| 00:00.36 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 00:00.55 | znoG | unpaidbill: i never would have thought that would make a difference, but it did! thanks ... it is weird though |
| 00:01.02 | znoG | by exporting CC i could see it compiled using gcc-4.1 |
| 00:01.43 | asdx | how can i enable colors in the cli |
| 00:01.47 | mchou | drmessano: you here? |
| 00:12.01 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 00:15.54 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
| 00:19.32 | Corydon76-dig | asdx: run Asterisk in a console |
| 00:19.40 | Corydon76-dig | asdx: with safe_asterisk, perhaps |
| 00:22.00 | jaytee | I run safe_asterisk and when I use Putty I have colors. |
| 00:24.37 | rvhi | hi, does anyone use spectralink phones, now polycom wireless phones? |
| 00:24.42 | rvhi | does it talk sip? |
| 00:24.54 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a7d1643b4855191c) |
| 00:25.23 | [TK]D-Fender | rvhi: Go look at its manual |
| 00:27.52 | rvhi | anyone knows? |
| 00:28.36 | asdx | Corydon76-dig: safe_asterisk? |
| 00:33.04 | asdx | how can i stress my asterisk server |
| 00:33.08 | asdx | sipp? |
| 00:33.27 | jaytee | meetme |
| 00:33.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:34.18 | [TK]D-Fender | Meetme with all channels using G.729 |
| 00:34.29 | Strom_C | pee in the PCI slots |
| 00:34.31 | jaytee | yeah, that'd do it. |
| 00:34.36 | jaytee | hahahaha |
| 00:34.55 | asdx | Strom_C: i can't |
| 00:35.19 | jaytee | I'm sure going pee-pee in the server room is a big no-no and in the employee handbook somewhere |
| 00:35.38 | asdx | my server is in usa, i live in paraguay |
| 00:35.44 | asdx | and i wouldnt do it anyway |
| 00:35.44 | rvhi | i didn't see it mention sip, but went through the mailinglist and see someone said that it can be used with * |
| 00:35.45 | asdx | :p |
| 00:35.46 | Strom_C | asdx: it was a joke |
| 00:35.54 | asdx | i know |
| 00:36.04 | asdx | xD |
| 00:36.39 | asdx | damn i act like a computer sometimes |
| 00:37.47 | asdx | how can i do it with meetme |
| 00:38.05 | asdx | dev2*CLI> meetme |
| 00:38.06 | asdx | kick lock mute unlock unmute list |
| 00:39.29 | jaytee | with meetme all the rtp streams go through your server so get enough users in conference and you've got a good load on the box. |
| 00:39.59 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 00:40.09 | asdx | ok |
| 00:40.15 | asdx | how can i generate those users |
| 00:40.18 | asdx | sipp? |
| 00:40.42 | jaytee | you buy a bunch of sip phones and hire people to sit at them :-) |
| 00:41.06 | asdx | yeah i want to buy a hardphone |
| 00:41.54 | jaytee | so who ya gonna call? Ghostbusters? |
| 00:42.06 | asdx | the police |
| 00:42.19 | asdx | they will never find me |
| 00:42.33 | asdx | or pizza hut |
| 00:43.56 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 00:52.04 | *** join/#asterisk DigitalIrony (n=eric@216.207.245.1) |
| 01:00.28 | *** join/#asterisk Gwayne (n=Gwayne@cm196.kappa245.maxonline.com.sg) |
| 01:03.07 | mikehime | well i got asterisk to connect to mysql via odbc |
| 01:03.12 | mikehime | but it won't return query results |
| 01:03.15 | mikehime | just returns null |
| 01:03.39 | mikehime | is there a trick? o.O |
| 01:03.51 | LoOoD | Any recommendations for a sip trunking provider? |
| 01:04.16 | mikehime | LoOoD: for land line termination? |
| 01:04.45 | drmessano | What other kind is there? |
| 01:04.53 | mikehime | lol just making sure i understood 'trunking' |
| 01:05.01 | mikehime | i don't come from a telephony background |
| 01:05.41 | [TK]D-Fender | ~trunk |
| 01:05.41 | jbot | extra, extra, read all about it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 01:05.42 | [TK]D-Fender | ~siptrunk |
| 01:05.43 | jbot | rumour has it, siptrunk is To use a SIP trunk with Asterisk simply add trunk=yes in sip.conf [general] or in the peer definition. |
| 01:05.43 | [TK]D-Fender | hrm... |
| 01:05.44 | *** join/#asterisk raytruz` (n=raytruz_@74-138-103-238.dhcp.insightbb.com) |
| 01:05.44 | asdx | i use teliax |
| 01:05.44 | asdx | it's pretty good |
| 01:05.44 | mikehime | i have had good experience with voip.ms |
| 01:05.44 | lanning | puts [TK]D-Fender in a trunk. |
| 01:05.58 | coppice | ~trunk is Where the horrible noises come from (at least for an elephant). |
| 01:05.58 | jbot | ...but trunk is already something else... |
| 01:06.26 | drmessano | WTFFFF |
| 01:06.47 | drmessano | siptrunk is WRONG |
| 01:06.56 | LoOoD | mike: we're running out of analog lines for incoming calls.. We either need to goto a PRI or sip trunking/virtual pri(?) |
| 01:07.01 | seanbright | drmessano: fix it |
| 01:07.05 | drmessano | jbot_: forget siptrunk |
| 01:07.28 | *** join/#asterisk DigitalIrony (n=eric@216.207.245.1) |
| 01:07.59 | drmessano | ~siptrunk |
| 01:07.59 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 01:08.03 | ManxPower | LoOoD: Why not do both? Have your analog lines roll over to a local sip provider number. |
| 01:08.04 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 01:08.24 | drmessano | Done. |
| 01:08.31 | ManxPower | drmessano: You have *no* sense of humor. |
| 01:08.57 | drmessano | That was supposed to be humorous? |
| 01:09.06 | seanbright | jbot_: siptrunk is also a whale's vagina |
| 01:09.29 | ManxPower | It is harmless to add that, as Asterisk ignores unknown options in it's config files, and it is sort of like taking someone on a Snipe hunt. |
| 01:10.10 | drmessano | ManxPower: And is also along the lines of the whole "misleading the users" and "making shit up" that you went a fucking tyrade at me about one night... |
| 01:10.20 | LoOoD | ManxPower: we just want to get rid of the analog lines all together. they've caused too much trouble. |
| 01:10.24 | drmessano | Sense of humor, yes.. less of one from being in here, that too |
| 01:10.38 | seanbright | ~slap drmessano |
| 01:10.39 | jbot | ACTION slaps drmessano, keep your grubby fingers to yourself! |
| 01:13.59 | ManxPower | drmessano: touche. |
| 01:14.08 | ManxPower | my stuff is *harmless*. |
| 01:15.18 | drmessano | I've never told anyone anything *harmful* |
| 01:15.35 | drmessano | or even directed someone to do anything incorrect on purpose |
| 01:15.38 | drmessano | But anyway |
| 01:16.29 | drmessano | Guess it's about *who* .. as always... It's only a "joke" or "correct" depending on *who" says it.. |
| 01:16.32 | drmessano | Fuck it |
| 01:16.38 | drmessano | Had enough |
| 01:16.40 | *** part/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 01:16.45 | *** part/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 01:19.22 | mikehime | ~o.O |
| 01:21.33 | unpaidbill | ~callweaver |
| 01:21.34 | jbot | callweaver is, like, a fork of Asterisk version 1.2. It was originally called OpenPBX, but changed its name to Callweaver. Essentially it's a big search and replace on ast to opbx, and then a big search and replace from opbx to cw, and then various GPL only addons included.... #callweaver ... http://www.callweaver.org/ |
| 01:22.00 | asdx | why did they fork it |
| 01:22.24 | unpaidbill | i dunno, but it has t.38 support supposedly |
| 01:22.37 | asdx | hm |
| 01:22.42 | unpaidbill | not passthrough either |
| 01:23.15 | unpaidbill | maybe 1.6 will have t.38 |
| 01:24.23 | file | 1.6.1 will have T38 termination and origination |
| 01:24.37 | file | actually even 1.6.0 will if I remember right |
| 01:26.14 | seanbright | you remember right |
| 01:26.27 | unpaidbill | oh baby |
| 01:26.30 | asdx | i want to drink this http://en.wikipedia.org/wiki/OpenCola |
| 01:26.37 | unpaidbill | i cant friggen wait for 1.6 |
| 01:26.38 | seanbright | asdx: it's not that good |
| 01:26.51 | asdx | seanbright: lets improve it xD |
| 01:26.57 | seanbright | asdx: meh |
| 01:26.58 | seanbright | :) |
| 01:27.45 | unpaidbill | i want to drink meat water |
| 01:27.50 | asdx | seanbright: did you drink it already |
| 01:27.57 | asdx | opencola |
| 01:27.59 | unpaidbill | http://dinnerinabottle.com/ it's fake though :( but it looks so real |
| 01:28.06 | seanbright | asdx: yessir |
| 01:28.11 | asdx | cool |
| 01:28.19 | seanbright | asdx: back in the day... like 8 or 9 years ago |
| 01:28.23 | seanbright | asdx: too sodium-y |
| 01:28.28 | asdx | lol |
| 01:30.28 | asdx | meat water o_O |
| 01:32.51 | unpaidbill | yeah mmmm, bacon and eggs |
| 01:33.41 | asdx | heh |
| 01:38.31 | asdx | can i dial directly with originate without a extension in my dialplan |
| 01:38.33 | asdx | ? |
| 01:41.11 | unpaidbill | why would you do that |
| 01:41.25 | asdx | to call the phone in my house |
| 01:41.27 | unpaidbill | like have it dial and put.. Playback in instead of extension? |
| 01:41.52 | asdx | yeah |
| 01:41.57 | unpaidbill | i dont think so |
| 01:42.05 | asdx | :( |
| 01:42.10 | unpaidbill | try it out, but i'm pretty sure you have to specify an extension or channel |
| 01:42.12 | seanbright | sure you can |
| 01:42.24 | seanbright | call files allow you to specify an application instead of an extension |
| 01:42.32 | unpaidbill | oh |
| 01:42.33 | asdx | cool |
| 01:42.33 | unpaidbill | nifty |
| 01:42.36 | seanbright | originate... i dunno |
| 01:42.42 | seanbright | may need to look that one up |
| 01:42.54 | unpaidbill | call files are handy buggers |
| 01:42.59 | asdx | Usage1: originate <tech/data> application <appname> [appdata] |
| 01:43.09 | asdx | Usage2: originate <tech/data> extension [exten@][context] |
| 01:43.15 | asdx | usage1 is what i want |
| 01:43.16 | asdx | heh |
| 01:43.21 | unpaidbill | well i'll be damned |
| 01:45.01 | *** join/#asterisk DigitalIrony (n=eric@216.207.245.1) |
| 01:47.44 | *** join/#asterisk nny_1 (n=Scott@64.20.141.77.dyn-e-pool14.pool.hargray.net) |
| 01:49.53 | mikehime | must one use ${EXTEN} to put DTMF into a variable? |
| 01:50.30 | nny_1 | i am remotely working on getting music on hold to play line in.. i got the latest alsa drivers loaded, all things seem to indicate should work (using /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw ) but since i am not on site (and their network admins didn't forward the vpn ports for this box i set up ><) anyone got another way to check it? |
| 01:50.39 | nny_1 | i have an extension set-up for it |
| 01:51.17 | nny_1 | not sure how to dial an extension directly from cli though |
| 01:54.04 | asdx | nny_1: dial? |
| 01:54.25 | asdx | Usage: console dial [extension[@context]] |
| 01:54.48 | asdx | i'm trying to dial from the cli without a extension |
| 01:55.00 | asdx | i just want to dial directly |
| 01:55.29 | nny_1 | asdx: ha |
| 01:55.32 | nny_1 | asdx: do what i did >< |
| 01:55.50 | nny_1 | at least in my dialplan dial 400 was dialing into the default context >< |
| 01:56.01 | nny_1 | the @context is what i didn't do |
| 01:56.41 | asdx | heh |
| 01:58.00 | nny_1 | <PROTECTED> |
| 01:58.12 | nny_1 | wonder if thats because it's broken or it's console >< |
| 01:59.44 | nny_1 | the howto says "at least one mp3 file needs to be in /var/lib/asterisk/[directory].. wonder if they mean an empty one |
| 02:00.14 | *** part/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:00.17 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:02.30 | *** part/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:02.57 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:04.45 | *** part/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:05.05 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:06.25 | nny_1 | hmm |
| 02:06.27 | nny_1 | <PROTECTED> |
| 02:06.50 | nny_1 | <PROTECTED> |
| 02:06.57 | nny_1 | wonder what is using it supposedly |
| 02:07.41 | seanbright | lsof |
| 02:08.54 | nny_1 | k thanks |
| 02:09.42 | nny_1 | says nothing |
| 02:09.45 | seanbright | lsof /dev/dsp |
| 02:09.48 | nny_1 | perms are asterisk:asterisk 777 |
| 02:09.49 | seanbright | that would be better |
| 02:09.49 | seanbright | heh |
| 02:10.02 | nny_1 | yah returns null |
| 02:10.12 | seanbright | then i dunno |
| 02:11.13 | nny_1 | hmm it does that if i put nothing.mp3 in /var/lib/asterisk/linein |
| 02:11.25 | nny_1 | if i dont it says -- Started music on hold, class 'linein', on channel 'Console/dsp' -- Stopped music on hold on Console/dsp |
| 02:14.25 | seanbright | lost interest after "k thanks" |
| 02:14.26 | seanbright | :) |
| 02:14.42 | nny_1 | :D |
| 02:14.54 | nny_1 | well i think it is working, but guess i need to test on site with a phone |
| 02:15.06 | seanbright | couldn't hurt |
| 02:16.43 | *** join/#asterisk moy (n=moyhu@189.169.82.208) |
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| 02:22.23 | *** part/#asterisk nny_1 (n=Scott@64.20.141.77.dyn-e-pool14.pool.hargray.net) |
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| 02:22.53 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 02:25.49 | sahafeez | question, if i have acd setup with 3 queues - and at some point the inbound call is forwarded to a cell phone, i see the caller id of the inbound call - however what i need to do is say what queue it came from, can that be done? |
| 02:26.59 | sahafeez | does that make sense |
| 02:27.01 | *** join/#asterisk shido6 (n=shido6@74-130-224-188.dhcp.insightbb.com) |
| 02:27.53 | lanning | the only thing you can do is play a prompt when the person answers. |
| 02:28.09 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 02:28.28 | sahafeez | what will the caller hear? |
| 02:29.00 | sahafeez | i need the guy on the cell phone to know it came from the support queue or the sales queue etc.. |
| 02:29.10 | lanning | what ever you want them to hear. |
| 02:30.06 | lanning | you can't just blindly send to the cell phone. you will have to dial the cell phone, and when it answers, you need to announce the call to the cell phone user |
| 02:30.18 | sahafeez | can i give the person on the cell the choice to voicemail the caller |
| 02:30.27 | sahafeez | okay.. |
| 02:30.27 | lanning | then they respond (press one to accept) and connect the call. |
| 02:33.15 | sahafeez | okay, can i do that on the agent level or does it have to be on the queue level. i only want this to happen when the call has been sent to a cell phone |
| 02:37.07 | sahafeez | i have been looking for an example of this.. |
| 02:38.30 | jeev | i was talking to one of the data center techs.. he was telling me polycom's suck |
| 02:38.30 | jeev | anyway |
| 02:38.40 | jeev | told me to get ATA's if i want to use regular phones.. wjhch ATA's |
| 02:38.50 | sahafeez | polycom works just fine. |
| 02:39.16 | jeev | hmm |
| 02:39.21 | jeev | i know i've been told a ton of things but |
| 02:39.31 | jeev | could my static have been caused by like |
| 02:39.40 | jeev | this is my set up.. or at least was (i lost my asterisk conf at the datacenter) |
| 02:39.48 | jeev | i had the phoen connected straight to the datacenter.. instead of a local asterisk box. |
| 02:39.54 | jeev | what do you think? upstream was 1mbit was no usage at all |
| 02:40.06 | jeev | polycom to server ~10ms, another ~10ms from server to wholesaler |
| 02:41.28 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 02:45.39 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:51.51 | *** join/#asterisk DigitalIrony (n=eric@216.207.245.1) |
| 02:59.40 | *** join/#asterisk john_fbac (i=johnfbac@202.sub-75-250-121.myvzw.com) |
| 02:59.41 | [TK]D-Fender | jeev: static can be caused by a jitter buffer overflow or at source. Thats typically it. |
| 03:00.59 | [TK]D-Fender | jeev: And if he wants to blame Polycom, he's a schmuck |
| 03:01.25 | [TK]D-Fender | jeev: Polycoms sound good where every other phone flakes out at PL. their jitter buffers are the best |
| 03:03.09 | jaytee | I <3 my Polycom IP330 |
| 03:07.20 | *** join/#asterisk Gregabyte_ (n=wintermu@216.207.245.1) |
| 03:10.33 | jeev | hmm |
| 03:10.41 | jeev | fender, what do you think about just a ATA |
| 03:10.49 | jeev | he suggested this http://www.vanaccess.com/show.asp?productid=223&catalogid=27 |
| 03:10.59 | jeev | man i messed up my server, formatted.. forgot to back up my asterisk and postfix MAIn config. |
| 03:11.03 | jeev | now i'm trying to recover ufs2 |
| 03:11.12 | jeev | only problem is i have the ufs2 recovery software for windows but no scsi on my windows box |
| 03:13.07 | [TK]D-Fender | jeev: Are you somehow mystically drawn to GARBAGE? |
| 03:13.37 | [TK]D-Fender | jeev: First Linksys Wiki, now this? |
| 03:13.43 | [TK]D-Fender | WiFi |
| 03:13.53 | jeev | hahaha |
| 03:13.56 | jeev | fender, typically.. yes. |
| 03:14.06 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 03:18.29 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:22.39 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.49) |
| 03:23.26 | BeeBuu | dial(local/XXX),what's the "local" mean? |
| 03:24.02 | Strom_C | local channel |
| 03:24.23 | BeeBuu | mean what? |
| 03:24.37 | Strom_C | ...look it up in the book |
| 03:24.43 | Strom_C | it lets asterisk loop back into itself |
| 03:24.45 | Strom_C | ~book |
| 03:24.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:25.02 | [TK]D-Fender | BeeBuu: its a channel with just dialplan executing. |
| 03:25.16 | BeeBuu | o,i got it! |
| 03:25.35 | BeeBuu | just make dialplan executing... |
| 03:25.56 | BeeBuu | [TK]D-Fender: thanks.are you a teacher? |
| 03:26.11 | [TK]D-Fender | BeeBuu: Depends on your point of view. |
| 03:26.46 | BeeBuu | your teach skill very good for me ( nut ). |
| 03:27.21 | docelmo | What is the directive to limit the amount of calls asterisk can do? |
| 03:27.29 | docelmo | is it in sip.conf? |
