| 00:01.52 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:05.24 | De_Mon | grrr, my phpagi is broke. its not reading the phpagi.conf but I duno why :( |
| 00:05.52 | *** join/#asterisk PhilCiccone (n=pciccone@ip66-104-145-162.z145-104-66.customer.algx.net) |
| 00:07.05 | PhilCiccone | I am in fairly desperate need of help on a new (now production) asterisk server. I have a PRI negotiaton problem that after 6 hours now I cannot solve. Anyone avail who is good in this area? |
| 00:07.14 | *** join/#asterisk ManxPower (n=manxpowe@015-797-116.area5.spcsdns.net) |
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| 00:12.17 | snuff-work | mm sfb2, you could raise that with devs.. generally i thought it would be more the form of 'sip set debug' and sip set no debug |
| 00:15.26 | De_Mon | FRACK |
| 00:15.36 | De_Mon | stupid permission errors |
| 00:15.39 | sfb2 | I guessed sip unset debug |
| 00:15.53 | sfb2 | or sip set debug as a toggle |
| 00:19.26 | ManxPower | sip no debug |
| 00:20.00 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:20.01 | ManxPower | at least for 1.0.x and 1.2.x |
| 00:20.45 | Phrozen_One | hows everyone doing tonight |
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| 00:24.03 | andresmujica | hi, i'm hackig app_addon_sql_mysql.c so i can pass the port option to mysql_real_connect but i'm getting problems trying to pass the port value to the MYSQL_exec function.. but it fails...... |
| 00:24.58 | [TK]D-Fender | bcnl, Still need help? |
| 00:28.08 | *** join/#asterisk Tond (n=t@74.122.241.161) |
| 00:28.08 | bcnl | [TK]D-Fender: no thanks, it got figured out |
| 00:28.29 | Tond | Hi does anyone know why i am getting this error: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available ? |
| 00:28.50 | Tond | the Mysql db is setup and all tables created and the config files are done correctly also |
| 00:29.15 | Tond | howevere asterisk is having problems connecting to MySql for my sip users and peers |
| 00:29.41 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 00:30.28 | snuff-work | you tried following a guide from like http://voip-info.org |
| 00:30.29 | snuff-work | ? |
| 00:30.38 | Tond | Yes |
| 00:30.50 | Tond | I even did ./configure --with-mysqlclient=/usr since mysql was installed using yum |
| 00:31.00 | Tond | in astersik-addons |
| 00:31.02 | snuff-work | you have unixodbc ? |
| 00:31.15 | Tond | no, MySql is installed on the localbox |
| 00:31.44 | Tond | in /var/lib/mysql/mysql.sock |
| 00:38.07 | [TK]D-Fender | bcnl, Yeah, you can't just start a patter with a CID restriction on it like that in the middle :) |
| 00:38.20 | [TK]D-Fender | bcnl, Need to start from priority 1 |
| 00:40.08 | *** part/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 00:45.25 | Uatec | oh that reminds me |
| 00:45.35 | Uatec | i need to get my asterisk box logging to my mssql server |
| 00:50.24 | Tond | when i run ./configure in astersik-addons, it says checking for asterisk.h... no |
| 00:50.34 | Tond | how come it can't find it? |
| 00:51.25 | Tond | i ahve asterisk instaleld |
| 00:56.12 | [TK]D-Fender | Tond, and HOW did you install it? |
| 01:01.34 | Tond | tk> i did everything and anything, i did, ./configure, make, make install |
| 01:01.44 | Tond | I did just make and make install |
| 01:02.14 | Tond | i even copied the asterisk.h file there since in the ./configure it was saying asterisk.h ... no |
| 01:02.24 | Tond | i don't knwo what the hell is wrong here... :S |
| 01:15.51 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 01:24.16 | *** join/#asterisk ta^3 (n=tacvbo@189.146.184.26) |
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| 01:32.01 | _DAW | is there a way to set sip t1 timer in polycom phones? |
| 01:56.10 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 01:56.24 | kiwoneka | good eveing to all |
| 01:57.43 | kiwoneka | is there a public place to get access to the latest software and bootrom for the polycom ip650 |
| 02:05.06 | *** join/#asterisk Ebola (n=Ebola@host86-143-168-67.range86-143.btcentralplus.com) |
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| 02:09.31 | Sci_05 | there is one site, but it might not be the latest kiwoneka |
| 02:09.56 | Sci_05 | they only want people that are register service people to have access to it |
| 02:10.38 | kiwoneka | i know, i am trying to get certified |
| 02:10.51 | Sci_05 | kiwoneka: check here http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+501 |
| 02:11.05 | Sci_05 | there is alink on that page to some polycom software |
| 02:11.24 | Sci_05 | kiwoneka: I need to get certitifed only its a pain to get done |
| 02:11.47 | Sci_05 | of course I don't have a lot of time to follow up and see whats gong on with it either |
| 02:12.00 | kiwoneka | thanks |
| 02:12.15 | kiwoneka | i ihave about 6 courses done |
| 02:12.46 | *** join/#asterisk DaPrivateer (i=Privatee@66.92.79.20) |
| 02:13.02 | Sci_05 | what one are you going for? I thought you only had to pass 2 tests to get access to the firmware? |
| 02:13.12 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 02:21.35 | kiwoneka | more than that |
| 02:21.56 | kiwoneka | some cources have like 5 modules |
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| 02:39.24 | ManxPower | Polycom has the one version before the most recent available to anyone via their web site |
| 02:39.39 | ManxPower | If you want the latest the you have to get it from a Polycom reseller |
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| 03:42.56 | *** join/#asterisk monstertruck (n=monstert@c-76-26-50-233.hsd1.fl.comcast.net) |
| 03:47.22 | monstertruck | anybody tried a grandstream gxw4008 ? |
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| 03:59.31 | sweeper | monstertruck: run away |
| 04:05.05 | monstertruck | sweeper, from grandstream? |
| 04:05.13 | sweeper | yes |
| 04:05.44 | monstertruck | sweeper, i dont need it for a long lasting application |
| 04:05.52 | monstertruck | is going to be in service for about a year |
