |
| 18:11.13 | hesco | with EXEC READ it did slow down to do the READ function |
| 18:11.36 | *** join/#asterisk wwalker (n=wwalker@72.249.1.66) |
| 18:11.44 | hesco | none of my attempts to print AGI functions directly to STDOUT have succeeded, unless wrapped in the EXEC. |
| 18:11.45 | WindowsUser | well, you're trying to write an AGI like you're in the dialplan |
| 18:12.04 | WindowsUser | are you reading in stdin? |
| 18:12.29 | hesco | In a dialplan I would have written that as: Read(ACCOUNTNUMBER,'',6,'',1,4) |
| 18:12.31 | wwalker | I've looked around a lot and can't find a decent comparison of app_conference and app_meetme. Does anyone have a pointer to a comparison of the two? |
| 18:12.35 | Lantizia | If I'm building an asterisk server that doesn't have ANY cards (so it's purely sip and iax only) do I need dahdi? or indeed anything but asterisk? |
| 18:12.50 | WindowsUser | ~agi |
| 18:12.51 | infobot | methinks agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 18:13.26 | hesco | yes, I've got code starting: while(<STDIN>){ copied character for character from the sample |
| 18:13.49 | WindowsUser | okay then EXEC read to ACCOUNTNUMBER |
| 18:13.58 | WindowsUser | and then GET VARIABLE ACCOUNTNUMBER |
| 18:13.58 | hesco | ok, trying that then |
| 18:13.59 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 18:14.26 | WindowsUser | okay then EXEC read to ACCOUNTNUMBER <-- i meant the syntax that worked before |
| 18:14.31 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 18:15.35 | hesco | is 'to' a part of that syntax? |
| 18:16.22 | WindowsUser | okay then EXEC read to ACCOUNTNUMBER <-- i meant the syntax that worked before <-- this is me trying to say dont read what i say as syntaxically correct |
| 18:17.01 | hesco | ok, thanks for that clarification |
| 18:22.00 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 18:23.23 | hesco | print "EXEC GET ACCOUNTNUMBER \n"; |
| 18:23.40 | hesco | WARNING[25307]: res_agi.c:1494 handle_exec: Could not find application (GET) |
| 18:23.52 | WindowsUser | what about GET VARIABLE ACCOUNTNUMBER |
| 18:24.18 | hesco | I had tried that first and got this: |
| 18:25.07 | phryk | svm_invictvs: It would be a flatrate |
| 18:25.07 | hesco | WARNING[25364]: res_agi.c:1494 handle_exec: Could not find application (GET) |
| 18:25.21 | WindowsUser | odd |
| 18:26.17 | hesco | I thought so too, b/c *CLI> agi show get variable gives me expected output |
| 18:26.56 | hesco | I thought so too, b/c `*CLI> agi show get variable` gives me expected output |
| 18:29.26 | hesco | Lantizia: I'm pretty sure you need dahdi for MeetMe(). Not sure what else. |
| 18:29.53 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 18:29.56 | hesco | Lantizia: make menuconfig will show you dependencies |
| 18:30.28 | phryk | Mh, I thought only the kernel had menuconfig |
| 18:30.58 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 18:30.59 | WindowsUser | its actually makeopts but theres an alias |
| 18:31.21 | *** join/#asterisk s0lid (n=s0lid@122.53.110.95) |
| 18:31.24 | hesco | actually perhaps that should have been `menuselect` |
| 18:32.54 | *** join/#asterisk exc_ess (n=xavier@69-165-165-12.dsl.teksavvy.com) |
| 18:33.17 | Lantizia | hesco, both do the same thing - but I don't see any dependancy things |
| 18:33.51 | hesco | as you use the arrow keys to go through the options, look at the bottom of the screen |
| 18:34.34 | Lantizia | yeah meetme depends on it (well zaptel but same thing) |
| 18:34.50 | Lantizia | hesco, I was more wondering about something I heard where zaptel is needed to provide a timing source |
| 18:34.57 | Lantizia | even if theres no card in the system |
| 18:35.48 | hesco | I've tried my AGI call using both EXEC and simply GET VARIABLE (var_name>. Both produce the same output, none. |
| 18:36.42 | hesco | Lantizia: I had heard that as well. And that it was critical to MeetMe(). Not sure what else requires it. |
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| 18:43.07 | Lantizia | Do I need the linux kernel sources if I'm just compiling dahdi and asterisk? Is the linux kernel sources only needed for libpri? |
| 18:44.03 | hesco | I believe that each of those packages require kernel headers to compile against |
| 18:44.06 | Psychobilly | kernel source or just kernel headers |
| 18:44.31 | Lantizia | so I don't need the source and exrtact it and all that |