| 03:28.14 | docelmo | I just found it.. maxcalls in asterisk.conf |
| 03:29.53 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:30.30 | docelmo | [TK]D-Fender with max calls if I try and send a call +1 of the directive will asterisk hand off congestion or will it queue up the call? Like for instance if I have 500 callfiles I execute them.. Tell asterisk it can only handle 499 when the 500 callfile is presented what will happen with it? |
| 03:31.27 | [TK]D-Fender | docelmo: No clue. It will try to run them all SIMULTANEOUSLY. You have the resources to hand off that much? |
| 03:31.56 | docelmo | yes.. I have been able to benchmark 534 at once with ulaw media |
| 03:32.36 | [TK]D-Fender | docelmo: Ouch..... well no idea on the concept of telling * to limit calls VS hitting the wall so to speak |
| 03:33.34 | docelmo | ya.. I am trying to figure out how hard it would be to make an internal queueing system in asterisk so if X amount of callfiles or originates are presented asterisk will throttle the output volume based on the maxcall variable or some other means |
| 03:34.27 | [TK]D-Fender | docelmo: Don't use call files then, use AMI to spool them in, and flag them with a var and watch AMI for their termination. |
| 03:35.43 | docelmo | Well that is what I am doing.. The thing is trying to keep track of how many calls are going on at one time on 4 boxes and make it intelligent enough so that when a channel free's up another call is dispursed into the queue |
| 03:36.34 | [TK]D-Fender | docelmo: Channel count ought to do, or a manual counter of (placed - Channel End Msg count) |
| 03:37.08 | docelmo | [TK]D-Fender mind chatting off channel for a minute? |
| 03:37.16 | [TK]D-Fender | docelmo: k |
| 03:37.40 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 03:37.59 | BeeBuu | [TK]D-Fender: can i load dialplan from a database? |
| 03:38.14 | [TK]D-Fender | BeeBuu: Yes, its in the book. |
| 03:38.29 | BeeBuu | cool. |
| 03:38.39 | BeeBuu | i read it now~ |
| 03:39.51 | docelmo | TK did you get the pm? |
| 03:39.53 | docelmo | nevermind |
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| 04:25.27 | [TK]D-Fender | ok, checkout time. Later all. |
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| 04:41.02 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
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| 05:11.02 | x_or | Is there something I need to configure to get a callerid to process correctly in my extensions.conf file? |
| 05:11.23 | Strom_C | what do you mean "process correctly"? |
| 05:11.25 | x_or | I put the variable in there when it is passed from voicepulse and the variable is empty when it reaches my AGI script. |
| 05:11.47 | x_or | Sorry, I used something like this: exten => _XXXX,n,Record(project/${CALLERIDNUM}.wav|3) ;; filename|silence|max|timeout |
| 05:11.52 | x_or | And, this is empty. |
| 05:12.07 | Strom_C | what version of asterisk are you using? |
| 05:12.18 | x_or | <PROTECTED> |
| 05:12.28 | Strom_C | you should have read the upgrade.txt file |
| 05:12.33 | x_or | Ok. |
| 05:12.39 | Strom_C | ${CALLERIDNUM} was deprecated in 1.2 and totally removed in 1.4 |
| 05:12.44 | x_or | OK. |
| 05:12.44 | Strom_C | use the CALLERID function instead |
| 05:12.47 | pputman | Callerid(num) |
| 05:12.55 | x_or | Ok. I'll look that up. |
| 05:16.12 | mikehime | anyone still around? |
| 05:16.17 | mikehime | =) |
| 05:17.39 | Strom_C | no |
| 05:17.41 | Strom_C | we're all dead |
| 05:17.49 | Strom_C | please leave a message after the maggot |
| 05:17.52 | Strom_C | *squish* |
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| 05:40.13 | asdx | how much calls per second can asterisk do, or concurrent calls at the same time... |
| 05:45.20 | unpaidbill | depends on the server |
| 05:46.00 | unpaidbill | depends on how you place the calls.. sip, zap, iax.. |
| 05:46.20 | unpaidbill | http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 05:46.23 | *** join/#asterisk Evol_M (n=IrcQnet@rrcs-76-79-244-108.west.biz.rr.com) |
| 05:46.27 | unpaidbill | there is some basic and generic info at that link |
| 05:46.32 | asdx | core 2 duo, 3.20 ghz, 2 gb ram, sip |
| 05:46.43 | Evol_M | hey, im doing call forwarding but both parties cant hear each other, any ideas? |
| 05:47.03 | asdx | unpaidbill: thx |
| 05:48.20 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 05:48.21 | asdx | would be interesting to see the performance of 1.4 compared to 1.6, etc |
| 05:48.35 | kaldemar | Evol_M: any NAT's on the media path? |
| 05:48.43 | asdx | is there changes in the core that makes 1.6 better in scalability etc? |
| 05:48.54 | asdx | performance/scalability |
| 05:49.04 | Evol_M | i put nat=yes in sip_nat.conf |
| 05:49.19 | unpaidbill | no clue asdx |
| 05:51.27 | kaldemar | Evol_M: that won't fix all nat issues. you'll get the best help if you give an accurate description of your setup and the problem. network and configuration wise. |
| 05:51.46 | Evol_M | well im behind a router |
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| 05:52.10 | Evol_M | with ports 10000-20000 and 5060-5060 |
| 05:53.34 | Evol_M | anything specific you need to know? |
| 05:53.55 | kaldemar | are you a phone or an asterisk server? where are the other phones? who's calling who? where is the call getting transferred? |
| 05:54.27 | Evol_M | ok i have a SIP account it comes in and gets forwarded to a external cell phone |
| 05:55.16 | kaldemar | do you have externip defined in sip.conf? |
| 05:55.32 | Evol_M | i have it in sip_nat.conf |
| 05:56.04 | kaldemar | so you're using freepbx? |
| 05:56.23 | Evol_M | kaldemar, http://pastebin.ca/1050839 |
| 05:56.39 | Evol_M | yes |
| 05:57.44 | kaldemar | then #freepbx is the place for you. |
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| 05:59.36 | kaldemar | but, if you're doing that stuff manually, take a sip debug trace of a call and look into the SDP parts of the SIP messages. if you have a private address there for the media, it's a nat issue that SHOULD be fixed with externip/externhost. |
| 06:00.42 | kaldemar | tcpdumping for the rtp packets is also a way to make sure that the audio is coming through your router. i assume you have forwarded rtp ports in your router. |
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| 07:03.56 | x_or | Does WaitExten have a limit of 4 digits that can be entered? |
| 07:04.04 | x_or | Is there a way to get more than that? |
| 07:06.27 | DigitalIrony | x_or: WaitExten doesn't really take digits itself |
| 07:07.13 | x_or | OK. So, is there a different command I should be using to gather input from the caller? |
| 07:07.16 | DigitalIrony | x_or: its more of a WaitExten(20) waits for 20 seconds for a user to enter an extension that you have some where else in the dial plan |
| 07:07.49 | DigitalIrony | x_or: how do you mean? like to store in a variable? or to dial an extension ? |
| 07:08.05 | x_or | Storing in a variable would be preferred. Looks like read()? |
| 07:08.39 | DigitalIrony | x_or: check out show application SET and READ in the asterisk CLI |
| 07:09.03 | x_or | OK. |
| 07:09.28 | DigitalIrony | x_or: if you just want it to call some one or to do something like run a macro etc....just make a new exten and when they dial that it will execute the application |
| 07:10.47 | DigitalIrony | x_or: but yes it is Read() to take input....you may want to make a Playback() before that though to prompt for it |
| 07:11.28 | x_or | It looks like Read(variable,filename) allows me to playback and set a variable all in one call, no? |
| 07:11.29 | DigitalIrony | you can also have READ play a prompt |
| 07:11.40 | DigitalIrony | your correct |
| 07:11.58 | DigitalIrony | But it depends on how you want to set it up.... |
| 07:12.04 | DigitalIrony | each way has its uses |
| 07:13.39 | x_or | OK, thanks for your help. This saved me a lot of time bouncing my thoughts off you. |
| 07:13.51 | DigitalIrony | x_or: anytime. |
| 07:13.53 | x_or | I'm very new to Asterisk. Wow, it is so cool. |
| 07:14.08 | DigitalIrony | x_or: I enjoy it. |
| 07:14.14 | kaldemar | and there's always the good old background app. |
| 07:14.17 | DigitalIrony | x_or: do you have any hardware |
| 07:14.24 | x_or | No, I'm using a virtual server. |
| 07:14.31 | DigitalIrony | x_or: ahhh |
| 07:15.08 | x_or | Why do you ask? |
| 07:15.09 | DigitalIrony | x_or: well once you get used to using voip if you can get your hands a cheap analog card it would be fun to play with |
| 07:15.44 | x_or | OK, I will consider that. What does that give me beyond just using Asterisk and a SIP line? |
| 07:16.35 | DigitalIrony | x_or: well not too much really, just gives you basic pots connection, which can depending be cheaper or more expensive where you live |
| 07:16.59 | x_or | I see. I live in Portland, OR, USA. |
| 07:18.09 | DigitalIrony | x_or: cool, i still don't know what it would be price wise for you, but think of it like this, if you already have a land line phone, you can basically have all your incoming calls routed through your asterisk box |
| 07:18.36 | DigitalIrony | x_or: you could basically make your own small little telco inside your house :P |
| 07:19.30 | x_or | That sounds fun. |
| 07:19.34 | x_or | How much is the hardware? |
| 07:19.52 | DigitalIrony | x_or: digium.com |
| 07:20.04 | DigitalIrony | x_or: can be quite expensive really |
| 07:20.21 | DigitalIrony | x_or: but there is always ebay and craigslist etc |
| 07:20.41 | x_or | ok. |
| 07:20.42 | DigitalIrony | x_or: or you might can try contacting one of our resellers, sometimes they have good deals |
| 07:21.45 | DigitalIrony | x_or: your simplest and probably best setup for you would be a TDM410 |
| 07:22.17 | DigitalIrony | x_or: with one or two FXO's and one or two FXS's |
| 07:24.47 | pputman | note to self: Make sure you're on the asterisk console before issuing commands that start with a ! |
| 07:26.45 | x_or | DigitalIrony: Thanks. |
| 07:39.31 | *** join/#asterisk ^shark_ (n=^shark_@41.222.2.65) |
| 07:42.49 | x_or | I'm struggling to get the read() command working. I have a dialplan like exten => s,n(start),Read(VERIFY,enter-pin,10) because I want to label it so I can jump back there if the PIN entry fails. But, now asterisk just falls through. This worked when I was using WaitExten because I would specify an extension and then use exten => _XXXX,n(start),... but now it does not. |
| 07:43.23 | x_or | Should I do something like exten => s,n,Set(EXTEN=1234) and then continue using exten => _XXXX,n,...? |
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| 07:46.06 | x_or | Oops, I guess it does work, I just used the wrong file. |
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| 08:17.48 | Vec | Does anyone have experiance using a data modem, in this config Data Modem > ATA (G711) > 100Mbit LAN > Asterisk > PRI ? |
| 08:23.01 | jblack | vec: Nope. you can try ulaw though. |
| 08:25.32 | Strom_C | Vec: you might get away with it if you jitterbuffer |
| 08:25.46 | Strom_C | but it's still a solution that strikes me as flaky |
| 08:31.27 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:31.55 | mandh | Hi all |
| 08:32.00 | mandh | 1009 (agent9) logged in on Zap/1-1 talking to Zap/2-1 (musiconhold is 'default') |
| 08:32.28 | mandh | but it is imposiible that agent log from zap chanell |
| 08:32.37 | mandh | how that happens ? |
| 08:34.48 | Vec | Strom_C : thanks |
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| 08:45.20 | DigitalIrony | Mandh: im not sure i understand you |
| 08:45.44 | mandh | DigitalIrony, ok i will re express the idea |
| 08:46.34 | mandh | DigitalIrony, after i exected the command show agents online |
| 08:46.38 | mandh | the result was |
| 08:47.45 | mandh | that some of agents 1009 : |
| 08:47.45 | mandh | 1009 (agent9) logged in on Zap/1-1 talking to Zap/2-1 (musiconhold is 'default' |
| 08:47.58 | mandh | but the agent 1009 is sip account , and can log only from sip network attached to the server , and cant login from any zap chanles |
| 08:48.04 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:48.38 | mandh | so how the agent 1009 loged from Zap/1-1 as the messeage says |
| 08:49.27 | DigitalIrony | mandh: im sorry i still don't understand...let me think about it for a sec |
| 08:49.59 | DigitalIrony | are you wanting someone on an analog connection to be able to login as an agent? is that what you are asking? |
| 08:52.16 | mandh | d |
| 08:52.56 | mandh | DigitalIrony, i have an queue and agents from 1000-1011, the agents logged from sip ip phone |
| 08:53.13 | mandh | and the asterisk server connected to Zap channels |
| 08:53.50 | mandh | so all incoming calls from Zap chanlles replied by agents that are alreday logged in to queue |
| 08:54.07 | mandh | the agent 1009 tell me that he cant log in |
| 08:54.16 | DigitalIrony | ahh |
| 08:54.20 | mandh | when i exectued the command |
| 08:54.34 | mandh | show agents online |
| 08:54.49 | mandh | one of the output lines was |
| 08:55.01 | mandh | 1009 (agent9) logged in on Zap/1-1 talking to Zap/2-1 (musiconhold is 'default') |
| 08:55.23 | DigitalIrony | hmmm |
| 08:55.24 | mandh | while other agents was like |
| 08:55.26 | mandh | 1007 (agent7) available at '700@customer_care' (musiconhold is 'default') |
| 08:55.39 | DigitalIrony | well |
| 08:55.54 | ^shark_ | mandh: attaching the channel configs for the 1009 would try and help |
| 08:56.17 | DigitalIrony | could be that agent9 possibly answered a call and transfered it back out to zap2 and until that call his hungup then his line will be tied up |
| 08:56.43 | DigitalIrony | and from that looks like some sort of transfer call loop |
| 08:56.49 | mandh | right |
| 08:56.53 | mandh | there was an loop |
| 08:56.54 | mandh | coz |
| 08:56.57 | DigitalIrony | try and hangup zap1 |
| 08:57.32 | mandh | when 1009 try to log agin ,new message tell him u are logged , and then new message tell him to log again |
| 08:57.59 | mandh | yes i hanguped zap1 but i wann to know why that happens to avoid it at future |
| 08:58.12 | DigitalIrony | a bad transfer |
| 08:58.24 | DigitalIrony | or a call that wasn't properly hungup in the dialplan |
| 08:58.37 | DigitalIrony | make sure you have a Hangup() where there should be |
| 08:58.50 | DigitalIrony | or else sometimes you get "wild" calls |
| 08:58.52 | mandh | yeah at queue |
| 08:59.20 | DigitalIrony | basically with queues |
| 08:59.22 | mandh | what u mean wild calls , what i fear that some ones can usge the channels to use them as gateway |
| 08:59.35 | DigitalIrony | mandh: that is possible |
| 08:59.44 | mandh | like on call zap1 and then try to call zap2 to internatiol call |
| 09:00.03 | DigitalIrony | mandh: make sure you are on the newest versions of astersik and zaptel to minimize security risks |
| 09:00.23 | DigitalIrony | mandh: you also need to lock down you outbound calls |
| 09:00.50 | DigitalIrony | mandh: no one calling in should be able to access an outbound call without using DISA or ro AUTH |
| 09:00.56 | DigitalIrony | *or AUTH |
| 09:01.12 | mandh | how to |
| 09:01.18 | mandh | i mean like what happens |
| 09:01.31 | DigitalIrony | it depends on your dialplan |
| 09:01.37 | DigitalIrony | if say |
| 09:01.40 | mandh | incoming calls and one of agents transfer him to new zap |
| 09:01.48 | DigitalIrony | yes |
| 09:02.06 | mandh | how to handle like this cases |
| 09:02.21 | DigitalIrony | or maybe some one discovered a security hole? |
| 09:02.28 | DigitalIrony | well |
| 09:02.48 | mandh | right |
| 09:02.49 | DigitalIrony | make sure that your incoming context doesn't have access to your internal context |
| 09:03.10 | DigitalIrony | and make an outgoing context for international calls |
| 09:03.12 | *** join/#asterisk oilinki3 (n=oil@ppp-124-121-246-136.revip2.asianet.co.th) |
| 09:03.16 | DigitalIrony | or things like that |
| 09:03.26 | mandh | second |
| 09:03.45 | DigitalIrony | and let only internal users or users who call in with and authenticate with DISA or AUTH to make outbound calls |
| 09:03.56 | DigitalIrony | but even that is risky |
| 09:04.12 | DigitalIrony | if someone wanted im sure they could brute force it |
| 09:07.12 | mandh | DigitalIrony, incoming context only play musci and then put the call at queue |
| 09:07.43 | mandh | i think using that the context is sperated from outgoing calls |
| 09:08.02 | mandh | but what fear me , features like transfer calls or things that i not know |
| 09:19.37 | DigitalIrony | mandh: then you should read up more on the dialplan, have you read the asterisk book? |
| 09:20.26 | mandh | DigitalIrony, yeah lot of the book |
| 09:20.52 | mandh | i see i need to read it agains and again |
| 09:24.00 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:24.13 | mandh | DigitalIrony, now i test transferd incomming call to outgoing call and it is work , it is security hole |
| 09:25.53 | pputman | mandh, Change the context of the person who is doing the transfer to not allow outbound calls. |
| 09:25.53 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 09:26.07 | pputman | And see if it resolves it |
| 09:29.52 | mandh | pputman, but i need that agents make outgoing calls |
| 09:30.12 | mandh | so how only banned outgoing calls from trnasferd only |
| 09:31.03 | pputman | mandh, you can configure it in your sip/iax/zaptel.conf whatever to not allow transfers |
| 09:31.29 | mandh | but i need make transer calls between agents |
| 09:31.32 | mandh | ?!!! |
| 09:32.06 | pputman | mandh, I'm sure there's a way to only allow people to transfer to certain extensions, but I don't know what it is. |