| 04:06.19 | monstertruck | im considering it because of ilbc support |
| 04:06.21 | sweeper | half that time will be spent gettting it to work imo :v |
| 04:06.43 | monstertruck | really? |
| 04:07.10 | monstertruck | which 8 port gateway would you recommend? |
| 04:07.20 | sweeper | well, I had a really, really, really bad experience with their FXO dealy |
| 04:08.02 | monstertruck | i havent used any, only experience with GS are the budgetone phones |
| 04:08.17 | monstertruck | have had 12 for about 5 months |
| 04:08.48 | monstertruck | not too many problems, they do have issues from time to time and need to reboot them |
| 04:09.20 | monstertruck | but then again those are at a remote clients site, so rebooting them is the easiest way to fix them |
| 04:10.12 | monstertruck | i would buy an spa 8000 but bandwidth is a problem |
| 04:12.37 | coppice | all the ills of the VoIP world seem to be blamed on grandstream. :-) |
| 04:13.14 | Strom_M | not all of them |
| 04:13.23 | Strom_M | merely 99.999% of them |
| 04:13.24 | monstertruck | haha |
| 04:13.47 | Strom_M | the rest can be conveniently blamed on superstring theory |
| 04:14.28 | Qwell | "I get bad audio quality on my SIP calls." "Do you have a grandstream?" "Yes" "Are you using it?" "No" "Is it plugged into the network?" "No" "Is it on?" "No..." "Replace it, and try again" |
| 04:15.54 | monstertruck | thats good to know .. then some of my problems havent been incorrect asumptions put into an agi script .. but instead superstring theory |
| 04:15.59 | monstertruck | im so relieved |
| 04:16.10 | monstertruck | :) |
| 04:17.37 | monstertruck | thats a bummer, given that i probably wont find any other gateway that supports ilbc or gsm |
| 04:18.20 | Strom_M | what about......g.729 |
| 04:18.21 | monstertruck | spa3102 supports g726-32, which is ok, but im not sure spa8000 supports it |
| 04:18.23 | Juggie | Qwell, why is DIAL allowed from agi, does it actually work properly at all? |
| 04:18.29 | Qwell | no idea |
| 04:18.34 | monstertruck | trying to stay away from licenses |
| 04:18.35 | Qwell | I don't use AGI |
| 04:18.41 | Strom_M | monstertruck: it's eight channels |
| 04:18.50 | Strom_M | and everything speaks it |
| 04:19.00 | Strom_M | also passthrough |
| 04:19.14 | Strom_M | but not iced tea |
| 04:19.28 | monstertruck | dial works very well from AGI |
| 04:19.37 | Qwell | Strom_M: You're in the South - it's sweet tea |
| 04:19.50 | monstertruck | most of my implementation is based on agi, it works well |
| 04:19.51 | Juggie | monstertruck, dial usually terminates the call when it exits though, does the g option work from agi? |
| 04:20.23 | Strom_M | Qwell: cohujibuggle |
| 04:20.27 | monstertruck | Juggie, works exactly as it works from a dialplan |
| 04:20.46 | monstertruck | you can call macros, transfers .. whatever |
| 04:21.02 | Juggie | monstertruck, intreasting, i thought i had problems with it before but maybe that was a long long time ago. |
| 04:21.37 | Juggie | the last thing that really bugged me w/ agi was the default build of php not being able to trap a sighup |
| 04:21.43 | Juggie | so you coudnt cleanup after a hangup |
| 04:22.01 | Juggie | not without starting the agi again in h as deadagi |
| 04:22.13 | Juggie | of course deadagi also works on an active channel for some reason, but doesnt work well. |
| 04:23.34 | monstertruck | Juggie, yeah, you are right about that, but having a clean up script running with deadagi is not so bad, except for having to start a new process |
| 04:23.40 | Juggie | yeah |
| 04:23.50 | monstertruck | if you are concious about resources |
| 04:24.34 | Juggie | well, at the time i used php/agi to write ivrs |
| 04:24.42 | Juggie | so it was handy to be able to clean up, write stats etc. |
| 04:24.51 | Juggie | its someone elses project now, and its also in c/fastagi |
| 04:24.55 | Juggie | so thats no longer a problem |
| 04:25.23 | Juggie | fastagi is much nicer, handle multiple calls in a single thread |
| 04:25.42 | Juggie | rather then one thread per call. |
| 04:27.28 | monstertruck | yup |
| 04:27.52 | monstertruck | on a different thread, spa8000 is not out yet |
| 04:28.01 | monstertruck | so im out of luck there |
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| 04:52.51 | nclx | Can anyone recommend a good quality VoIP provider in the US which will interface with my existing asterisk system, I am looking for multiple concurrent calls (at least 6) |
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| 05:32.27 | remmo | moan |
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| 05:35.07 | kingsob | my iax providor says this should be the 1st line in my extensions.conf file for incoming calls.. exten => 6135551212,1,SetAccount(6135551212), but i'm getting this error in my logs... pbx.c:1797 pbx_extension_helper: No application 'SetAccount' for extension ... |
| 05:36.37 | kingsob | any ideas? |
| 05:39.48 | kingsob | is it even necessary?? if i just rmeove the line it all works great |
| 05:40.59 | snuff-work | mm just use this.. |
| 05:41.17 | snuff-work | set(CDR(accountcode)=${EXTEN}) |
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| 06:55.21 | [TK]D-Fender | kingsob, No. |
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| 07:43.56 | bkruse_home | ~seen jcmoore |
| 07:44.16 | jbot | jcmoore <n=jcmoore@unaffiliated/tgrman> was last seen on IRC in channel #asterisk, 36d 3h 20m 28s ago, saying: 'GotoIf($[!${ISNULL(${BridgePIN})]?3:5)'. |
| 07:44.16 | bkruse_home | jbot come on! |
| 07:44.17 | jbot | oh alright then.. |
| 07:45.33 | bkruse_home | hmm |
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| 12:00.58 | twodelta | hey guys - is this an appropriate channel for discussing branches/patches? |
| 12:02.53 | twodelta | (we're after a little help on how to merge a branch (oej/cancel_elsewhere_1.4) into the current 1.4.7.1 build (as a diff patch? - not very familiar with svn :) |
