| 00:00.20 | *** join/#asterisk sponger (n=sean@204.14.240.25) |
| 00:01.56 | mltlnx | ManxPower: thanks, still investigating |
| 00:13.05 | *** part/#asterisk NateHB (n=NateHB@72.34.90.74) |
| 00:19.53 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 00:24.22 | yziquel | How do you get the From: field to be set to a given username, overriding the field value given by a client phone? |
| 00:25.11 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:25.25 | yziquel | fromuser doesn't seem to work, and the f flag in Dial() is of no effect... |
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| 01:06.38 | yziquel | what does 'rport' in the Via: field mean? |
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| 01:29.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:36.46 | Jumpie | sup fellas |
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| 01:45.40 | yziquel | anyone willing to help out, or direct to documentation on authentication with SIP? |
| 01:45.47 | yziquel | or direct me? |
| 01:59.54 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/session) |
| 02:01.02 | yziquel | could someone please point me to a tutorial on how to get authenticated with an SIP proxy? |
| 02:01.44 | Mango | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf should be able to help you |
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| 02:02.44 | yziquel | Mango: thank you very very much. |
| 02:03.03 | Mango | if you have any questions I can try to answer them :) |
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| 02:22.37 | kb3ien | any clueful ways to sync 2 asterisk machine's voicemail directories? |
| 02:24.25 | Mango | ah damn I just saw an article about that yesterday |
| 02:24.28 | Mango | let me see if I can find it again |
| 02:25.05 | kb3ien | i'm seriously thinking of adding hooks to vrrpd to handle the worst of the 'is my copy the most current' issues. |
| 02:25.20 | Mango | exten => s,n,Dial(SIP/${EXTEN}@us3) perhaps? |
| 02:25.23 | Mango | whoops |
| 02:25.26 | Mango | http://opensourcemadness.blogspot.com/2007/03/centralizing-asterisk-voicemail.html |
| 02:25.28 | kb3ien | or not. |
| 02:26.28 | kb3ien | thats a neat dodge, have only one voicemail system, and dont sync it. |
| 02:26.45 | Mango | ah, really? :P |
| 02:27.04 | kb3ien | well, i'm more interested in hardware failover, but thanks. |
| 02:27.12 | Mango | np |
| 02:27.13 | Mango | good luck |
| 02:27.30 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 02:28.39 | kb3ien | i'm thinking of adding an option to vrrpd to allow a command to be executed on becomeing the master, to scan the rsynced-in dirs and select one as the most likely to be current candidate, then cp it into the live dir. |
| 02:30.39 | box2 | are you using linux-ha? because it contains some features like that |
| 02:31.19 | kb3ien | any suggested reading? |
| 02:31.55 | box2 | http://www.linux-ha.org/ |
| 02:35.49 | leifmadsen | I'd just put the voicemail in a DB, and then it doesn't matter where it lives |
| 02:35.54 | leifmadsen | ODBC voicemail storage |
| 02:37.19 | *** join/#asterisk haryv (i=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
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| 02:44.47 | leifmadsen | FYI: http://leifmadsen.wordpress.com/2009/07/30/suzanne-bowen-interviews-leif-madsen-asterisk-rockstar/ |
| 02:45.52 | box2 | heh |
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| 02:56.44 | kb3ien | ODBC is SUCH a beast to build and then it requires replication... |
| 02:56.52 | kb3ien | i'm done; vrrpd needed this feature. |
| 02:57.16 | leifmadsen | really? I find it pretty straight forward |
| 02:57.41 | ManxPower | leifmadsen! leifmadsen! leifmadsen! Autograph! Autograph? |
| 02:57.44 | kb3ien | added S&M to vrrpd, -S CMD and -M CMD. executed when the box becomes a standby or master respectively. |
| 02:58.01 | leifmadsen | ManxPower: for you! never! |
| 02:58.01 | *** join/#asterisk twisted (n=twisted@c-71-207-231-57.hsd1.al.comcast.net) |
| 02:58.01 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:58.03 | kb3ien | i'll even publish. |
| 02:58.08 | twisted | hola |
| 02:58.18 | leifmadsen | WHAT?! |
| 02:58.32 | twisted | is stalking leif |
| 02:58.42 | kb3ien | if you have any howto build ast with odbc on jaunty, let me know. |
| 02:58.54 | kb3ien | oneday i'll try it. |
| 02:58.59 | kb3ien | again *shudder* |
| 02:59.07 | leifmadsen | kb3ien: see the asterisk book, chapter..... 10 I think |
| 02:59.13 | leifmadsen | it's really not very hard |
| 02:59.25 | ManxPower | waves to twisted |
| 02:59.27 | leifmadsen | 'apt-get install unixODBC-dev ; ./configure ; make install' |
| 02:59.34 | twisted | Hola Manx |
| 02:59.35 | leifmadsen | (or whatever the pkg name is in ubuntu |
| 03:00.44 | twisted | pokes leifmadsen |
| 03:02.46 | leifmadsen | yes?! |
| 03:02.56 | twisted | hi? |
| 03:03.00 | leifmadsen | HI! |
| 03:03.12 | twisted | I have come to insert the windows 2000 disc |
| 03:04.29 | leifmadsen | nice! |
| 03:04.31 | leifmadsen | in A:? |
| 03:04.39 | twisted | yes. I will cut to fit |
| 03:05.51 | twisted | hm |
| 03:06.01 | twisted | suddenly i feel a sense of rejection |
| 03:14.31 | WindowsUser | :( sfa bug tracker |
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| 03:42.01 | mmlj4 | "VGCP is a proprietary layer 2 link protocol working at between IP stack and nic driver for VoIP anti-blocking." # do what? that gets me only to my cable modem, what kind of scam is this? |
| 03:43.31 | mmlj4 | "meanwhile disguises sip and rtp packets into normal allowed data packets such as DNS and TFTP, and makes two-way encryption and decryption driven by user-customized policy" |
| 03:43.47 | mmlj4 | layer 2? smart folks, indeed |
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| 03:51.10 | MrNeutron | Hello, everyone. |
| 03:53.48 | MrNeutron | There's a server-side asterisk log visualization tool that shows you each step of the call in a different row of an HTML table... |
| 03:54.00 | MrNeutron | I know it uses asterisk-java, but I can't remember the name of the software.... |
| 03:54.48 | MrNeutron | Does anyone happen to know what I'm talking about? Could you tell me what it's called? |
| 04:00.46 | *** part/#asterisk MrNeutron (n=mrneutro@75-148-215-209-Houston.hfc.comcastbusiness.net) |
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| 04:03.07 | kb3ien | okay vrrpd patches submitted. rsync nearly there, where are the berkeley Db files by default? |
| 04:03.36 | jaytee | think they're in /var/lib/asterisk |
| 04:03.58 | jaytee | but you can check in asterisk.conf |
| 04:04.33 | kb3ien | right you are, |
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| 05:21.57 | aces1up | anyone here have the sip firmware package for a cisco 7912 phone, i don't have an account with cisco. |
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| 05:43.37 | picillo | Hello all. |
| 05:44.12 | picillo | I was hoping to get a quick answer to what this may mean on my Asterisk build of 1.2.9.1. |
| 05:44.15 | picillo | In the end Digium support helped out. The problem was fixed by disabling |
| 05:44.31 | picillo | not that. |
| 05:44.35 | picillo | This.. |
| 05:45.21 | picillo | http://pastebin.com/m78dbbe0a |
| 05:52.24 | WindowsUser | whats rejecting the frames? |
| 05:53.59 | *** part/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
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| 06:27.58 | MACscr | is there a command to debug a particular trunk? Im trying to figure out why one isn't registering |
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| 06:30.59 | kaldemar | MACscr: in case of sip peers, sip set debug ip <ip_address> |
| 06:31.23 | MACscr | kaldemar: what's odd is that when I tried that, it said the peer wasn't found |
| 06:31.46 | MACscr | I have a fqdn setup versus an ip for the actual sip trunk if that makes a diff |
| 06:32.33 | kaldemar | did you start debug for an ip address or for a peer? |
| 06:33.49 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 06:36.40 | MACscr | I just set it for the ip |
| 06:36.50 | MACscr | but it shows the rest of the command options like it wasn't enough info |
| 06:37.03 | MACscr | also, the data keeps scrolling away because of all the extra crap |
| 06:37.04 | kaldemar | show the command and the output |
| 06:37.14 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 06:37.14 | MACscr | which is really frustrating about asterisk debugging in general |
| 06:37.14 | kaldemar | ~pb |
| 06:37.15 | 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 |
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| 06:40.06 | MACscr | nvm, I forgot to add the 'ip' part |
| 06:40.36 | MACscr | anyway, now the cli is just sitting there. Is there a command I can use to just get that particular trunk to try to register? |
| 06:40.48 | *** join/#asterisk CrashSys (n=james@azrael.crashsys.com) |
| 06:41.11 | kaldemar | you can't affect the peer's behavior from asterisk |
| 06:41.16 | MACscr | one of my sip trunks are working just fine |
| 06:41.32 | MACscr | the other 3 are from icall and not working |
| 06:42.15 | kaldemar | they don't have to register to you for calls to work if you know their address |
| 06:43.44 | VooDooNOFX | The other day, I was told I could use the same * server to host 2 seperate companies phone systems, but I'm confused on how to actually accomplish this. Can anyone give some advice about doing it? |
| 06:44.46 | VooDooNOFX | I've got 2 seperate SIP trunks via username/password auth, but I need to figure out how to create the seperate company extensions |
| 06:45.10 | VooDooNOFX | CompanyA = 100, 101, 102 and CompanyB = 100, 101, 102 |
| 06:45.13 | MACscr | VooDooNOFX: it can be done, but its not simple |
| 06:45.28 | VooDooNOFX | MACscr: that's what i'm finding :D |
| 06:45.32 | MACscr | you really should subcontract the job if you really are not experienced with asterisk |
| 06:45.34 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:45.45 | VooDooNOFX | I need some guidance on it |
| 06:45.45 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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| 06:46.00 | MACscr | no, you need much more than that =P |
| 06:46.17 | MACscr | if your trying to use all gui for it, its not possible |
| 06:46.30 | VooDooNOFX | yeah, I'm using the config files. gui won't do it |
| 06:46.31 | MACscr | not if its done right |
| 06:46.43 | kaldemar | VooDooNOFX: you need to separate the phone devices in your dialplan |
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| 06:47.32 | *** part/#asterisk CrashSys (n=james@azrael.crashsys.com) |
| 06:48.57 | VooDooNOFX | instead of using [100], I was using names like [CompanyA100], etc |
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| 06:50.22 | VooDooNOFX | the asterisk-gui doesn't even seem to allow templates for extensions |
| 06:51.14 | kaldemar | you can use whatever names you want to. |
| 06:51.34 | kaldemar | they don't have to be numbers. extensions.conf, aka dialplan, is where you define numbers. |
| 06:52.19 | kaldemar | then you define separate contexts for the companies. |
| 06:52.28 | WindowsUser | uhm |
| 06:52.46 | VooDooNOFX | so I'll start by defining the sip trunks for each company. Then create a seperate dial plan for each company. Then add the phones from each company to their seperate dialplans. Is that the generic process? |
| 06:52.59 | WindowsUser | VooDooNOFX: so i have a phone, and dialing 100 will work for CompanyA based on how its configured in sip.conf? |
| 06:53.34 | kaldemar | you add the phone devices also to sip.conf (if SIP is what you use). |
| 06:54.01 | WindowsUser | VooDooNOFX: yea that process sounds right |
| 06:54.18 | VooDooNOFX | yeah, we're 100% sip based atm |
| 06:56.49 | VooDooNOFX | would it be smart, or easier to have extensions.conf include other files, so I can seperate the companies easily? |
| 06:56.54 | VooDooNOFX | or just put it all in one giant file? |
| 06:57.17 | VooDooNOFX | I'm thinking the same type of approach apache has to virtualhosts on some linux distros |
| 06:58.43 | WindowsUser | make it work how you want it |
| 06:59.00 | jkroon | can asterisk #include wildcards? |
| 06:59.28 | WindowsUser | if its a split file you can easily give CompanyA a copy of like companya.extensions |
| 07:00.57 | MACscr | kaldemar: so there is absolutely know way to trigger registration for a trunk? that seems retarded |
| 07:01.17 | kaldemar | jkroon: yes |
| 07:01.33 | jkroon | cool. |
| 07:01.40 | VooDooNOFX | what would the syntax for that look like? |
| 07:01.53 | kaldemar | #include foo*.conf |
| 07:02.20 | VooDooNOFX | and it should perform similar to the apache include, where it does a simple text replacement on the current line |
| 07:03.16 | kaldemar | VooDooNOFX: yes |
| 07:05.38 | VooDooNOFX | ok, i've got to re-read the astbook then. I read through it once, but I need a refresher I think |
| 07:05.55 | VooDooNOFX | thanks for the advice guys, I'd really like to learn how to do this |