| 09:32.21 | mandh | imagine one make incoming call from zap1 to agetns, agent one reply and need to transfer the call to agent2 |
| 09:35.14 | *** join/#asterisk dieno (i=771e694a@gateway/web/ajax/mibbit.com/x-fe4f604e27089518) |
| 09:36.06 | pputman | mandh, however, if you trust your agents enough for them to be able to dial outbound, I would assume they wouldn't be transferring people to external calls. |
| 09:36.51 | mandh | right , by the way i become not turst any one :-D |
| 09:41.50 | DigitalIrony | mandh: maybe better if you hire a consultant :P |
| 09:42.53 | mandh | DigitalIrony, are u ready ;) |
| 09:42.53 | Fanee | tranfering calls between agents are quite easy |
| 09:43.05 | Fanee | just make a local intercom dial plan |
| 09:43.18 | Fanee | and set agents context to that |
| 09:44.18 | *** part/#asterisk BeeBuu (n=beebuu@219.132.188.49) |
| 09:44.30 | DigitalIrony | mandh: sorry I can't do it :P I work for digium, and im not allowed to be a consultant |
| 09:44.54 | mandh | but i wannt agnets can transfer call but i wann that the transferd call cant be transfred outgoing |
| 09:45.15 | Fanee | yes its not much hard |
| 09:45.18 | mandh | DigitalIrony, oh very fine great company i hope to do there |
| 09:45.21 | DigitalIrony | you would have to make a macro basically to see where the call is comming from |
| 09:45.25 | Fanee | you can control your context |
| 09:45.42 | DigitalIrony | readup on gotoif |
| 09:45.44 | DigitalIrony | and macros |
| 09:45.50 | Fanee | you can also filter it by callerid |
| 09:45.59 | DigitalIrony | exactly |
| 09:46.49 | DigitalIrony | mandh: maybe you would benefit from our bootcamp |
| 09:47.04 | DigitalIrony | mandh: they usually teach some very basic dial plan security there |
| 09:47.10 | mandh | DigitalIrony, what bootcamp |
| 09:47.25 | DigitalIrony | mandh: digium has an asterisk 5 day training course |
| 09:47.34 | mandh | free ;) |
| 09:47.36 | mandh | or? |
| 09:47.38 | DigitalIrony | mandh: nope |
| 09:47.44 | trnzmeta | wishful thinking |
| 09:47.46 | DigitalIrony | mandh: costly but worth it |
| 09:47.47 | trnzmeta | free |
| 09:47.49 | mandh | i am in syria , where is the course? |
| 09:48.04 | DigitalIrony | mandh: you will have to check on digium.com they move around |
| 09:48.19 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 09:48.23 | mandh | oh very fine |
| 09:48.28 | mandh | may the come near our country |
| 09:48.54 | DigitalIrony | http://www.digium.com/en/training/locator/results.php |
| 09:49.41 | DigitalIrony | im not very good with geography so im not sure |
| 09:50.06 | mandh | dont say that u dont hear or know about syria |
| 09:50.13 | DigitalIrony | i know about it |
| 09:50.22 | mandh | from where u are? |
| 09:50.23 | DigitalIrony | i just can't see where it is and whats close to it in my head |
| 09:50.26 | DigitalIrony | US |
| 09:50.27 | DigitalIrony | AL |
| 09:50.36 | mandh | oh very far |
| 09:51.03 | DigitalIrony | yes |
| 09:51.14 | DigitalIrony | but we have bootcamps in other countries |
| 09:52.53 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:53.00 | DigitalIrony | there are spain, korea, south africa |
| 09:53.12 | DigitalIrony | italy |
| 09:53.25 | DigitalIrony | those are probably closer than here or las vegas |
| 09:54.54 | DigitalIrony | but I don't know if they will be doing one in the middle east |
| 09:56.28 | DigitalIrony | mandh: but just sit in here and read other peoples problems and ask questions, thats probably the best way to learn |
| 09:56.35 | DigitalIrony | mandh: that and make mistakes |
| 09:56.41 | mandh | right |
| 09:57.13 | mandh | it is very nice IRC and i think best method to learn by practise |
| 09:57.16 | mandh | and see real problems |
| 09:57.23 | DigitalIrony | yes |
| 09:57.46 | DigitalIrony | i have learned alot here doing support |
| 09:57.52 | DigitalIrony | i learn something new everday |
| 09:57.53 | mandh | now for exmaple fanee give me an hint to use macros and like that |
| 09:58.15 | mandh | and i found i can solve the transfered prolem by using like that GotoIf($[$["${ENUM:0:3}" = "SIP"] |
| 09:59.00 | DigitalIrony | that could work |
| 09:59.17 | mandh | what u think about performance |
| 09:59.35 | mandh | can asterisk will work for large call center with 100 agents? |
| 09:59.40 | DigitalIrony | but also becareful because if some one were to register as a sip user remotely they could use that too |
| 09:59.54 | DigitalIrony | mandh: i have customers he have thousands of callers |
| 10:00.09 | mandh | with only one server |
| 10:00.15 | DigitalIrony | mandh: oh no |
| 10:00.21 | DigitalIrony | mandh: many servers |
| 10:00.29 | mandh | fine |
| 10:00.46 | mandh | and what u think about one server |
| 10:00.54 | mandh | with not lot of calls |
| 10:01.02 | DigitalIrony | mandh: but a powerful enough server could probably handle 100 agents, with enough bandwidth and the right codecs |
| 10:01.10 | mandh | oh that is very very fine |
| 10:01.53 | DigitalIrony | mandh: i don't know though....you would probably want to use a TC400 card |
| 10:02.02 | DigitalIrony | and have multiple CPU's |
| 10:02.19 | DigitalIrony | would be better to have a few servers....like a front end back end setup |
| 10:02.58 | DigitalIrony | but if you have 100 agents, you wouldn't want to not have many calls |
| 10:03.04 | DigitalIrony | you would loose money paying agents |
| 10:03.26 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 10:03.38 | DigitalIrony | mandh: are you talking 100 agents all at once? |
| 10:04.44 | DigitalIrony | well its time for me to go now |
| 10:04.47 | DigitalIrony | good luck |
| 10:04.53 | DigitalIrony | good night/morning |
| 10:06.42 | mandh | DigitalIrony, sure for not coz the server only have 12 chanlles (zap) but i ask about that for knwoldge |
| 10:07.12 | mandh | DigitalIrony, good night /morn |
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| 10:52.05 | Delvar | yay now me 2 |
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| 10:58.45 | TallAndy | Using call files you can tell an agi script to be run... but can you set variables also like with Originate? |
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| 11:42.56 | seanbright | TallAndy: yup |
| 11:43.11 | seanbright | TallAndy: see sample.call in the source tarball for an example |
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| 12:02.00 | dieno | [TK]D-Fender Hello :D |
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| 12:24.20 | Weezey | anyone besides fring build a SIP client for the iPhone? |
| 12:29.12 | TallAndy | nice one thanks seanbright |
| 12:29.18 | [TK]D-Fender | Weezey: http://www.google.ca/search?hl=en&q=iphone+SIP+client&btnG=Google+Search&meta= , http://swik.net/iphone+sip |
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| 12:43.08 | Schreiber1337 | Has anyone had problems with callers hiting 0 to exit voicemail? |
| 12:43.12 | dieno | and for SIP Client for Blackberry Curve ? |
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| 12:43.37 | lmadsen | Schreiber1337: nada |
| 12:43.42 | ManxPower | Schreiber1337: only if I didn't set it up correctly. |
| 12:44.59 | Schreiber1337 | ManxPower: It's been working fine for 8 months and then this morning, I get "sent into invalid extension '0'" |
| 12:45.28 | lmadsen | soemthing changed then |
| 12:45.54 | ManxPower | Schreiber1337: odd, since it jumps to exten => o "oh, not zero" |
| 12:46.29 | Schreiber1337 | Hmm... let me look at that. |
| 12:48.27 | Schreiber1337 | ManxPower: This is how I have it setup... http://www.spectrumcontrol.com/zero/pastbin/1.txt |
| 12:48.57 | Schreiber1337 | ManxPower: o and 0 should go to the same place then, I think... |
| 12:49.16 | ManxPower | Schreiber1337: "o" has nothing to do with "0" in the dialplan |
| 12:49.33 | ManxPower | Schreiber1337: chances are you have a context issue |
| 12:50.08 | ManxPower | Schreiber1337: PB the CLI output of the failed call |
| 12:50.58 | Schreiber1337 | ManxPower: http://www.spectrumcontrol.com/zero/pastbin/2.txt |
| 12:51.15 | ManxPower | Schreiber1337: BTW, n+101 does not work in 1.4 |
| 12:51.33 | ManxPower | that appears to be an empty paste |
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| 12:52.27 | Schreiber1337 | ManxPower: Sorry.. try it again. |
| 12:53.03 | ManxPower | Schreiber1337: count your "m"s |
| 12:54.19 | [TK]D-Fender | YUP,.... 1337 |
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| 12:59.18 | Schreiber1337 | ManxPower: Yup.. that was it. Thanks... what replaced n |
| 12:59.40 | Schreiber1337 | ManxPower: what replaced n+101 |
| 12:59.58 | lmadsen | status variables like DIALSTATUS |
| 13:00.26 | lmadsen | see examples at www.asteriskdocs.org |
| 13:00.30 | ManxPower | Schreiber1337: see /path/to/asterisk/src/doc |
| 13:00.39 | ManxPower | ..er. |
| 13:00.52 | ManxPower | Schreiber1337: see /path/to/src/asterisk/doc/upgradt.txt |
| 13:00.57 | lmadsen | lol |
| 13:01.06 | lmadsen | ManxPower: its gonna be that kind of day huh? |
| 13:01.12 | ManxPower | there should be two upgrade.txt files in recent 1.4 releases |
| 13:01.21 | ManxPower | lmadsen: Every day is that kind of day. |
| 13:01.36 | Schreiber1337 | lol... thanks everyone.. |
| 13:01.47 | ManxPower | idly notes he is looking for a position teaching Asterisk. |
| 13:02.12 | lmadsen | well today is worse for me than usual since i injured the tip if my right ring finger so now i'm one finger typing with my right hand... that'll slow ya down... |
| 13:02.40 | ManxPower | lmadsen: I almost had a knuckle ripped off yesterday by a falling beam |
| 13:02.53 | ManxPower | lmadsen: what, you are not an expert at one handed dyping? |
| 13:02.57 | ManxPower | typing, even. |
| 13:03.27 | lmadsen | heh no... I use all my fingers and thumbs when I type, and have done that since I was a kid |
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| 13:35.27 | dieno | can any on etell me what is the best boook to learn dialplans and etc |
| 13:37.44 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 13:38.49 | _ShrikE | ~book |
| 13:38.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:39.07 | dieno | :D thnx |
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| 13:45.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:45.18 | raytruz` | 1.0 G force is just the speed the earth pulls us to it right? |
| 13:45.56 | pelaofeliz | raytruz`: yep, I think that's right |
| 13:46.13 | raytruz` | k |
| 13:46.15 | raytruz` | thanks |
| 13:51.16 | x86 | ok this is very very annoying... I've got a Toshiba phone system with trunk ports going to a Rhino channel bank, and then over T1 into an Asterisk phone system, so that the two systems can call each other |
| 13:51.44 | x86 | when I make a call from Toshiba --> channel bank --> Asterisk, I randomly get a beeping sound almost like call waiting |
| 13:52.12 | x86 | there is no call waiting on the trunk ports, any idea what this could be? |
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| 13:55.20 | x86 | ManxPower: I checked the timing, channel bank is setup as slave, and asterisk is setup as master |
| 13:55.44 | x86 | ManxPower: only difference that I can see as far as T1 configuration is that the channel bank is setup for loopstart, and asterisk is using kewlstart |
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| 14:04.01 | rwaite | hey, i'm having a problem with some polycom 330 sip phones are asterisk ... i have two lines on this phone, 231 and 233. if i leave a vmail in 233's box, asterisk send a notify to both extensions for the vmail ... how can i limit it to only send a notify to 233 if the vmail is in 233's box? |
| 14:04.47 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 14:04.54 | rwaite | i can see the packets with sip debug on - i originally thought it was a problem with the polycom config, but it looks like i'm doing something wrong. |
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| 14:07.44 | rwaite | :( |
| 14:13.23 | rwaite | please somebody help, this issue is really hard for me |
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| 14:16.56 | peterpen | Hey, I know you _can't_ have more than one phone connect as the same extension and expect them all to ring (and the first to pick up - WINS! woo) |
| 14:17.21 | peterpen | However might such a thing be possible with some kind of proxy - and how easy is that? |
| 14:17.56 | peterpen | At the moment I'm doing extensions in a ring group and a 60 failover to voicemail - its messy and poo |
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| 14:21.01 | mikealeonetti | in a situation where a company needs about 20-50 phone lines, what kind of carrier and hardware is necessary for that? |
| 14:22.31 | x86 | peterpen: setup a queue, make all of your users agents of that queue |
| 14:23.07 | peterpen | x86, no I want the ability to have an extension be registered twice |
| 14:23.11 | peterpen | or more |
| 14:23.25 | [TK]D-Fender | peterpen: You can't |
| 14:23.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:23.41 | peterpen | ok so I get a proxy to register then, and get the phones registered to the proxy? |
| 14:23.49 | [TK]D-Fender | peterpen: You need to make multiple ACCOUNTS in sip.conf and have your extension dial ALL of them together. |
| 14:24.10 | peterpen | yeah I already do that now, it works but I hate it |
| 14:24.17 | [TK]D-Fender | peterpen: Dial(SIP/100a&SIP/100b&SIP/100c) |
| 14:24.26 | [TK]D-Fender | peterpen: TFB <- thats the way it works. |
| 14:24.38 | peterpen | <peterpen> yeah I already do that now, it works but I hate it |
| 14:24.51 | x86 | 09:22 < x86> peterpen: setup a queue, make all of your users agents of that queue |
| 14:24.55 | [TK]D-Fender | peterpen: Feel free to hate it... you just aren't going to change it. |
| 14:24.58 | x86 | those are your two options |
| 14:25.01 | peterpen | can I get a proxy to register instead, and get the phones registered to the proxy? |
| 14:25.14 | x86 | why? |
| 14:25.31 | peterpen | not why, usually its 'yes', 'no', 'don't know' |
| 14:25.37 | x86 | [TK]D-Fender: any ideas on my "call waiting" tone randomly on my trunk ports between asterisk and my toshiba PBX? |
| 14:25.47 | ManxPower | x86: turn it off |
| 14:25.50 | [TK]D-Fender | x86: nope. |
| 14:25.59 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 14:26.00 | x86 | ManxPower: no call waiting enabled |
| 14:26.15 | [TK]D-Fender | peterpen: Go find out in your proxy's channel if it can fake this out. These proxy's are not supported here. |
| 14:26.44 | peterpen | I don't know of any proxies! that was just an idea |
| 14:26.48 | [TK]D-Fender | peterpen: But * cannot do this. If the proxy can fake everything, then fine. |
| 14:27.12 | artz22 | hi eveybody..here I am with latency problems again... even with phones working in ulaw. echo test does alright, little latency, but is there a reason to explain latency about 1 sec ? |
| 14:28.05 | [TK]D-Fender | peterpen: So since you don't even know of any proxies, even the attempt seems out of scale for you to attempt. |
| 14:28.17 | x86 | ManxPower: would kewlstart on one end and loopstart on the other be causing the problem? |
| 14:28.18 | [TK]D-Fender | peterpen: You've been given the approach you should follow. |
| 14:28.22 | mwalling | whats wrong with doing it like everyone else? |
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| 14:32.51 | x86 | ManxPower: also, on my PSTN T1, users are reporting "static" and occaisional dropped calls... any ideas on that one? |
| 14:33.04 | rwaite | yay i figured it out |
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| 14:42.56 | lmadsen | lol... wish rwaite had told us what he did to fix it |
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| 15:01.21 | muiro | Just a quick question here. Just asking advice from people woh are in the business. I'm trying to make sense of some CDR data, mostly I'm trying to look at trunk usage (to look for overlap, etc). Can anyone suggest a methodology for doing this? |
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| 15:08.38 | Schreiber1337 | Can someone help me with syntax? |
| 15:08.51 | Schreiber1337 | What Chr do you use to Concatenate a stringÂ… Dial(SIP/?${MACRO_EXTEN}) |
| 15:08.52 | Schreiber1337 | What would the ? beÂ… |
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| 15:09.10 | lmadsen | Schreiber1337: the string you want to concatonate |
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| 15:09.21 | lmadsen | Dial(SIP/foobar${MACRO_EXTEN}) |
| 15:09.49 | Schreiber1337 | lmadsen: foobar? |
| 15:10.07 | lmadsen | foobar is just a random string I made up |
| 15:10.37 | lmadsen | you concatenate just by putting the strings beside each other |
| 15:10.52 | lmadsen | its not like other programming languages where you need a special char or anything |
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| 15:11.58 | Schreiber1337 | Lmadsen: so Dial(SIP/${MACRO_EXTEN}) in a macro would dial the extension that called the macro? |
| 15:12.09 | lmadsen | yes |
| 15:12.21 | Schreiber1337 | lmadsen: Thank you... |
| 15:12.27 | lmadsen | ${MACRO_EXTEN} would contain the extension that called the macro |
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| 15:14.32 | ManxPower | I normally do, as the first line of my macros that contain a Dial, an exten => s,1,Goto(${MACRO_EXTEN},1) then start the rest of my macro with exten => _X.,1,Whatever. |