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| 12:14.51 | _Raptor_ | hello |
| 12:16.31 | _Raptor_ | can anyone tell me something about this problem: |
| 12:16.41 | _Raptor_ | [2007-07-14 14:11:13] WARNING[20955]: chan_misdn.c:2917 misdn_request: Could not create channel on port:1 with extensions:0913191XXXXX |
| 12:16.41 | _Raptor_ | [2007-07-14 14:11:13] NOTICE[20955]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'mISDN' (cause 0 - Unknown) |
| 12:17.39 | *** join/#asterisk k31th (n=keith@cartman.nzsolutions.net) |
| 12:18.41 | k31th | how can i check to see if a UDP port is forwarded ? I am attempting to forward 5060 and the other sip ports thru a nat router however i suspect its the router at fault here. Is there some way i can check? if this was tcp i would use telnet... |
| 12:18.41 | Nugget | telnet is eeeeeeevil! |
| 12:21.04 | fetcher | k31th: you might try 'netcat' (nc) to send a test packet, and see if an ICMP Unreachable comes back (may need to run tcpdump at the same time), but that isn't foolproof. Many routers will silently filter packets without returning an Unreachable |
| 12:21.58 | k31th | damn |
| 12:22.03 | fetcher | k31th: there's nothing as straightforward as telnet, since UDP has no concept of a "connection" (anything beyond fire-and-forget has to be implemented by the application) |
| 12:22.22 | k31th | yeah its connectionless right |
| 12:22.34 | k31th | surly udp is old hat? |
| 12:23.38 | fetcher | it has advantages for VoIP and other realtime streaming, where TCP's behavior (designed for bulk data transfer) is less than helpful |
| 12:24.39 | fetcher | e.g. if a single packet is lost, TCP will keep retrying it, freezing up the entire connection until the missing one is acknowledged. Not good for voice, where a brief drop-out would be much preferred |
| 12:24.45 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 12:25.55 | _Raptor_ | [2007-07-14 14:24:42] WARNING[21119]: misdn_config.c:664 _build_port_config: misdn.conf: "max_incoming=1" (section: default) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". |
| 12:26.14 | _Raptor_ | can anyone explain to me why 1 is out of range for maximum incoming calls? |
| 12:26.24 | _Raptor_ | and the same for outgoing |
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| 12:43.24 | tzafrir_home | fetcher, what you say there is not accurate: SIP is only used for signalling. SIP's audio is passed as RTP in a separate connection which is indeed UDP for the reasons you mentioned |
| 12:44.07 | tzafrir_home | SIP (according to specs) can be sent over TCP. Asterisk doesn't support this yet, though |
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| 13:48.17 | rob0 | -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Unable to do INITIAL ProSLIC powerup on module 0 |
| 13:49.03 | rob0 | This was working, only thing that has changed: no phone plugged in on FXS module 0. |
| 13:49.21 | Strom_M | that shouldn't make a difference |
| 13:49.35 | rob0 | perhaps a failed FXS module? |
| 13:51.05 | rob0 | hmmm, looking at the config, no, the failed one is the one with the phone. |
| 13:51.14 | _DAW | Hell all, any polycom users know if it is possible to adjust the Sip T1 timer? I am testing one over a satellite and it is an issue. |
| 13:51.38 | Strom_M | rob0: is that the only fxs module? |
| 13:51.48 | Strom_M | _DAW: how so? |
| 13:52.04 | *** join/#asterisk kickbackit (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 13:52.19 | _DAW | The latency of the satellite is causing retransmits cause the timer is only set at 500ms or so. |
| 13:52.32 | Strom_M | "only"?? |
| 13:52.43 | Strom_M | total round-trip latency on a phone call should never exceed 400ms |
| 13:53.18 | _DAW | I know, but this is vsat so 500ms is actually a great round trip. Its usually 6-700. |
| 13:53.33 | _DAW | and actually works pretty well excluding this timer issue. |
| 13:54.53 | _DAW | Ive tried several other devices that allow changing the timer and that fixes it right up. Just could not find it in the polycom. |
| 13:58.08 | rob0 | 2 FXS, 1 FXO, 1 empty slot |
| 14:02.27 | *** join/#asterisk Strom_C (n=strom@12.175.45.209) |
| 14:02.31 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 14:06.55 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 14:07.59 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:11.15 | *** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au) |
| 14:12.31 | Penggu | hi all |
| 14:13.08 | kiwoneka | good morning to all |
| 14:13.33 | Penggu | could asterisk be used as a sip client to accept calls from say, another asterisk PBX server, for the purposes of accepting various inputs by callers (including audio) for further processing? |
| 14:13.44 | Penggu | eg an IVR, but separate from the geenral PBX server itself |
| 14:13.47 | Penggu | would be an over kill? |
| 14:14.39 | Strom_C | yes |
| 14:14.41 | Strom_C | that's doable |
| 14:14.44 | Penggu | eg. i was thinkig to make a 'bell timer' or 'paging' server (with a sound card), taht will for eg accept calls, take the recoding, and then play it over the sound card (to the PA system), or have ian interactive menu to set up a schedule, etc |
| 14:14.57 | Strom_C | it's called a feature server |
| 14:15.31 | Penggu | would asterisk itself be recommended for that, or are there (Easy to use, well documented) programs out there? |
| 14:15.45 | Strom_C | use asterisk |
| 14:16.54 | Penggu | the other alternative i would have is to set up wnidoze xxx and use scheduled tasks (with all the remoteness that comes with it), but asterisk/linux with a web interface would be elegant |
| 14:17.18 | Penggu | and could be squashed into a small footprint |
| 14:17.32 | Strom_C | yeah, i can't see why you'd want to do that with windows :) |
| 14:18.28 | Penggu | Strom_C: u know anything about setting up an embedded device, small scale? eg buy a 'bare box' with a bit og grunt enough to be a server and just chuck on some things? |
| 14:18.54 | Penggu | like, a dsl modem/router without the dsl part |