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| 07:08.18 | VooDooNOFX | btw, can anyone tell me how to flash an aastra 55i's bootloader code? I update the firmware, but the bootloader seems to redownload the stupid packet8 firmware on every boot. |
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| 07:18.41 | JerJer | ugh - GSM Gateway hell :( |
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| 08:11.46 | Snoogan | if anyone's awake in channel, may i get your thoughts on the best distro to run asterisk with freepbx?? |
| 08:12.21 | box2 | the one you are most familiar with is my vote |
| 08:13.11 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:15.52 | Snoogan | well i'm running asterisk with debian lenny at the moment, with freepbx. Taken a bit to get running, but its doing some strange stuff. Think its mostly the dahdi stuff causing grief |
| 08:16.22 | Snoogan | I'm thinking maybe i should try it under Centos, or go for the AsteriskNow Distro |
| 08:16.36 | Snoogan | might have a play with both and see how they compare |
| 08:19.18 | kaldemar | strange stuff meaning what? |
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| 08:22.03 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 08:22.03 | *** join/#asterisk LeddyHM (n=NONE@mail.artica.net) [NETSPLIT VICTIM] |
| 08:22.03 | *** join/#asterisk scoof (n=scoof@irq.hestdesign.com) |
| 08:22.03 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) |
| 08:25.59 | *** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 08:26.16 | Hatrix | Hello, we just upgraded to astersik 2.4.26, running on debian lenny, kernel 2.6.26-2-686, dahdi 2.2.0 rc5, compiled with THREAD debugging and NO optimizations because we have problemes segfaulting with app_queue, and now we have SEGFAULTS in libc-2.7!! |
| 08:26.31 | Hatrix | Jul 31 09:03:57 logitravel-voip2 kernel: [6557288.850791] asterisk[4670]: segfault at fe28fe30 ip b7d783b3 sp b5c7d19c error 4 in libc-2.7.so[b7d02000+155000] |
| 08:26.43 | Hatrix | HOW can this possible be? segfault in libc? this machines gives us lot's of trouble, we replaced the complete hardware, it's multicore, it has 4 gig of ram, and we have lot's of problems ... |
| 08:27.34 | Hatrix | i posted this as well in asterisk-bugs, but i am not sure if it belongs there |
| 08:32.07 | Snoogan | kaldemar: to start with, bad echos through the TDM410P card. Looking into how to tweak the echo cancelling, otherwise I'll buy the echo cancelling module for the card |
| 08:34.00 | Snoogan | when we try to call out using a dahdi channel, sometimes the line goes dead and when we hangup, to doesn't actually disconnect. You have to wait a few minutes before you can get a dial tone again to call |
| 08:34.16 | *** join/#asterisk yang (i=yang@freenode/sponsor/cacert.assurer.yang) |
| 08:34.25 | Snoogan | phantom calls are also coming in. Just got one then. Rings for a few times and then stops, or if you answer there is noone there. |
| 08:34.48 | box2 | ghost in the machine @_@; |
| 08:36.29 | Snoogan | seems that way :) |
| 08:37.21 | Snoogan | never had much success with asterisknow, but i know a lot more about linux & asterisk now, so might give it another crack |
| 08:45.16 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:fdf7:18b3:d48a:74d6) |
| 08:50.34 | *** join/#asterisk arossouw (n=arossouw@dsl-146-28-60.telkomadsl.co.za) |
| 08:50.54 | arossouw | hello ,what does REC mean on a Wildcard TE122 Card? |
| 08:51.08 | arossouw | it goes rec, then blue |
| 08:51.26 | AdvoWork | Hi there. does anyone know of a free uk voice suite to replace the US voice recordings/prompts on trixbox? |
| 08:55.38 | *** join/#asterisk qu1ckkkk (n=leonja51@dsl-240-205-46.telkomadsl.co.za) |
| 09:02.07 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 09:04.30 | *** join/#asterisk roshan08 (i=d2d40557@gateway/web/freenode/x-aynlkiampmzkggqi) |
| 09:05.17 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 09:05.41 | roshan08 | we intend to make a help center in which we will redirect calls from intended customers to our people on their cell phones, is asterisk suitable for that |
| 09:17.51 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 09:26.25 | *** join/#asterisk s0lid (n=s0lid@203.177.143.137) |
| 09:26.59 | tamiel | roshan08: yes you can dot that |
| 09:27.24 | tamiel | dot/do/ |
| 09:32.23 | roshan08 | tamiel: can you provide me some link so that i can get to know what sort of hardware i need to buy for that |
| 09:35.08 | roshan08 | tamiel: i intend to have 50 simultaneous calls in peak hours |
| 09:35.39 | tamiel | roshan08: it depends on telecom protocols you'll use |
| 09:35.51 | tamiel | roshan08: 50 calls is ok |
| 09:36.11 | tamiel | roshan08: I have more than 120 ISDN calls in peak hours |
| 09:36.23 | tamiel | roshan08: in one asterisk |
| 09:36.32 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 09:36.50 | tamiel | roshan08: (with pri isdn) |
| 09:37.54 | tamiel | roshan08: and 2 digium cards TE412p on this server. |
| 09:38.12 | roshan08 | tamiel: can you start from scratch and explain what you did, i have download a book from the site but could not grab much |
| 09:39.21 | tamiel | roshan08: I have : E1 <-----> Asterisk with 2x TE412p cards |
| 09:39.46 | tamiel | roshan08: E1 is in mix mode so I can receive and transmit calls |
| 09:43.54 | *** join/#asterisk Psychobilly (n=moi@adsl8-176.kln.forthnet.gr) |
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| 10:02.56 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 10:18.31 | *** join/#asterisk bmg505 (n=leon@196-209-79-155-rndf-esr-5.dynamic.isadsl.co.za) |
| 10:19.27 | *** part/#asterisk roshan08 (i=d2d40557@gateway/web/freenode/x-aynlkiampmzkggqi) |
| 10:22.32 | *** join/#asterisk Psychobilly (n=moi@adsl8-176.kln.forthnet.gr) |
| 10:49.57 | Gnarfy | Is it possible to get Dial to use a callerid, but still have CDR/csv display the callerid of the inbound call? |
| 10:58.52 | kaldemar | core show application ForCDR |
| 10:59.19 | kaldemar | core show application ForkCDR, sorry |
| 11:01.06 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 11:04.07 | Gnarfy | hmm, yeah that would work - but was hoping to keep it to 1 CDR line: considering storing the callerid before and restoring it after the dial |
| 11:11.24 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 12:00.10 | Hatrix | can anybody point me to some understanding what this means: |
| 12:00.10 | Hatrix | utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 12:00.12 | Hatrix | ? |
| 12:00.30 | Hatrix | I have it lot's and lot's of times in my logs |
| 12:01.45 | dongs | which part of that is unclear to you??? |
| 12:05.17 | *** join/#asterisk Gnarfy (n=mirc@213.144.244.115) |
| 12:25.11 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:26.03 | Hatrix | dongs: what ast_carefulwrite is trying to write, or how i can figure out which pipe it is that seems to be broken, for what is this used? to communicate with whom/what? |
| 12:36.59 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 12:39.11 | dongs | ... if you give that much of a shit, open the god damn source file and read it? |
| 12:39.43 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:40.22 | Hatrix | dongs: thank you very much for your suggestion, is it full moon today or what? |
| 12:40.59 | dongs | np |
| 12:41.18 | dongs | alternatively you can load asterisk into an awesome and user-friendly debugger like 'gdb' and have at it. |
| 12:45.35 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 12:47.12 | dongs | i wonder if quality of opensores software is directly (inversely?) proportional to the awesomeness of debug/development tools? hehe. |
| 12:52.39 | [TK]D-Fender | opensores definitely calls for being patched.... |
| 12:56.43 | dongs | ya, its 2009 and asterisk still doesnt support sip session timers.. |
| 12:56.46 | dongs | awesomebeans |
| 12:56.50 | dongs | patches have been around to do it for years. |
| 13:00.26 | file | support for SIP session timers was added starting in 1.6.0 |
| 13:00.51 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:01.18 | dongs | rite. |
| 13:03.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:03.57 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:04.47 | Katty | :> |
| 13:04.54 | Katty | GOOD MORNING!!!!! |
| 13:04.55 | [TK]D-Fender | Katty: Mew. |
| 13:05.02 | Katty | hugs [TK]D-Fender |
| 13:05.17 | _ShrikE | Morning Katty! |
| 13:05.22 | Katty | it's friday!!! :> |
| 13:05.27 | Katty | hugs _ShrikE |
| 13:08.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.55 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 13:11.53 | timeshell_atwork | Happy Fivesday! |
| 13:12.14 | ariel_ | morning... when did the sun come out? |
| 13:13.08 | jaytee | morning Katty |
| 13:14.11 | ariel_ | goes out to get a very large Espresso to see if he can wake up...Morning everyone |
| 13:19.07 | Katty | hugs on jaytee |
| 13:19.19 | Katty | ariel_: oooh! i want a hot chocolate and a blueberry muffin while you're out :> |
| 13:19.54 | ariel_ | ok |
| 13:24.15 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 13:26.00 | *** join/#asterisk jtodd (i=tj9lozs9@ns.fox-den.com) |
| 13:26.00 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 13:29.11 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
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| 13:32.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:33.47 | *** join/#asterisk javb (n=Owner@190.166.139.124) |
| 13:34.12 | javb | Hi.. does anyone knows about at least top 5 CDR report tools for Asterisk? |
| 13:34.18 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 13:34.56 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 13:39.37 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:40.50 | timeshell_atwork | Anybody know of any free tools like fop (other than fop?) |
| 13:43.38 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 13:47.32 | [TK]D-Fender | timeshell_atwork: isymphone, HUDLite |
| 13:53.10 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 13:54.21 | *** join/#asterisk mib_mib (n=mib_mib@c-24-30-94-70.hsd1.ga.comcast.net) |
| 13:55.01 | *** join/#asterisk evangelion (n=manzy_ze@ufficio.sonic.it) |
| 13:55.07 | evangelion | hello |
| 13:55.16 | evangelion | anyone who tested chan_skype around? |
| 13:59.39 | dongs | sounds illegal. |
| 14:00.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:02.52 | [TK]D-Fender | Ho informed... |
| 14:02.54 | [TK]D-Fender | how* |
| 14:04.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:04.39 | dongs | surely you arent talkign about me are you. |
| 14:05.28 | *** join/#asterisk nny_atg (n=Scott@64.203.237.47) |
| 14:05.43 | [TK]D-Fender | dong got to keep up with news... |
| 14:05.53 | dongs | Typical configuration for 30 ChanSkype channels: Dual Xeon 3GHz, 6 Gb memory. haha |
| 14:06.28 | [TK]D-Fender | evangelion: And that is a less stable implementation... SFA just hit public beta |
| 14:07.40 | evangelion | [TK]D-Fender: i'm testing it right now, anyway i could not get * to register my account |
| 14:08.04 | dongs | obviously, the problem is on your end then. |
| 14:08.05 | evangelion | skype login user foo fails silently |
| 14:08.17 | dongs | i mean if a software is released, evne in beta form |
| 14:08.25 | dongs | one would expect it to function at least to perform basic tasks. |
| 14:08.27 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:08.30 | evangelion | maybe |
| 14:08.43 | dongs | (duno about lunix software, but thats how it generally is in the real software development world). |
| 14:08.52 | evangelion | well i'm here looking for another user's experience |
| 14:10.27 | grandpapadot | Hey guys, I'm trying to trim the last characater off of a dial string, anyone see my obvious mistake? ---- exten => s,n,Dial(${GROUP:0:LEN(${GROUP})-1}|${ARG2}|rd) |
| 14:10.52 | nny_atg | any way to run a pattern loop test with asterisk running? |
| 14:11.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:11.21 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:13.36 | [TK]D-Fender | grandpapadot: You aren' referencing a function properly there, and you're using deprecated delimiters |
| 14:14.33 | anonymouz666 | why does asterisk make a DNS query when you place a call SIP<->SIP? it's a query TYPE A for... XXX-0931bd78.hostname.com where XXX is one for src and another to dst number? |
| 14:14.58 | dongs | look in teh sores and find out. |
| 14:15.40 | kaldemar | anonymouz666: do you have srvlookup=yes in sip.conf? |
| 14:16.33 | anonymouz666 | kaldemar: no sir. |
| 14:16.51 | anonymouz666 | I do have srvlookup=no in sip.conf. |
| 14:21.58 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 14:22.32 | anonymouz666 | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg225498.html |
| 14:22.38 | anonymouz666 | this is a mistery |
| 14:23.46 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:29.33 | Brixius | Question on http://pastebin.com/d291dc30a when I dial the 7005550001 the call will go through to priority 2, but if I dial any other # it will continue starting on line 2 |