| 15:14.39 | ManxPower | that way the CDRs will be correct. |
| 15:15.48 | *** join/#asterisk hoonteke (n=kevin@cpe-069-134-205-184.nc.res.rr.com) |
| 15:20.57 | hoonteke | I'm learning asterisk: I have an asterisk user (?) named 'hoonteke' that goes to my soft phone. I also have a mailbox named 'hoonteke'. How do I access my mailbox without calling myself? (I have the messages as per email, but would be nice to perform other administrative tasks, etc.) |
| 15:21.39 | *** join/#asterisk comprookie2000 (n=david@gentoo/contributor/comprookie2000) |
| 15:22.25 | pputman- | hoonteke, read about the voicemailmain application |
| 15:22.31 | *** part/#asterisk tristanbob_ (n=tristanr@ubuntu/member/tristanbob) |
| 15:23.47 | hoonteke | pputman-: that's the documentation I think I'm reading right now, but the problem is that the documentation seems to have numbered mailboxes (that are *not* the same as usernames), and a month ago I thought I'd be cute and make my mailbox the same as my username. Is that in there and I just glossed over it? |
| 15:24.00 | ManxPower | hoonteke: it is usually a bad idea to use alpha mailboxes |
| 15:24.20 | justdave | hmm, does the 1.4.21 release have settings you need to tweak for IAX to keep working? (changed defaults or anything) |
| 15:24.37 | hoonteke | ;-) yep. I'm with ya now on that, ManxPower. The question is how do I fix that? |
| 15:24.42 | justdave | my box seems to stop listening for incoming IAX after upgrading |
| 15:24.47 | ManxPower | justdave: no. Any fixes would be in the code, not in a config ilfe. |
| 15:25.00 | ManxPower | justdave: is this a production box? |
| 15:25.11 | justdave | that's what I thought, didn't see anything mentioning iax config in the changelog |
| 15:25.15 | justdave | yeah, it is. |
| 15:25.26 | justdave | I rolled back to the previous version to get it working |
| 15:25.37 | ManxPower | justdave: report on bugs.digium.com and revert to your old version |
| 15:26.44 | hoonteke | ManxPower: so, numbered mailbox from now on. Do I have to do anything special to clear out my 'hoonteke' mailbox? Can I just rename the mailbox, remove /var/spool/asterisk/voicemail/default/hoonteke/ and reload? |
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| 15:27.09 | ManxPower | you can just remove the directory and do a reload, restart or "reload app_voicemail.so" |
| 15:27.16 | hoonteke | great, thanks. |
| 15:27.37 | ManxPower | Before you do any of that, remove the mailbox from /etc/asterisk/voicemail.conf |
| 15:27.41 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:28.19 | x86 | ManxPower: ok, I was looking at the channel bank when the user experienced the "call waiting" tone, and it said "<-- Signal" and under each channel it had an L |
| 15:28.30 | hoonteke | ManxPower: right there with ya. Thanks! |
| 15:28.57 | x86 | ManxPower: does that mean loss of signal? the T1 never dropped... also, it's a Rhino channel bank, and the Comm light flashed just for a quick second when the message displayed |
| 15:29.02 | justdave | I upgraded the remote office that wasn't connecting first before I downgraded the main office |
| 15:29.20 | justdave | and that remote office is now running the version the main office was when it wouldn't accept iax connections |
| 15:29.38 | justdave | and I can still connect going back the other way |
| 15:29.45 | justdave | which is what was making me wonder if it's config. |
| 15:30.01 | justdave | but they are different OS versions, too (that remote office is on rhel4, the main office is on rhel5) |
| 15:30.10 | x86 | ManxPower: I power cycled the channel bank hoping that would fix it... not sure if it fixed it or not yet... |
| 15:30.55 | ManxPower | x86: I can't imagine what might be wrong unless your description of "call waiting sound" is not accurate. |
| 15:33.14 | *** join/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 15:38.19 | Schreiber1337 | Could someone look at my 1.2 -> 1.4 dialplan conversion and see if it looks correct? |
| 15:38.21 | Schreiber1337 | http://www.spectrumcontrol.com/zero/pastbin/0.txt |
| 15:38.38 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 15:39.08 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 15:39.22 | kamanashisroy | hi, how can I flush my sip phone display ? I am using sendtext to send data .. is there anything like flush in dtmf ? |
| 15:39.30 | *** join/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 15:44.45 | *** part/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 15:49.28 | ecrist | can someone tell me, for polycom phones, does the following dialplan make sense: 911|91[2-9]xx[2-9]xxxxxx|9[2-9]xx[2-9]xxxxxx|[8]xx|[7]xxx|9011xxx.T |
| 15:50.19 | fskrotzki | ~book |
| 15:50.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:54.10 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:00.44 | [TK]D-Fender | ecrist: I highly recommend using a "math all" Polycom plan, and letting * do all the deciding. I don't like the phone being too smart for its own good. I also dislike dial prefixes like "9". Its sooooo '80s |
| 16:01.06 | [TK]D-Fender | kamanashisroy: Send it MORE, and that "more" being blank |
| 16:01.31 | ecrist | [TK]D-Fender: example? |
| 16:01.57 | [TK]D-Fender | ecrist: *.T|x.T|*.T |
| 16:02.01 | [TK]D-Fender | ecrist: *.T|x.T|#.T |
| 16:02.13 | kamanashisroy | [TK]D-Fender: thank you for your answer .. I am trying with \r\n .. anyway I shall be able to test things again after 3 hours :-P |
| 16:02.21 | ManxPower | eric2: You'll have to press Dial or wait for the timeout on the phone before the call will be sent. A better way is to design a dialplan that does not rely on timeouts. |
| 16:03.29 | ecrist | [TK]D-Fender: very cool. I'm fighting with our VoIP provider right now over 10-digit dialing. I want both 91+areacode+number and 9+areacode+number to work. |
| 16:03.38 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 16:03.38 | ecrist | is that difficult to do? |
| 16:04.11 | ManxPower | ecrist: Dude, remove the 9 in the dialplan before sending to provider |
| 16:04.12 | [TK]D-Fender | ecrist: Your users should have to care about dialing... what you send to your provider is up to YOU in your dialplan. |
| 16:04.13 | ecrist | speak of the devil. |
| 16:04.34 | ecrist | ManxPower: no. |
| 16:04.37 | [TK]D-Fender | ecrist: They could dial 123 and you'll spit 947 to the PSTN just because you felt like it. Your users don't have to care. |
| 16:04.51 | [TK]D-Fender | ecrist: You need to be smart with your dialplan. |
| 16:05.08 | [TK]D-Fender | shouldn't* |
| 16:05.20 | ecrist | [TK]D-Fender: I understand, but you didn't answer my quesiton. |
| 16:05.26 | ManxPower | ecrist: Best of luck. EVERYONE else removes the 9 in the dialplan, and they also add a leading 1 if required, all in extensions.conf |
| 16:05.56 | [TK]D-Fender | ecrist: Your provider always wants 1 + 10? |
| 16:06.29 | ecrist | [TK]D-Fender: they're willing to change it. they claim it's too difficult to allow *both* 1 + 10 and just 10. |
| 16:06.40 | Qwell | so then don't send both |
| 16:06.54 | [TK]D-Fender | ecrist: its even BETTER that they force 11 digit dialing. Go for it. |
| 16:07.22 | [TK]D-Fender | ecrist: You can let YOUR users do whatever they want. just have a match for both and tweak the # before sending it out. |
| 16:07.35 | ecrist | my point is that in Minneapolis, we have 4 local area codes. with that comes 10 digit dialing. we shouldn't have to dial a 1 before local number. |
| 16:08.02 | Qwell | yes you should |
| 16:08.10 | ecrist | Qwell: no, we shouldn't. |
| 16:08.19 | Qwell | if you're dialing the area code, you must dial 1 |
| 16:08.31 | [TK]D-Fender | ecrist: DOESN'T MATTER that your providers demands it... this isn't something you have to enforce upon your USERS. |
| 16:08.44 | ecrist | [TK]D-Fender: we don't control the asterisk server, so it would have to be done on the phones. |
| 16:08.54 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 16:08.57 | Qwell | so then why are you in here asking for help with Asterisk? O.o |
| 16:08.59 | [TK]D-Fender | ecrist: Well.... ICK |
| 16:09.01 | outtolunc | what you accept from your users and what you send to your provider are 2 (or more <G>) different things |
| 16:09.26 | ecrist | [TK]D-Fender: great, is it something I can do on the phone? If not, that's fine. |
| 16:09.30 | [TK]D-Fender | ecrist: Fine, go read SIP 2.2.2+'s admin guide on dial substitution. You CAN do this direct on the phone if you want to. |
| 16:09.56 | ecrist | Qwell: if you read, I was in here asking about a dial plan for asterisk and whether something was possible. |
| 16:10.05 | [TK]D-Fender | ecrist: And next time make it clear that AREN'T controlling * yourself. This wasted a lot of time. |
| 16:11.33 | ManxPower | What kind of jerk comes to #asterisk and can't actually configure anything with Asterisk? |
| 16:11.56 | russellb | because they want to learn, and figure out where to start |
| 16:12.02 | ecrist | [TK]D-Fender: our level of control varies. We can request changes to almost anything, but there are limits. I'm apparently pushing those limits. |
| 16:12.12 | ManxPower | russellb: This isn't #voip-tutorial |
| 16:12.15 | Qwell | russellb: he isn't using (or planning on using) Asterisk... |
| 16:12.21 | russellb | oic. |
| 16:12.24 | [TK]D-Fender | ecrist: They suck. Haven't considered getting your own *? |
| 16:12.37 | Qwell | using, in a typical sense. he's a user |
| 16:12.42 | russellb | doesn't read scrollback |
| 16:12.44 | outtolunc | well at least he now knows that if he puts an asterisk box between his users and his provider .. this is simple <G> |
| 16:12.55 | [TK]D-Fender | Qwell: if he isn't in control, he isn't using *. |
| 16:13.07 | Qwell | [TK]D-Fender: right, by user, I mean how you and I refer to users |
| 16:13.18 | ecrist | [TK]D-Fender: consider it every time I work with them. is it something that's feasible and worthwhile for ~15 users? |
| 16:13.23 | Qwell | ie; "The users are complaining again!" |
| 16:13.36 | pputman- | user with an l |
| 16:13.37 | [TK]D-Fender | Qwell: just I say about trixbox users. They aren't using Asterisk. TRIXBOX is using Asterisk, and the users are using trixbox. |
| 16:13.52 | Qwell | [TK]D-Fender: in that specific case, I'd say it's using them |
| 16:14.04 | [TK]D-Fender | feels kinda dirty |
| 16:15.21 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 16:15.50 | ecrist | you know though, folks, if you guys were a bit more friendly, things would be a lot easier. I apologize that I'm not up on all the terms, I've admitted, many times, to being a new VoIP user. We've been trying to balance having both a self-hosted * server and having someone host it for us. |
| 16:16.38 | mchou | lol |
| 16:16.39 | *** join/#asterisk grulk (n=grulk@201.220.136.117) |
| 16:17.55 | *** join/#asterisk geek_cl (n=lletelie@200.75.18.211) |
| 16:18.34 | [TK]D-Fender | ecrist: Well advising you is really hard when we know that all the rules of the product this channel is for don't apply to you. |
| 16:19.16 | [TK]D-Fender | ecrist: Either way, you've got an option for where you are now. |
| 16:19.54 | ecrist | [TK]D-Fender: all/never/always are strong words. I'm mostly here because 1) nobody's in #voip, 2) my provider uses * and I'm trying to get some insight as to how their system operates and what their capabilities are |
| 16:20.03 | ecrist | [TK]D-Fender: very true. |
| 16:22.12 | grulk | Hi guys. I am trying to build a system for a small office, about 20 people (each one with a phone). I have 4 analog lines (FX*). What I want to do is to put does 4 lines into a digium card. I was thinking of the digium TDM400P(Any recommendations??) and connect a ip phone for the 20 employees. I was looking at polycom webpage but I see very fancy models that have many thing that wont be necesary for me. Does anyone know of a polycom sip phone or any othe |
| 16:22.12 | grulk | r branch, that allow me to redirect calls, make and received, etc.. all the basic stuff. |
| 16:22.26 | [TK]D-Fender | ecrist: So yeah, your provider could be smarter, but they clearly don't want to be. You could take back control yourself if you were so inclined. |
| 16:22.29 | grulk | and not get to complicated or fancy on the way |
| 16:22.38 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:22.50 | *** join/#asterisk doolph (n=doolph@200.75.196.191) |
| 16:22.54 | [TK]D-Fender | grulk: Where are you located? |
| 16:23.04 | grulk | Central America |
| 16:23.11 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 16:23.15 | chandoo | hi |
| 16:23.23 | chandoo | need help in configuring asterisk |
| 16:23.34 | doolph | help, I am using opvxa1200 module, but I don't have zap support with asterisk |
| 16:23.34 | chandoo | what i have to put in service providers |
| 16:23.49 | grulk | [TK]D-Fender, but I can get the phones and the card from anywhere. That is not a problem. |
| 16:23.50 | chandoo | i use phone power voip service |
| 16:23.51 | [TK]D-Fender | grulk: ok, Polycom might still be a great value for you then. You should probably be looking at the IP 330 running off a PoE Switch : www.telephonydepot.com |
| 16:23.58 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 16:24.41 | ManxPower | doolph: there are OpenVox drivers for Asterisk, I believe. You'll have to talk to them as almost nobody here uses their hardware |
| 16:24.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:25.06 | ManxPower | chandoo: Are you using FreePBX, Trixbox, AsteriskNow or AsteriskGUI? |
| 16:25.18 | chandoo | asterisk gui |
| 16:25.31 | ManxPower | chandoo: then you should be asking this on #asterisk-gui |
| 16:25.40 | ManxPower | just like it says in the /topic if this channel |
| 16:25.47 | chandoo | oh is there such channel |
| 16:25.59 | pputman- | nah, april fools |
| 16:26.06 | pputman- | err, june |
| 16:26.25 | ManxPower | chandoo: you would have to remove all your config files, scripts, and settings for us to be able to help you and you might as well not be using a GUI then. |
| 16:26.29 | [TK]D-Fender | last I heard OpenVox cards were 100% clones of Digium cards. Should take any extra special drivers. |
| 16:26.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:26.50 | doolph | ManxPower the driver is loaded |
| 16:27.02 | doolph | zaptel 198840 5 opvxa1200 |
| 16:27.04 | ManxPower | doolph: great. Now contact Openvox |
| 16:27.17 | ManxPower | I thought you said you did not have zaptel. |
| 16:27.31 | doolph | well I don't have the zap command at asterisk cmd |
| 16:27.48 | ManxPower | doolph: then zaptel was not installed when you installed Asterisk. |
| 16:27.57 | [TK]D-Fender | doolph: Verify "ztcfg -vvvv" then load chan_zap manually |
| 16:28.01 | nny_1 | going to be making a web front end to do some VERY basic tasks (change which dial out macro each extension uses), changing variables for CELL1 CELL2 etc, uploading audio (that's pretty much done) and manipulating the MOH sources. If anyone has any knowledge of other projects related to this let me know. Please note: I am not trying to make asterisk-gui/trixbox etc |
| 16:28.18 | doolph | how to load chan_zap manually? |
| 16:28.27 | ManxPower | doolph: load chan_zap.so in the CLI. |
| 16:28.27 | nny_1 | fwiw i am probably going to be using a php frontend that manipulates entries in asteriskdb |
| 16:28.30 | doolph | 16 channels to configure. |
| 16:29.10 | doolph | well |
| 16:29.14 | doolph | it said |
| 16:29.22 | doolph | its loaded |
| 16:29.23 | doolph | but its not |
| 16:29.34 | ManxPower | doolph: prove it. |
| 16:29.42 | doolph | <PROTECTED> |
| 16:29.43 | ManxPower | "core show modules" will show you what is loaded. |
| 16:29.48 | chandoo | ManxPower: i am brand new in with asterisk, some one suggested gui, so i am using it, This is the first time i am doing configuration so i wont be having any config files |
| 16:29.55 | ManxPower | doolph: We can't help you if you are using a GUI. Y |
| 16:30.02 | doolph | I am not using a gui |
| 16:30.09 | ManxPower | chandoo: all GUIS come with default config files. |
| 16:30.25 | [TK]D-Fender | doolph: "zap show channels", and PASTEBIN your configs. |
| 16:30.26 | chandoo | ManxPower: let me delete then |
| 16:30.30 | ManxPower | doolph: merge all your zapata*.conf files |
| 16:30.43 | ManxPower | chandoo: perhaps you misunderstood. You should GO TO #ASTERISK-GUI. |
| 16:30.48 | doolph | elastix*CLI> zap show channels |
| 16:30.48 | doolph | No such command 'zap show' (type 'help' for help) |
| 16:30.54 | doolph | dang |
| 16:31.29 | ManxPower | doolph: 1) merge all your zapata*.conf files, then put the resulting zapata.conf on PB so we can see it. |
| 16:31.36 | doolph | ok I'll try |
| 16:31.58 | ManxPower | doolph: you must be using 1.2. |
| 16:31.59 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:32.11 | doolph | 1.4.19 |
| 16:32.14 | ManxPower | because in 1.4 the command is not "zap show channels" |
| 16:32.16 | chandoo | ManxPower: i am ready to use command line |
| 16:32.28 | doolph | what it is |
| 16:32.30 | rwaite | is there a way to have asterisk send a 'sip notify' to multiple phones if they are all set to subscribe to the same mwi |
| 16:32.31 | ManxPower | chandoo: then you should reinstall Asterisk from source, then read the Book. |
| 16:32.33 | ManxPower | ~book |
| 16:32.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:32.49 | chandoo | ManxPower: i installed from source |
| 16:32.53 | rwaite | right now its only sending the notify to the most recent subscriber |
| 16:32.55 | chandoo | compiled it |