| 18:44.36 | *** part/#asterisk rwong (n=ricky@www.roflwaffle.com) |
| 18:44.47 | Lantizia | just installing the headers will do for dahdi and * ? |
| 18:45.23 | exc_ess | Hey all, quick question. I have a dialplan that, when you call in externally, plays a greeting message and then transfers you to the extension you dial. Unfortunately I don't hear a ringtone while it's ringing the extension. I do get ringtones when calling extensions internally. I'm pretty sure that I used to get ringtones when calling from outside, before upgrading to 1.6 |
| 18:45.39 | exc_ess | I tried adding the "r" flag to the Dial() command, but it didn't help |
| 18:46.45 | exc_ess | The relevant section of my dialplan is here: http://pastebin.com/m21df4664 |
| 18:47.02 | hesco | Lantizia: I think that is right. Test it, if that does not work, grab the entire source instead |
| 18:47.09 | Lantizia | ok |
| 18:52.50 | Micc | Are there any unified communications apps for asterisk? I know asterisk is one in itself in a way, but I mean something that ties it all together using asterisk? |
| 18:54.01 | *** join/#asterisk lolo2 (n=joel@c-69-180-160-4.hsd1.mn.comcast.net) |
| 18:55.08 | lolo2 | I have followed this tut: http://www.757.org/~joat/wiki/index.php/Inbound_PSTN_Calls_to_Asterisk,_via_Gizmo5_and_GoogleVoice and am wondering if i can have two callers on thepbxc at the same time or if the way i set it up does not allow me two? |
| 19:02.37 | *** join/#asterisk chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:02.39 | exc_ess | No one has any idea how to get a ringtone when dialing an extension through an IVR? I feel like this is fairly standard stuff, and I'm just missing some small point |
| 19:03.57 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:04.10 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:04.15 | WindowsUser | what are you calling? |
| 19:04.49 | exc_ess | WindowsUser: me? Well the relevant part of my dialplan is here http://pastebin.com/m21df4664 |
| 19:05.20 | exc_ess | It plays a greeting, and asks you to dial an extension. When you dial that extension you can't hear a ringtone over the line. |
| 19:05.55 | WindowsUser | so are you calling SIP/someone or DAHDI/someone? or Skype/something? |
| 19:06.15 | exc_ess | SIP/someone |
| 19:06.47 | Micc | exc_ess, if the cli shows it is ringing, the caller usually hears it ringing. |
| 19:07.07 | exc_ess | That's what I would have thought, also that's the behaviour I used to have |
| 19:07.12 | exc_ess | I've just migrated the system to a new computer |
| 19:07.33 | exc_ess | The CLI does show it ringing, btw |
| 19:07.44 | Micc | exc_ess, what is your progressinband set to? |
| 19:07.47 | Micc | in your sip.conf |
| 19:08.20 | Micc | What kind of device are you calling in from? |
| 19:08.21 | exc_ess | Not set at all |
| 19:08.28 | Micc | try progressinband=yes |
| 19:09.24 | exc_ess | Micc: progressinband didn't seem to help |
| 19:09.37 | Micc | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband |
| 19:10.01 | exc_ess | Also, it doesn't work if I call in from my cellphone externally, or if I dial a special internal extension which puts me into the [external] context |
| 19:10.04 | Micc | exc_ess, your calling device should receive the ringing sip packet and generate the tone itself. |
| 19:10.40 | Micc | exc_ess, it may be a nat/firewall issue eating the ring packet. |
| 19:11.03 | exc_ess | No NAT internally, though |
| 19:11.07 | exc_ess | nor firewalls |
| 19:11.16 | exc_ess | so I could see that being a problem when I actually call from outside |
| 19:11.34 | exc_ess | but when I dial my special "make me external" extension you wouldn't think it would be |
| 19:11.50 | Micc | progressinband=yes should really do it. Do you have an answer in the dialplan first? |
| 19:12.26 | Micc | No nat? that is extremely rare these days. |
| 19:12.36 | Micc | Is it a dedicated or virtual server? |
| 19:13.06 | Micc | you should put progressinband=yes on the sip trunk as well as the phone you are dialing. |
| 19:13.19 | exc_ess | It's a dedicated sever |
| 19:13.20 | Micc | but it probably doesn't need it on the phone you are dialing. |
| 19:13.22 | exc_ess | In the office |
| 19:13.27 | exc_ess | that's why there's no NAT internally |
| 19:13.33 | Micc | exc_ess, oh, I see what you mean. |
| 19:13.53 | Micc | exc_ess, what is your incoming line device? |