| 14:29.56 | Brixius | wait, it will not continue on line 2 |
| 14:30.43 | Brixius | also let me rephrase that, to priority 2 |
| 14:30.53 | Brixius | (It's early...) |
| 14:33.01 | Stese | Does NoOp not need ()? |
| 14:33.06 | kaldemar | Brixius: what is the question? |
| 14:33.19 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:33.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:33.20 | kaldemar | Stese: no |
| 14:33.36 | Stese | then it's not that... :P over to you ;) |
| 14:34.16 | Brixius | Ok, I have a dialplan that should answer all incomming #'s, but if it's for example purposes 7005550001 it should do something else that was a goto origionally, but no sense pasting the part where I'm playing back tt-monkies to test this. |
| 14:34.22 | [TK]D-Fender | Brixius: Becasue you don't have a priority 1 to match |
| 14:34.37 | kn0x | so anyone going to cluecon? |
| 14:34.51 | Brixius | ok, that makes sense I guess. |
| 14:35.05 | Brixius | I'll switch _X. to start at 1 |
| 14:35.30 | Brixius | like I said, it's early and it's been a while since I wrote a dialplan. |
| 14:37.56 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:37.56 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.04 | Brixius | bangs head on desk for being an idiot... |
| 14:39.05 | jaytee | You better stop, look around, Here it comes, here it comes, here it comes, here it comes, Here comes your nine-teenth nervous breakdown. |
| 14:39.19 | leifmadsen | rocks out |
| 14:39.30 | Brixius | no, I'm up to at least 23 by now |
| 14:39.33 | Brixius | :) |
| 14:39.36 | jaytee | hehe |
| 14:40.11 | [TK]D-Fender | I've only had one small one... caused by resisting the urge to kill a coworker.. |
| 14:40.36 | [TK]D-Fender | I shouldn't be so repressed, but the dry-cleaning bill would become astronomical |
| 14:40.45 | jaytee | :-) |
| 14:41.48 | Brixius | does anyone else think the command :q! should exit asterisk console. :) |
| 14:42.46 | afink | no |
| 14:42.59 | *** join/#asterisk shido6 (n=shido6@dsl-67-212-24-92.acanac.net) |
| 14:43.25 | afink | even though I have tried wq many times |
| 14:43.28 | afink | it still doesn't work |
| 14:43.40 | Brixius | that was the other one I type in alot |
| 14:44.14 | kn0x | [TK]D-Fender: where do you work? |
| 14:44.26 | [TK]D-Fender | afink: You can create a CLI alias for the. |
| 14:44.37 | [TK]D-Fender | kn0x: At my office... |
| 14:44.43 | kn0x | -_- |
| 14:44.44 | [TK]D-Fender | kn0x: Sometimes at home... |
| 14:45.02 | kn0x | what do you do? |
| 14:46.06 | [TK]D-Fender | kn0x: IT for a plumbing manufacturer/distributor |
| 14:46.14 | kaldemar | Brixius: 1.6.2 will have cli aliases for that. :) |
| 14:46.37 | kaldemar | oh, i'm echoing with some distortion. |
| 14:46.49 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 14:47.38 | [TK]D-Fender | kaldemar: Alias' have been around since 1.6.0 |
| 14:49.35 | kaldemar | res_clialiases is introduced with 1.6.2 |
| 14:51.13 | [TK]D-Fender | kaldemar: Maybe a more formal replacement method... |
| 14:51.35 | kaldemar | there is a separate configuration file for it. |
| 14:54.22 | dongs | lol @ chan_skype requiring X server and VNC |
| 14:54.27 | dongs | thats hilarious |
| 14:55.21 | [TK]D-Fender | ~skype |
| 14:55.22 | infobot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 14:55.26 | [TK]D-Fender | ^^^^ |
| 14:55.28 | [TK]D-Fender | HACKS |
| 14:55.49 | [TK]D-Fender | dongs: We've been laughing at these implementations since their inception |
| 15:01.14 | grandpapadot | Thanks, TK. Hey, would "${SIPPEER(${MEMBER}:status)}" != "UNKNOWN" be a valid and good way to check for an online peer if qualify is enabled? |
| 15:02.43 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:04.29 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 15:05.57 | *** join/#asterisk Akiyuki (n=Jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 15:06.11 | Akiyuki | Is it possible to stream mgcp calls? |
| 15:06.35 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 15:06.59 | grandpapadot | Way cool, this works great for making sure a peer exists and is online: ExecIf($[ "${SIPPEER(${ROUTER}:status)}" != "UNKNOWN" && "${SIPPEER(${ROUTER}:context)}" == "${MACRO_CONTEXT}" ]|Set|ROUTERS=${ROUTERS}SIP/${METRIC}@${ROUTER}&) |
| 15:07.52 | ThoMe | hello. |
| 15:07.56 | ThoMe | have a little proble: |
| 15:07.57 | ThoMe | [Jul 31 16:57:54] NOTICE[24545]: chan_sip.c:13885 handle_request_invite: Sending fake auth rejection for user "01786323765" <sip:01786323765@dus.net>;tag=as40b43fc2 |
| 15:08.10 | ThoMe | i have never seen "sending fake auth".. ? |
| 15:08.13 | ThoMe | any ideas? |
| 15:08.53 | *** join/#asterisk robert_penz2 (n=quassel@mk093111018099.a1.net) |
| 15:09.57 | robert_penz2 | Hi, my asterisk has just been misused for an DDOS attack. I've stoped the VM with it. The only stuff thats looks not normal in the logfiles is |
| 15:10.25 | robert_penz2 | Jul 31 13:42:40 asterisk -- MARK -- |
| 15:10.25 | robert_penz2 | Jul 31 14:02:40 asterisk -- MARK -- |
| 15:10.25 | robert_penz2 | Jul 31 14:16:34 asterisk init: Failed to open console: Permission denied |
| 15:10.25 | robert_penz2 | Jul 31 14:16:35 asterisk asterisk[7958]: rc_avpair_new: unknown attribute 1490026597 |
| 15:10.25 | robert_penz2 | Jul 31 14:16:35 asterisk last message repeated 6 times |
| 15:10.26 | dongs | lol, someone finally figured out how to write a bandwidth lens for SIP? |
| 15:10.26 | robert_penz2 | Jul 31 14:16:36 asterisk exiting on signal 15 |
| 15:11.37 | robert_penz2 | using asterisk (1:1.4.17~dfsg-2ubuntu1) shiped with hardy |
| 15:11.49 | robert_penz2 | I didn't find anything on google |
| 15:11.55 | robert_penz2 | can someone help me please? |
| 15:12.02 | *** join/#asterisk dominic1 (n=Miranda@213.221.82.242) |
| 15:12.27 | dominic1 | Hello guys, can i use completely different table layouts with adaptive? |
| 15:12.36 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:12.39 | Zeeek | wake up |
| 15:13.23 | robert_penz2 | the server which is only running asterisk did send following |
| 15:13.31 | robert_penz2 | 16:08:25.652144 IP 188.40.xxx.xxx.17208 > 91.13.126.179.10600: UDP, length 172 |
| 15:13.31 | robert_penz2 | 16:08:25.654125 IP 188.40.xxx.xxx.16004 > 194.24.158.2.35705: UDP, length 172 |
| 15:13.31 | robert_penz2 | 16:08:25.655069 IP 188.40.xxx.xxx.16634 > 87.156.87.6.10616: UDP, length 172 |
| 15:13.32 | Zeeek | join us in 45 minutes for #voip-users-conference http://VUC.me - Skype for Asterisk Opens Beta, free SIP calls via GV+GV |
| 15:14.31 | dongs | wat |
| 15:14.34 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-42-42.dhcp.mrqt.mi.charter.com) |
| 15:14.41 | Gnarfy | robert_penz2: how are you sure its asterisk related and not just the box that has been hacked? |
| 15:15.01 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 15:16.45 | grandpapadot | Hey guys, I'm getting a console warning "Missing }" on this line, anyone see my oversight? exten => s,n,Set(GROUP_LENGTH=${MATH(${GROUP_LENGTH}-1,int)}) |
| 15:17.31 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 15:17.49 | grandpapadot | Nevermind, found it, can't count. |
| 15:18.00 | *** join/#asterisk mrkiko (n=mrkiko@ad2.z1.infracom.it) |
| 15:18.21 | mltlnx | hi, is there anyway to generate a SUBSCRIBE for a device. It appears that a reload wipes out the subscriptions and there does not appear to be a way to set the re-SUBSCRIBE timer lower on the device. |
| 15:18.24 | robert_penz2 | Gnarfy: only asterisk port open by firewall |
| 15:18.32 | mrkiko | Hi all! I'm a newbie. can someone tell me how to write an interactive voice responder? Or better - cansomeone give me a manual page or something to read??? |
| 15:18.40 | robert_penz2 | and only asterisk running on that VM |
| 15:19.22 | grandpapadot | On call queues, has anyone had success making remote peers members of call queues in 1.4.26? i.e., member => 806@someremoteasteriskbox ??? It works the first time and then the queue member gets stuck in a "ringing" state until you reload app_queue |
| 15:19.26 | mltlnx | http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
| 15:19.56 | mrkiko | thank you |
| 15:20.55 | *** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net) |
| 15:22.20 | eric_hill | Is there an elegant way for a user on [asterisk-box-1] to transfer a voicemail to a user on [asterisk-box-2]? |
| 15:22.36 | Akiyuki | Does anyone know how to convert rtp dump into a wav format? |
| 15:22.47 | mltlnx | Why are subscriptions not persistent through a reload? |
| 15:23.04 | grandpapadot | Akiyuki: sniffing someones RTP stream are we? |
| 15:23.06 | [TK]D-Fender | mltlnx: Because what they subscribe to may no longer exist... |
| 15:23.21 | eric_hill | Akiyuki: http://www.cs.columbia.edu/irt/software/rtptools/ |
| 15:23.31 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:23.32 | Akiyuki | grandpapadot, i can play the rtp back through wireshark, but was hoping to have a command line option |
| 15:23.47 | Akiyuki | eric_hill, thats where im at now, but i dont see the option to convert it to a wav file |
| 15:24.13 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/session) |
| 15:24.48 | mltlnx | [TK]D-Fender: I see, but what does one do when there device will only re-subscribe every 3600 (not configurable). I can of course reboot the phones, but thats not practical. |
| 15:25.48 | angryuser | Akiyuki, hello try to use siworkbench, you can extract rtp streams from there and import to audacity as ulaw/whatever 8000hz |
| 15:25.50 | angryuser | sipworkbench* |
| 15:26.12 | Akiyuki | angryuser, thanks, i will look intot hat... looking for a way to stream rtp packets on a shoutcast or equiv server |
| 15:26.17 | [TK]D-Fender | mltlnx: What ver of *? |
| 15:26.31 | mltlnx | 1.6.1.0 |
| 15:26.51 | angryuser | Akiyuki, in real time ? sip workbench dont know hoq to do that |
| 15:27.06 | Akiyuki | angryuser, oh, is there a program that can do it from the command line then? |
| 15:28.16 | angryuser | Akiyuki, try to search around audiojack http://jackaudio.org/ not sure about converting |
| 15:28.44 | dongs | how is that rtptrash questioneven asterisk related. |
| 15:28.45 | mrkiko | ok... where can I find a reference for those instructions, such as background(), agi() and so on? |
| 15:28.48 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:29.48 | Stese | heres a question.... what is with the sound tt-weasels ? |
| 15:30.02 | [TK]D-Fender | mrkiko: "core show application background" <- etc |
| 15:30.16 | dongs | that was from before markster got out of the closet. |
| 15:30.20 | *** join/#asterisk thieums (n=mathieu@LPuteaux-156-16-101-43.w80-12.abo.wanadoo.fr) |
| 15:30.38 | [TK]D-Fender | Stese: Weasels..... have eaten our phone system |
| 15:30.53 | Stese | yes.... i have that on an extension now :P |
| 15:31.13 | mrkiko | [TK]D-Fender. sorry... I don't understand ... the answer |
| 15:31.27 | dongs | does asterisk even support wideband? |
| 15:31.35 | [TK]D-Fender | mrkiko: Enter into * CLI and read the instructions <- |
| 15:31.41 | Stese | Who recorded the Male voices? |
| 15:31.48 | mltlnx | [TK]D-Fender: Do you think it would be possible to regenerate a SUBSCRIBE using sipp or something of that sort? |
| 15:31.48 | mrkiko | ok |
| 15:31.58 | [TK]D-Fender | dongs: G.722 |
| 15:32.02 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 15:32.37 | dongs | rite. i was asking if it does. |
| 15:33.18 | dongs | anyway sleeptime. |
| 15:33.18 | [TK]D-Fender | dongs: yes |
| 15:35.00 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 15:35.19 | ThoMe | [TK]D-Fender: hi :-) |
| 15:35.21 | ThoMe | hmm |
| 15:35.21 | ThoMe | [Jul 31 17:34:58] NOTICE[18389]: chan_sip.c:14777 handle_request_invite: Sending fake auth rejection for user "01786323765" <sip:01786323765@dus.net>;tag=as424a25c8 |
| 15:35.32 | ThoMe | any ideas why asterisk said "sending fake" ? |
| 15:36.23 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:39.17 | angryuser | ThoMe, maybe asterisk using random info in that rejection so the person who is trying to brute force would not know that he is hitting good peer. Try to do a sip debug to be sure. |
| 15:39.22 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 15:40.12 | kaldemar | ThoMe: http://downloads.asterisk.org/pub/security/AST-2009-003.html |
| 15:43.57 | ThoMe | kaldemar: hi |
| 15:44.05 | ThoMe | kaldemar: i have asterisk 1.4.26 |
| 15:44.34 | ThoMe | "correct in 1.4.24.1" |
| 15:46.33 | kaldemar | exactly. |
| 15:46.58 | kaldemar | so don't worry about it. |
| 15:50.31 | *** join/#asterisk rgrasell (n=ryan@75.128.3.166) |
| 15:50.45 | rgrasell | can somebody help? My asterisk server wont isten on 5060 |
| 15:51.32 | *** join/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 15:51.39 | jelly-bean1 | how do i stay logged into a hunt group? |
| 15:51.48 | jelly-bean1 | or queue |
| 15:52.34 | jelly-bean1 | if i hang up it logs me out |