| 16:32.57 | ManxPower | chandoo: then go read the book |
| 16:33.12 | doolph | what should be the command |
| 16:33.19 | chandoo | ManxPower: what book you recommend |
| 16:33.25 | pputman- | wow |
| 16:33.28 | ManxPower | rwaite: mailbox=themailbox in each sip peer |
| 16:33.36 | ManxPower | chandoo: I recommend Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:34.02 | chandoo | thanks ManxPower |
| 16:34.08 | ManxPower | doolph: maybe you should go into /path/to/src/asterisk/doc and read the two upgrade files, which tell you all the changes for the CLI and everything else. |
| 16:34.08 | rwaite | ManxPower, but they also have their own mailbox |
| 16:34.23 | [TK]D-Fender | doolph: pastebin the entire output of "reload chan_zap.so" and pastebin your configs. |
| 16:34.27 | ManxPower | rwaite: mailbox=mymailbox@mymailboxcontext,theothermailbox@theothercontext |
| 16:34.31 | [TK]D-Fender | rwaite: you have have multiple entries |
| 16:34.41 | [TK]D-Fender | ManxPower: 1 line each, IIRC |
| 16:34.57 | rwaite | the problem with that, i have one mailbox i want everyone to be able to have as line 2 on their sets |
| 16:35.02 | ManxPower | [TK]D-Fender: maybe that is new in 1.4 |
| 16:35.20 | rwaite | if i put multiple mailbox entries, then both lines will show as having a message even if only one of them does |
| 16:35.23 | ManxPower | rwaite: not going to happen if you have all the lines in each phone register as the same user |
| 16:36.00 | rwaite | what's the best way to set up a common mailbox then, that when they delete the message, all the other phones will stop displaying the mwi |
| 16:36.22 | ManxPower | exactly the way I just told you |
| 16:36.36 | *** join/#asterisk _thomas__ (i=bd18b2c4@gateway/web/ajax/mibbit.com/x-c82fb3be51df18de) |
| 16:36.41 | *** part/#asterisk _thomas__ (i=bd18b2c4@gateway/web/ajax/mibbit.com/x-c82fb3be51df18de) |
| 16:36.56 | grulk | [TK]D-Fender, My 4 lines uses a RJ-11 connector so I guess they are analog FX* lines(correct me if I am wrong). I will buy the TDM400P. The sip phones will be IP 330, but what I cannot fine anywhere is the PoE switch you talk about. Does a normal PoE switch will make the job or do I need a special one for the polycom phones? |
| 16:37.12 | grulk | sorry TDM800P |
| 16:37.26 | rwaite | ManxPower: if i do that, which i did, then on the phone if you leave a message in one box, asterisk will send a sip notify for -both- extensions. so the phone says there is vmail waiting in both, when really its only in one. |
| 16:37.54 | [TK]D-Fender | grulk: Lookup the D-Link DES-1228P |
| 16:38.05 | ManxPower | rwaite: NOT. IF. YOU. DEDICATE. A. LINE APPEARANCE. TO. THE. OTHER. MAILBOX. |
| 16:38.19 | rwaite | i did that!!! |
| 16:38.24 | grulk | thanks |
| 16:38.30 | rwaite | 233 is the general line, and 231 is the private one |
| 16:38.49 | rwaite | asterisk sends a sip notify to both 231 AND 233 if i leave a vmail in 233 |
| 16:38.51 | ManxPower | Ok, so in the sip.conf [231] section put mailbox=theothermailbox |
| 16:39.08 | rwaite | this is with mailbox=231@dma,233@dma in the sip.conf |
| 16:39.13 | ManxPower | in [233] mailbox=therealmailbox |
| 16:39.29 | ManxPower | rwaite: no, in this situation you only put one mailbox per peer. |
| 16:39.50 | rwaite | but i want two per peer? the 231 as well as the 233 |
| 16:39.57 | ManxPower | the put two mailboxes on one mailbox line was when you had not bothered to tell us enough detauils. |
| 16:40.00 | rwaite | if this is not possible, please let me know |
| 16:40.07 | ManxPower | you want two peers, that is what I said. |
| 16:40.45 | rwaite | so you're saying to make a sip entry for, lets say, 235 that has mailbox=231@dma,233@dma |
| 16:40.46 | ManxPower | [233] mailbox=233@dma and [231] mailbox=231@dma |
| 16:40.51 | grulk | [TK]D-Fender, I read about the switch. So it will end with a TDM800P (for the RJ-11 lines), 20 Polycoms IP 330, and a D-Link DES-1228P for connecting them. That Is all I will need beside asterisk |
| 16:41.02 | rwaite | ManxPower: that is exactly how my configuration is. |
| 16:41.11 | [TK]D-Fender | grulk: Thats all. |
| 16:41.30 | grulk | [TK]D-Fender, thanks, really apreciate your help |
| 16:41.31 | ManxPower | rwaite: Best of luck with this. I have to get back to paying work. |
| 16:41.38 | doolph | wow |
| 16:41.44 | nny_1 | is the ip330 web interface any more useful than the 601 series? |
| 16:41.45 | doolph | I still cannot find the error |
| 16:41.47 | rwaite | ugh, you keep telling me to do what i'm telling you i already tried |
| 16:42.07 | doolph | this is really werid |
| 16:42.37 | ManxPower | doolph: best of luck. Too bad we didn't get that pastebin of zapata.conf before I had to leave. |
| 16:42.59 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 16:44.20 | *** join/#asterisk devilbues (n=psousa@213.63.47.23) |
| 16:44.31 | doolph | http://pastebin.ca/1051199 my zapata.conf |
| 16:44.53 | *** join/#asterisk bmg505 (n=leon@196-209-76-21-tbnb-esr-2.dynamic.isadsl.co.za) |
| 16:45.06 | devilbues | hi can anybody tell me how do I build a rpm from asterisk source? |
| 16:45.09 | doolph | http://pastebin.ca/1051200 my zapata-channels.conf |
| 16:45.34 | [TK]D-Fender | devilbues: #redhat |
| 16:46.36 | [TK]D-Fender | doolph: And you still did not provide the CLI output I asked for. |
| 16:46.56 | doolph | you mean load chan_zap.so? |
| 16:47.04 | rwaite | figured it out. make a 233a, 233b, and 233c, all with mailbox=233@context |
| 16:47.27 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 16:47.47 | doolph | http://pastebin.ca/1051203 load chan_zap.so |
| 16:48.13 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 16:48.43 | doolph | ummm |
| 16:48.56 | doolph | at my /var/log/asterisk/full says [Jun 19 11:47:01] WARNING[2978] chan_zap.c: Unable to specify channel 1: Device or resource busy |
| 16:49.10 | doolph | thats the error!! |
| 16:49.21 | [TK]D-Fender | doolph: and I told you to do "ztcfg -vvvv" FIRST. You should then restart * afterwards |
| 16:49.53 | doolph | same |
| 16:49.57 | doolph | I just did it |
| 16:50.41 | [TK]D-Fender | doolph: Ok, you're going to seriously wake up and PASTEBIN EVERYTHING. |
| 16:50.43 | doolph | is says 16 channels to configure. |
| 16:51.40 | *** join/#asterisk NOT_guru (n=NOT_guru@24-241-103-142.static.stls.mo.charter.com) |
| 16:51.45 | doolph | what does this means |
| 16:51.46 | doolph | ERROR[3013] config.c: *********** YOU SHOULD REALLY READ THIS ERROR *********** |
| 16:52.09 | NOT_guru | I am guessing it means you should read the log before or after that line =D |
| 16:52.13 | NOT_guru | sorry |
| 16:52.30 | NOT_guru | ah this is compiling |
| 16:52.48 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:53.43 | doolph | I think its an error of the card or the computer |
| 16:53.51 | *** join/#asterisk Schreiber1337 (n=Schreibe@spectrumcontrol.com) |
| 16:54.23 | NOT_guru | anyone here working with cisco 79X0 endpoints? |
| 16:54.36 | NOT_guru | 7940 or 7960 |
| 16:54.51 | NOT_guru | trying to figure out if I can reprogram the softbuttons |
| 16:54.57 | *** join/#asterisk Schreiber1337 (n=Schreibe@spectrumcontrol.com) |
| 16:54.59 | NOT_guru | but my searches have proven fruitless |
| 16:55.10 | NOT_guru | might just be time to buy some aastra phones |
| 16:55.53 | NOT_guru | if this is not a topic for this channel please let me know to " move along, nothing to see here " |
| 16:55.56 | NOT_guru | =) |
| 16:55.59 | doolph | what could be the error for this message? Unable to open channel 1: Device or resource busy |
| 16:56.14 | NOT_guru | I would personally guess |
| 16:56.24 | NOT_guru | unable to open channel 1 |
| 16:57.24 | NOT_guru | as ifis this the first time you have used this card |
| 16:57.44 | NOT_guru | have you ensured the card is functioning |
| 16:57.57 | doolph | well it was working |
| 16:58.05 | NOT_guru | you may even just have the card in a slot thats sharing IRQ's |
| 16:58.21 | NOT_guru | ok so the card has been known to work |
| 16:58.23 | NOT_guru | thats good |
| 16:58.36 | doolph | 169: 3003242 IO-APIC-level opvxa1200 |
| 16:58.36 | doolph | 177: 2983217 IO-APIC-level opvxa1200 |
| 16:58.42 | NOT_guru | so your rebuilding what was a working system? |
| 16:58.42 | doolph | both are different irq |
| 16:58.48 | NOT_guru | good deal |
| 16:58.51 | smth | hi, how can I get only those 'Channel' infos which generated by my sending 'originagte' throught AMI? |
| 16:59.27 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 17:00.04 | doolph | [Jun 19 11:59:06] ERROR[4071] chan_zap.c: Signalling requested on channel 1 is FXO Kewlstart but line is in FXS Kewlstart signalling |
| 17:00.08 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 17:00.29 | Idle | I seem to be getting a lot of errors with zaptel and a PRI |
| 17:00.36 | Idle | its dropping a LOT of calls |
| 17:00.44 | Idle | I've tried national, and dms100 switching |
| 17:00.54 | Idle | dms100 works better, but I still get the exact same errors |
| 17:01.02 | Idle | [Jun 19 10:58:37] ERROR[3101]: chan_zap.c:8248 zt_pri_error: !! Got reject for frame 102, retransmitting frame 102 now, updating n_r! |
| 17:01.06 | Idle | always that |
| 17:01.07 | [TK]D-Fender | smth: When you do your originate, set a channel variable. Then when you dump your channels, look for it. |
| 17:05.35 | rwaite | man asterisk is cool |
| 17:06.22 | Idle | manpages do rule |
| 17:06.22 | rwaite | i agree |
| 17:06.22 | rwaite | in your guys (or girls?) opinion, should i upgrade from 1.2 to 1.4 |
| 17:06.38 | anonymouz666 | rwaite: do you need to? |
| 17:07.00 | coppice | I think everyone should upgrade to Vista |
| 17:07.04 | rwaite | i dont think so, and to be frank, i'd like to stay on 1.2 since that's what etch has packaged |
| 17:07.06 | Idle | PRI on asterisk seems to suck pretty hard |
| 17:07.21 | pputman- | Idle, what type of card do you have? |
| 17:07.23 | Strom | Idle: how so? |
| 17:07.26 | rwaite | but i dunno the benefits of 1.4 really, if it would be more stable, or work better |
| 17:07.29 | anonymouz666 | coppice: no, because it's pretty slow here. |
| 17:07.41 | Idle | pputman-: digium, PCIE, dual port |
| 17:08.17 | pputman- | Idle, if you contact digium tech support they provide free installation support to get it working. |
| 17:08.19 | [TK]D-Fender | Idle: no.. its just you. |
| 17:08.30 | Idle | pputman-: actually, thats not a bad idea |
| 17:08.36 | pputman- | usually not |
| 17:08.49 | [TK]D-Fender | Idle: And yes you should work that out with Digium support. PCI-E is new for them... thing may be going wrong that don't with PCI, etc. |
| 17:09.05 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 17:09.21 | Idle | [TK]D-Fender: I'm 99% sure that its something on the lines |
| 17:09.22 | Idle | :S |
| 17:09.34 | Idle | we have people talking, no issues, then some calls get rejected |
| 17:09.46 | Idle | so, q931 strikes again... yay nortel |
| 17:09.58 | [TK]D-Fender | Idle: If you're messing with your signalling,t hen somethings wrong... you should have 1 mode, period, and you should know for certainwhat it is when you provisioned your circuit frmo the telco |
| 17:11.31 | smth | [TK]D-Fender, I mean when I on the call. how can i get the channel info which belongs to this call session? |
| 17:12.00 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:12.01 | Idle | [TK]D-Fender: yea, I know |
| 17:12.07 | Idle | I had 'national' before |
| 17:12.11 | [TK]D-Fender | smth: Looks like english, but I can't understand a thing you're saying. Pleas try to rephrase that. |
| 17:12.12 | Idle | 90% of calls failed with national |
| 17:12.18 | Idle | 10% fail under dms100 |
| 17:12.21 | [TK]D-Fender | Idle: .... PASTEBIN is your friend. |
| 17:12.34 | [TK]D-Fender | Idle: and go ask your telco |
| 17:12.46 | Idle | yea, I know, I will |
| 17:13.00 | pputman- | make sure you're on zaptel 1.4.11 too |
| 17:13.11 | Idle | I am about to call digium first, BUT, is there a way to get my serial number without looking at it? its 3 floors down and in a closet... :S |
| 17:13.20 | pputman- | no |
| 17:15.36 | *** join/#asterisk s0lid (n=s0lid@124.106.140.114) |
| 17:15.42 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-24-227.dhcp.embarqhsd.net) |
| 17:16.38 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 17:19.13 | smth | [TK]D-Fender, after a call was originated on AMI, I hope to know which is the new Channel created by me according to the response or events. |
| 17:30.28 | [TK]D-Fender | smth: When you create the call file, set that unique variable, and then doa channel dump for it. |
| 17:31.03 | *** join/#asterisk FuriousGeorge (n=brian@ool-457f216e.dyn.optonline.net) |
| 17:31.08 | FuriousGeorge | hey all |
| 17:31.52 | FuriousGeorge | has anyone had hangup detection issues exacerbated with the newest version (1.4.21). I've always had them, now they are worse, and when i get them the CLI becomes unresponsive now |
| 17:32.42 | FuriousGeorge | ive always had to restart asterisk daily, but since 1.X improvement in reliability seems to have peaked between 1.2 and 1.4 and very much plateaued, for me |
| 17:33.33 | jets | hrm i had several problems deploying 1.4.21 this week, i haven't had a chance to debug the problem |
| 17:33.34 | ManxPower | FuriousGeorge: you should never have to restart Asterisk daily |
| 17:33.46 | jets | but after a phone call the process hung and was unresponsive |
| 17:33.48 | *** join/#asterisk Serge8888 (n=Serge@76.66.214.130) |
| 17:33.55 | jets | so i threw it back to 1.4.18 the version it was at |
| 17:34.03 | ManxPower | and people wonder why I'm afraid to upgrade |
| 17:34.06 | FuriousGeorge | ManxPower: ive used asterisk for years, and i dont think ive had 30 days of uptime on any server that is at all moderately busy |
| 17:34.49 | ManxPower | FuriousGeorge: We get about 180 - 300 days of uptime |
| 17:35.02 | ManxPower | FuriousGeorge: Oh! Are you using analog? |
| 17:35.06 | FuriousGeorge | ManxPower: i should elaborate |
| 17:35.07 | FuriousGeorge | :) |
| 17:35.17 | FuriousGeorge | asterisk wont crash, but i need to restart it |
| 17:35.33 | jets | do you have a lot of hung channels and thats your need to restart? |
| 17:35.34 | FuriousGeorge | yeah, analog is the bane of my existence |
| 17:35.55 | Serge8888 | Hi All, pls help, I have stuck chanel (SIP), tried to send hangup command from manager api - did not help. is any way to kill the channel exept restart? thnks |
| 17:36.16 | FuriousGeorge | jets: when they become too many to soft hangup, yes, or when the cli becomes unresponsive, which is any time any channel gets hung now with 1.4.21 |
| 17:36.32 | FuriousGeorge | Serge8888: what version of asterisk? |
| 17:37.12 | smth | [TK]D-Fender, I got the way ,really appreciate |
| 17:37.17 | *** join/#asterisk [tasty]freeze (n=yamahabr@204-181-48-126.skybest.com) |
| 17:37.18 | Serge8888 | FuriousGeorge: 1.4.19.1 |
| 17:37.46 | FuriousGeorge | Serge8888: when soft hangup fails i usually restart |
| 17:38.17 | [tasty]freeze | Theres a bit of an echo using my spa3000 on my side, it seems to be real sensitive to all the noise on my phones mic; the other side does not notice this, is there a setting i need to tweak? |
| 17:38.31 | [tasty]freeze | This is only on a PSTN call |
| 17:38.34 | Serge8888 | cannot restart it is live system with calls |
| 17:39.23 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:39.52 | jets | hrm |
| 17:40.39 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:41.21 | trapa | I have a question about multiple sip lines. We have a bunch of inbound-only sip lines. I have made these work by having multiple register => commands in the [general] section of the sip.conf file ... How would i be able to detect which line it is that is ringing? |
| 17:42.00 | FuriousGeorge | ive started to document my epic struggle with asterisk over the years here... |
| 17:42.06 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:42.08 | FuriousGeorge | http://forums.digium.com/viewtopic.php?t=22719 |
| 17:42.14 | ManxPower | trapa: normally it would auto matically be routed to an exten => line that matches the DID |
| 17:42.21 | FuriousGeorge | could it be because ive chosed amd_64 as the arch? |
| 17:42.29 | FuriousGeorge | *chosen? |
| 17:42.54 | trapa | Matches teh DID? it's not a DID line. It's a second regisztration to another seperate voip provider |
| 17:43.15 | ManxPower | trapa: if it's inbound then there is a phone number call to get to you |
| 17:43.23 | trapa | So right now it goes to the default context. |
| 17:43.31 | trapa | Yeah .. There's two actually but soon to be more. |
| 17:43.51 | ManxPower | it should go to whatever context= is in the matching friend/user/peer section of sip.conf |
| 17:43.58 | trapa | It's a bunch of different 800 numbers that a publizied in different places ... What i'm suppose to do is change the caller id to say which company the caller called |
| 17:44.15 | FuriousGeorge | ...and im not too proud to outsource either, it just seems really problematic to set up: "GIG: make asterisk run for two months with no intervention, and no daily restarting" |
| 17:44.27 | FuriousGeorge | must employ techniques i can reproduce |
| 17:44.32 | ManxPower | FuriousGeorge: any reason to not compile as intel 32-bit? |
| 17:44.52 | FuriousGeorge | ManxPower: no, i suppose i should try it |
| 17:45.15 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 17:45.16 | trapa | Hrm it doesn't seem to be doing that ... what does it use to match to the matching friend/user/peer ? |