| 19:13.58 | exc_ess | I had an Answer() originally, which I thought might be the problem, so I took it out, but it didn't seem to make a difference |
| 19:14.22 | Micc | You should have an answer in before playing audio. |
| 19:14.25 | exc_ess | Our internal asterisk server is hooked up only via the internet to our external SIP provider |
| 19:14.33 | Micc | but I think background and playback do an answer if you didn't do it. |
| 19:14.38 | exc_ess | I guess they must |
| 19:14.44 | exc_ess | but I can put it back in to be explicit |
| 19:15.00 | Micc | exc_ess, then you probably have nat between you and your sip provider. |
| 19:15.05 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:15.15 | Micc | exc_ess, I like to be explicit. |
| 19:15.18 | exc_ess | Yeah, except that that doesn't explain why I don't get ringback when I do the call internally |
| 19:15.31 | exc_ess | I set up ext 198 to GoTo(exposed,s,1) |
| 19:15.59 | exc_ess | So if I call that from an internal phone it should be equivalent to calling in externally, but not have NAT issues |
| 19:16.03 | Micc | exc_ess, do you have an extension that calls Dial? |
| 19:16.36 | exc_ess | Yeah, that's how the extensions work |
| 19:16.53 | Micc | exc_ess, what parameters are you passing to the dial command? |
| 19:17.11 | Micc | exc_ess, also, what type of phones are you using internally? |
| 19:18.00 | exc_ess | Analogue phones hooked up to Linksys SPA-2102s |
| 19:18.14 | exc_ess | My whole dialplan is here: http://pastebin.com/m103c86a7 |
| 19:19.01 | exc_ess | The only flag I pass to Dial right now is the 't' flag. I tried passing the 'r' flag but it didn't make a difference |
| 19:28.11 | exc_ess | Micc: Is it possible that it's a bug in my Asterisk installation? I'm on a Mac now, and was also having issues where if I enabled converting voicemail to "wav" it would play a static burst instead of the soundfile. So there might be flakinesses tehre |
| 19:30.04 | WindowsUser | maybe your wav player doesn't support 8khz sample rate |
| 19:30.41 | Micc | exc_ess, what order did you build asterisk and dahdi in? |
| 19:31.49 | exc_ess | I used macports, which did the install |
| 19:31.55 | exc_ess | so I think it must have done dahdi first |
| 19:32.07 | exc_ess | I remember that the last thing it compiled was asterisk |
| 19:32.50 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:32.56 | exc_ess | But why should that matter? I don't have any "hardware device interfaces" |
| 19:33.27 | Micc | exc_ess, It matters a lot. Thats the main problem I find with new installations and reinstallations. |
| 19:33.41 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 19:33.42 | Micc | exc_ess, make sure dahdi and dahdi_tools are installed first. |
| 19:33.59 | Micc | exc_ess, you don't need most of it, but for some reason it is needed for some basic asterisk functionality. |
| 19:34.21 | Micc | esc_ess, and its good to install dahdi_dummy too. |
| 19:34.25 | WindowsUser | how does moh determine which file to play? |
| 19:34.28 | Micc | what version of asterisk is this? |
| 19:34.38 | WindowsUser | every time i use hold its fpm-world-mix |
| 19:34.48 | exc_ess | 1.6.1.1 |
| 19:35.03 | Micc | WindowsUser, from the config file for moh and the musicclass defined in your dialplan. |
| 19:35.17 | Micc | or in sip.conf for an account. |
| 19:35.46 | Micc | WindowsUser, I haven't played with that in a long time though. Just look at the music on hold page on voip-info.org |
| 19:36.06 | exc_ess | Micc: The only files in my install that have dahdi in their names are "chan_dahdi.conf.sample" and "spy-dahdi.gsm". I'm guessing that means it's not installed |
| 19:36.19 | Micc | WindowsUser, there is a command like "moh show" that will show you which classes and the files you have defined. |
| 19:37.06 | Micc | exc_ess, if you didn't build it and install it yourself then its probably not installed. Although there could be a package to install it. |
| 19:37.10 | WindowsUser | oh i probably just need to turn random on |
| 19:37.22 | Micc | But its a pretty particular thing that needs to compile with your kernel sources. |
| 19:37.53 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 19:37.54 | Micc | WindowsUser, moh show classes |
| 19:37.55 | exc_ess | Micc: Looks like they don't have a dahdi package in macports, so I'd have to do that by hand |