| 15:53.18 | *** part/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 15:53.29 | *** join/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 15:54.05 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:56.13 | Stese | does anyone know of a decent .gsm player? |
| 15:56.18 | rgrasell | My asterisk server isnt listenning on 5060. help? |
| 15:57.37 | Katty | jaytee: lunch. |
| 15:57.41 | Katty | jaytee: hungry |
| 15:59.01 | KavanS | Stese, asterisk will play those files for you |
| 15:59.13 | KavanS | Stese, that's how I've played back gsm myself |
| 15:59.18 | jaytee | Katty, go eat! |
| 15:59.30 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:59.35 | Stese | KavanS > I knwo, I want to play them outisde |
| 15:59.40 | Katty | jaytee: don't wanna eat alone )= |
| 15:59.56 | KavanS | Stese, ok...I've never played gsm outside of asterisk...I've tried with asterisk before |
| 15:59.58 | Stese | s/outisde/outside |
| 16:00.00 | jaytee | Katty, by the time I got there it would dinner tomorrow |
| 16:00.06 | Katty | i know :< |
| 16:00.13 | KavanS | heh, just eat lunch at your desk? |
| 16:00.51 | Qwell | Stese: sox |
| 16:00.53 | rgrasell | can someone help? asterisk isnt listening on 5060. |
| 16:01.07 | Katty | KavanS: ummm. |
| 16:01.09 | Katty | KavanS: no. |
| 16:01.28 | Stese | rgrasell > Firewall (external or iptables) |
| 16:01.30 | KavanS | Katty, umm...ok... |
| 16:02.11 | rgrasell | Stese: i dont have any, this is just running on my laptop |
| 16:02.18 | kaldemar | rgrasell: what version? did you configure it to do so? |
| 16:02.26 | Stese | your laptop running Linux? |
| 16:02.36 | KavanS | ahh nice, gotta love ignore |
| 16:02.42 | KavanS | took me a moment to figure it out... |
| 16:02.45 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:02.51 | rgrasell | im running ubuntu, i just intalled asterisk form the ubuntu repo |
| 16:03.00 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:03.08 | Stese | then Ubuntu will probably have a firewall running... |
| 16:03.18 | kaldemar | rgrasell: edit sip.conf to listen to a valid address if there are none. |
| 16:03.30 | kaldemar | Stese: ubuntu doesn't block any ports by default |
| 16:03.34 | rgrasell | stese: nope. no iptables |
| 16:04.31 | Stese | kaldemar > really? fair enough... |
| 16:04.49 | rgrasell | my sip.conf: http://pastebin.com/m3dc0a56d |
| 16:04.52 | kaldemar | Stese: really. :) |
| 16:05.27 | rgrasell | 192.168.1.5 is my ip on my networkd |
| 16:05.36 | Stese | kaldemar > Seems you learn something every day... though when in here, it's more than 1ce a day :P |
| 16:06.03 | Stese | anyway |
| 16:06.04 | Stese | gtg |
| 16:06.05 | Stese | laters |
| 16:06.08 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:06.35 | kaldemar | rgrasell: does "module show like sip" in cli list chan_sip.so? |
| 16:07.25 | Katty | i need a new boyfriend. |
| 16:07.26 | Katty | mine sucks. |
| 16:07.35 | Katty | grumpy. |
| 16:09.35 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 16:09.36 | rgrasell | kaldemar: yes it does. use count is 0, but that makes sense i guess. |
| 16:10.21 | kaldemar | do a "module unload chan_sip.so" and a "module load chan_sip.so" to see if there are any errors or warnings |
| 16:10.59 | kaldemar | "core set verbose 10" first to get useful output |
| 16:11.18 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:11.20 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 16:11.33 | kaldemar | feel free to pastebin results |
| 16:11.36 | kaldemar | ~pb |
| 16:11.36 | 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 |
| 16:11.49 | *** join/#asterisk Lantizia (n=cyor@212.57.229.111) |
| 16:12.00 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 16:12.10 | Lantizia | Lo, has anyone here ever accessed a Snom phone over a COM port? |
| 16:13.07 | rgrasell | kaldemar: http://pastebin.com/m394c12f3 |
| 16:13.37 | rgrasell | no errors i believe... |
| 16:13.52 | kaldemar | yes. how did you verify that it doesn't listen to port 5060? |
| 16:14.05 | rgrasell | nc. and lsof -i |
| 16:15.33 | rgrasell | kaldemar: any other config files you need to see>? |
| 16:16.39 | kaldemar | nope. netstat -nlp | grep asterisk will tell you what is listens to. |
| 16:16.50 | kaldemar | i'd say that UDP 5060 is one of the listed ones. |
| 16:17.12 | kaldemar | you don't really get much proof with netcat. |
| 16:19.00 | rgrasell | kaldemar: reloading the moduel did it! thanks :) |
| 16:19.07 | rgrasell | it too ka minute but it worked |
| 16:21.46 | *** join/#asterisk s0lid (n=s0lid@61.28.187.210) |
| 16:22.06 | *** part/#asterisk thieums (n=mathieu@LPuteaux-156-16-101-43.w80-12.abo.wanadoo.fr) |
| 16:23.15 | *** part/#asterisk rgrasell (n=ryan@75.128.3.166) |
| 16:28.48 | *** join/#asterisk bobJR (n=bob@adsl-150-237-62.tys.bellsouth.net) |
| 16:29.44 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 16:34.02 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 16:34.09 | jelly-bean1 | is it not possible to stay logged in to a group in Asterisk 1.6?? |
| 16:34.41 | grey-monkey | Does anyone know how to do an AgentCallbackLogin in Asterisk 1.6? |
| 16:34.46 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:34.52 | grey-monkey | It looks like it was taken out of version 1.6 |
| 16:38.23 | brah | There was this website where you could set where you were, and where you wanted to call to. |
| 16:38.23 | brah | And it would tell you what digits to press. Anybody know it? |
| 16:43.56 | *** part/#asterisk Akiyuki (n=Jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 16:52.59 | *** join/#asterisk shaggy63 (n=shaggy63@c-24-12-41-126.hsd1.in.comcast.net) |
| 16:55.36 | shaggy63 | I'm having a problem on asterisk 1.4.25.1. When an ext receives a voicemail and the user presses 8 to forward the voicemail, if they choose to add a comment to it then voicemailmain never hangups up and the voicemail file grows very large over a few days until it fills the drive. |
| 16:55.48 | ThoMe | hm |
| 16:55.48 | ThoMe | [Jul 31 18:55:18] NOTICE[4620]: chan_sip.c:14726 handle_request_invite: Call from '' to extension '0851987798890' rejected because extension not found. |
| 16:56.01 | ThoMe | but the 0851987798890 is availaible |
| 16:56.22 | Qwell | define "available" |
| 16:57.11 | Qwell | show us that extension in the context it's looking in |
| 16:57.14 | ariel_ | wow that is a long name for an extension |
| 16:57.24 | ThoMe | Qwell: http://paste.keks.be/647/txt |
| 16:57.39 | shaggy63 | softhangup won't even hangup the sip channel |
| 16:57.40 | Qwell | and show us that it's looking in that context |
| 16:57.54 | *** join/#asterisk vector_xyz (n=vecy@unaffiliated/t3rminator) |
| 16:58.09 | Qwell | ThoMe: it very likely isn't matching your peer |
| 16:58.35 | vector_xyz | hey guys im not really working with Asterisk but im working on a Telephone Software.. to fetch caller id for example.. but i have no way of testing .. i mean i have no landline... is there any fake phone call simulator or so ? anywhere |
| 16:58.39 | ThoMe | Qwell: http://paste.keks.be/648 |
| 16:59.45 | [TK]D-Fender | vector_xyz: Buy an ATA |
| 17:00.04 | [TK]D-Fender | ThoMe: FAIL... |
| 17:00.19 | [TK]D-Fender | ThoMe: Qwell asked you to prove the call is LOOKING at that context... |
| 17:00.39 | ThoMe | Qwell: the last http://paste.keks.be/649 |
| 17:01.06 | [TK]D-Fender | ThoMe: You aren't looking at the CALL. |
| 17:01.08 | vector_xyz | ATA ? |
| 17:01.11 | [TK]D-Fender | ~ata |
| 17:01.12 | infobot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 17:01.25 | ThoMe | [TK]D-Fender: hi. i dont know what you mean :-( |
| 17:01.42 | Qwell | ThoMe: sip debug |
| 17:01.51 | [TK]D-Fender | ^^^ |
| 17:01.58 | WindowsUser | vector_xyz: its the FXS part you want |
| 17:02.14 | kn0x | [TK]D-Fender: Any reason why VoiceMailMain would recognize the dtmf for Mailbox: prompt but nothing for password? |
| 17:02.35 | kn0x | Im getting that auth failed for mailbox '1234' with password '' |
| 17:02.47 | kn0x | had them try without pound and all |
| 17:02.48 | [TK]D-Fender | kn0x: Not a DTMF issue |
| 17:02.50 | kn0x | weirdest thing |
| 17:02.51 | *** join/#asterisk Shinsaku (n=Shinsaku@chello084010157123.chello.pl) |
| 17:03.05 | kn0x | [TK]D-Fender: I realize that... obviously dtmf works if it works for user |
| 17:03.11 | kn0x | just weird |
| 17:03.15 | [TK]D-Fender | kn0x: PASTEBIN <- |
| 17:03.23 | kn0x | kay sec |
| 17:03.24 | vector_xyz | where can i purchase it i kinda need it asap.. radio shack or so ? |
| 17:03.27 | ariel_ | look at your voicemail.conf file to see what password is in there |
| 17:03.37 | [TK]D-Fender | vector_xyz: depends where you are |
| 17:03.40 | WindowsUser | vector_xyz: what country and if US/Canada what side? |
| 17:03.58 | ariel_ | what side... nice |
| 17:04.24 | [TK]D-Fender | is on the underside... |
| 17:04.24 | WindowsUser | west side/east side woot |
| 17:04.24 | vector_xyz | USA, Central, Michigan |
| 17:04.24 | vector_xyz | i have Vonage VOIP but they use SIP |
| 17:04.26 | ariel_ | is almost at the furthest end of one of those |
| 17:04.29 | vector_xyz | and my software needs to use the standard |
| 17:04.31 | [TK]D-Fender | vector_xyz: www.telephonydepot.com |
| 17:04.42 | ariel_ | vonage voip has an ata |
| 17:04.50 | [TK]D-Fender | vector_xyz: You said you want to test your CID device. Your vonage ATA isn't good enough? |
| 17:05.03 | WindowsUser | vector_xyz: you have a vonage part that hooks to a handset? thats what you need |
| 17:05.07 | shaggy63 | anyone know of a place to start with my problem? |
| 17:05.07 | vector_xyz | well when i get incoming call i get no results because vonage is SIP |
| 17:05.27 | WindowsUser | odd |
| 17:05.30 | vector_xyz | and caller id data is transfered differently on SIP protocol |
| 17:05.32 | ariel_ | vector_xyz: really |
| 17:05.36 | WindowsUser | I get CID on incoming SIP calls |
| 17:05.46 | WindowsUser | vector_xyz: the ATA converts it normally |
| 17:05.52 | vector_xyz | i do to, but not on 'YAC' for example 'Yet another Caller Id' tool which doesnt support SIP |
| 17:05.55 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 17:06.04 | ariel_ | so do I, and the ata give me the callerID between ring 1 and 2 to my phones |
| 17:06.10 | vector_xyz | hold on i have umm from vonage a router (motorola) thats about it |
| 17:06.33 | shaggy63 | I think he means he has no way interface it into his computer to read it to his software |
| 17:06.34 | vector_xyz | but see i get caller id on my phone.. but not on my computer screen in my C++ application |
| 17:06.37 | timeshell_atwork | Is there a reason why my login to the forum at digium.com doesn't work with the same login I use to report issues at asterisk.org? |
| 17:06.47 | vector_xyz | i however do get incoming data from landline phones, but not sip protocol |
| 17:06.54 | ariel_ | vector_xyz: you have to someway to plug into your ata |
| 17:07.03 | ariel_ | via a modem/voice phone maybe |
| 17:07.15 | [TK]D-Fender | vector_xyz: You said you want to fake dialtone and receive callerID. this a POTS, not SIP |
| 17:07.23 | [TK]D-Fender | vector_xyz: SIP has nothing to do with this |
| 17:07.31 | WindowsUser | vector_xyz: is the modem plugged in to the ATA? |
| 17:07.45 | *** join/#asterisk Keisuke (n=webchat@logram-project.org) |
| 17:07.49 | vector_xyz | well i need a software to fake those calls because i cannot test with vonage.. its not giving me the correct landline protocol, if it is giving then i wouldnt need an simulator, emulator |
| 17:08.11 | vector_xyz | my 56K modem is plugged into the 'router' that vonage gave me if thats what you are refering to as ATA |
| 17:08.19 | vector_xyz | its just a router which i can access over 192.168.0.1 |
| 17:08.37 | WindowsUser | how do you use your vonage phone service? |
| 17:08.47 | kn0x | [TK]D-Fender: -- Incorrect password '' for user |
| 17:08.47 | kn0x | '113' (context = default) |
| 17:08.59 | vector_xyz | i have a small phone jack in their router |
| 17:09.01 | WindowsUser | Handset -> Modem -> Correct port on the vonage box for a handset |
| 17:09.05 | shaggy63 | I'm having a problem on asterisk 1.4.25.1. When an ext receives a voicemail and the user presses 8 to forward the voicemail, if they choose to add a comment to it then voicemailmain never hangups up and the voicemail file grows very large over a few days until it fills the drive. |
| 17:09.05 | WindowsUser | ^ do this |
| 17:09.07 | vector_xyz | that is where my 56K Modem is connected to |
| 17:09.13 | vector_xyz | yes |
| 17:09.16 | vector_xyz | WindowsUser: thats correct |
| 17:09.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:09.30 | Qwell | vector_xyz: What does any of this have to do with Asterisk? |
| 17:09.34 | vector_xyz | Handset > Computer Modem (with 2 ports 1 phone, 1 vonage line) > Vonage Router |
| 17:09.56 | vector_xyz | well you guys are the experts in the phone field and i just cant figure out something :x |
| 17:10.00 | *** join/#asterisk lucasb (n=bussey@s154-5-252-231.bc.hsia.telus.net) |
| 17:10.02 | ariel_ | shaggy63: how is the call coming into your system? |