| 14:19.01 | Penggu | and may be perhaps with a sound card |
| 14:19.04 | Strom_C | for your feature server? |
| 14:19.07 | Penggu | yeh |
| 14:19.15 | Strom_C | why must this be a separate device in the first place? |
| 14:19.50 | Penggu | well, in our case, the pbx is somewhere |
| 14:19.56 | Penggu | and the pa system is somewhere else |
| 14:20.21 | Penggu | (could prolly get a phone with auto-answer and hook up the ear piece to the audio input, hey?) |
| 14:22.20 | MrMister2 | I have paging working fine with a extension that dials "console/dsp" but want to be able to control access to it. what would be my options? |
| 14:22.46 | MrMister2 | only allow some extensions to dial the paging extension or allow all but ask for a code. |
| 14:22.53 | Strom_C | MrMister2: control access to it just as you control access to any other part of your dialplan |
| 14:22.54 | MrMister2 | any ideas on how to do it? |
| 14:23.23 | MrMister2 | I'm using Trixbox since it is easier for me but not afraid to mess with the dialplan if I get some hints :) |
| 14:23.34 | Strom_C | oh christ |
| 14:23.38 | MrMister2 | LOL |
| 14:23.39 | Strom_C | you don't want to mess with that dialplan |
| 14:23.54 | Strom_C | it'll make your head spin and then you'll mess it up anyway |
| 14:24.03 | Strom_C | ask in #trixbox or learn asterisk :) |
| 14:24.12 | MrMister2 | ok. any ideas on how to go abou it then? |
| 14:24.16 | *** join/#asterisk kickbackit (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 14:24.29 | Strom_C | step 1: remove trixbox |
| 14:24.31 | MrMister2 | well, I'm trying to learn * since that way I won't be limited by trixbox |
| 14:24.44 | Strom_C | step 2: learn how to use contexts |
| 14:24.49 | Strom_C | step 3: there is no step 3 |
| 14:25.11 | MrMister2 | I'm actually going about it the worst way since I'm testing stuff on astlinux :) |
| 14:28.13 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 14:35.02 | kickbackit | My asterisk box is behind the corporate firewall along with 99% of the phones. I have 4 phones that are remote offices. I have pointed those phones at the firewall server and forwarded UDP ports 5004-5060 and 10000-20000 to the voip server. The external phones can get audio from the voip server, but cannot complete calls or hit keys when the ivr prompts for them. Am I missing a port? |
| 14:35.54 | Strom_C | if they cannot complete calls, then how are they completing calls to the IVR? |
| 14:36.15 | kickbackit | If they dial a land line they cannot get audio or send audio to the land line |
| 14:36.38 | kickbackit | I have an extention that sends them to the public ivr |
| 14:36.43 | kickbackit | they can hear that fine |
| 14:36.45 | Strom_C | can they complete calls and talk to sip phones internally? |
| 14:37.00 | kickbackit | along with anything coming from the voip such as music on hold etc |
| 14:37.23 | *** join/#asterisk msetim (n=marcos@201-14-60-90.ctame706.dsl.brasiltelecom.net.br) |
| 14:37.37 | kickbackit | I have only tried remote to remote phone since no one is in the office yet |
| 14:38.06 | Strom_C | go into the office or wait till there is someone there |
| 14:38.11 | Strom_C | then continue asking |
| 14:38.39 | kickbackit | We have a queue/agent ivr setup, with callback logins. It prompts for a passcode wich the remote phones cannot enter. |
| 14:38.47 | Strom_C | yes |
| 14:38.47 | kickbackit | this something with dtmfmode? |
| 14:38.57 | Strom_C | it might be, or it might be a UDP issue |
| 14:39.21 | Strom_C | without knowing whether they can successfully talk with a human on the inside, it's tough to narrow it down |
| 14:40.00 | perf3kt | is there going to be an update on the book? |
| 14:40.15 | Strom_C | perf3kt: yes |
| 14:40.17 | Strom_C | very soon |
| 14:41.08 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 14:41.09 | perf3kt | cool, will it be free>? of course I don't mind buying I have the first oreily |
| 14:41.27 | Strom_C | perf3kt: i would assume so, but I don't know offhand |
| 14:41.29 | Penggu | is that the tfot book? |
| 14:41.41 | Strom_C | didn't have a chance to ask Leif when I saw him yesterday :) |
| 14:42.06 | perf3kt | penggu: that's what I was talking about so yeah I hope so |
| 14:42.50 | Penggu | 2nd ed? |
| 14:42.56 | Penggu | or vol2? |
| 14:43.22 | perf3kt | so I have a theory question |
| 14:43.31 | perf3kt | there is * cli - you guys |
| 14:43.31 | Strom_C | ask away |
| 14:43.47 | perf3kt | then *now with the gui interface |
| 14:44.03 | perf3kt | then a couple of others freepbx, elastix, trixbox |
| 14:44.09 | perf3kt | why? |
| 14:44.40 | Strom_C | the others were written before the digium gui |
| 14:44.57 | perf3kt | okay |
| 14:45.08 | Strom_C | they're mostly attempts to kludge a GUI onto a GUI-less piece of software |
| 14:45.11 | perf3kt | but you guys the cli groups still dispise all gui |
| 14:45.23 | Strom_C | the digium GUI is a fundamental change in the asterisk architecture |
| 14:45.40 | perf3kt | users.conf? |
| 14:45.44 | Strom_C | yes |
| 14:46.06 | perf3kt | see my months on the boards hav not been in vain |
| 14:46.25 | Strom_C | heh |
| 14:46.37 | Strom_C | well, the digium GUI actually looks promising |
| 14:46.44 | perf3kt | but the digum gui is the only one liek that |
| 14:46.57 | Strom_C | i've been playing with the asterisk appliance recently, and it's actually quite usable |
| 14:47.13 | tzanger | Strom_C: howso? |
| 14:47.18 | tzanger | the gui reads and writes native formats |
| 14:47.27 | tzanger | I I bet |
| 14:47.28 | perf3kt | it just seems that with the wealth of knowledge that the cli group has it would onl p advance the digium gui |
| 14:47.54 | perf3kt | yeah they have the conf editor, sorts according to the context |
| 14:48.15 | Strom_C | perf3kt: i'd like to play around with the gui more and see what I can do with it |
| 14:48.20 | perf3kt | but the groups are so separate |
| 14:48.22 | Strom_C | i just havent had the time to do so |
| 14:48.29 | perf3kt | i understand |