| 19:37.59 | exc_ess | ugh |
| 19:38.09 | WindowsUser | aye |
| 19:38.10 | Micc | and moh show files |
| 19:38.25 | WindowsUser | yea i can tab surf the console |
| 19:38.32 | WindowsUser | thx tho |
| 19:38.42 | Micc | exc_ess, yeah its a bit of a pain, you'll need kernel-devel and kernel-src too I think. |
| 19:38.51 | exc_ess | Micc: Ah well, I think what this means is that I won't be fixing anything tonight. Damn I was hoping it was something trivial. Thanks Micc. |
| 19:39.00 | exc_ess | Why is nothing in life simple? |
| 19:39.04 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 19:39.07 | Micc | WindowsUser, if your file isn't listed in moh show files then it won't play it. |
| 19:39.47 | Micc | exc_ess, sex is simple. You could do it even if you didn't know how. |
| 19:40.36 | [TK]D-Fender | Micc: Probably just fuck it up... and then you're screwed ;) |
| 19:40.45 | exc_ess | Good point. I should probably be testing that theory out instead of being in the office on a Saturday |
| 19:41.06 | exc_ess | On that note, bye all. Thanks for all the help |
| 19:41.27 | Micc | exc_ess, good luck with your simple task. ;) |
| 19:42.02 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:42.16 | Micc | TKD-Fender, I did get that client's polycom's working with the sonicwall. |
| 19:42.24 | Micc | They are very happy now. |
| 19:43.01 | Micc | I had to turn off all the SIP settings that it had. I guess it just screws things up more for some reason. |
| 19:43.35 | [TK]D-Fender | exc_ess: Micc Funny, that sounds just like the very first thing I told you ;) |
| 19:43.59 | *** join/#asterisk sah-work (n=Bawbatos@99-40-7-14.lightspeed.sntcca.sbcglobal.net) |
| 19:44.34 | Micc | Yup, I had to turn off the NAT setting in there too. |
| 19:44.40 | Micc | not just transforms. |
| 19:45.24 | Micc | Now their only problem is the dsl modem is in bridge mode which means it won't use the QoS settings. |
| 19:45.45 | Micc | It'll cost them another 300$ for the sonicwall premium that does QoS. |
| 19:46.02 | [TK]D-Fender | SonicWALL = pricey toaster |
| 19:46.21 | Micc | TKD-Fender, do you have any experience with the sonicwall QoS? Is it worth it for them to get that or should they just double NAT it? |
| 19:47.43 | Micc | I'm working on talking points to present to 3 VC's on tuesday. |
| 19:48.29 | Micc | Business is good, but it could be better with a quick boost of capital. |
| 19:49.21 | Micc | I'm not sure how much detail I should go into on VoIP. |
| 19:49.36 | [TK]D-Fender | Micc: Never really dealt with QoS personally |
| 19:50.17 | Micc | TKD, I would have though you use it all the time. Its a must for our customers. |
| 19:50.44 | Micc | I suppose it wouldn't be a problem if your devices are all internal. |
| 19:51.01 | *** join/#asterisk lolo2 (n=lolo@c-69-180-160-4.hsd1.mn.comcast.net) |
| 19:51.58 | [TK]D-Fender | Micc: Most of mine are very low volume or use hardware for PSTN |
| 19:52.17 | lolo2 | can someone pls help. If i have two simultaneous callers the first connects just fine but the second one just keeps ringing the odd things is that the log tells me its playing the sounds the the 2nd caller |
| 19:52.38 | lolo2 | and the first caller |
| 19:53.54 | Micc | lolo2, pastebin the appropriate section of your dialplan and your sip.conf accounts involved(trunks and devices) |
| 19:59.55 | lolo2 | my sip http://pastebin.com/m17312fd6 |
| 20:00.25 | lolo2 | i am going google voice to gizmo5 and asterisk to gizmo5 |
| 20:01.57 | [TK]D-Fender | lolo2: pastebin the failed call with SIP DEBUG enabled |
| 20:08.27 | lolo2 | http://pastebin.com/d6e2e1ef1 |
| 20:09.29 | *** join/#asterisk SuPrSluG (n=SuPrSluG@96.243.10.231) |
| 20:11.05 | [TK]D-Fender | lolo2: I don't see an outgoing call. |
| 20:11.16 | lolo2 | they are 2 incomming calls |
| 20:11.26 | lolo2 | i removed one number as it is not mine |
| 20:11.55 | lolo2 | one the first caller gets in |
| 20:12.12 | [TK]D-Fender | lolo2: is your * on a public IP? |
| 20:12.46 | lolo2 | yes |
| 20:14.26 | [TK]D-Fender | lolo2: I'd still specify nat=no for your peer, and "canreinvite=no" under [general] |
| 20:14.31 | [TK]D-Fender | lolo2: Just to be sure. |
| 20:14.46 | lolo2 | ok |
| 20:20.03 | lolo2 | ok made the changed but still same thing |
| 20:20.51 | Micc | lolo2, how are the two calls being made and from what device? cell phones? |