| 17:10.11 | WindowsUser | vector_xyz: thats odd, it should work |
| 17:10.13 | [TK]D-Fender | vector_xyz: Go buy another ATA |
| 17:10.44 | shaggy63 | a local sip phone |
| 17:10.52 | ThoMe | Qwell: my tracelog: http://paste.keks.be/650/txt |
| 17:10.57 | ariel_ | vector_xyz: you have a phone line from the phone company? |
| 17:10.58 | shaggy63 | its only when they go to prepend to the message before they forward it |
| 17:11.06 | vector_xyz | here this is from a forum post "Call Vonage customer services and tell them you want US style caller id. The default for the UK is caller 'ESPI FSK with PR'. It needs to be the default US setting instead." |
| 17:11.16 | vector_xyz | ariel_: nope :/ |
| 17:11.23 | Qwell | ThoMe: Found no matching peer or user for '83.125.8.80:5060' |
| 17:11.27 | ariel_ | but your vonage is a phone line |
| 17:11.32 | ariel_ | like one from the telco |
| 17:11.57 | ariel_ | just plug that one to your modem and it's just like plugging into the phone company's jack |
| 17:12.05 | ThoMe | Qwell: <--- SIP read from 83.125.8.80:5060 ---> ? |
| 17:12.09 | ariel_ | dont' think it's sip that is done by the router |
| 17:12.21 | ThoMe | ahjo |
| 17:12.23 | Qwell | ThoMe: Your peer entry in sip.conf isn't being matched. Figure out why |
| 17:12.28 | robert_penz2 | Hi, my asterisk has just been misused for a DDOS attack. Only Asterisk is running on this VM, no ssh or other stuff. I've stoped the VM with it. The only stuff thats looks not normal in the logfiles is |
| 17:12.31 | ThoMe | Qwell: i think "you said" :-) hii :-) |
| 17:12.32 | [TK]D-Fender | ThoMe: Found no matching peer or user for '83.125.8.80:5060' |
| 17:13.18 | robert_penz2 | Jul 31 14:16:34 asterisk init: Failed to open console: Permission denied |
| 17:13.18 | robert_penz2 | Jul 31 14:16:35 asterisk asterisk[7958]: rc_avpair_new: unknown attribute 1490026597 |
| 17:13.18 | robert_penz2 | Jul 31 14:16:35 asterisk last message repeated 6 times |
| 17:13.19 | robert_penz2 | Jul 31 14:16:36 asterisk exiting on signal 15 |
| 17:13.26 | shaggy63 | pressing pound while doing a comment on the voicemail before forwarding it doesn't even do anything |
| 17:13.48 | robert_penz2 | using asterisk (1:1.4.17~dfsg-2ubuntu1) shiped with hardy |
| 17:13.51 | WindowsUser | vector_xyz:so the caller id style is wrong? looks like you do have to buy another ata then :-/ |
| 17:13.52 | vector_xyz | Well the main problem is this reading online... blogs on this issue they all stated that SIP Protocol sends Caller ID information before the phone rings, regular landline sends data after the first ring... so software that are existing 99% do not work with an VOIP service if you have one... so im just wondering if there is a tool i can test emulate 'a software that is' that i can download which will send data to my 56k modem and emulate |
| 17:14.06 | robert_penz2 | hardy = ubuntu 8.04 |
| 17:14.11 | ThoMe | Qwell: but i dont know why no matched. yesterday i have revieved calls :_( |
| 17:14.20 | WindowsUser | vector_xyz: my ata sends out caller id at the proper time for bell caller id |
| 17:14.24 | robert_penz2 | any idea. didn't find anything on google |
| 17:14.26 | ThoMe | Qwell: its my entry http://paste.keks.be/649 |
| 17:14.27 | WindowsUser | kinda annoying actually :-/ |
| 17:14.36 | [TK]D-Fender | robert_penz2: And 1.4.26 is current. Almost 10 releases behind |
| 17:15.04 | ThoMe | ah now |
| 17:15.04 | ThoMe | have now "port=5060" |
| 17:15.05 | vector_xyz | WindowsUser: could you link me to an ATA i mean i still dont know what an ATA is all i have is a Motorola Router Vonage gave me |
| 17:15.05 | robert_penz2 | [TK]D-Fender: but thats the version shipped with the distribution and it is up2date |
| 17:15.05 | ariel_ | vector_xyz: no |
| 17:15.18 | robert_penz2 | [TK]D-Fender: is there know anything and ubuntu is not updating the package? |
| 17:15.33 | [TK]D-Fender | robert_penz2: 1.4.26 begs to differ. If your repo manager feels that 10 versions ago is "current" well... no comment |
| 17:15.38 | Qwell | robert_penz2: distros are very slow to release updates. |
| 17:15.40 | shaggy63 | can I disable the ability to forward voicemails? |
| 17:15.44 | hardwire | wheres manx?! |
| 17:15.55 | ThoMe | Qwell: but, when i have more as one host (its a cluster (dus.net)) |
| 17:15.59 | WindowsUser | vector_xyz: http://ncix.com/products/?sku=21847&vpn=PAP2T-NA&manufacture=CISCO |
| 17:16.00 | ThoMe | Qwell: how i can say "all host" any? |
| 17:16.04 | [TK]D-Fender | shaggy63: vi app_voicemail.c |
| 17:16.19 | robert_penz2 | Qwell: i don't need new features, the question is there a security hole in the version I'm using? |
| 17:16.23 | Qwell | Yes |
| 17:16.30 | Qwell | and many bug fixes... |
| 17:16.53 | ariel_ | WindowsUser: he will have the same issue. I think he is trying to tap into the sip stream not the analog stream. If he is he is going at this the wrong way. |
| 17:16.56 | *** join/#asterisk manxpower (n=eric@84.sub-70-222-54.myvzw.com) |
| 17:17.01 | shaggy63 | ok, I mean without recompiling |
| 17:17.05 | robert_penz2 | Qwell: one that leads to asterisk beein misused for DDOS? |
| 17:17.07 | Qwell | hardwire: ask and you shall receive. |
| 17:17.12 | WindowsUser | oh |
| 17:17.13 | Qwell | robert_penz2: maybe. |
| 17:17.23 | hardwire | manxpower: tehy be strippin mah bits! |
| 17:17.45 | Qwell | hardwire: That'll be $49.99 btw |
| 17:17.50 | manxpower | before. first. cup. of. coffee. |
| 17:17.55 | [TK]D-Fender | I concur.... hardwire is definitely a few bits short of a full byte :p |
| 17:18.11 | Qwell | [TK]D-Fender: he's a few bits short of a NIBBLE |
| 17:21.37 | Keisuke | hi all |
| 17:22.13 | Keisuke | sorry for my english, i'm french and i have a bad english.... |
| 17:22.31 | Keisuke | i have a problem on asterisk |
| 17:22.51 | hardwire | nibbles on [TK]D-Fender |
| 17:22.51 | hardwire | I'll make a word out of you! |
| 17:22.57 | hardwire | ok.. that's enough of that. |
| 17:23.42 | hardwire | actually.. the TOS is in place.. weird. |
| 17:23.45 | hardwire | it was just not working! |
| 17:23.47 | hardwire | just two seconds ago! |
| 17:23.54 | hardwire | sigh. |
| 17:24.07 | Keisuke | i can called with every line, but i've no sound at all |
| 17:24.08 | ThoMe | Qwell / [TK]D-Fender: works now with: http://paste.keks.be/651/txt <<is it correct? |
| 17:24.19 | ThoMe | Qwell / [TK]D-Fender: and thank you very much for you help! |
| 17:24.25 | [TK]D-Fender | ThoMe: You just said it works... |
| 17:24.30 | [TK]D-Fender | ThoMe: What do YOU think? |
| 17:24.47 | Keisuke | if i'm in my local network everything works |
| 17:24.54 | ThoMe | [TK]D-Fender: yes, works. i mean, is it posible shorter or i need three entrys? is a cluster from dus.net :-( |
| 17:25.18 | [TK]D-Fender | Keisuke: Read : |
| 17:25.20 | [TK]D-Fender | ~sipnat |
| 17:25.20 | 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 |
| 17:25.22 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:25.25 | *** part/#asterisk nny_atg (n=Scott@64.203.237.47) |
| 17:26.10 | Keisuke | thanks [TK]D-Fender |
| 17:26.46 | *** join/#asterisk ingenius (n=alektro@vpn.itshidden.com) |
| 17:27.49 | manxpower | hardwire: Are you supposed to have QoS on that link? |
| 17:28.20 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:28.26 | hardwire | manxpower: no.. but we were hopping express traffic prio would help. |
| 17:28.31 | hardwire | hopeing |
| 17:28.32 | hardwire | hoping |
| 17:28.35 | hardwire | dangit |
| 17:31.30 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 17:32.19 | shaggy63 | I'm having a problem on asterisk 1.4.25.1. When an ext receives a voicemail and the user presses 8 to forward the voicemail, if they choose to add a comment to it then voicemailmain never hangups up and the voicemail file grows very large over a few days until it fills the drive. Even pressing pound won't hang it up. |
| 17:35.29 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 17:35.45 | buttons840 | i forgot, what is the syntax to line wrap in the dial plan? |
| 17:36.52 | [TK]D-Fender | buttons840: There is none |
| 17:37.06 | buttons840 | that's why i can't find it in the book then, thanks |
| 17:37.13 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:38.29 | shaggy63 | is there another way to hangup a stuck sip call besides restarting asterisk? |
| 17:38.52 | [TK]D-Fender | shaggy63: Soft hangup / redirect to dead-end, etc |
| 17:39.01 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 17:39.16 | shaggy63 | soft hangup won't hang it up |
| 17:39.56 | *** part/#asterisk Keisuke (n=webchat@logram-project.org) |
| 17:40.58 | *** part/#asterisk shaggy63 (n=shaggy63@c-24-12-41-126.hsd1.in.comcast.net) |
| 17:43.56 | generalhan | tzafrir_laptop: you around ? |
| 17:44.08 | tzafrir_laptop | generalhan, yes |
| 17:44.52 | generalhan | tzafrir_laptop: i was wondering if you could tell me, do the Xorcom astribank devices synch the timing from digital and analog interfaces to a single clock ? |
| 17:45.16 | tzafrir_laptop | on the same system: yes |
| 17:45.58 | generalhan | tzafrir_laptop: wonderful. against many objections i am going to continue to run faxes through asterisk and our PRI line, i was thinking of going with Sangoma cards and sync cable, but i think i like the idea of the channel bank type device ! |
| 17:46.23 | tzafrir_laptop | (with potentially the exception of the FXO modules) |
| 17:46.41 | generalhan | tzafrir_laptop: thats ok, this is entirely PRI and FXS |
| 17:46.51 | manxpower | hardwire: It was worth a try at least. |
| 17:51.21 | generalhan | tzafrir_laptop: well thanks for that confirmation. appreciate your time ! |
| 17:52.02 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 17:53.46 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:55.20 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 17:57.24 | *** join/#asterisk clemahieu (n=none@63.169.105.141) |
| 18:08.27 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:08.57 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:10.27 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:11.29 | *** join/#asterisk exc_ess (n=xavier@69-165-165-12.dsl.teksavvy.com) |
| 18:14.35 | grandpapadot | Hey guys, is there an AMI command that will show the asterisk version? |
| 18:15.25 | [TK]D-Fender | grandpapadot: COMMAND <- |
| 18:17.48 | grandpapadot | (sigh) Thanks, TK. I'm giving up for the day, stupid has set in. |
| 18:19.45 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 18:19.48 | eppigy | hello |
| 18:19.50 | eppigy | i am dave |
| 18:20.08 | exc_ess | Hi all, it's possible that I'm missing some key point and that this question doesn't make sense: I have some Linksys SPA-2102 ATAs hooked up to my asterisk server. The optimal behaviour I would like to see is that when I pick up my analogue phone it immediately connects me to the Asterisk server, which sends me a dial tone and from there I get to do whatever fancy dialplan magic I want. Unfortunately the SPA-2102 only actually connects to |
| 18:20.45 | manxpower | exc_ess: That is the way all ATAs and SIP phones work. |
| 18:21.09 | exc_ess | manxpower: Damn. I was hoping I was just wrong |
| 18:21.24 | manxpower | The device provides the dialtone and collects digits until it thinks it should send the digits. Most devices can be configured with a dialplan on the device to not make the user wait for a timeout or press # to send the call. |
| 18:22.28 | exc_ess | manxpower: Are there any tricks where, for instance, I make pressing ANY key immediately SIP connect to the asterisk box, triggering an application there that then handles the subsequent keypresses? |
| 18:22.48 | [TK]D-Fender | exc_ess: And you can have the ATA act as a Bat-Phone, but that means having to write an IVR or such to do you input which doesn't add much to your life, adn will mess with your IVR's, etc |
| 18:23.08 | [TK]D-Fender | exc_ess: It also blocks you from using your ATA's internal features in pr-dial |
| 18:23.23 | coppice | to behave as a bat phone you will need an exceptionally wide band codec |
| 18:24.06 | *** join/#asterisk shaggy63 (n=shaggy63@c-24-12-41-126.hsd1.in.comcast.net) |
| 18:24.23 | shaggy63 | what is the difference between exten => and exten = ? |
| 18:24.33 | Qwell | coppice: how wide would it need to be to hear color/texture? |
| 18:25.01 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 18:25.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:25.03 | exc_ess | One issue, though, is that the SPA-2102 defines a whole bunch of "*XX" extensions. I think this blocks me from setting my own internal "*XX"s that overlap because the SPA-2102 absorbs the number |
| 18:25.33 | WindowsUser | exc_ess: save your dialplan on the SPA 2102 and see what making it S0 does |
| 18:25.42 | manxpower | exc_ess: Usually people that want the dialtone and stuff provided by Asterisk are doing something wrong. |
| 18:25.58 | exc_ess | Does this mean I should use, say, a "#" prefix for internal functions (e.g. eavesdrop on existing channel) |
| 18:26.09 | manxpower | exc_ess: the SPAs are great devices. Every feature can be disabled. |