| 17:45.19 | trapa | host? |
| 17:45.26 | ManxPower | trapa: the ONLY thing registration does is tell the provider what IP is associated with which userid/password, and optionaly request what number the call be destined to when it comes in. |
| 17:46.15 | ManxPower | trapa: it can match host, but it's very unreliable in real world setups, as multiple accounts could come from the same host, you normally want to match on the [thisistheincominguserid] and secret= |
| 17:46.23 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 17:46.24 | ManxPower | trapa: I think you really need to read The Good Book |
| 17:46.31 | kamh | hi |
| 17:46.42 | kamh | short question.... |
| 17:46.48 | FuriousGeorge | i like the part where asterisk smote the proprietary pbx |
| 17:47.07 | kamh | is asterisk a registrar or proxy server? |
| 17:47.10 | trapa | I have read the good book, but the thing is, that we're not making outbound calls with this line, so it gets a little confusing when i know i don't need to use oubound stuff .. |
| 17:47.33 | ManxPower | kamh: Asterisk is neither, it is a B2BUA |
| 17:48.07 | ManxPower | FuriousGeorge: I suggest it because Asterisk on Intel 32-bit is a million times better tested than AMD-64 8-) |
| 17:48.11 | kamh | hmm... 'couse on voip-info.org there is written taht ast is a registrar or location but NOT the proxy |
| 17:48.26 | [TK]D-Fender | kamh: B2BUA <- |
| 17:48.29 | FuriousGeorge | ManxPower: first thing im doing thonight |
| 17:48.53 | ManxPower | FuriousGeorge: I don't know enough about AMD-64 to know if it can emulate/support Intel-32 |
| 17:49.33 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:49.40 | kamh | I asked this question because i have noticed that my all voice traffic is going through asteris |
| 17:49.45 | FuriousGeorge | ManxPower: it can, but im wondering if i might be better off just going all 32bit |
| 17:49.57 | kamh | so I think that it may be the proxy |
| 17:50.08 | FuriousGeorge | i can also set up a chroot enviornment for asterisk to run in, but it sounds too complicated |
| 17:50.12 | ManxPower | kamh: would be that the SIP traffic or the RTP traffic? |
| 17:50.17 | *** part/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 17:50.21 | kamh | RTP |
| 17:50.48 | kamh | I've noticed the RTP packets |
| 17:51.17 | ManxPower | kamh: then you have something setup that requires Asterisk to stay in the media path, like call recording, DTMF based call control, transcoding or disabling reinvites. Asterisk is not, has never been, and will never be a SIP proxy. |
| 17:52.25 | kamh | thank You for Your answer :] |
| 17:55.37 | Idle | fun, its like playing 'stump the digium support guy'... :( |
| 17:55.59 | Idle | I just want these phones to work... I think it may be a clock drift because it only seems to fail right before its about to reset the lines |
| 17:56.48 | *** part/#asterisk ecrist (n=ecrist@snipe.secure-computing.net) |
| 17:57.33 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 18:00.34 | *** join/#asterisk Corazu (n=Corazu@bas3-toronto12-1128688788.dsl.bell.ca) |
| 18:06.24 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 18:15.14 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 18:15.21 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:18.59 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 18:19.10 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 18:21.04 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 18:21.35 | jaytee | Time Warner Telecom is having major problems today in the midwest. |
| 18:23.06 | Strom_C | this is why you shouldn't buy your telephone service from the cable TV company |
| 18:23.23 | rwaite | done and done |
| 18:23.27 | jaytee | they are both our telco and our ISP |
| 18:23.43 | Qwell | ITSP |
| 18:23.49 | Qwell | and ISP |
| 18:24.01 | jaytee | nope |
| 18:24.05 | jaytee | ISP only |
| 18:24.13 | Qwell | So they're providing you copper? |
| 18:24.28 | jaytee | two T1's as PRI |
| 18:24.48 | NovceGuru | since when did time warner off t1s :( |
| 18:24.52 | NovceGuru | offer* |
| 18:25.32 | jaytee | dunno, but at least since we switched from SBC (AT&T) in 2005 |
| 18:25.41 | Idle | Qwell: I'm stumping your tech support guys with this issue :P |
| 18:25.50 | Qwell | O.o |
| 18:25.57 | Idle | I know :P |
| 18:26.41 | Idle | stupid thing is getting bad messages... disconnects with reason code 95 |
| 18:26.56 | Idle | meaning, asterisk is sending bad data... they think it may be CID related.. |
| 18:27.46 | Idle | can I set zapata to not send CID outbound on span1, but to send it out on span2 ? |
| 18:28.10 | Idle | or is that a global setting? |
| 18:28.16 | jaytee | Time Warner's NOC says they had to take 2 OC-192 circuits down that make up their backbone to Chicago and all the traffic is being routed through "narrower tubes :-) " or as they like to say, "are traversing other circuits with lower bandwidth capabilities." |
| 18:28.57 | Idle | jaytee: theyre just trying to say that net neutrality cant work... |
| 18:29.09 | rwaite | the internet is a series of tubes |
| 18:29.42 | sakajawebe | I believe you can set the CID right before you do a dial in your dialplan |
| 18:29.44 | jaytee | "if we let everyone access whatever content they want on the internet for a fixed price then the terrorist will win!!!!" |
| 18:29.56 | x86 | lawlz |
| 18:30.00 | sakajawebe | so it doens't have to be global |
| 18:30.09 | Idle | sakajawebe: of course, but, I want it to not even send it.. not send a blank one.. :P |
| 18:30.13 | jaytee | oops, terrorist should have been plural |
| 18:30.40 | sakajawebe | well, can you set it to empty string then? |
| 18:30.56 | mchou | anybody here a EE? My ringer coils are buzzing like hell |
| 18:31.07 | mchou | any way to fix that noise? |
| 18:31.12 | Qwell | replace it |
| 18:31.18 | jaytee | WD-40 |
| 18:31.19 | *** join/#asterisk john_fbac (n=johnfbac@66.35.148.33) |
| 18:31.23 | Idle | sakajawebe: but does that prevent zaptel from sending cid alltogether? |
| 18:31.33 | mchou | jaytee: WD-40?? |
| 18:31.38 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 18:31.55 | mchou | jaytee: how does that stop the noise? |
| 18:32.01 | jaytee | mchou, never heard of it? it's almost as big a miracle as duct tape. |
| 18:32.20 | mchou | jaytee: look at my question |
| 18:32.25 | mchou | jaytee: how does that stop the noise? |
| 18:32.43 | jaytee | mchou, nevermind. |
| 18:32.55 | jaytee | do what Quell advised, replace them |
| 18:33.04 | *** join/#asterisk B1ST (n=overdose@d51A57C9F.access.telenet.be) |
| 18:33.49 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:34.55 | sakajawebe | usecallerid=no in zapata.conf ? |
| 18:34.58 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 18:35.31 | sakajawebe | Idle: or restrictcid=yes in zapata.conf ? |
| 18:36.20 | sakajawebe | but that would then be a general setting |
| 18:36.22 | Idle | sakajawebe: yea, can I set that per channel? |
| 18:36.36 | Idle | er, per span |
| 18:36.59 | sakajawebe | hrhm, if you change it again afer the channel(s)=X command you can change it for the next set of channels |
| 18:37.22 | zeeesh | can we run asterisk and freeswitch at the same machine ? |
| 18:37.27 | B1ST | hi, just a (probably noob) question, how can i enable domain support in asterisk? |
| 18:37.32 | Idle | sakajawebe: I may try that |
| 18:37.38 | lmadsen | zeeesh: possibly... with careful configuration |
| 18:37.41 | B1ST | telenetPC*CLI> sip show domains |
| 18:37.41 | B1ST | SIP Domain support not enabled. |
| 18:37.46 | sakajawebe | everything above the channel => command is supposed to apply to those channels, so if you cahnge something after the channels=> then it will different for the next set |
| 18:38.02 | lmadsen | B1ST: I think there is an autodomain=yes option in sip.conf |
| 18:38.12 | lmadsen | or you can define the domains you are going to handle |
| 18:38.17 | sakajawebe | Idle, take luck! |
| 18:38.21 | lmadsen | domain=leifmadsen.com |
| 18:38.22 | B1ST | in asterisk.conf? |
| 18:38.24 | lmadsen | for example |
| 18:38.28 | Qwell | Idle: take many lucks |
| 18:38.37 | lmadsen | B1ST: read my first comment again |
| 18:38.42 | Idle | Qwell: wot? |
| 18:38.43 | B1ST | i did not setted a domain on my debian box |
| 18:38.51 | B1ST | so i do not need that, right? |
| 18:38.58 | lmadsen | you do not need domain support |
| 18:39.03 | B1ST | ok, cool |
| 18:39.06 | Idle | sakajawebe: yea, i know about the channels thing ;) |
| 18:39.23 | sakajawebe | ah sorry then |
| 18:39.31 | sakajawebe | ya never know what they know ya know |
| 18:39.45 | Idle | yep, I know, always good to cover the bases |
| 18:39.55 | Idle | sometimes people miss the most basic of things |
| 18:41.33 | sakajawebe | yep |
| 18:46.19 | *** join/#asterisk metfan2007 (n=jc@201.103.64.129) |
| 18:46.55 | *** part/#asterisk hoonteke (n=kevin@cpe-069-134-205-184.nc.res.rr.com) |
| 18:48.54 | metfan2007 | Hi all!!! I'm having some problems with Polycom 330 phones when I try to make a call using "New Call" button or transfer a call. After some sip debug I found that only when I use this "methods" the phone sends the next header to asterisk: |
| 18:49.10 | metfan2007 | Proxy-Authorization: Digest username="1102", realm="asterisk", nonce="4fd3029c", uri="sip:11@192.168.2.99:5060;user=phone", response="1238fcf7f12422443cf04623d1eb1655", algorithm=MD5 |
| 18:49.16 | *** join/#asterisk maqr (n=maqr@httpcraft/hax) |
| 18:49.59 | maqr | is there any reason why outgoing calls from my sip phone wouldn't recognize dtmfs? i called another phone system, and apparently i can't press "1" or anything else :) |
| 18:50.15 | metfan2007 | if you can see the uri part sends "11@192.168...." instead of "1102@192..." any idea??? |
| 18:50.49 | maqr | my call is going outbound through Dial(), maybe there's a better way to do it that will send dtmf? |
| 18:50.54 | metfan2007 | If I pess Line 1 or 2 in the phone I can make calls because the phone does NOT sends the Proxy-Authorization header |
| 18:53.12 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:53.33 | maqr | DTMF incoming works fine, but i can't send it out |
| 18:53.58 | maqr | i have 'dtmfmode=rfc2833' set in my sip.conf in the general section |
| 18:54.16 | jaytee | how is your DTMF configured on the phones? in-audio, rfc2833 or SIP INFO? |
| 18:54.22 | Idle | I've been talking to this guy for like, 3 hours |
| 18:54.31 | Idle | I guess maybe 2... hm |
| 18:54.46 | maqr | jaytee: i have no idea, i didn't even know that was a phone option |
| 18:54.57 | jaytee | yeah, it is on SIP phones |
| 18:55.32 | jaytee | and you can enable it to use combinations. |
| 18:55.37 | jaytee | check your phone's manual |
| 18:56.32 | maqr | jaytee: i hvae a polycom 330, and it doesn't seem to be in the menu |
| 18:56.41 | maqr | jaytee: is there some way to just have asterisk tell the phone which mode to use? |
| 18:57.06 | jaytee | maqr, the DTMF setting in your sip.conf |
| 18:57.44 | jaytee | the DTMF setting can be set both globally and per SIP device if I'm not mistaken but then I'd verify that with someone else here because I'm often mistaken |
| 18:58.27 | maqr | jaytee: 'rfc2833' is the general setting, which works fine inbound, so it should also be applying that setting outbound, right? |
| 18:58.32 | maqr | that's what vitelity told me to use |
| 18:59.03 | jaytee | and that's what I'm using. what kind of channel is the outbound? analog FXO or digital PRI? |
| 18:59.15 | *** join/#asterisk ddunavant (n=David@pool-96-231-68-230.washdc.east.verizon.net) |
| 19:00.17 | maqr | jaytee: umm, i guess digital? it's my vitelity sip account calling my cell phone |
| 19:03.09 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:03.35 | maqr | hmm, the admin guide does mention it, but it looks like i need to edit the actual config files on the phone |
| 19:04.21 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:08.01 | *** join/#asterisk ddunavant (n=David@pool-71-191-28-243.washdc.east.verizon.net) |
| 19:08.06 | maqr | i guess i need a boot server to config this |
| 19:08.10 | maqr | that's less than idea |
| 19:08.11 | maqr | *ideal |
| 19:08.24 | ManxPower | you'll eventually use a boot server if you use polycoms |
| 19:08.59 | [TK]D-Fender | maqr: "less than ideal" depends if you want to get the value out of your phones. |
| 19:09.29 | Nugget | you'll eventually quit your job and join a monastary to escape if you use ciscos. |
| 19:09.54 | maqr | [TK]D-Fender: with all the options on the web interface, you'd think "DTMF mode" would be one of them |
| 19:10.03 | maqr | but it looks like you need a boot server for that |
| 19:10.06 | [TK]D-Fender | maqr: They all use RFC2833 |
| 19:10.18 | [TK]D-Fender | maqr: so jsut set that in your phone's sip.conf entry |
| 19:10.29 | [TK]D-Fender | maqr: this is not something you have to worry about on the phone side |
| 19:10.34 | maqr | [TK]D-Fender: but i can't get to my phone's sip.conf without a boot server :/ |
| 19:10.34 | maqr | oh |
| 19:10.41 | maqr | [TK]D-Fender: well, i set that, and i can't send dtmf |
| 19:10.45 | maqr | i can receive it fine though |
| 19:10.52 | [TK]D-Fender | maqr: sip.conf, not sip.cfg <- |
| 19:10.58 | metfan2007 | anyone knows why plolycom phones tries to send the Proxy-Authorization header when I press a soft key to place a call? and when using a hard key it doesn't? |
| 19:10.59 | maqr | yeah, it's set there |
| 19:11.24 | [TK]D-Fender | maqr: then taht is yout *ITSP* peer thats wrong. |
| 19:11.27 | maqr | [TK]D-Fender: you sure? it's vitelity, and they told me to set dtmfmode in [general] |
| 19:11.35 | maqr | hmm |
| 19:11.52 | [TK]D-Fender | maqr: NO. Set it in the PEER <- |
| 19:13.02 | maqr | [TK]D-Fender: that doesn't do it either |
| 19:13.22 | maqr | if i call my cell, i can hear my cell's dtmf on the voip side, but i can't hear my sip phone's dtmf on the cell phone |
| 19:14.12 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 19:14.59 | ManxPower | maqr: what is the [thissection] named for your phone? |
| 19:15.05 | ManxPower | in sip.conf |
| 19:20.03 | maqr | ManxPower: well, they all have "dtmfmode=rfc2833" now, i just put it in every section |
| 19:20.54 | lmadsen | that sounded like a great use for templates maqr :) |
| 19:22.32 | [TK]D-Fender | maqr: http://www.google.ca/search?hl=en&q=vitelity+dtmfmode&btnG=Google+Search&meta= |
| 19:23.40 | [TK]D-Fender | maqr: 2nd link seems to say something pretty clear |
| 19:24.08 | [TK]D-Fender | "11 Apr 2008 ... All Vitelity has told me is that I should be using dtmfmode=inband" |
| 19:24.31 | [TK]D-Fender | JFGI , and don't assume that EVERYBODY uses rfc2833 |
| 19:25.19 | maqr | [TK]D-Fender: oh, i have no idea how i overlooked that, thank you |
| 19:28.27 | *** join/#asterisk catpants (n=catpants@c-71-228-179-232.hsd1.al.comcast.net) |
| 19:39.40 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 19:46.39 | *** join/#asterisk trapa (n=2264581@207.230.238.94) |
| 19:47.48 | ipstatic | If you are using the ring strategy rrmemory, does asterisk obey agent priorities? |
| 19:47.57 | ipstatic | in queues.conf |
| 19:48.43 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 19:49.46 | *** join/#asterisk john_fbac (n=johnfbac@66.35.148.33) |
| 19:49.52 | [TK]D-Fender | ipstatic: it obeys within each priority grouping. |
| 19:51.18 | *** join/#asterisk underguiz (n=undergui@unaffiliated/underguiz) |
| 19:52.16 | ipstatic | [TK]D-Fender: how can I set this grouping? |
| 19:52.37 | [TK]D-Fender | ipstatic: this isthe priority itself, be it the agent, or the caller. |
| 19:53.40 | ipstatic | [TK]D-Fender: You confused me with that statement, let me explain what is going on |
| 19:54.19 | ipstatic | agent 1 has a priority of 10, agent 2 has a priority of 50. call 1 goes to agent 1, however call 2 goes to agent 2, even when agent 1 is off the phone |
| 19:54.25 | ipstatic | shouldn't call 2 go to agent1 since they are off the phone and have a lower priority? |
| 19:54.43 | ipstatic | they are in the same queue, dynamically added |
| 19:55.24 | [TK]D-Fender | ipstatic: not sure... |
| 19:55.47 | ipstatic | voip-info says that will happen, but they mention the ring strategy is ringall |
| 19:58.03 | maqr | [TK]D-Fender: btw, vitelity requires rfc2883 inbound and inband outbound, just for reference |
| 19:59.28 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 20:00.30 | *** join/#asterisk [netman] (n=netman@12.Red-81-36-134.dynamicIP.rima-tde.net) |
| 20:00.38 | [netman] | hi all |
| 20:00.43 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 20:02.35 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.mn.comcast.net) |
| 20:12.17 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 20:14.25 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-130-112.dsl.teksavvy.com) |
| 20:19.44 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 20:21.02 | dlynes | maqr: nice to remain consistent, isn't it? |
| 20:22.51 | maqr | dlynes: i'm not smart enough to question them yet :) |
| 20:31.57 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 20:32.35 | Schreiber1337 | Has anyone upgraded a Ubuntu 6.10 server to 8.04 running Asterisk? |
| 20:36.40 | brodiem | never used noobuntu but good chance it'll break and need to be recompiled |
| 20:37.12 | brodiem | due to new glibc |
| 20:38.15 | Schreiber1337 | brodiem: don't be a hater... thanks for your input |
| 20:39.37 | *** join/#asterisk ZX81 (n=matt@203-79-98-34.adsl.paradise.net.nz) |
| 20:39.39 | brodiem | heh couldn't resist my bad |