| 20:21.06 | lolo2 | one from cell one from land line |
| 20:21.30 | Micc | lolo2, does gizmo support multiple calls per account? |
| 20:22.02 | lolo2 | i belive so. If it did not would i still gets the messages thats its playing to both users? |
| 20:22.53 | Micc | lolo2, try adding defaultuser=17473118234 |
| 20:23.06 | Micc | lolo2, if your using 1.6 I think username has changed to defaultuser |
| 20:23.27 | lolo2 | im useing 1.4 |
| 20:25.45 | lolo2 | i have username=17474749813 |
| 20:26.26 | Micc | ok, thats fine then. |
| 20:26.43 | Micc | I would make sure gizmo supports multiple calls at once using the same account. |
| 20:26.47 | Micc | It could be them that is misconfigured. |
| 20:28.06 | lolo2 | i belize they do becouse by random chance i got it to work but then it never worked again |
| 20:28.51 | Micc | Thats not really a confirmation. |
| 20:28.58 | [TK]D-Fender | lolo2: so #2 simply never works? |
| 20:28.58 | lolo2 | lol sorry |
| 20:29.20 | lolo2 | never -1 time |
| 20:29.48 | lolo2 | and yes allways the 2nd caller |
| 20:32.57 | lolo2 | i had the same thought as micc but then i was confused as to how the log would show that the second caller is getting the sounds... if gizmo5 only allows one at a time when the log would not show that right? |
| 20:37.29 | *** join/#asterisk Tagor (n=none@s55928c6d.adsl.wanadoo.nl) |
| 20:38.07 | Tagor | I'm using APF + Asterisk. The Asterisk manual says I only have to open port 5060 for sip. But I noticed my hardware phone is connected on port 5062. Which ports do I need to configure in APF? |
| 20:39.00 | [TK]D-Fender | APF? |
| 20:39.30 | Tagor | APF firewall |
| 20:39.43 | Tagor | It's a firewall that works with iptables |
| 20:42.28 | [TK]D-Fender | Tagor: * only needs 5060 +your RTP range |
| 20:42.40 | lolo2 | d_fender: if i use a softphone and dial my gizmo number directly and a cell and dial the did it works?? |
| 20:43.05 | [TK]D-Fender | lolo2: .... I can't tell the question part of that... |
| 20:43.10 | Tagor | [TK]D-Fender >> What's the RTP range? Is that the range used for the clients? |
| 20:43.25 | [TK]D-Fender | Tagor: SIP = signalling, RTP = VOICE |
| 20:43.34 | [TK]D-Fender | Tagor: * typically uses 10000-20000 |
| 20:43.43 | lolo2 | why does it work with one soft phone and one cell but not one land line and 1 cell? |
| 20:45.04 | [TK]D-Fender | lolo2: does the cell fail in scenario #2? |
| 20:46.31 | lolo2 | they both get connected to the pbx if its cell + softphone or landline + softphone |
| 20:46.44 | lolo2 | just not land line + cell |
| 20:47.25 | Xetrov` | anyone know of a good stable windows mobile sip client? |
| 20:47.57 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 20:50.20 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 20:52.16 | *** join/#asterisk ethicx (n=Administ@adsl-146-66-194.mia.bellsouth.net) |
| 20:52.25 | *** join/#asterisk DarkRift (n=dark@65.92.171.53) |
| 20:53.08 | ethicx | im trying to compile asterisk but when I run make I get this error make:Warning: File 'Makefile' has modification time 2e+08 s in the future ...is this something to do with my clock? |
| 20:56.00 | ethicx | yup it does.\ |
| 20:56.16 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 21:00.14 | Tagor | Is it possible to set maxmessage of the voicemailbox to 0 for unlimited? Or how can I remove the limitation? |
| 21:00.36 | WindowsUser | set it to 9999? |
| 21:00.56 | WindowsUser | it saves msg0000 and up, i think it'd peak at msg9999 |
| 21:01.00 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:01.14 | Tagor | hmm, oke thanks :) |
| 21:11.18 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-186-211-209.bflony.east.verizon.net) |
| 21:12.49 | ethicx | can anyone help me...I finished installing asterisk but when I go to /etc/asterisk/ I have nothing there!! why is this? |
| 21:13.07 | *** join/#asterisk catojo (n=catojo@189.24.3.107) |
| 21:15.27 | ethicx | anyone? |
| 21:18.39 | WindowsUser | so copy them from somewhere? |
| 21:19.11 | WindowsUser | theres a configs directory in the source tarball |
| 21:20.40 | ethicx | k |
| 21:20.44 | [TK]D-Fender | ethicx: because you didn't do "make samples" like the giant advertisement at the end of "make install" told you |
| 21:21.08 | [TK]D-Fender | ethicx: Which would have copied them over |
| 21:21.21 | WindowsUser | i was wondering about that |
| 21:21.25 | ethicx | ohhh =( damn it.. |