| 18:26.30 | [TK]D-Fender | exc_ess: So remap the codes the SPA uses or disable them alltegether |
| 18:26.41 | exc_ess | manxpower: Do you have a good documentation resource for those? Mine didn't come with a manual and I couldn't seem to find a good one |
| 18:26.53 | exc_ess | manpower: So all the config I've done with them has been fairly "shot in the dark" |
| 18:27.03 | [TK]D-Fender | exc_ess: the web GUI on those is so brain-dead you could probbely configure them while deceased |
| 18:27.22 | exc_ess | But, for instance, the dialplan syntax isn't defined |
| 18:27.25 | WindowsUser | the gui doesn't explain the dialplan syntax |
| 18:27.30 | WindowsUser | well its got the bits |
| 18:27.36 | manxpower | exc_ess: for the most part disabling feature codes should be pretty easy to figure out. there is extensive docs on the internet regarding designing a dialplan for the phone. |
| 18:27.43 | [TK]D-Fender | WindowsUser: MGCP standard. JFGI <- |
| 18:28.47 | exc_ess | Well I've turned off every supplementary feature (I had done already) |
| 18:28.58 | exc_ess | And it still absorbs "*XX" numbers |
| 18:29.19 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:29.29 | manxpower | exc_ess: are you sure the dialplan on the phone ALLOWS *XX numbers? |
| 18:29.41 | manxpower | All roads lead to the dialplan. The use the dialplan, Luke! |
| 18:29.45 | exc_ess | Well it's "S0" now |
| 18:30.03 | WindowsUser | does that call the asterisk when you pick up the phone? |
| 18:30.08 | manxpower | that won't do any good if you don't do all the extra work on |
| 18:30.12 | manxpower | asterisk to make S0 work |
| 18:30.41 | timeshell_atwork | Is hudlite still maintained? There's not much information on it out there... |
| 18:30.41 | exc_ess | Ah, which I didn't. Anyway, I think I'm at the very least now aware of WHAT I need to fix, thanks for the pointers |
| 18:30.58 | exc_ess | I guess I need to figure out these SPA dialplans, and make it work for my purposes |
| 18:31.05 | shaggy63 | what is the difference between exten => and exten = ? |
| 18:31.21 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 18:31.34 | [TK]D-Fender | shaggy63: > |
| 18:31.48 | WindowsUser | > is next to ? on the keyboard |
| 18:31.52 | shaggy63 | hehe, funny man |
| 18:31.55 | WindowsUser | are you sure its not a typo? |
| 18:31.55 | timeshell_atwork | I can't find a link for its server download either. |
| 18:32.00 | [TK]D-Fender | WindowsUser: depends on your keyboard |
| 18:32.13 | shaggy63 | Well its seems to me that exten = and exten=> do the same thing |
| 18:33.26 | manxpower | shaggy63: they do. |
| 18:33.29 | shaggy63 | I see both in the extensions.conf |
| 18:33.49 | manxpower | I think they added = support because to many people were forgetting the > |
| 18:34.13 | [TK]D-Fender | manxpower: AFAIK its always worked both ways |
| 18:34.37 | manxpower | [TK]D-Fender: how far back is "always"? |
| 18:34.45 | Sargun | Alright, I don't know anything about the UK, do US POTS phones work there? |
| 18:34.45 | *** join/#asterisk Psychobilly (n=moi@adsl185-58.kln.forthnet.gr) |
| 18:34.53 | [TK]D-Fender | manxpower: 0.9 at least.... |
| 18:35.12 | *** join/#asterisk mltlnx (n=mltlnx@x40342696.ip.e-nt.net) |
| 18:35.29 | [TK]D-Fender | manxpower: Its interchangeable just about everywhere |
| 18:35.30 | manxpower | Sargun: As I understand it the UK has two types of POTS lines. 3-wire, which would require an adapter to convert to 2-wire and 2-wire which works pretty much like a US phone. |
| 18:35.50 | Sargun | manxpower, what's 3-wire? |
| 18:36.02 | [TK]D-Fender | Sargun: 50% more wires than 2-wire |
| 18:36.06 | manxpower | Sargun: 3-wires are used for the phone line. |
| 18:36.21 | manxpower | [TK]D-Fender: and 30% fewer calories! |
| 18:36.27 | [TK]D-Fender | manxpower: \o/ |
| 18:36.42 | manxpower | I do not know how common the 3-wire standard is. |
| 18:36.51 | *** part/#asterisk shaggy63 (n=shaggy63@c-24-12-41-126.hsd1.in.comcast.net) |
| 18:40.02 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 18:40.36 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 18:45.30 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 18:49.19 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:50.01 | *** join/#asterisk werdan7 (i=w7@freenode/staff/wikimedia.werdan7) |
| 18:50.29 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 18:52.29 | Gnarfy | With cdr.conf -> unanswered=yes, is there any way to give the calls which show up by using Dial a distinctive mark to match them to the inbound leg (like uniqueid from the inbound call)? Matching on just originating channel isn't "unique" enough. |
| 18:56.06 | sumasuma | Gnarfy: unique with has a different meaning |
| 18:56.23 | sumasuma | Gnarfy: unique with asterisk has a different meaning, don't take it as unique |
| 18:57.05 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 18:58.20 | Gnarfy | sumasuma, i know the channel identifier isnt unique, which is why i wanted to give a mark to the CDR which shows up with Dial - but i can't find anything in the book, nor does something like set userfield show up for the Dial cdr rules |
| 18:58.38 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 18:58.40 | Gnarfy | rules/lines |
| 19:00.04 | sjobeck | hey, hi all, can i ask a half related half unrelated (or seemingly so any way) question. i have a totally overbuilt proliant, 3gb ram, dual proc's, 15k drives, the works, 2 yrs old, with openSUSE 11.1 on it & asterisk 1.4. it basically idles all day. it runs perfectly, and i mean perfectly, for 2-8 hrs at a time & then randomly just hangs. i cant see anything in the logs, dmesg, etc. i have no way to see what is going on. where to sta |
| 19:01.39 | WindowsUser | ewwwww |
| 19:01.42 | sjobeck | i know |
| 19:01.43 | sjobeck | right |
| 19:01.49 | WindowsUser | why are you using opensuse |
| 19:02.02 | sjobeck | i dont know where to begin. in 20 yrs never seen this before. there is like 0 clues. |
| 19:02.06 | sjobeck | it just hangs |
| 19:02.22 | WindowsUser | try and be running top when it hangs |
| 19:02.29 | sjobeck | long story, the OS was chosen 6 yrs ago & we just haven't wiped the mirror for centOS 5.x yet |
| 19:02.34 | sjobeck | k |
| 19:02.35 | sjobeck | i will |
| 19:02.40 | sjobeck | and what will i see or look for ? |
| 19:03.01 | WindowsUser | load might spike realllly high for whatever reason |
| 19:04.47 | sjobeck | k |
| 19:04.48 | sjobeck | right |
| 19:04.49 | sjobeck | thx |
| 19:05.15 | sjobeck | if i leave it in a "screen" for later (since it takes like hrs to hang) can or will i still be able to see that then when i rejoin screen |
| 19:05.43 | ThoMe | can anybody call me to my number 0049 851 98 77 98 890 ? i would like test on my asterisk - please? |
| 19:05.44 | WindowsUser | well run the screen on something else |
| 19:05.48 | WindowsUser | not on the box that hangs.... |
| 19:06.09 | ThoMe | WindowsUser: can you help me please? |
| 19:06.14 | sjobeck | screen via SSH then ? |
| 19:07.08 | Gnarfy | sjobeck: if your hp proliant has something like ilo or other remote management, you can always use that to check the actually screen output remotely |
| 19:07.58 | sjobeck | indeed it does. been using it to flip the virtual power button off/on to reset it the last 3 days |
| 19:07.58 | sjobeck | :( |
| 19:07.59 | sjobeck | :) |
| 19:08.12 | ThoMe | Gnarfy: can you call me? short? |
| 19:08.54 | Gnarfy | Its not my server i'm working on, rather not O.o |
| 19:09.02 | sjobeck | thx guys ! |
| 19:09.03 | sjobeck | as always |
| 19:09.04 | sjobeck | cheers |
| 19:09.14 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.50) |
| 19:09.16 | edwin_quijada | Hi! |
| 19:09.38 | edwin_quijada | I am cheking my asterisk and I have a RED alarm without any cause |
| 19:09.57 | edwin_quijada | so when I call to my asterisk I get a busy tone |
| 19:11.24 | edwin_quijada | so any cluees for starting with this issue? |
| 19:11.34 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 19:11.35 | WindowsUser | well, what gives the red alarm? nagios? |
| 19:11.53 | *** join/#asterisk korcan (n=korcan@adsl-76-234-161-17.dsl.sfldmi.sbcglobal.net) |
| 19:12.11 | Alfio | edwin_quijada why dont you give us more details |
| 19:18.10 | edwin_quijada | WindowsUser: ztscan i did |
| 19:18.27 | edwin_quijada | t1 card just for IVR |
| 19:18.57 | edwin_quijada | asterisk 1.4 |
| 19:19.09 | *** join/#asterisk Lantizia (n=cyor@212.57.229.111) |
| 19:19.30 | edwin_quijada | there is a way to do a test for whole system |
| 19:19.30 | Lantizia | Lo, is it possible to get the MAC address of a registered phone from the asterisk CLI? |
| 19:19.54 | [TK]D-Fender | Lantizia: No. |
| 19:20.01 | Lantizia | thought as much |
| 19:20.13 | [TK]D-Fender | Lantizia: to * its just IP. It won't see that. |
| 19:20.23 | [TK]D-Fender | Lantizia: You'll have to nmap it or such |
| 19:20.30 | Lantizia | not on the same network |
| 19:20.38 | Lantizia | it's ok I'll just ask them :D |
| 19:22.58 | *** join/#asterisk DarkLogik (n=darklogi@darklogik.com) |
| 19:25.43 | edwin_quijada | zttool says that is RED |
| 19:26.11 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 19:26.39 | Micc | Does anyone know if there is a sip notify message you can send to a polycom ip450 to reload the directory? |
| 19:27.11 | Micc | TKD-Fender, BTW, thanks for the help with the sonicwall yesterday. It looks like its working fine now. |
| 19:29.40 | Micc | is everyone out to lunch? |
| 19:29.47 | Micc | or taking off for the weekend already? |
| 19:29.53 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 19:34.45 | Gnarfy | Having each dial do a macro with NoOp(${ARG1}), which is the uniqueid of the inbound call. Then matching the lastapp from the CDR lines to the incoming calls... guess thats not best practice :x |
| 19:38.56 | *** join/#asterisk errotan (n=errotan@5403E6F2.catv.pool.telekom.hu) |
| 19:39.11 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 19:42.25 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:43.01 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 19:43.03 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:fdf7:18b3:d48a:74d6) |
| 19:43.46 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 19:54.25 | *** part/#asterisk T` (n=total@pdpc/supporter/student/T) |
| 19:55.25 | *** join/#asterisk greenroom (n=support@72-17-198-10.wpb.fdn.com) |
| 19:55.51 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 19:56.08 | greenroom | Could anyone point me in the best way to go about finding an Asterisk Developer or Consultant? |
| 20:00.37 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 20:04.18 | sumasuma | greenroom: How is the pay per hour ? |
| 20:04.44 | sumasuma | greenroom: This is also one of the good place to find good consultants |
| 20:05.12 | greenroom | It depends on if they can do what we need. |
| 20:05.35 | greenroom | We're looking to have someone create an AMD module that can play a message while doing the machine detection in the background. |
| 20:05.51 | greenroom | This way there is no silence at the beginning and the message plays right away. |
| 20:07.32 | *** join/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 20:07.50 | *** join/#asterisk tamiel (n=tamiel@ip-230.net-82-216-103.rev.numericable.fr) |
| 20:13.18 | exsync | asterisk-biz mail list |
| 20:18.15 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 20:24.07 | *** join/#asterisk galeras (n=galeras@190.24.217.121) |
| 20:26.59 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:29.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:40.08 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 20:40.51 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:40.51 | sumasuma | greenroom: AMD is not very good with asterisk |
| 20:40.56 | sumasuma | exernal solutiong might help |
| 20:41.07 | andresmujica | hi, is possible to make an AGI script to answer a QUEUE? |
| 20:41.10 | sumasuma | Try Sangoma NetBorder |
| 20:41.21 | Sargun | Does digium make that 1 port FXO card? |
| 20:41.39 | sumasuma | andresmujica: What you mean by answering a queue ? |
| 20:41.45 | manxpower | Sargun: not in years. Intel discontinued the chip they used. |
| 20:41.46 | sumasuma | As a agent ? |
| 20:41.52 | andresmujica | sumasuma: yeap, as an agent... |
| 20:42.01 | sumasuma | Yes |
| 20:42.23 | iflux | has heard rumors about the sangoma usb-fxo supporting echo-can soon.. |
| 20:42.27 | Sargun | will use: http://www.voipsupply.com/sangoma-usbfxo |
| 20:42.32 | Sargun | possibly. |
| 20:42.45 | Sargun | iflux, will it require a hardware change, or software upgrade? |
| 20:42.55 | iflux | firmware upgrade from what I hear |
| 20:43.00 | Sargun | Ok |
| 20:43.12 | sumasuma | Give the agi script in an extension and Place the agent Dial to that Local AGI |
| 20:43.19 | iflux | infact.. I wouldn't be surprised if it were a free firmware upgrade.. |