| 20:39.51 | ZX81 | anyone from xorcom around? |
| 20:43.49 | ZX81 | anyone have an idea: if I have a xorcom unit plugged into USB, the computer won't start bios. If I unplug in and connect it once debian starts loading, it works fine |
| 20:44.42 | brodiem | try disabling USB as an auto-boot device in your bios |
| 20:45.23 | ManxPower | tzafrir_laptop |
| 20:49.05 | ZX81 | brodiem: done |
| 20:49.59 | ZX81 | also tried disabling usb legacy support |
| 20:50.22 | ZX81 | it doesn't even get to the boot stage, just stays blank, keyboard doesn't work etc |
| 20:50.23 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:51.39 | brodiem | ZX81: hm possibly try going through a usb hub instead if you've got one |
| 20:51.53 | ZX81 | yep, actually 3 units, powered usb hub |
| 20:52.08 | ZX81 | if I remove the astribanks it works fine, even with the hub in there |
| 20:52.20 | *** join/#asterisk adiel (n=adiel@bg-fw2out.monmouth.com) |
| 20:52.25 | ZX81 | also tried removing the UPS comms cable (USB) to no avail |
| 20:52.48 | ZX81 | also moved xorcom units to the UPS so they share the same ground with pc |
| 20:53.06 | ZX81 | and connected the case to the ground pins on the xorcoms |
| 20:53.14 | ZX81 | in case there was a ground loop |
| 20:53.14 | brodiem | bios upgrade? |
| 20:53.19 | *** join/#asterisk KenLee (n=k3leland@bg-fw2out.monmouth.com) |
| 20:53.25 | ZX81 | heh maybe |
| 20:53.34 | ZX81 | brand new supermicro though |
| 20:54.10 | ZX81 | can check, problem is its a production machine with around 100 rooms of a hotel on it |
| 20:54.12 | ZX81 | :) |
| 20:54.35 | ZX81 | I've organised a scheduled outage for this arvo, so will try everything at once |
| 20:55.29 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:55.58 | ZX81 | its all good while I'm here - I'm more worried about what will happen if they have a power cut while I'm not :) |
| 20:57.15 | *** join/#asterisk Qb3rt (n=eric@palpatine.privatedns.com) |
| 20:57.25 | *** part/#asterisk adiel (n=adiel@bg-fw2out.monmouth.com) |
| 20:57.25 | Qb3rt | . |
| 20:58.05 | Qb3rt | what card do you guys recommand: iam looking for a fxo card 1 port pci |
| 21:00.14 | ZX81 | looking to upgrade in the future or just 1 port forever? |
| 21:00.16 | [TK]D-Fender | Qb3rt: what kind of usage? |
| 21:00.26 | Qb3rt | small house pbx |
| 21:00.30 | Qb3rt | 1 line |
| 21:00.41 | Qb3rt | 1 port for ever |
| 21:01.51 | [TK]D-Fender | Qb3rt: Linksys SPA-3102 |
| 21:01.55 | *** join/#asterisk Zemmad (n=calibisi@cpe-66-69-164-90.sw.res.rr.com) |
| 21:02.29 | Qb3rt | this is not pci!? |
| 21:03.20 | Qb3rt | i want it to be fxo |
| 21:03.20 | brodiem | yep it'll fit right in the slow |
| 21:03.21 | brodiem | slot |
| 21:03.21 | brodiem | lol |
| 21:03.41 | Zemmad | i need some help here...been trying to setup with voipjet but receiving the call reject....no authority error |
| 21:03.48 | [TK]D-Fender | Qb3rt: No, its not PCI, and YES, it is an FXO interface |
| 21:04.01 | Zemmad | i'm assuming that my account is not created on their side...is that the case? |
| 21:04.49 | Qb3rt | need a pci fxo card please what do you recommand |
| 21:06.04 | [TK]D-Fender | abWhy does it have to be PCI? |
| 21:06.05 | [TK]D-Fender | Qb3rt: * |
| 21:06.05 | brodiem | Qb3rt: single fxo pci cards are only like $20 for the digium card |
| 21:06.05 | brodiem | x100 or whatever it is |
| 21:07.21 | Qb3rt | Fender: yesy please cuz iam running out of space |
| 21:07.49 | [TK]D-Fender | Qb3rt: .... kinda sad. |
| 21:07.51 | Zemmad | any help?? |
| 21:08.07 | Qb3rt | Fender: hehe |
| 21:08.08 | [TK]D-Fender | Qb3rt: Find, TDM410P then. |
| 21:08.39 | tzafrir_laptop | ZX81, basically the issue is that the BIOS gets a strange idea that it needs to boot from that device |
| 21:09.27 | [TK]D-Fender | Zemmad: might help if you told us where it is that you get that error |
| 21:10.14 | *** part/#asterisk Qb3rt (n=eric@palpatine.privatedns.com) |
| 21:10.40 | brodiem | you're welcome |
| 21:12.10 | Zemmad | Fender: this error i receive when i try to terminate a call through voipjet |
| 21:12.41 | brodiem | here comes |
| 21:12.45 | Zemmad | Fender: this is the error line " WARNING[15372]: chan_iax2.c:7484 socket_process: Call rejected by 208.72.186.69: No authority found" |
| 21:12.56 | [TK]D-Fender | Zemmad: bad peer setup or dial. pastebin your iax.conf masking ONLY passwords, and your dialplan, along with the error |
| 21:12.58 | [TK]D-Fender | ~pb |
| 21:12.58 | jbot | [~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 |
| 21:13.00 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 21:13.03 | *** join/#asterisk ZX81_ (n=matt@203-79-98-34.adsl.paradise.net.nz) |
| 21:15.17 | Zemmad | Fender: http://pastebin.com/m12de3122 |
| 21:16.53 | [TK]D-Fender | Zemmad: and the rest? |
| 21:20.56 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 21:21.13 | Zemmad | Fender: http://pastebin.com/d75077d69 |
| 21:24.15 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:24.38 | [TK]D-Fender | Zemmad: host=test.voipjet.com <- I wonder if this is indeed the proper URI for live use |
| 21:25.04 | Zemmad | Fender: it's their test server |
| 21:26.34 | *** join/#asterisk d-k-t (n=dt@125.120.133.213) |
| 21:26.45 | [TK]D-Fender | Zemmad: comment out "username" |
| 21:27.06 | *** join/#asterisk galeras (n=galeras@201.244.245.108) |
| 21:27.21 | Zemmad | Fender: ok |
| 21:28.33 | Zemmad | Fender: same error |
| 21:29.26 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-10-123.lns10.syd7.internode.on.net) |
| 21:29.59 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:30.48 | [TK]D-Fender | Zemmad: and have you asked them? |
| 21:31.11 | galeras | Please, give me some clue about why my fxo ports doesn't hangup inbound calls. |
| 21:31.35 | galeras | besides asterisk show no channels in use |
| 21:32.08 | Zemmad | Fender: tried contacting them with no answer to my emails |
| 21:34.35 | *** part/#asterisk fogo (n=fogo@72.8.104.15) |
| 21:35.43 | [TK]D-Fender | Zemmad: maybe your account isn't active there yet |
| 21:36.49 | *** join/#asterisk whye (n=whye@unaffiliated/whye) |
| 21:37.42 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 21:37.42 | jaytee | galeras, what kind of FXO card is it? |
| 21:37.47 | Zemmad | Fender: i was thinking the sam |
| 21:37.52 | Zemmad | same |
| 21:38.44 | galeras | jaytee: digium tdm404 |
| 21:39.45 | jaytee | with fxs_ks or fxs_ls signalling? |
| 21:40.19 | galeras | jaytee: with fxs_ks signalling? |
| 21:40.31 | galeras | jaytee: with fxs_ks or fxs_ls signalling |
| 21:40.48 | galeras | jaytee: sorry, with jaytee: with fxs_ks signalling |
| 21:41.53 | jaytee | is it for both outbound calls and inbound calls or from just one originating point? |
| 21:46.53 | galeras | jaytee: is for all inbound calls. if the callee (from asterisk) hangs up, the channel remains busy until the caller hangs up. the weird thing is asterisk thinks channel is available (zap show channel) |
| 21:49.54 | jaytee | what country are you in? |
| 21:49.58 | galeras | Colombia |
| 21:50.55 | jaytee | what is the loadzone= set to in your zaptel.conf file? |
| 21:51.45 | galeras | jaytee: loadzone = us |
| 21:52.01 | jaytee | and is this a new install or did it work before and then just start happening? |
| 21:52.31 | galeras | jaytee: is a new install |
| 21:55.12 | galeras | jaytee: it seems is a problem with disconnect supervision, i tried to call from two phones (without asterisk) and the behaviour is the same. |
| 21:55.35 | jaytee | how many channels on the card and if more that 1 FXO do they all exhibit the same behavior? |
| 21:56.31 | galeras | jaytee, so far i tested only one line, one port, please give me a sec to test other ports/lines |
| 21:56.47 | *** join/#asterisk ryoohki (n=ryoohki@208.96.15.252) |
| 21:58.06 | ryoohki | our system is down - can anyone help? rhel 4.4 + asterisk 1.2.18 |
| 21:58.39 | ryoohki | a new user added a second entry with the same mac address because out phones take two extensions |
| 22:00.45 | russellb | O.o |
| 22:00.50 | russellb | what in asterisk 1.2 uses mac addresses? |
| 22:01.00 | ManxPower | ryoohki: If you want help you need to start providing details. What MAC? Where? What phone? What is down? Is the console responsive? |
| 22:01.13 | [TK]D-Fender | russellb: Abso-frakking-lutely nothing :) |
| 22:01.13 | ManxPower | russellb: The SIP userid, if you are smart. |
| 22:01.20 | russellb | [TK]D-Fender: i know :-p |
| 22:01.30 | russellb | ManxPower: hm, i suppose that's one way to do it :) |
| 22:01.52 | russellb | i suggest a hammer |
| 22:01.58 | russellb | (pending further details of what is actually wrong) |
| 22:01.59 | jaytee | lol |
| 22:02.02 | ManxPower | russellb: it's better than [robertdobbs] or [1234] |
| 22:02.29 | [TK]D-Fender | ryoohki: * does not care about MAC's. And phones don't take "extensions". |
| 22:02.41 | russellb | eh, i don't really care what people call their peers... i just want stuff to work |
| 22:02.44 | [TK]D-Fender | ryoohki: An extension is something you dial in your dialplan. |
| 22:02.54 | russellb | you guys can worry about best practices :) |
| 22:04.58 | unpaidbill | it cares about macs if you make it use chan_sccp! or does that not count |
| 22:05.59 | ManxPower | russellb: wrong words lead to wrong troubleshooting |
| 22:06.09 | [TK]D-Fender | unpaidbill: It certainly does not! SCCP is a lie made up to scare small children :p |
| 22:06.16 | unpaidbill | haha |
| 22:06.22 | unpaidbill | i wish that was all it is :P |
| 22:06.35 | ManxPower | unpaidbill: So you can't run SCCP across networks? (as you don't know the MAC if the device is not on the local network) |
| 22:06.39 | russellb | chan_skinny > chan_sccp |
| 22:06.43 | russellb | (in 1.4 anyway) |
| 22:06.53 | [hC] | You can run skinny/sccp across networks |
| 22:06.57 | unpaidbill | manx, well.. technically it isnt really macs i guess |
| 22:07.12 | [hC] | and chan_sccp-b has come a long way in the last 2 months actually. We finally got support for missed call logs! |
| 22:07.17 | unpaidbill | it's the MAC of the phone with SEP prefix sent in some kind of authentication request |
| 22:07.28 | unpaidbill | chan_sccp-b has been good to me |
| 22:07.34 | unpaidbill | it's the goddamn cisco phones that are the problem! |
| 22:07.43 | unpaidbill | and my 7935 that locks up on registration :( |
| 22:08.20 | [hC] | once we have a working CONF button for 7960/70's in skinny mode, i'll be satisfied |
| 22:08.33 | unpaidbill | i just use sip for those |
| 22:08.43 | unpaidbill | even though SCCP works better (as far as Ui is concerned) |
| 22:08.56 | unpaidbill | it really pisses me off the small changes cisco made to the sip firmware |
| 22:08.56 | ManxPower | ryoohki: Which Asterisk GUI are you using? |
| 22:09.16 | [hC] | sip is so annoying, and the firmware is missing a bunch of stuff, and its buggy, on cisco's |
| 22:09.19 | unpaidbill | it's like they did it to slap you in the face for using si |
| 22:09.28 | [hC] | thats exactly what they did |
| 22:09.54 | unpaidbill | well next time im near the cisco HQ i'm gonna take a dump in front of their main enterance |
| 22:10.49 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 22:11.29 | galeras | jaytee: same behaviour for 3 differents lines on 3 differents ports |
| 22:11.30 | [TK]D-Fender | ok, out for a while.. later |
| 22:11.32 | unpaidbill | you work on chan_Sccp-b hc? |
| 22:11.48 | [hC] | unpaidbill: i dont develop it, no, but i do use it |
| 22:11.59 | unpaidbill | you dont use 7935s by any chance, do you? |
| 22:12.30 | [hC] | unpaidbill: those are the cordless phones right? |
| 22:12.41 | [hC] | unpaidbill: no i dont unfortunately. |
| 22:12.43 | unpaidbill | nah, the cisco/polycom bastard children for conferencing |
| 22:12.50 | [hC] | oh |
| 22:12.53 | [hC] | nope, none of those either |
| 22:12.59 | unpaidbill | darn :) |
| 22:13.02 | [hC] | i use ip4000's for conferencing |
| 22:13.17 | unpaidbill | yeah, the 7935 is end of life too |
| 22:13.35 | unpaidbill | damn thing cant register on my box though, it registers, then immediately locks up |
| 22:15.20 | [hC] | That sucks |
| 22:15.26 | [hC] | im not even sure where i'd start with that |
| 22:15.30 | [hC] | kinda hard to debug |
| 22:15.50 | [hC] | Here's an unrelated question.. Why is it that all asterisk distro's are based on top of centos? Is there a reason? |
| 22:17.27 | ManxPower | [hC]: I imagine because they don't understand that Mandriva is far superior. 8-) |
| 22:17.30 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-43-102.lns10.syd7.internode.on.net) |
| 22:17.41 | [hC] | ManxPower: ;) |
| 22:17.54 | unpaidbill | yeah pretty much, i kind of gave up on it |
| 22:18.17 | unpaidbill | hey i put asterisk on my centos box |
| 22:18.20 | Qwell | unpaidbill: send me one |
| 22:18.32 | Qwell | (a 7935) |
| 22:18.34 | unpaidbill | qwell will you send it back? :P |
| 22:19.07 | unpaidbill | they dont make them anymore |
| 22:19.14 | Qwell | eh, I don't really have time lately to add it anyways |
| 22:19.14 | unpaidbill | it's kind of a waste of time to fix imo |
| 22:20.07 | unpaidbill | i didnt know you worked with the chan_sccp-b project |
| 22:20.13 | Qwell | I don't |
| 22:20.15 | Qwell | it's junk |
| 22:20.17 | unpaidbill | oh |
| 22:20.25 | Qwell | I maintain chan_skinny though |
| 22:20.27 | unpaidbill | chan_skinny right, i forgot that is there :P |
| 22:20.37 | unpaidbill | i cant remember why i had to use sccp |
| 22:20.41 | unpaidbill | but there was some skinny problem |
| 22:20.44 | [hC] | Qwell: do you have call log lists working on chan_skinny? missed, placed, received, etc? |
| 22:20.53 | Qwell | [hC]: that's all done on the phone |
| 22:20.58 | Qwell | s/on/by/ |
| 22:21.01 | Qwell | ... |
| 22:21.03 | Qwell | stupid bot |
| 22:21.32 | ManxPower | Qwell: stupid too short of search strings |
| 22:21.33 | [hC] | Qwell: yeah, but it requires a certain command order from the server to store them properly. sccp-b only got it working recently. up until then it said "unknown caller" |
| 22:21.51 | [hC] | Qwell: something about having to send an ACK packet before the call was answered, or hung up, or something. Im not sure exactly. |
| 22:22.48 | ManxPower | When I played with SCCP/Skinny phones with Asterisk, it just seemed about as useful as a bicycle in Alaska -- sure you can use it, but only a few days a year and it really isn't very USEFUL. |
| 22:23.04 | ryoohki | ManxPower: we're back up due to back ups made in the form of tar files |
| 22:23.18 | ryoohki | ManxPower: now out voice mail files are gone |
| 22:24.14 | ManxPower | ryoohki: But that is not the answer to the questions I asked. |
| 22:24.38 | ManxPower | The most important of which is what GUI are you using to manage Asterisk? |
| 22:26.20 | jaytee | ManxPower, what GUI do you use? vim or nano? |
| 22:26.44 | ManxPower | jaytee: usually 'joe' |
| 22:26.57 | ManxPower | Of if I'm VPN'd in, jEdit. |
| 22:30.10 | ManxPower | *** ryoohki added to /ignore |
| 22:32.06 | russellb | *** ManxPower added to /ignore |
| 22:32.07 | russellb | :-p |
| 22:32.35 | *** mode/#asterisk [+b russellb!*@*] by Qwell |
| 22:32.38 | ManxPower | russellb: I think that is for the best. 8-) |
| 22:32.38 | *** mode/#asterisk [-b russellb!*@*] by Qwell |
| 22:32.59 | *** mode/#asterisk [+b %Qwell!*@*] by russellb |
| 22:33.14 | *** mode/#asterisk [-b %Qwell!*@*] by russellb |
| 22:33.53 | Qwell | russellb: ...nub |
| 22:33.56 | *** mode/#asterisk [-o Qwell] by russellb |
| 22:34.10 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:34.15 | russellb | hehe |
| 22:34.27 | *** mode/#asterisk [-o Qwell] by ChanServ |
| 22:34.29 | Qwell | :( |
| 22:34.36 | russellb | pwnt |
| 22:34.53 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:34.59 | jaytee | I feel like I'm back in homeroom period in my sophomore year |
| 22:35.01 | ManxPower | Tonight on WWE: Russel "Killer of Drums" B. .vs. Qwe "mom had a sense of humor" ll. |
| 22:35.38 | *** mode/#asterisk [-o Qwell] by ChanServ |
| 22:35.46 | Qwell | fail |
| 22:35.55 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:36.00 | russellb | O.O |
| 22:36.05 | putnopvut | bows |
| 22:36.23 | ManxPower | That was supposed to be WWE SMACKDOWN |
| 22:36.26 | Qwell | russellb: put that back :p |
| 22:36.34 | russellb | how did you do that without access? |
| 22:36.42 | russellb | or did putnopvut do it ... |
| 22:36.49 | putnopvut | russellb: I did it :p |
| 22:37.23 | *** join/#asterisk geek_cl (n=lletelie@200.75.18.211) |
| 22:37.31 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:38.05 | *** kick/#asterisk [Qwell!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (putnopvut) |
| 22:38.08 | *** kick/#asterisk [putnopvut!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 22:38.12 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 22:38.24 | russellb | hides |
| 22:38.26 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 22:38.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:38.35 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 22:39.08 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-53-41.lns10.syd7.internode.on.net) |
| 22:39.28 | Qwell | russellb: +O too, kthx |
| 22:39.40 | russellb | o.o |
| 22:39.48 | Qwell | autoop |
| 22:40.49 | Qwell | (AOP)? |
| 22:41.07 | Qwell | they changed around all the perm stuff.. no clue what (AOP) is |
| 22:41.34 | russellb | yeah, i just noticed how it's different |
| 22:41.38 | russellb | shrugs |
| 22:41.40 | russellb | whateva |
| 22:41.44 | *** join/#asterisk TheSov (n=a@dsl092-128-140.chi1.dsl.speakeasy.net) |