| 21:21.34 | ethicx | thx. |
| 21:21.38 | WindowsUser | I compiled asterisk twice yesterday and didn't notice that |
| 21:21.40 | WindowsUser | but hey |
| 21:21.44 | WindowsUser | <-- :) |
| 21:21.44 | ethicx | yup |
| 21:23.18 | ethicx | i got them now..thx |
| 21:26.42 | AlmightyOatmeal | [Aug 1 16:25:33] WARNING[47787]: chan_sip.c:12696 handle_response_invite: Received response: "Forbidden" from '"Jamie Ivanov" <-- i get that when i try to dial the 3 digit extension for another SIP user.. any info/advice would be great |
| 21:29.28 | AlmightyOatmeal | any ideas? |
| 21:30.01 | [TK]D-Fender | AlmightyOatmeal: Yeah, where is the failed call with SIP DEBUG in a pastebin for us to look at? |
| 21:30.56 | AlmightyOatmeal | 2 sec |
| 21:33.36 | AlmightyOatmeal | http://pastebin.ca/1515174 |
| 21:35.18 | [TK]D-Fender | AlmightyOatmeal: THE ENTIRE CALL |
| 21:35.44 | AlmightyOatmeal | i'm trying but i can't just dump the debug text to a file can i? |
| 21:35.55 | AlmightyOatmeal | i can only paste whats in the ssh buffer |
| 21:36.26 | [TK]D-Fender | AlmightyOatmeal: Get a bigger buffer |
| 21:38.22 | AlmightyOatmeal | http://pastebin.ca/1515184 is that any better? |
| 21:39.24 | [TK]D-Fender | AlmightyOatmeal: No. Go get the entire call |
| 21:40.25 | AlmightyOatmeal | http://pastebin.ca/1515187 |
| 21:40.31 | AlmightyOatmeal | had putty log the entire output to a file heh |
| 21:41.04 | [TK]D-Fender | AlmightyOatmeal: Go into your putty config and just enlarge the scrollback to 2000 lines or so. |
| 21:41.30 | AlmightyOatmeal | or just tell putty to dump everything to a log file like i did |
| 21:41.39 | AlmightyOatmeal | that is *everything* |
| 21:44.30 | AlmightyOatmeal | or try http://pastebin.ca/1515189 |
| 21:44.36 | [TK]D-Fender | AlmightyOatmeal: Contact: <sip:s@192.168.1.50> <-- you have not set your system up properly to work behind NAT |
| 21:44.38 | [TK]D-Fender | ~sipnat |
| 21:44.38 | 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 |
| 21:44.39 | [TK]D-Fender | ^^^^^^ |
| 21:44.52 | AlmightyOatmeal | [TK]D-Fender: yes i have, everything else works just fine |
| 21:45.06 | [TK]D-Fender | AlmightyOatmeal: That contact begs to differ |
| 21:45.06 | AlmightyOatmeal | the user i'm trying to call is outside my network though connecting in via sip |
| 21:46.18 | AlmightyOatmeal | the user, 'deshi' that i'm trying to call is behind a nat on a remote network |
| 21:46.46 | AlmightyOatmeal | otherwise all of my other inbound/outbound calls and extensions work just fine, its just his extension |
| 21:47.09 | [TK]D-Fender | AlmightyOatmeal: Well the remote side says "forbidden" and appears to be from an * box |
| 21:47.18 | [TK]D-Fender | AlmightyOatmeal: we'd have to see their configs too |
| 21:47.39 | AlmightyOatmeal | hmm |
| 21:47.48 | AlmightyOatmeal | afaik he's using a linksys spa phone adapter |
| 21:48.16 | [TK]D-Fender | AlmightyOatmeal: lines 217 & 230 |
| 21:48.25 | AlmightyOatmeal | of my latest paste? |
| 21:48.30 | [TK]D-Fender | AlmightyOatmeal: AlmightyOatmeal Yes |
| 21:48.35 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:48.38 | [TK]D-Fender | AlmightyOatmeal: Where is shows the UA rather blatantly |
| 21:50.06 | AlmightyOatmeal | i dont see anything on likes 217 and 230 of my last 2 pastebins |
| 21:50.10 | AlmightyOatmeal | lines* |
| 21:50.11 | *** join/#asterisk ingenius (n=alektro@188-213-17-190.fibertel.com.ar) |
| 21:50.20 | [TK]D-Fender | AlmightyOatmeal: http://pastebin.ca/1515187 |
| 21:50.26 | [TK]D-Fender | AlmightyOatmeal: Not your last it seems |
| 21:50.42 | AlmightyOatmeal | 230 says # |
| 21:50.43 | AlmightyOatmeal | Call-ID: 7b9c43457644faf253ae515257bd984d@192.168.1.50 |
| 21:50.52 | AlmightyOatmeal | oh 234 |
| 21:50.58 | AlmightyOatmeal | i'll have to talk to him about that then |
| 21:51.11 | AlmightyOatmeal | thanks for the heads up |
| 21:51.26 | [TK]D-Fender | AlmightyOatmeal: 83 & 90 on the latest |
| 21:51.51 | AlmightyOatmeal | ty |
| 21:52.23 | AlmightyOatmeal | thats not *my* asterisk box is it? |
| 21:52.37 | [TK]D-Fender | AlmightyOatmeal: the To: header should also have been a hint |
| 21:52.51 | [TK]D-Fender | AlmightyOatmeal: To: <sip:s@71.90.85.228> <- only * tells another system to contact them at "s" |
| 21:52.52 | AlmightyOatmeal | shrugs |
| 21:53.04 | AlmightyOatmeal | um |
| 21:53.19 | AlmightyOatmeal | hmm |