| 20:43.23 | andresmujica | sumasuma: so i can execute an AGI application through the queue... hmmm gonna explore that... i'm assuming it's the AGI option availabe at the QUEUE app, right? or should i create an exten pointing to the AGI and make that exten a queue member? |
| 20:43.26 | sumasuma | i mean the interface |
| 20:43.28 | iflux | but I have no idea.. |
| 20:44.06 | sumasuma | andresmujica: that is correct, you need give the interface as local and asterisk will dial to that interface which is local |
| 20:44.08 | sumasuma | and goes to the agi |
| 20:44.17 | Sargun | iflux, are you by chance the marketing guy at sangoma |
| 20:44.36 | andresmujica | sumasuma: great. thnks. |
| 20:45.25 | sumasuma | andresmujica: I assume all call as soon as it arrives in the queue will land up in that agi |
| 20:45.28 | sumasuma | you need to take care |
| 20:45.46 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 20:45.52 | andresmujica | sumasuma: that's the idea. |
| 20:48.00 | Katty | stretches |
| 20:48.21 | *** join/#asterisk cb22 (n=federico@moinmoin/developer/federico) |
| 20:48.59 | iCEBrkr | Qwell: You alive over here? |
| 20:49.08 | Qwell | nope |
| 20:49.11 | iCEBrkr | Qwell: http://www.cyberdyne.org/~icebrkr/photo_album/thumbnails.php?album=60 |
| 20:49.12 | iCEBrkr | ok |
| 20:49.13 | iCEBrkr | nevermind |
| 20:49.15 | iCEBrkr | lol |
| 20:49.20 | cb22 | Hi! I have a bit of a different question... Is it possible to get the dial command to cut out the first few seconds of a call? I've tried googling and all that comes up are people with sound issues |
| 20:49.39 | iCEBrkr | cb22: eh? |
| 20:50.14 | cb22 | iCEBrkr: basically I'm using a cheap ethernet ATA FXO, and when i dial through it, you can hear the actual dialing |
| 20:50.41 | cb22 | like as if you were pressing the keys on a normal phone, i'd like to cut this out (only start passing audio lets say 3 seconds after initiating a call) |
| 20:51.03 | Sargun | cb22, doesn't the dialing sound make you a little bit nostalgic? |
| 20:51.04 | iCEBrkr | cb22: That's kinda odd. |
| 20:51.27 | cb22 | Sargun: haha the first time it did... not so much after that |
| 20:51.28 | manxpower | cb22: Asterisk does not pass audio until dialing is finished. |
| 20:51.30 | iCEBrkr | Sargun: like waiting for a modem to answer or international call? lol |
| 20:51.43 | manxpower | cb22: what version of Asterisk are you using? |
| 20:51.45 | Qwell | ~cheap |
| 20:51.45 | infobot | i guess cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:51.52 | Qwell | Don't use a cheap ATA.. |
| 20:51.53 | iCEBrkr | manxpower: I was actually thinking it's some sort of bleed through :) |
| 20:52.13 | iCEBrkr | Yeah, I found Cheap + Asterisk doesn't work so well. |
| 20:52.22 | cb22 | manxpower: yeah, but the thing is that this thing answers as soon as it gets the command to dial |
| 20:52.29 | cb22 | i'm running 1.6 |
| 20:52.43 | manxpower | cb22: You're going to have to live with the issue.\ |
| 20:52.45 | cb22 | Qwell: i know, i know. it actually works really well, this issue aside. quality is great |
| 20:52.46 | Qwell | cb22: The device is broken. Fix the problem. Don't try to hack around it. |
| 20:53.15 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 20:53.23 | Sargun | cb22, Make it so your calls get patched into a meetme conference, and then it spawns a call that will dial out, wait a second, and then patch the call into the meetme conference |
| 20:53.45 | cb22 | Sargun: that sounds like an epic workaround. my hat off to you |
| 20:53.50 | cb22 | i'll give that a show |
| 20:53.53 | iCEBrkr | Sargun: That's sick |
| 20:53.55 | cb22 | shot* |
| 20:54.16 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:54.19 | Sargun | sees everyone here with competence facepalm |
| 20:54.49 | Katty | humm. |
| 20:54.53 | manxpower | iCEBrkr: you realize now he's going to spend the next 12 hours pestering everyone on the channel trying to figure out how to make that work. |
| 20:55.06 | Sargun | cb22, That's like saying, "I want to fly" and then going to a cliff with a mattress at bottom, and jumping. |
| 20:55.08 | iCEBrkr | manxpower: Hey man, it wasn't my idea. |
| 20:55.12 | Sargun | haha |
| 20:55.14 | manxpower | Thanks for ruining our day. |
| 20:55.14 | cb22 | manxpower: nah, relax |
| 20:55.18 | iCEBrkr | Points at Sargun |
| 20:55.36 | iCEBrkr | I'm so far removed from Asterisk anymore. |
| 20:55.37 | iCEBrkr | :( |
| 20:55.43 | Sargun | is wearing an "I'm with stupid shirt" -- pointing to manxpower |
| 20:55.47 | iCEBrkr | My Asterisk box as 17 new vmails that I haven't listened to |
| 20:55.59 | Sargun | barely touches his asterisk |
| 20:56.14 | cb22 | lemme try this quick... i'll let you know if it works :P |
| 20:56.23 | Sargun | It's been running for like 6 months |
| 20:56.30 | Sargun | cb22, it's not quick, or simple |
| 20:56.39 | iCEBrkr | Sargun: Exactly.. My machine just runs. |
| 20:56.42 | manxpower | Sargun: I'm feeling stupid. I'm trying to figure out why some tracks that I play in WMP11 from a Windows Media server slow down the speed and pitch in about 5 seconds until it sounds like someone is on Qualudes. |
| 20:56.50 | cb22 | Sargun: i was thinking of using app konference instead of meetme |
| 20:56.59 | Sargun | app_conference would work |
| 20:57.07 | Sargun | hasn't played with it much |
| 20:57.17 | iCEBrkr | See. App_conference is new |
| 20:57.30 | Sargun | too new. |
| 20:57.44 | manxpower | you could add D(wwwwww) to the dial line and see if that helps. |
| 20:57.51 | WindowsUser | whats the horribly had idea? |
| 20:57.54 | Qwell | manxpower: it's SIP, presumably |
| 20:57.58 | Sargun | WindowsUser, windows. |
| 20:58.00 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 20:58.00 | cb22 | considering my asterisk installation is on a netbook, nothing is too new |
| 20:58.19 | Sargun | WindowsUser, some overly complicated thing I suggested. |
| 20:58.22 | WHYS | I'm trying to remove realtime, but not sure I've got it. when I enterthe console, and do a reload, it takes 45 seconds before anything shows up on the screen. |
| 20:58.26 | manxpower | Qwell: I assume so too, but if Asterisk is passing the audio it must think an answer happened on the SIP device. |
| 20:59.00 | manxpower | WHYS: that is commonly DNS issues with the IPs on the machine not being able to determine the hostnames. |
| 20:59.02 | Qwell | manxpower: no, the ATA is just stupid and sends an answer/sets up audio as soon as it gets a call |
| 20:59.16 | WHYS | Humm, I'll check.... |
| 20:59.17 | cb22 | Qwell: yep, it's a very stupid ATA |
| 20:59.20 | Sargun | # w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 20:59.20 | Sargun | # W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 20:59.25 | WindowsUser | cb22: what ATA? |
| 20:59.42 | cb22 | grandstream 488 |
| 20:59.47 | manxpower | Qwell: Yup, the D(wwwwwww) will make asterisk "dial" a bunch of pauses, which should prevent audio during that time. |
| 20:59.54 | manxpower | Sargun: WRONG "w" |
| 20:59.59 | Sargun | manxpower, ahhh |
| 21:00.11 | Sargun | manxpower, so, like Dial(wwwwwwww${EXTEN}) |
| 21:00.26 | manxpower | look up the D() option to Dial. "w" means .5 second pause |
| 21:00.41 | manxpower | Sargun: No! |
| 21:00.48 | Sargun | manxpower, Oh, whoa. |
| 21:00.53 | manxpower | what did I just say about the D() option to Dial? |
| 21:00.56 | Sargun | manxpower, I just looked it up, cool, never knew about that. |
| 21:01.15 | Sargun | manxpower, you said that after I said that |
| 21:01.31 | manxpower | It would be an easy thing to try. |
| 21:01.54 | iCEBrkr | I really should finish my call manager for Asterisk |
| 21:01.55 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 21:01.56 | iCEBrkr | sighs |
| 21:02.00 | manxpower | and if asterisk is pausing chances are it's not passing audio. I doubt it will make any difference, but it is an easy fast thing to try. |
| 21:02.04 | iflux | this dude has serious spare tires.. http://thereifixedit.com/wp-content/uploads/2009/07/diegoj-tiretransport.jpg |
| 21:02.48 | manxpower | ~phones |
| 21:02.49 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 21:03.05 | manxpower | and now I retire to the pool |
| 21:03.15 | iCEBrkr | ha |
| 21:03.24 | cb22 | haha |
| 21:03.25 | iCEBrkr | I have another 30mins |
| 21:03.32 | iCEBrkr | and then an hour drive home |
| 21:06.01 | iflux | http://thereifixedit.com/wp-content/uploads/2009/07/scottw-motorbike.jpg <-- wow.. just.. wow. |
| 21:06.04 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:06.21 | Qwell | iflux: take it elsewhere |
| 21:06.30 | iflux | wrong channel |
| 21:06.38 | afink | just got to kansas city woohoo |
| 21:06.41 | iflux | was supposed to be on /query |
| 21:08.53 | [TK]D-Fender | :O |
| 21:09.45 | iflux | sup fender? |
| 21:10.13 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 21:11.00 | [TK]D-Fender | iflux: SSDD |
| 21:11.37 | Katty | afink: you're only a few hours from me now! |
| 21:11.44 | iflux | sup katty? |
| 21:13.14 | afink | cool where are you katty? |
| 21:14.04 | Katty | couple hours south of stl |
| 21:14.08 | Katty | iflux: hello. |
| 21:19.10 | WHYS | manxpower, seems like I can get to dns fine, resolve all hosts and hosts.conf is fine. I can't find any networking issues. I though my delay problems were database related. perhaps not |
| 21:19.37 | WHYS | *thought |
| 21:23.53 | *** join/#asterisk shaggy63 (n=shaggy63@c-24-12-41-126.hsd1.in.comcast.net) |
| 21:24.02 | shaggy63 | I found the soltion to my problem |
| 21:24.43 | shaggy63 | changed format=WAV to format=wav in voicemail.conf |
| 21:34.33 | *** join/#asterisk kbukhari (n=kashif@203.99.174.33) |
| 21:34.37 | kbukhari | hello |
| 21:35.06 | kbukhari | exten => _X.,1,Set(LOCALNUMBER=${EXTEN:-7}) |
| 21:35.06 | kbukhari | exten => _X.,2,Dial(SIP/${LOCALNUMBER}) |
| 21:35.06 | kbukhari | exten => _X.,3,Hangup |
| 21:35.13 | kbukhari | is working fine for me |
| 21:36.20 | kbukhari | now i want 0569810003-9 forward 9810001 and 9810002 |
| 21:36.23 | kbukhari | any help ? |
| 21:37.00 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:37.25 | ScarEye | guys whats the big difference between the polycom ip 320 and 321 ? |
| 21:38.10 | [TK]D-Fender | ScarEye: 321 has more RAM to support the newer firwares. Don't touch the 320/330 unless you get a noticable drop in price for them |
| 21:38.24 | [TK]D-Fender | ScarEye: I DID tell you this :) |
| 21:38.43 | ScarEye | I know but he said. that the 321 is not certified |
| 21:38.49 | ScarEye | and he wants me to get the 331 |
| 21:38.54 | ScarEye | sorry |
| 21:38.57 | ScarEye | 320 |
| 21:38.58 | mchou | [TK]D-Fender: is there a way to have the polycom handset volume level persistent? If I hang up and pick up again the volumme always defaults to 50%. |
| 21:39.29 | [TK]D-Fender | ScarEye: 100% same configuration... |
| 21:39.34 | ScarEye | he said the 321 is not certified and for us to get the 320's |
| 21:39.41 | ScarEye | and I already ordered the 321's |
| 21:39.43 | [TK]D-Fender | mchou: sip.cfg > "persist" |
| 21:40.24 | [TK]D-Fender | ScarEye: Not certified != not work... Ask for reasons and background about what might fail and what they are doing about it |
| 21:41.59 | kbukhari | now i want 0569810003-9 forward 9810001 and 9810002 |
| 21:42.02 | kbukhari | exten => _X.,1,Set(LOCALNUMBER=${EXTEN:-7}) |
| 21:42.07 | kbukhari | exten => _X.,2,Dial(SIP/${LOCALNUMBER}) |
| 21:42.10 | kbukhari | exten => _X.,3,Hangup |
| 21:42.13 | kbukhari | is working fine for me |
| 21:42.16 | kbukhari | now i want 0569810003-9 forward 9810001 and 9810002 |
| 21:42.20 | kbukhari | any help ? |
| 21:44.12 | *** part/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 21:44.47 | MACscr | grr, im not find any official docs that say the pap2t-na supports t38 |
| 21:44.56 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 21:45.31 | [TK]D-Fender | MACscr: Last I heard from coppwas negative on that count |
| 21:45.52 | [TK]D-Fender | Coppice* |
| 21:46.22 | MACscr | [TK]D-Fender: yeah, some people in forums are saying it does, but there are no config options for it. Can ya recommend a cheap ata I can use for my fax machine? |
| 21:46.46 | [TK]D-Fender | SPA-2102 at a minimum |
| 21:48.42 | MACscr | maybe its time for me to get a new router/ata/wireless. Do they have ones that do all 3? |
| 21:49.00 | WindowsUser | ASUS has something |
| 21:49.15 | [TK]D-Fender | MACscr: You are making increasingly unwise decisions in this area..... |
| 21:49.26 | MACscr | [TK]D-Fender: lol, ok |
| 21:49.48 | WindowsUser | AX-112W |
| 21:50.36 | MACscr | well, its just that I don't have QOS setup at my home office, so I figured an all in one might be the better way to go. My actual pbx is remotely colocated |
| 21:52.00 | WindowsUser | all in one leads to all is broken |
| 21:52.16 | MACscr | yes, that can be the case |
| 21:52.46 | MACscr | you can say that about anything though that does multiple functions |
| 21:53.01 | *** join/#asterisk sjobeck (n=Adium@32.158.192.46) |
| 21:53.14 | *** part/#asterisk sjobeck (n=Adium@32.158.192.46) |