| 14:48.48 | Strom_C | perf3kt: well also, any GUI is going to limit your flexibility |
| 14:48.59 | perf3kt | honestly the only reason I went to the gui |
| 14:49.19 | perf3kt | is because I'm not a linux guy, and the install with regard to dependencies and all was mind-boggling |
| 14:49.31 | Infested | heh |
| 14:49.54 | perf3kt | but on the other hand you say I can pop is a cd that installs * and everything, okay |
| 14:50.14 | perf3kt | but that is only offered with the gui guys |
| 14:50.37 | Strom_C | well, what do you want - flexibility or ease of use? |
| 14:50.38 | Strom_C | pick one :) |
| 14:51.03 | Strom_C | flatten the learning curve and you lose flexibility |
| 14:51.07 | perf3kt | what loss of flexibility is there? |
| 14:52.10 | Strom_C | well, the gui dictates a certain method of doing things |
| 14:52.15 | *** join/#asterisk friedrich| (n=friedric@e177244140.adsl.alicedsl.de) |
| 14:52.20 | mvanbaak | a gui only allows you to do stuff that is implemented in the gui and it will only allow you to fiddle with it in the way the gui author decided |
| 14:52.51 | *** join/#asterisk kickbackit (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 14:55.06 | perf3kt | and I'm just asking to gain understanding |
| 14:55.39 | kickbackit | Strom_C, in my situation is there a better method then just forwarding the packets on to the voip? Maybe something like a Sip Proxy? |
| 14:55.55 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:55.55 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:56.03 | Strom_C | kickbackit: that might be a very good idea |
| 14:56.09 | Strom_C | hi blitzrage |
| 14:56.14 | blitzrage | yo yo |
| 14:56.37 | blitzrage | anyone know if I was right about the 's' priority for bcnl's issue? |
| 14:56.50 | Strom_C | blitzrage: recap |
| 14:56.52 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 14:56.55 | blitzrage | that's what the issue looked like, and gave me an idea of how to use the 's' priority |
| 14:57.40 | blitzrage | he was trying to match on a CID at a certain priority, but had 3 different CID matches that he wanted to use at the same priority |
| 14:57.44 | blitzrage | pretty sure I was right :) |
| 14:58.46 | Strom_C | so something like exten => s/3115552368,n,NoOp(catsex) ? |
| 14:59.03 | blitzrage | yes |
| 14:59.07 | blitzrage | but replace 'n' with 's' |
| 14:59.14 | blitzrage | (for 'same') |
| 14:59.55 | Strom_C | oh, i haven't heard about that |
| 15:00.14 | blitzrage | ya... I didn't know about that priority until I was writing the SECOND edition of TFoT :) |
| 15:00.25 | Strom_C | haha |
| 15:00.34 | Strom_C | when are you leaving town? |
| 15:01.29 | Strom_C | i was thinking of a waffle house run before my flight out of huntsville :) |
| 15:01.35 | blitzrage | I'm already in Toronto :) |
| 15:01.37 | blitzrage | I left yesterday |
| 15:01.42 | Strom_C | hah, ok |
| 15:01.48 | Strom_C | oh that's right, durh |
| 15:01.52 | blitzrage | hehehe |
| 15:02.05 | Strom_C | you were rushing out of the office at 4ish |
| 15:02.42 | Qwell | Strom_C: when you leaving? |
| 15:02.55 | Strom_C | Qwell: 2:50 IIRC |
| 15:03.08 | file | Strom_C: Der Waffle Haus? |
| 15:03.18 | Strom_C | DAS WAFFLE HAUS |
| 15:03.20 | mvanbaak | oh no! ze germanz ! |
| 15:03.24 | Strom_C | LA CASA DE WAFFLE |
| 15:03.37 | Strom_C | Qwell: 2:49 !!! |
| 15:03.47 | Qwell | off-by-one error |
| 15:03.53 | Strom_C | yes |
| 15:04.19 | mvanbaak | I have that with the asterisk count here |
| 15:04.31 | mvanbaak | came home after helping my lil bro move to his new house |
| 15:04.42 | mvanbaak | to find out on of my asterisk dev boxen has died |
| 15:05.18 | *** join/#asterisk friedrich| (n=friedric@e177244140.adsl.alicedsl.de) |
| 15:05.27 | *** join/#asterisk Op3r (n=op3r@121.97.214.210) |
| 15:05.28 | *** join/#asterisk wunderkin (i=wunderki@ip68-2-61-64.ph.ph.cox.net) |
| 15:05.32 | mvanbaak | those round big rolls on the motherboard. I have a row of them without roof now |
| 15:06.28 | Qwell | blown caps? |
| 15:07.24 | perf3kt | must be a dell |
| 15:07.25 | perf3kt | lol |
| 15:08.07 | Qwell | it's not on fire |
| 15:08.10 | Qwell | must be a gateway |
| 15:09.28 | mvanbaak | Qwell: yeah, blown caps |
| 15:09.48 | mvanbaak | hate it when that happens |
| 15:09.54 | mvanbaak | and no, it was not a dell |
| 15:10.06 | mvanbaak | aopen motherboard (intel p4) |
| 15:13.13 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:14.58 | perf3kt | is the book dated on ways to obtain the * source code and dependencies? |
| 15:15.09 | [TK]D-Fender | perf3kt, Yes |
| 15:15.15 | [TK]D-Fender | perf3kt, www.asterisk.org |
| 15:15.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:15.26 | Strom_C | damnit, I completely forgot to go to chick-fil-a this week |
| 15:15.33 | [TK]D-Fender | perf3kt, there are readme's in the source tarball as well. READ them. |
| 15:15.49 | Qwell | Strom_C: you have chick-fil-a in CA :p |
| 15:16.02 | Strom_C | Qwell: like way the hell out in Ontario |
| 15:16.05 | Qwell | heh |
| 15:16.22 | Strom_C | it's decent, but not worth driving an hour into the inland empire for |
| 15:16.52 | Qwell | there's one in Torrence, and Redondo |
| 15:16.57 | Strom_C | oh |
| 15:17.01 | Strom_C | that's closer |
| 15:17.04 | Qwell | a bit :p |
| 15:17.09 | Strom_C | i only knew about the one in Ontario |
| 15:17.28 | Qwell | Cerritos, Long Beach |
| 15:17.31 | perf3kt | anyone use webmin? |
| 15:17.36 | Qwell | Orange, Santa Ana |
| 15:17.59 | [TK]D-Fender | perf3kt, I have and forget it as far as * is concerned |
| 15:18.06 | Strom_C | i've only eaten at chick-fil-a once |
| 15:18.10 | Qwell | really? |
| 15:18.12 | Strom_C | at the philadelphia airport |
| 15:18.12 | Strom_C | ys |
| 15:18.14 | Qwell | I love that place, heh |
| 15:18.15 | [TK]D-Fender | perf3kt, You are going to have to really read and do this yourself. |
| 15:18.22 | Qwell | there's one about... |
| 15:18.26 | perf3kt | tk: just saw it on the ftp |
| 15:18.30 | Qwell | 1/2 mile from me |