| 22:42.20 | mvanbaak | <PROTECTED> |
| 22:42.22 | mvanbaak | ;) |
| 22:43.16 | *** mode/#asterisk [+o mvanbaak] by russellb |
| 22:43.18 | russellb | w00t |
| 22:43.38 | *** join/#asterisk edoceo (n=edoceo@c-71-197-244-147.hsd1.or.comcast.net) |
| 22:43.39 | mvanbaak | whoa ! |
| 22:43.49 | mvanbaak | hides for mvanbaak |
| 22:44.39 | edoceo | How would I selectively change the ring count for two of 34 extensions? |
| 22:44.44 | *** part/#asterisk galeras (n=galeras@201.244.245.108) |
| 22:44.55 | edoceo | All use a macro that has Dial(ARGV1, 20) |
| 22:45.12 | edoceo | I want some to Dial(ARGV1, ARGV2) and if ARGV2 is not present use 20 |
| 22:45.50 | mvanbaak | edoceo: use a combination of Set() and IF() |
| 22:46.39 | mvanbaak | IF($[${TIMEOUT} = ""]?foo:bar) |
| 22:46.46 | mvanbaak | something like that |
| 22:47.10 | edoceo | mvanbaak: Alright! A new realm for me! current dialplan has none of that! Thanks |
| 22:49.37 | lmadsen | oh I love using IF, EXISTS, ISNULL with Set and Exec |
| 22:50.01 | lmadsen | you can do some crazyness with that combo :) |
| 22:50.26 | mvanbaak | lmadsen: amen |
| 22:51.03 | edoceo | Ok you two don't rub it in ;) |
| 22:51.05 | lmadsen | i wish I had access to one of my dialplans :) |
| 22:51.17 | edoceo | So, how about this one... |
| 22:51.18 | lmadsen | edoceo: no, we're saying learn how to use that to your advantage |
| 22:51.26 | lmadsen | it's quite powerful |
| 22:51.28 | edoceo | Couldn't I just hire you? |
| 22:51.34 | lmadsen | sure you could |
| 22:51.40 | edoceo | rate? |
| 22:51.45 | edoceo | in USD |
| 22:51.47 | ryoohki | ManxPower: we did a full restore and restarted everything and now we're back to 2008-06-18 and everything is working again. we're using the web interface |
| 22:51.56 | mvanbaak | 00:51 <@ lmadsen> i wish I had access to one of my dialplans :) |
| 22:51.57 | lmadsen | lots of people in here are capable of being hired |
| 22:52.17 | mvanbaak | lmadsen: get a copy of 'Asterisk, the future of telephony' |
| 22:52.18 | mvanbaak | ;) |
| 22:52.51 | edoceo | Yea - I read some of that book - quite good - wish I had more time to focus only on * |
| 22:53.21 | edoceo | I know I can make a phone ring two or three extensions at the same time... |
| 22:54.02 | edoceo | Can I use the Set DB stuffs to make dial rules where exten 700 can dial *834 and enable 701 to ring simultanious the *843 to disable? |
| 22:54.12 | edoceo | I'd have to use lots of IF commands huh? |
| 22:54.33 | mvanbaak | hhmm |
| 22:54.36 | mvanbaak | not really |
| 22:54.57 | edoceo | Yea, my clients want auto-attendant (is that the right term?) but don't want that big box |
| 22:54.58 | mvanbaak | it will be a combination of GotoIf(DB |
| 22:55.07 | mvanbaak | with GotoIf(IF |
| 22:55.34 | edoceo | They want assistant phone to indicate to them which extens are in use so assistant knows when to answer someone elses DID |
| 22:56.11 | mvanbaak | gheh, I just lit another smoke, to find out there are 8 death smokes on my ashtray |
| 22:59.27 | edoceo | This looks OK? |
| 22:59.27 | edoceo | exten => s,n,Set(RINGTIME=${IF($[$TIMEOUT=""] ? 20 : $TIMEOUT)}) |
| 22:59.46 | edoceo | Use 20s if no $TIMEOUT specified? |
| 22:59.49 | *** join/#asterisk cyberdeath (n=cyberdea@67.131.149.209) |
| 23:00.38 | cyberdeath | Does anyone know if it is possible to have asterisk flash over to a 3 way, dial another number, then flash back and hangup ("dirty" call fwding with 1 line)? |
| 23:01.42 | cyberdeath | I basically want to allow the caller to press a number during a voice menu and have it connect the person with another outside line. |
| 23:02.33 | mvanbaak | I'm going to bed |
| 23:08.52 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 23:09.58 | *** part/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 23:11.12 | *** join/#asterisk hardhatpat (n=pat@c-24-21-228-126.hsd1.mn.comcast.net) |
| 23:11.49 | hardhatpat | I am running a fairly simple asterisk implementation ... but after it has been running for 12-36 hours, when I dial the DID connected to my asterisk box i just get a busy signal, unless i reboot asterisk |
| 23:11.57 | hardhatpat | has anyone run into this before? |
| 23:15.12 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 23:15.16 | chandoo | hi |
| 23:15.21 | chandoo | i configured asterisk and ekiga and i can listen to default 1000 call successfully, and what next now |
| 23:15.36 | russellb | ~nowwhat |
| 23:15.36 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 23:16.03 | chandoo | jbot_: yes from source |
| 23:16.15 | chandoo | and i am using ekiga sip client |
| 23:16.33 | chandoo | can i make calls from my machine to outside phones |
| 23:17.37 | mchou | chandoo: depends what you mean by "phones" |
| 23:18.00 | mchou | chandoo: you should be able to make calls to SIP phones :) |
| 23:18.36 | mchou | for PSTN phones you still need someone/something to terminate the call for you |
| 23:18.50 | chandoo | mchou: like to my mobile phone from pc |
| 23:19.16 | mchou | chandoo: for PSTN phones you still need someone/something to terminate the call for you |
| 23:19.26 | mchou | that includes mobile |
| 23:19.54 | chandoo | what is that something you mean, |
| 23:20.07 | hardhatpat | chandoo, a gateway |
| 23:20.30 | mchou | HW or voip provider who can provide call termination |
| 23:20.34 | hardhatpat | someone who has a server connected to a PRI |
| 23:20.48 | mchou | yup yup |
| 23:20.49 | chandoo | mchou: let me tell you what i have, i have voip service from Phone Power and a desktop with amd 64bit with 3gb ram, |
| 23:21.32 | mchou | I dont know what Phone Power is |
| 23:21.45 | chandoo | voip service provider |
| 23:21.57 | chandoo | that is what i have as a regular phone |
| 23:22.05 | chandoo | like vonage |
| 23:22.27 | hardhatpat | chandoo, will they let you send calls via IAX or SIP to the PSTN through them? |
| 23:23.07 | mchou | hardhatpat: hehe, dont forget BYOD :) |
| 23:23.21 | chandoo | hardhatpat: how to find out, i have a web account and i can see what features i have |
| 23:23.39 | mchou | hardhatpat: try hooking * to vonage, be in for a world of hurt :) |
| 23:24.08 | hardhatpat | chandoo, probably they wont let you ... i looked at their website and it dosent look like they want you sending calls from * |
| 23:24.23 | hardhatpat | i use www.voipstreet.com ... they are pretty cheap $.0145/minute |
| 23:24.52 | mchou | hardhatpat: how's the voice quality and reliability for voipstreet? |
| 23:25.01 | chandoo | cant i have a gateway in my machine |
| 23:25.01 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 23:25.24 | hardhatpat | mchou, no complaints ... i just use the default bandwidth=high and it sounds better than my blackberry... |
| 23:25.34 | mchou | hardhatpat: haha |
| 23:25.47 | mchou | hardhatpat: that's not exactly saying much :) |
| 23:26.14 | hardhatpat | mchou, yeah tell me about it ... the BB leaves something to be desired when it comes to voice quality |
| 23:26.15 | mchou | hardhatpat: blackberry shouldnt be used as a baseline |
| 23:27.13 | hardhatpat | mchou, did you read what i wrote when i joined the channel? about my * server needing to be rebooted every 12-36 hours? |
| 23:27.25 | mchou | hardhatpat: nope, sorry |
| 23:27.48 | hardhatpat | when it just sits running, after a while i will need to reboot * ... is this a common thing? |
| 23:27.50 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
| 23:27.59 | mchou | hardhatpat: hell no |
| 23:28.13 | chandoo | hardhatpat: cant i have gateway setup in my machine |
| 23:28.36 | hardhatpat | chandoo, what do you mean? |
| 23:29.11 | hardhatpat | [Jun 19 16:28:43] NOTICE[5386]: chan_sip.c:13865 handle_request_invite: Call from '' to extension '5033341400' rejected because extension not found. |
| 23:29.32 | hardhatpat | and if i 'reload' it just starts working again |
| 23:29.32 | chandoo | hardhatpat: all i want is to make calls from my machine to other phones, what i need? do i need any hardware |
| 23:29.58 | hardhatpat | chandoo, you need $$ to pay a voip provider other than your current gateway (unless they will accept calls from *) |
| 23:30.15 | chandoo | hardhatpat: pardon my silly question , i am new to asterisk |
| 23:30.30 | hardhatpat | chandoo, I figured, everyone starts somewhere. |
| 23:30.48 | hardhatpat | if your voip provider dosent provide you instructions on how to setup * they probably wont accept calls from it |
| 23:30.53 | mchou | hardhatpat: how $ does voipstreet charge for DIDs? |
| 23:31.10 | mchou | hardhatpat: well, I wouldnt go that far :) |
| 23:31.46 | mchou | hardhatpat: it just means it isn't supported. Which is different than "impossible" :) |
| 23:32.11 | chandoo | i think i buy a pci modem and use it as softphone is the other alternative to make outside calls |
| 23:32.17 | mchou | hardhatpat: i.e. too costly to support |
| 23:32.41 | mchou | chandoo: huh? |
| 23:32.57 | hardhatpat | mchou, voipstreet has two different plans: per minute and unlimited incoming |
| 23:33.23 | hardhatpat | per minute: $3 setup fee and $2.95/month + $.0145/minute |
| 23:33.27 | chandoo | do voipstreet allow calls what we are taking about |
| 23:33.52 | hardhatpat | or $5 setup fee and $9.95/month for unlimited origination still $.0145 for termination |
| 23:34.03 | mchou | hardhatpat: the per minute part, that's inbound and outbound?? |
| 23:34.05 | hardhatpat | chandoo, thats ALL voipstreet does |
| 23:34.21 | hardhatpat | yeah $.0145/minute for inbound and outbound |
| 23:34.40 | mchou | hardhatpat: I dont think that's such a great dea |
| 23:34.44 | mchou | deal* |
| 23:34.56 | chandoo | can i also have a regular phone? do they send modem too |
| 23:35.03 | mchou | ideally for inbound it should be $0 |
| 23:35.05 | hardhatpat | I WAS paying junction networks $2/month for my DID + $6 setup fee + $.029/minute |
| 23:35.31 | hardhatpat | mchou, I know ... because they get paid $.01/minute for incoming |
| 23:35.41 | hardhatpat | mchou, who do you use? |
| 23:35.43 | mchou | hardhatpat: ever look at diamondcard.us? |
| 23:36.18 | mchou | hardhatpat: cheesy web site, but great service |
| 23:36.33 | hardhatpat | hows the support? |
| 23:36.40 | mchou | hardhatpat: not too bad |
| 23:36.48 | mchou | they work saturdays :) |
| 23:36.50 | hardhatpat | that is the ONE thing that I like about voipstreet, is any time i call the support # I get a person |
| 23:36.59 | chandoo | right now i am paying $21/mo for unlimited calls |
| 23:37.06 | chandoo | from Phone Power |
| 23:37.12 | hardhatpat | i dont leave a message and wait for a call back, i talk to someone |
| 23:37.18 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:37.20 | hardhatpat | chandoo, thats a LOT |
| 23:37.36 | hardhatpat | mchou, do you know if they will port DIDs? |
| 23:37.43 | mchou | hardhatpat: it's pretty much the same for diamondcard (although I've only called them once) |
| 23:37.50 | chandoo | hardhatpat: LOL that is what i am doing since |
| 23:38.11 | mchou | hardhatpat: they will port your own # (not sure about DIDs) |
| 23:38.41 | mchou | hardhatpat: I assume that's what you meant |
| 23:38.45 | chandoo | one confusion, if i take service from voipstreet do they voip modem? so that i can connect regular phone |
| 23:38.56 | hardhatpat | mchou, I had my DID ported from junctionnetworks to voipstreet, they will port it from voipstreet to diamondcard.us ... right? |
| 23:39.20 | mchou | hardhatpat: I dont know what you mean.... |
| 23:39.23 | chandoo | for home usage |
| 23:39.41 | hardhatpat | chandoo, you are talking about an ATA ... you could set one up for your home phones if you wanted ... but asterisk would handle it |
| 23:39.42 | mchou | hardhatpat: where did the "original" phone # come from? |
| 23:39.50 | hardhatpat | junctionnetworks |
| 23:40.05 | hardhatpat | then it was ported to voipstreet, and if i left voipstreet i would leave with that # |
| 23:40.11 | mchou | hardhatpat: you'd probably have to ask |
| 23:40.31 | hardhatpat | mchou, as far as i understand the porting legislation ... they HAVE to port it |
| 23:40.45 | chandoo | hardhatpat: do i need to have special hardware to setup, |
| 23:40.49 | hardhatpat | $24/year is a good deal |
| 23:41.06 | hardhatpat | chandoo, you might be able to use the ATA that your current provider sent you, or you may be hitting up eBay |
| 23:41.31 | chandoo | ATA means voip modem am i right |
| 23:41.36 | hardhatpat | mchou, i serioulsy have to fix my problem with asterisk not accepting calls after a while ... where should I look? |
| 23:41.50 | hardhatpat | chandoo, if by voip modem you mean, ATA ... yes |
| 23:41.54 | chandoo | right now i have linksys broadband router with two phone connections |
| 23:42.15 | dlynes | Does anyone know of a company called 'Canadian Operator' where you need to dial '090' to reach them? |
| 23:42.56 | hardhatpat | how did you setup your linksys router to talk to your provider? |
| 23:44.37 | chandoo | hardhatpat: Internet <-- BroadBand connection(cable vision)<-- linksys router (Phone Power)<-- Dlink Wireless router(for wifi) |
| 23:44.57 | *** join/#asterisk frieze (n=frieze@pool-71-251-13-242.nycmny.fios.verizon.net) |
| 23:45.12 | *** join/#asterisk giovani2 (n=giovani@unaffiliated/giovani) |
| 23:45.47 | chandoo | my dektop is connected to linksys router with two 1GB nic |
| 23:45.55 | hardhatpat | hold on |
| 23:46.04 | giovani2 | hi, I'm using trixbox 2.4.2.0 -- and any internal call (to voicemail, echo test, etc) is producing no audio from the server -- no obvious errors in the full log, server claims to be playing recording, but pcap shows nearly no RTP packets from server during call |
| 23:46.20 | giovani2 | outbound/inbound calls to the PSTN via a voip provider on sip work fine |
| 23:46.26 | giovani2 | all of the handsets are local, no firewall |
| 23:46.30 | hardhatpat | chandoo, did phone power sent you the linksys router? |
| 23:46.32 | chandoo | Uniden Phone is connected to linksys router phone slots |
| 23:46.37 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 23:46.39 | chandoo | hardhatpat: yes |
| 23:46.43 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:47.02 | hardhatpat | chandoo, are they going to want it back? |
| 23:47.14 | giovani2 | I'd appreciate any advice someone can offer |
| 23:47.59 | frieze | anyone have chan_mobile running with 1.4.21? |
| 23:48.02 | bbryant | ~sipnat |
| 23:48.03 | jbot | [~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 |
| 23:48.10 | bbryant | giovani2: that should help |
| 23:48.14 | chandoo | even if they want it back i still have old linksys voip modem and other voip modem from motorola |
| 23:48.15 | giovani2 | bbryant, there's no NAT |
| 23:48.26 | giovani2 | as I said, handsets are local -- directly on the switch with the server |
| 23:48.26 | chandoo | hardhatpat: |
| 23:48.51 | giovani2 | in fact, the only place where there is nat, which is outbound, the audio is working perfectly fine |
| 23:49.04 | bbryant | giovani2: ah, try #freepbx |
| 23:49.06 | hardhatpat | chandoo, hold on im going to go out for a smoke |
| 23:49.15 | bbryant | #asterisk doesn't support trixbox |
| 23:49.15 | blitzrage | is it trying to reinvite the media and failing? |
| 23:49.17 | giovani2 | bbryant, you think this is specific to freepbx? |
| 23:49.25 | chandoo | hardhatpat: i will be back from 40min jogging |
| 23:49.39 | blitzrage | I'd suggestion making sure canreinvite=no on all phones |
| 23:49.47 | hardhatpat | chandoo, pm me and ill help you out later |
| 23:50.01 | chandoo | okay hardhatpat |
| 23:50.06 | giovani2 | blitzrage, alright, I'll try that -- how would I determine if it's trying to reinvite? |
| 23:50.07 | blitzrage | s/on all phones/for all devices configured in sip.conf |
| 23:50.14 | blitzrage | s/on all phones/for all devices configured in sip.conf/ |
| 23:50.19 | blitzrage | lol |
| 23:50.29 | blitzrage | I'd suggestion making sure canreinvite=no on all phones |
| 23:50.31 | blitzrage | s/on all phones/for all devices configured in sip.conf/ |
| 23:50.52 | blitzrage | you'll have to figure out how trixbox does that |
| 23:51.08 | giovani2 | I'll wade through it |
| 23:51.42 | scooby2 | Does a call show up as Agent Dump if they get hung up on and would that not show up as a connected/disconnected call? |
| 23:51.55 | *** join/#asterisk map7 (n=user@124-168-127-223.dyn.iinet.net.au) |
| 23:53.49 | map7 | I'm new to asterisk and have a problem where I cannot dial out to mobile phones, but I can call local phones. Which configuration file should I start at? |
| 23:54.20 | Idle | ugh, playing 'stump the digium tech', aparently... stupid PRI |
| 23:54.29 | giovani2 | blitzrage, is it not telling that this only happens when calling the server, and not on outbound calls? or rather, not even on calls between local extensions? |
| 23:54.42 | frieze | for 1.4.21 should I be checking addons out of trunk on the svn? |
| 23:55.01 | blitzrage | it's telling it to not redirect the media between the phones, but rather, keep the audio running through Asterisk |
| 23:55.14 | giovani2 | no no, I didn't mean that |
| 23:55.20 | giovani2 | nevermind |
| 23:58.25 | giovani2 | blitzrage, that was already in both of the extension definitions |
| 23:59.34 | giovani2 | what I meant earlier was, I'm wondering if it's some problem with audio playing in asterisk -- since both extensions can call each other and hear each other fine, as well as to outbound sip connections -- it's just internal audio that doesn't play |