| 21:53.21 | [TK]D-Fender | AlmightyOatmeal: there's a reason I gave you 2 line #'s. Becase the first showed it is a READ |
| 21:53.50 | AlmightyOatmeal | do you think thats a config issue with my box or his then? |
| 21:53.58 | AlmightyOatmeal | i didn't realize he was using asterisk to connect to me |
| 21:54.02 | [TK]D-Fender | AlmightyOatmeal: He's refusing you. God knows why. |
| 21:54.10 | AlmightyOatmeal | that bastard *cries* |
| 21:54.11 | AlmightyOatmeal | hehe |
| 21:54.14 | [TK]D-Fender | AlmightyOatmeal: As bad as you can screw up your system, he can match you... |
| 21:54.42 | AlmightyOatmeal | well my system isn't screwed up as far as i can tell |
| 21:55.11 | AlmightyOatmeal | i'll have to work with him on that |
| 21:55.18 | [TK]D-Fender | AlmightyOatmeal: I have no idea what your proxy is doing to the packets I'm seeing so I'm still skepticle |
| 21:55.29 | [TK]D-Fender | AlmightyOatmeal: it LOOKS bad. |
| 21:55.41 | AlmightyOatmeal | oh? |
| 21:56.00 | AlmightyOatmeal | how does it look bad? |
| 21:56.08 | AlmightyOatmeal | pouts |
| 21:56.12 | [TK]D-Fender | AlmightyOatmeal: AlmightyOatmeal bad contact headers, etc, the fact it considers local IP's "NAT", etc |
| 21:56.26 | AlmightyOatmeal | that is odd |
| 21:57.18 | AlmightyOatmeal | i have localnet=192.168.1.0/24 which should tell it my local ip's are not nat, right? |
| 21:59.16 | [TK]D-Fender | AlmightyOatmeal: depends on the peer definition as well.... what are you running that proxy for? |
| 22:00.05 | AlmightyOatmeal | the proxy is my sip provider? |
| 22:00.16 | [TK]D-Fender | AlmightyOatmeal: the sipproxd you run though |
| 22:01.02 | AlmightyOatmeal | ah that, siproxd is running on my router becuase i've been having trouble with registering with my sip provider and etc |
| 22:01.39 | [TK]D-Fender | AlmightyOatmeal: Probably for doing the rest of the NAT config wrong. Have * do its own job. it obfuscates your debugging |
| 22:02.55 | AlmightyOatmeal | pf seems to be hurting SIP and RTP packets |
| 22:04.05 | AlmightyOatmeal | i'll try it without the proxy and see what happens again |
| 22:04.18 | AlmightyOatmeal | gf wants to go to the store for potato salad :P |
| 22:04.54 | [TK]D-Fender | AlmightyOatmeal: Words can barely describe how tragically white that sounds.... |
| 22:05.10 | AlmightyOatmeal | you wouldn't believe how white i am :P |
| 22:05.29 | [TK]D-Fender | AlmightyOatmeal: Oh, I wouldn't bet on that ;) |
| 22:05.48 | AlmightyOatmeal | i'm so white i glow in the dark :P |
| 22:05.58 | AlmightyOatmeal | but potato salad beacons |
| 22:06.12 | AlmightyOatmeal | afk(god+i_hate_my_life); |
| 22:07.40 | [TK]D-Fender | beckons <- |
| 22:07.50 | [TK]D-Fender | AlmightyOatmeal: Watch out for that flashing light! |
| 22:21.24 | *** join/#asterisk Katty (n=Katty@mail.copi-rite.com) |
| 22:21.34 | Katty | mew |
| 22:24.19 | [TK]D-Fender | kattMew. |
| 22:29.17 | Tagor | I have one context [voicemail] in my voicemail.conf. I changed the emailbody and serveremail but it won't use it for some reason. Anybody who knows what I'm doing wrong? |
| 22:29.25 | Tagor | And yes I did reload/restart Asterisk |
| 22:34.05 | *** join/#asterisk shido6 (n=shido6@dsl-67-212-24-92.acanac.net) |
| 22:35.30 | *** join/#asterisk sjobeck (n=Adium@pool-98-108-149-215.ptldor.fios.verizon.net) |
| 22:35.47 | *** part/#asterisk sjobeck (n=Adium@pool-98-108-149-215.ptldor.fios.verizon.net) |
| 22:36.32 | *** join/#asterisk marl_scot (n=marl_sco@78.151.77.116) |
| 22:37.38 | AlmightyOatmeal | [TK]D-Fender: har |
| 22:37.41 | AlmightyOatmeal | :P |
| 22:38.03 | marl_scot | hi folks, can anyone shed some light on why the following keeps failing? exten => 01415312345,n,,Set(CALLERID(num)=${IF($["${DB(cidlookup/${CALLERID(num)})}" = ""]?${CALLERID(num)}:${DB(cidlookup/${CALLERID(num)})})}) * just bombs out with 'No application '' for extension' :( |
| 22:39.18 | marl_scot | lol, its ok, just spotted it after i hit return!!! i have 2 commas after the 'n' :( |
| 22:39.24 | marl_scot | sorry to bother u |
| 22:41.39 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:41.54 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:48.21 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 22:56.12 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:00.05 | *** join/#asterisk propellerhead (n=yogurt2u@host183.190-136-104.telecom.net.ar) |