| 15:18.38 | perf3kt | tk: and I have been reading and have been doing it myself |
| 15:18.41 | Qwell | on the corner of the main street |
| 15:18.57 | Qwell | Strom_C: I'd recommend Zaxby's |
| 15:19.01 | perf3kt | tk: dosesn't help when the book that paid 40 to be a rsource is dated |
| 15:19.06 | Strom_C | what about chick-fil-b |
| 15:19.06 | Qwell | if you like chicken sandwiches and such |
| 15:19.13 | Strom_C | where's zaxby's? |
| 15:19.18 | blitzrage | perf3kt: it IS 2 years old, and a new version is coming out in August |
| 15:19.23 | Qwell | there's one on University |
| 15:19.26 | *** join/#asterisk Chris-NB (n=chris@ip.tech.t-mobile.at) |
| 15:19.29 | Qwell | over by the parkway |
| 15:19.43 | Qwell | (and, of course, one on the same corner right here as Chick-fil-a |
| 15:19.48 | blitzrage | perf3kt: and it's not dated for 1.2, which is why the 2nd edition covers 1.4 (1.4 did not exist when we wrote the first edition) |
| 15:19.51 | [TK]D-Fender | perf3kt, and the book is also available for FREE and yes new versions come out. You consider * getting better as being BAD?! |
| 15:19.55 | perf3kt | tk: and like I was telling the guys earlier its such a turn off when I wanna get into the program, but I'm stopped by installing, compiling, depencies and etc |
| 15:19.58 | Qwell | there's a starbucks right next to the one on university :p |
| 15:20.23 | perf3kt | tk: well sorry I didn't know it was FREE until after I got on here |
| 15:20.28 | [TK]D-Fender | perf3kt, You should have simply gone to the SOURCE : www.asterisk.org Its all listed there. |
| 15:20.33 | Strom_C | EERF |
| 15:20.38 | blitzrage | FORZA 2! |
| 15:20.41 | Strom_C | FREEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE |
| 15:20.41 | Qwell | there's also one right by digigraph, near the airport |
| 15:20.44 | [TK]D-Fender | perf3kt, "Life sucks but rarely swallows" |
| 15:20.51 | blitzrage | I DON'T WANT TO MEET YOUR MOM |
| 15:21.15 | [TK]D-Fender | blitzrage, I JUST WANT |
| 15:21.15 | Qwell | ??? |
| 15:21.15 | blitzrage | ! ! ! |
| 15:21.15 | Strom_C | Qwell: you fail |
| 15:21.15 | Strom_C | sorry |
| 15:21.29 | file | Qwell always fails! |
| 15:21.31 | perf3kt | tk: I'm not complaing at all, but don't try to treat me like some non reading n00b |
| 15:21.47 | Strom_C | perf3kt: [TK]D-Fender can be a bit irascible at times |
| 15:21.54 | [TK]D-Fender | perf3kt, s'ok |
| 15:22.02 | perf3kt | tk: no wonder so many people go to the guis |
| 15:22.07 | blitzrage | perf3kt: ya.... gotta get thick skin to hang out in the asterisk community |
| 15:22.11 | Strom_C | and by "at times" I mean "pretty much 99.999% of the time" |
| 15:22.22 | [TK]D-Fender | Strom_C, So who's playing the "kettle" today? ;) |
| 15:22.33 | Strom_C | the other .001% can be conveniently blamed on superstring theory |
| 15:22.58 | perf3kt | strom: that's cool, I dont' want to be given anythign I wanna earn my stripes |
| 15:23.04 | Penggu | will all these play nicely together: ast-1.2.21.1, zaptel-1.2.18, ast-addons-1.2.7, libpri 1.2.5 ? <-- i believe these are the latest version from the 1.2x branch.. |
| 15:23.44 | Strom_C | [TK]D-Fender: i'm auditioning for the part of "KETTLE" but they may cast me as "POT" |
| 15:23.46 | [TK]D-Fender | Penggu, www.asterisk.org lists all the currect versions right next to each other, go check and you'll know for sure |
| 15:23.48 | Qwell | Strom_C: there's also Lenny's, if you like subs |
| 15:24.00 | Qwell | Lenny's is pretty awesome |
| 15:24.01 | file | mmm Lenny's |
| 15:24.03 | Strom_C | i think chicken sandwiches have caught my fancy |
| 15:24.11 | [TK]D-Fender | Strom_C, You "POT"? I dunno... you're way too uptight ;) |
| 15:24.13 | Strom_C | but if even file is drooling over lenny's... |
| 15:24.38 | Penggu | [TK]D-Fender: ah yer.. didnt see that list on the RHS.. |
| 15:24.39 | *** join/#asterisk friedrich| (n=friedric@e177244140.adsl.alicedsl.de) |
| 15:24.46 | Qwell | in'n'out > lenny's though |
| 15:24.54 | Qwell | in'n'out > pretty much everything |
| 15:26.19 | Strom_C | yes |
| 15:26.29 | Strom_C | the instant i get back home i'm going to in-n-out |
| 15:26.59 | Qwell | I'm gonna tell Kevin that I need to go to Astricon, on grounds that I haven't had in-n-out in 9 months |
| 15:27.14 | [TK]D-Fender | Strom_C, but that'll only take a minute or so ;) |
| 15:27.17 | Strom_C | i'd approve that |
| 15:27.25 | Strom_C | [TK]D-Fender: ? |
| 15:27.47 | Strom_C | [TK]D-Fender: you do know what in-n-out is, right? |
| 15:27.59 | file | is there an in-n-out in phx? |
| 15:28.03 | Qwell | file: yes |
| 15:28.09 | [TK]D-Fender | Strom_C, perhaps not the one you're alluding to. |
| 15:28.19 | file | well if that's not a reason to go I don't know what is |
| 15:28.31 | Qwell | file: exactly |
| 15:28.38 | Qwell | I should speak |
| 15:28.53 | Strom_C | "Asterisk asterisk ok let's all go to in-n-out" |
| 15:29.00 | Qwell | They cater |
| 15:29.03 | Qwell | hmm |
| 15:29.04 | Qwell | They cater |
| 15:29.08 | Strom_C | ooh |
| 15:29.09 | Qwell | I should tell Sokol :p |
| 15:29.11 | Strom_C | They cater |
| 15:29.15 | Strom_C | !!! |
| 15:29.18 | file | They cater?!? |
| 15:29.24 | Strom_C | They cater !1111 |
| 15:29.32 | Qwell | They have a semi-truck mobile restaurant :) |
| 15:29.37 | Qwell | it's pretty friggen awesome |
| 15:29.47 | Strom_C | also, this year we should call the convention "Phoneix" just to confuse the dyslexics |
| 15:30.04 | Qwell | Hooked on Phoneix worked for me |
| 15:30.16 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:30.21 | [TK]D-Fender | hukt on fonix werkt 4 me! |
| 15:31.51 | Strom_C | http://multigeeks.com/pics/strom.jpg |
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| 15:44.58 | asterisknerds | <PROTECTED> |
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| 15:45.23 | [TK]D-Fender | asterisknerds, you don't say.... |
| 15:45.27 | macTijn | <PROTECTED> |
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| 15:55.50 | asterisknerds2 | <PROTECTED> |