| 21:54.09 | WindowsUser | also theres probably no combo ATA, Draft-N, Gigabit switch, Router combo |
| 21:54.39 | MACscr | right, that's what I figured. I could care less about N though |
| 21:56.05 | WindowsUser | oh theres also http://www.cisco.com/en/US/products/ps10028/index.html |
| 21:56.55 | [TK]D-Fender | "Would you like fries with that, sir?" |
| 21:58.24 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 21:59.29 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:11.26 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 22:17.57 | cb22 | Sargun: haha, i got it to work :) |
| 22:18.09 | cb22 | a horrible hack if there ever was one, but it works |
| 22:18.09 | Sargun | cb22, what did you do |
| 22:18.56 | cb22 | well, i used app_conference and system(). system() calls a script that puts a .call file in /var/spool/asterisk/outgoing that calls the ATA and the conference |
| 22:19.06 | cb22 | then it waits for 3 seconds, then it runs the conference |
| 22:20.30 | Qwell | ... |
| 22:21.07 | cb22 | Qwell: ...? |
| 22:27.45 | cb22 | ok... so on a what-i'm-doing-is-retarded scale, from 1 to 10, what would this be? |
| 22:28.43 | Qwell | the fact that you managed to pull it off... 4 |
| 22:28.49 | Qwell | I've seen worse |
| 22:29.02 | Qwell | If you hadn't succeeded, I would call it a clear 7 |
| 22:29.55 | cb22 | haha, to be honest, i didn't think i'd be able to get it to work myself. first i thought it would be easy, then i thought it would be impossible, then i read http://www.mail-archive.com/asterisk-users@lists.digium.com/msg181140.html |
| 22:30.25 | Qwell | Nothing is impossible. |
| 22:31.09 | mchou | WindowsUser: Do you have (or have you ever used) the asus ax-112w? |
| 22:31.15 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 22:32.27 | mchou | well, I must say I'm slightly disapponted with this polycom habdset |
| 22:32.32 | mchou | handset* |
| 22:32.41 | mchou | nedd bigger buttons! |
| 22:33.07 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 22:33.10 | Sargun | Need bigger button |
| 22:33.24 | mchou | Sargun: yeah yeah |
| 22:33.27 | *** join/#asterisk clemahieu (n=none@63.169.105.141) |
| 22:33.50 | Sargun | Qwell, I think cb22 just showed us how easy asterisk is to use |
| 22:34.44 | Sargun | I'd say when I first started playing with asterisk it was an 8 on the difficult scale |
| 22:35.24 | cb22 | Sargun: Same, but after a bit of time, everything just started to 'fit' into place |
| 22:35.29 | mchou | Sargun: what's the difficulty now? |
| 22:35.38 | Sargun | mchou, 4? |
| 22:35.45 | Sargun | 4, 5, or 6. |
| 22:35.47 | cb22 | I'm still a big Asterisk noob though |
| 22:35.54 | Sargun | It really depends on the part of asterisk you're messing with |
| 22:36.02 | Sargun | mchou, voip-info, the wiki, has helped A LOT |
| 22:36.04 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 22:36.14 | Sargun | mchou, the biggest problem was the lack of documentation |
| 22:36.32 | mchou | Sargun: no doubt, but that wiki also has losts of cruft |
| 22:36.38 | mchou | lots* |
| 22:36.49 | Sargun | mchou, Crappy documentation is better than no documentation |
| 22:37.01 | mchou | Sargun: in fact most of the cruft leads noobs astray |
| 22:37.31 | mchou | Sargun: "* for Dummies" for me was a beeter book |
| 22:37.36 | mchou | better* |
| 22:37.51 | mchou | Sargun: succinct, to the point |
| 22:38.30 | mchou | not like "the future of telephony," a meandering morass |
| 22:39.09 | mchou | and I can see why Mark Spencer wrote the forward for "* for dummies" |
| 22:39.17 | drmessano | I was just reading about the Fonality Fight Club.. I honestly though the only Fonality Fight Club was the IT department trying to secure their trixbox from being hacked.. by fonality |
| 22:39.19 | Qwell | Sargun: in some cases I would very much disagree with that |
| 22:39.19 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 22:39.29 | *** join/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 22:39.36 | Qwell | Sargun: sometimes crappy documentation is *worse* than no documentation |
| 22:39.44 | *** part/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 22:39.48 | Sargun | needs to buy: 4 x http://www.usr.com/products/modem/business-product.asp?sku=USR3453c and 2 x http://sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html |
| 22:39.53 | Sargun | Qwell, Alright, I agree. |
| 22:40.00 | mchou | drmessano: the first rule.... |
| 22:40.05 | Sargun | Qwell, but voip-info isn't THAT CRAPPY |
| 22:40.12 | drmessano | Just read the voip-wiki.. Shit is mostly old.. which makes it WRONG |
| 22:40.18 | mchou | drmessano: we dont talk about fonality fight club |
| 22:40.24 | drmessano | err voip-info |
| 22:40.26 | Qwell | Sargun: sometimes it is |
| 22:40.40 | *** part/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 22:40.55 | drmessano | A lot of the info on voip-info is so out of date it's irresponsible to keep |
| 22:41.05 | mchou | yeah, I agree |
| 22:41.23 | mchou | voip-info is ministry of dis-information |
| 22:41.33 | mchou | beats chemical ali |
| 22:41.59 | drmessano | I dare say the sample files in Asterisk now offer more than half the content on the site.. which says a lot, but yet, not much |
| 22:43.30 | mchou | drmessano: so my friend finally got a googlevoice invite (he was on waitlist) |
| 22:43.36 | drmessano | The problem is.. it's not cool to submit content to someone elses site.. Why do that when I can put up my own failure of a site that features 3 useful pages of info |
| 22:45.19 | *** part/#asterisk mwalling (n=mwalling@you.dontlike.us) |
| 22:46.02 | drmessano | The real answer to the asterisk documentation dilemna is better small CMS that objectively present the content to search engines and that dynamically accept feedback on the overall value of said contact, which would allow a properly indexed, distributed, peer reviewed set of documentation |
| 22:46.29 | drmessano | s/contact/content/ |
| 22:47.11 | manxpower | Oh! Eureka is on tonight! |
| 22:47.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:47.49 | *** join/#asterisk alrs (n=lars@170.206.224.54) |
| 22:49.06 | Chainsaw | manxpower: I got all excited until I realised you're in the US :/ |
| 22:49.10 | Chainsaw | manxpower: Will have to wait here. |
| 22:49.18 | drmessano | Google already goes pretty far into allow you to set the relevance of a site.. if only there was a kickass feedback system.. |
| 22:49.23 | manxpower | These days I mostly use voip-info to look at sample config files or other asterisk docs. |
| 22:49.29 | manxpower | Chainsaw: bittorrent it tomorrow |
| 22:49.38 | Chainsaw | manxpower: Heh, obviously ;) |
| 22:50.13 | manxpower | My tivo will catch it. |
| 22:51.30 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 22:51.36 | drmessano | I think if someone could spend a month trolling google and voting up and down sites to influence the pagerank and then send a report to every asshole who claims they have a decent asterisk documentation site showing them how bad their content was rated and how it's going to shove down their page rank thanks to someone actually bothering to rank the validity of the content.. hang on, flying pig |
| 22:52.15 | manxpower | maybe someone could instead just work with the owner of voip-info.org to audit the entire site. |
| 22:52.45 | alrs | crazy talk: it's a wiki |
| 22:52.48 | manxpower | use UPGRADE*.txt as most of the reference for changes between versions |
| 22:52.51 | drmessano | I think the wiki model for documentation of software that is far from static will never work |
| 22:53.30 | alrs | drmessano: the magic of a wiki is that it is a dynamic document |
| 22:53.39 | drmessano | No, thats the problem |
| 22:53.40 | alrs | just sayin' |
| 22:53.50 | drmessano | Thats 100% of the problem |
| 22:54.13 | manxpower | alrs: I don't think it's crazy talk. |
| 22:54.15 | alrs | there's some ogre out there that reverts edits back to 1.0 behavior? |
| 22:54.40 | manxpower | How does wikipedia address that issue? |
| 22:54.53 | drmessano | Look at this channel.. Ask someone about SIP NAT issues at 2PM EST on a weekday in here.. you will get 10 different answers, some of which is so far off you can't even google where they misheard whatever it is they are spewing |
| 22:55.11 | drmessano | Thats 10 people editing the same wiki page. |
| 22:55.37 | drmessano | Give them 10 pages on 10 sites, index and rank them, let the users decide who has the correct content |
| 22:55.46 | manxpower | NAT setup/issues has not changed much since like 1.0 |
| 22:56.01 | manxpower | drmessano: a bad example, but I see your point.; |
| 22:56.15 | drmessano | Not a bad example at all |
| 22:56.34 | drmessano | Thats actually something somewhat STATIC that you will get 10 arguments about |
| 22:56.40 | drmessano | Move on to dialplan |
| 22:56.49 | manxpower | there are bunches of examples of apps converted into functions, apps and functions being deprecated, etc. that stuff changes often it seems |
| 22:58.30 | drmessano | My point in using SIP NAT for an example was that the documentation is pretty simple.. but sit in here for two days and tell me how much misinformation is spread.. take that to a wiki with constant edits, and theres no telling how correct it will be one moment to the next.. let them have their own content and let some index/rank system give it value, and you're onto something |
| 22:59.03 | drmessano | Do the same for any other less-than-static functions and you can see popularity across a time period to gauge the freshness and accuracy of the cotent |
| 22:59.07 | drmessano | content |
| 22:59.28 | drmessano | Wiki is dry erase board in an office of 21 yr olds |
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| 22:59.54 | manxpower | drmessano: *I* can't even understand half of that crap sip/nat docs. But I blame the people that write those docs. Most of them have too much information. 90% of the tim I don't care if it's cone nat, one to one nat, one to many nat, or cat nat. It's all set up the same. |
| 23:01.38 | drmessano | Yes, but you throw a simple page out there: |
| 23:01.41 | drmessano | ~sipnat |
| 23:01.41 | 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 |
| 23:02.10 | drmessano | and people will read that 10 different ways.. and even tell you how wrong it is.. "because my friend told me..." |
| 23:03.03 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 23:04.18 | drmessano | Which is where my idea about ranking comes in.. which was sort of the idea behind a wiki.. peer review, dynamic content, self policing.. but you cant police content in a sandbox without 3 of those same 10 people who had 10 different resolves for a specific issue being the very ones that moderate it.. Unless we have a wiki with devs babysitting it all day |
| 23:04.22 | drmessano | Oh yay |
| 23:05.26 | drmessano | Maybe a Diggipedia would work.. |
| 23:05.58 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
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| 23:17.44 | _ShrikE | Can someone answer a question about polarity reversal and answer supervision? Is the reversal supposed to be a brief flash, or a permanent change of polarity? |
| 23:22.04 | drmessano | Brief |
| 23:27.02 | _ShrikE | drmessano: Thanks |
| 23:28.35 | *** join/#asterisk `RobertW (n=RobertW@ip70-178-13-110.ks.ks.cox.net) |
| 23:29.15 | `RobertW | I'm completely new to asterisk and I've read the manual.. but I couldn't figure out if you're able to dial out from console - possible? |
| 23:29.33 | drmessano | yes |
| 23:29.41 | drmessano | "originate" |
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| 23:52.05 | *** join/#asterisk luch (n=Dwayne@64.42.227.97) |
| 23:53.56 | luch | i have PRI -> CCM -> ATA -> FXO -> Asterisk, for some reason the it stops detecting dtmf digits as soon as the first one is pressed. I also have plain incoming PSTN FXO lines and dtmf works perfectly |
| 23:54.21 | andresmujica | hi, i've got a queue with a Local Channel as member, pointing to an AGI script. If the caller dials a number it should be able to enter a different queue ... is this possible ?? |
| 23:55.13 | luch | i hear a tiny echo following the first digit then it seems to stop detection |
| 23:55.47 | manxpower | luch: you have such a complicated setup I'm surprised that it works at all. |
| 23:56.06 | manxpower | any chance of connecting the CCM and Asterisk directly with SIP? |
| 23:56.57 | luch | manxpower: works perfectly, the calling part anyway. I have channel banks with fxo ports and the ccm is remote, i felt this was my best option at the time |
| 23:59.05 | luch | manxpower, its funny, see we have a large network of company interconnected and the ccm handles number it knows are internal bi itself, meaning they do not go to pstn, if I dial from any number that is not so routed by the call manager, like goes out on pstn it works fine |
| 23:59.22 | luch | is it possible that some sort of echo cancellation is causing this. |
| 23:59.44 | luch | sometime it will recognize the first two digits, other time only one |