| 23:02.40 | Tagor | I have a file mailbox.gsm in /var/lib/asterisk/sounds. But I keep getting this error: file.c:602 ast_openstream_full: File mailbox does not exist in any format |
| 23:02.51 | Tagor | Does anyone know what I did wrong? |
| 23:03.41 | WindowsUser | is that the right dir? |
| 23:03.57 | Tagor | I thought so. The wiki says |
| 23:05.21 | WindowsUser | try using it more specifically Playback(/var/lib/asterisk/sounds/mailbox) |
| 23:07.57 | Tagor | Hmm that works |
| 23:08.18 | Tagor | Have you got any idea where I can set the default location? |
| 23:12.33 | *** join/#asterisk manxpower (n=eric@134.sub-70-221-216.myvzw.com) |
| 23:17.59 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-41.tricom.net) |
| 23:20.51 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 23:26.01 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:42.26 | *** join/#asterisk keebler64 (n=chris@adsl-75-17-124-183.dsl.rcsntx.sbcglobal.net) |
| 23:42.46 | keebler64 | Does anyone run Asterisk on an EEE PC?\ |
| 23:45.15 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:45.26 | Qwell | keebler64: there's no reason it wouldn't run |
| 23:46.17 | keebler64 | I know it would runn |
| 23:46.21 | keebler64 | Im installing it now. |
| 23:46.33 | keebler64 | Just want to know how many concurrent calls it can support. |
| 23:46.34 | WindowsUser | why are you running asterisk on an eeepc? |
| 23:46.45 | Qwell | keebler64: just like any other platform - the answer is "it depends" |
| 23:46.47 | keebler64 | Because I want to? |
| 23:46.48 | WindowsUser | 4? 10? |
| 23:47.10 | keebler64 | WindowsUser: Because I don't feel like setting up my desktop to test my new network. |
| 23:47.16 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-41.tricom.net) |
| 23:47.22 | WindowsUser | the only people that run things soley because they want to are OS/2 and Haiku users |
| 23:47.36 | WindowsUser | everyone else has some sort of method to thier madness |
| 23:47.39 | keebler64 | Originally I had it running on my WindPC server. |
| 23:47.49 | WindowsUser | ah |
| 23:47.49 | keebler64 | But again, don't feel like setting it up. |
| 23:48.05 | WindowsUser | what model of eeepc? |
| 23:48.11 | keebler64 | since the EEEPc and WIND are similar setups. I figured it would be suitable to make comparative tests. |
| 23:48.14 | keebler64 | 900 |
| 23:48.19 | keebler64 | It actually have the celery |
| 23:48.21 | keebler64 | not hte atom |
| 23:48.24 | keebler64 | has |
| 23:48.32 | keebler64 | Running FreeBSD |
| 23:48.46 | WindowsUser | cool |
| 23:48.51 | keebler64 | Typing on the floor, so my grammar is a bit skewed. |
| 23:49.11 | keebler64 | had asterisk running fine on the Wind the past year. |
| 23:49.32 | keebler64 | Supporting 6 concurrent calls. (never tried more than that.) |
| 23:52.15 | *** join/#asterisk PaulAviles (n=paviles@dsl-7-36.cofs.net) |
| 23:52.25 | PaulAviles | hello all |
| 23:52.29 | PaulAviles | good day |
| 23:52.53 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 23:53.26 | PaulAviles | can anyone suggest a good open source browser based soft phone? |
| 23:54.03 | PaulAviles | if such thing exist... |
| 23:54.38 | keebler64 | also, anyone on the asterisk+skype beta? Just got the invite the other day. |
| 23:54.51 | afink | never heard of anything like that |
| 23:55.12 | afink | Paul have u tried xlite? |
| 23:55.14 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 23:55.34 | PaulAviles | afink: : yes, but we are looking for a web based one... |
| 23:55.44 | PaulAviles | xlite works great though.. |
| 23:57.48 | afink | cool idea. I'm sure someone has tried something similar before |
| 23:58.07 | PaulAviles | keebler64 : we did get the beta the other day but got dissapointed as you will need a commercial skype account |
| 23:58.26 | PaulAviles | afink : that I what I thought... |
| 23:58.37 | keebler64 | ah. Well, its a shame I got it so late. I just quit my job last week. |
| 23:58.49 | WindowsUser | is it possible to see which file is being played by the music on hold system? |
| 23:59.04 | PaulAviles | I have not installed yet |
| 23:59.16 | afink | just watch the cli on verbose 3 or so |
| 23:59.16 | PaulAviles | is an rpm though |
| 23:59.33 | PaulAviles | why do you care on the moh? |
| 23:59.37 | WindowsUser | the rpm is intended for asterisknow only btw |
| 23:59.50 | afink | should say somthing like playing moh class default |