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| 16:05.12 | Strom_C | They cater |
| 16:07.39 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 16:08.44 | Qwell | Strom_C: they totally cater |
| 16:09.10 | Strom_C | ok, seriously pulling myself away from the pc now |
| 16:09.13 | Qwell | http://www.in-n-out.com/cookout_trailer.asp |
| 16:09.14 | Strom_C | shower tiems |
| 16:13.04 | nclx | I have two Snom320 phones registered as SIP clients on my LAN to asterisk, they have usernames of : 710 and 720 respectively. I haven't altered anything else from the default dialplan aside from sip.conf, I can dial the demo from each phone, should I be able to dial 710 from 720 and ring that phone? Right now I can't. I'm just learning though. |
| 16:13.36 | wunderkin | probably wrong time and day to ask but has anyone here ever compiled asterisk for openwrt or similar? i can't ever get it to find ncurses when it compiles in the editline directory |
| 16:18.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:23.17 | [TK]D-Fender | nclx, Noy you shouldn't be able to dial them with the"out of the box" sample extensions.conf. |
| 16:23.17 | Penggu | nclx: you need to add those numbers to an accessable context in extensions.conf |
| 16:23.33 | [TK]D-Fender | nclx, You need to create extensions to do this |
| 16:24.25 | Strom_C | wunderkin: i just used openwrt's asterisk package |
| 16:24.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:25.14 | wunderkin | all i see with openwrt is 1.0.x |
| 16:26.10 | kickbackit | I have been staring at sip proxy's for too long... Anyone have suggestions on one to use? |
| 16:26.36 | [TK]D-Fender | kickbackit, OpenSER |
| 16:26.49 | [TK]D-Fender | kickbackit, Do you really need one? |
| 16:26.51 | Strom_C | wunderkin: i recall that a year ago there was a 1.2.x package |
| 16:27.04 | Strom_C | so i suppose you're running a really old version of openwrt |
| 16:28.02 | wunderkin | ok well there is 1.2.14 in backports... but still anything i find is old.. no its not really old |
| 16:28.18 | kickbackit | Well I wonder if its going to make my situation easier. My asterisk box is behind a firewall, but I have external clients. If I put the proxy on the firewall box I should be able to forward traffic to it without messing around witht he firewall's port forwarding. |
| 16:28.22 | wunderkin | there has to be something wrong i'm doing compiling it |
| 16:28.43 | Strom_C | you're sure you have libncurses5-dev or equivalent installed? |
| 16:29.01 | wunderkin | i do but this would be from the toolchain and not the host computer |
| 16:29.28 | wunderkin | i even specified the directory to the library in the configure line for it |
| 16:29.29 | kickbackit | I installed openser, but the sit housing the documentation is timing out :S |
| 16:29.46 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 16:31.02 | [TK]D-Fender | kickbackit, SO not workth it. Just port forward. You'll install all sorts of new work and problems for a "zero" problem |
| 16:31.39 | kickbackit | ok, I have ports forwarded now, but have issues with the external client being able to dial out |
| 16:31.42 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-150-67.dsl.ipltin.ameritech.net) |
| 16:31.56 | [TK]D-Fender | kickbackit, here : ... |
| 16:31.58 | [TK]D-Fender | ~sipnat |
| 16:31.58 | 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 |
| 16:35.00 | kickbackit | thanks.... Looks like this is exactly the situation I have. |
| 16:35.37 | [TK]D-Fender | kickbackit, I wrote it to cater to pretty much all of the usual circumstances and layed it out so each part of the example was seperate. |
| 16:35.53 | Strom_C | laid out, not layed out ;) |
| 16:36.21 | [TK]D-Fender | Strom_C, http://multigeeks.com/pics/strom.jpg |
| 16:36.35 | Strom_C | cohujibuggle! |
| 16:37.28 | Strom_C | my god, [TK]D-Fender, can't you READ a DICTIONARY?! |
| 16:38.08 | kickbackit | Does the bindaddress matter in this case? I note the internal and external ip, but no bindaddress in the example |
| 16:38.08 | [TK]D-Fender | Strom_C, Yes, I am your God, bow before me! |
| 16:38.33 | shido6 | ZzZz |
| 16:38.46 | Strom_C | GNU/Boring |
| 16:39.34 | Strom_C | cohujibuggle! |
| 16:40.47 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 16:40.49 | *** join/#asterisk friedrich| (n=friedric@85.177.244.140) |
| 16:40.49 | [TK]D-Fender | Strom_C, you are indeed "incorrigible" ;) |
| 16:41.20 | Strom_C | i didn't say incorrigible |
| 16:41.24 | Strom_C | I said cohujibuggle |
| 16:41.41 | Qwell | incohujibugglable |
| 16:41.51 | [TK]D-Fender | lol |
| 16:42.09 | Strom_C | k@-h@`-j@-b@g-@l |
| 16:42.57 | kickbackit | [TK]D-Fender, clients on the local lan, even though their external traffice is natted, traffic to the server should be direct. So in their sip.conf section nat=no.... Correct? |
| 16:43.14 | [TK]D-Fender | kickbackit, local direct to *? |
| 16:44.06 | kickbackit | I have phones that are all behind the firewall, and 4 external phones that need to go through the firewall. |
| 16:44.24 | kickbackit | the * server is sitting on the same network as the local phones. |
| 16:44.58 | [TK]D-Fender | kickbackit, Ok, any phone OUTSIDE your LAN, that IS behind its own NAT requires "nat=yes", and "qualify=yes". Clear? |
| 16:45.11 | kickbackit | yep |
| 16:45.14 | Strom_C | clear as mud |
| 16:45.19 | kickbackit | an bindaddress? |
| 16:45.23 | kickbackit | and* |
| 16:45.31 | [TK]D-Fender | kickbackit, no, leave bindaddr=0.0.0.0 |
| 16:45.51 | [TK]D-Fender | kickbackit, that will let * talk on any IP that it has available to it |
| 16:45.59 | kickbackit | ok |
| 16:46.08 | kickbackit | now getting an error from a peer |
| 16:46.10 | kickbackit | <PROTECTED> |
| 16:46.38 | [TK]D-Fender | kickbackit, you have multiple SIP phones behind the SAME remote NAT? |
| 16:47.20 | kickbackit | I have 85 phones on a 192.168.0.x address, the * is 192.168.0.5 |
| 16:47.42 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-135.ph.ph.cox.net) |
| 16:47.50 | [TK]D-Fender | kickbackit, ummm... clarify please |
| 16:48.02 | kickbackit | I changed nat=no in sip.conf and I can call out |
| 16:49.03 | kickbackit | Ok my remote phone issues are resolved |
| 16:49.23 | kickbackit | now I have to drive in to the office and test the local phones |
| 16:49.24 | [TK]D-Fender | kickbackit, those phones local to * should have "nat=no" |
| 16:49.43 | kickbackit | that they do, and remote phones nat=yes |
| 16:50.03 | kickbackit | but when I turned the nat=yes on in the general section I could not talk to my voip provider |
| 16:50.47 | [TK]D-Fender | kickbackit, you need to set your PROVIDER's entry to "nat=no" |
| 16:51.05 | [TK]D-Fender | kickbackit, and you should have "nat=yes" under [general] |
| 17:00.02 | kickbackit | Audio quality on the remote phone has improved a ton. Everything seems ok on the remote side of things. Thanks guys. |
| 17:01.29 | coppice | can you purchase audio quality by the ton? it comes in kilos here |
| 17:01.47 | kickbackit | depends on the codec |
| 17:02.30 | coppice | you mean Alaw would offer kilos of quality, and ulaw would offer tons? |
| 17:03.25 | kickbackit | yeah I checked sip show channel, and in the report there I increased the audio quality from 1 ton to 2.... |
| 17:03.34 | kickbackit | codec was ulaw |
| 17:05.33 | coppice | recent power supplies have lots of SATA power connectors, and not nearly enough good old molex connectors to serve the needs of these analogue telephony cards :-( |
| 17:06.31 | [TK]D-Fender | coppice, You mean like.. 1? |
| 17:07.28 | kickbackit | that reminds me, I am supposed to find a card for our * server to allow us to connect a t1. My boss is cheap and does not want to spend money on the ones that digum currently offers. What is an older card that I might pick up on ebay? |
| 17:07.31 | coppice | if you have only 1 card, then 1 might be enough. still a problem though, since they come in bunches, with shorts leads between each connector |
| 17:08.04 | [TK]D-Fender | coppice, And extensions/splitters cost like 50$.... oops.... /100 |
| 17:08.22 | [TK]D-Fender | kickbackit, BAD IDEA |
| 17:08.36 | Strom_C | ~cheap |
| 17:08.45 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 17:08.45 | [TK]D-Fender | kickbackit, Cheap out and you'll be hoping to win a lifetime supply of KY |
| 17:08.46 | coppice | well, yeah, except I don't have any. I replaced a dead PSU, and find I don't have enough connectors. this is somewhat annoying me |
| 17:08.47 | kickbackit | agreed |
| 17:08.47 | [TK]D-Fender | ~ygwypf |
| 17:08.47 | jbot | well, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 17:10.26 | coppice | ygwypf is stupid. you don't get any mor than you pay for is far more accurate |
| 17:10.42 | coppice | expect for the "mor" bit :-\ |
| 17:11.09 | Strom_C | ~cohujibuggle |
| 17:11.10 | jbot | hmm... cohujibuggle is gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb |
| 17:11.29 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 17:11.31 | [TK]D-Fender | coppice, I agree |
| 17:12.34 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 17:12.55 | [TK]D-Fender | ~gs |
| 17:12.56 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 17:15.06 | *** join/#asterisk Strom_M (n=pocketir@m055e36d0.tmodns.net) |
| 17:20.42 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 17:21.00 | chiardon | Hello all!! |
| 17:22.20 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:22.39 | Strom_M | hi |
| 17:26.38 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 17:26.55 | *** join/#asterisk dulzuroso (n=kvirc@eu85-84-137-219.clientes.euskaltel.es) |
| 17:26.58 | dulzuroso | hiii |
| 17:27.07 | dulzuroso | anybody know what addon i should use to liseng an inbound call and redirect to the phone that the caller dial in that moment? |
| 17:27.46 | Strom_M | ..... |
| 17:27.55 | Strom_M | i dont understand you |
| 17:28.00 | dulzuroso | well |
| 17:28.04 | *** join/#asterisk monstertruck (n=monstert@c-76-26-50-233.hsd1.fl.comcast.net) |
| 17:28.10 | dulzuroso | i want to call to a DID that i have |
| 17:28.30 | monstertruck | hey guys, im trying to implement a gateway to gsm |
| 17:28.39 | dulzuroso | and i wish asterisk take that call, and lisend to me to dial a numer where i want to call |
| 17:28.42 | monstertruck | anybody had any experience with PhoneLabs Dock-N-Talk? |
| 17:28.49 | juuva | monstertruck: don't try, just bye one |
| 17:29.09 | dulzuroso | then i wish asterisk call this numer using a diferent voip acount, and joint it |
| 17:29.16 | Strom_M | dulzuroso: look at the DISA app |
| 17:29.23 | dulzuroso | thanx |
| 17:29.42 | juuva | I've had nightmares about one voip-gsm -gateway solution we have tried to make work.. |
| 17:30.00 | monstertruck | juuva, using which hardware? |
| 17:30.09 | dulzuroso | there is no disa addons |
| 17:30.10 | dulzuroso | :( |
| 17:30.50 | monstertruck | juuva, i found this Phonelabs Dock-N-Talk browsing .. it advertises GSM to Wired PSTN |
| 17:31.13 | monstertruck | which doesnt seem to be a bad idea, i can use a regular zap channel |
| 17:31.21 | monstertruck | and the DNT does the rest |
| 17:31.35 | kiwoneka | afternoon to all |
| 17:32.26 | *** join/#asterisk griels (i=user@60.67-246-213.ippool.namesco.net) |
| 17:32.53 | chiardon | dulzuroso . . . they tell you to look at the DISA functionality in * to do what you want |
| 17:33.00 | juuva | for xample eurotech sels sip-gsm -routers which would eliminate need for zap-channel |
| 17:33.18 | dulzuroso | ou, oks |
| 17:33.19 | dulzuroso | :D |
| 17:33.27 | kiwoneka | please please someone help and direct me to a place i can download the new bootrom and sip for the ip650 |
| 17:34.01 | juuva | but I haven't tried those.. and as far as I know , there is grandstream hardware inside + gsm module |
| 17:34.23 | _DAW | kiwoneka: your vendor |
| 17:34.32 | kiwoneka | ebay? |
| 17:34.38 | kiwoneka | :) |
| 17:35.09 | chiardon | juuva... you are right tha gran works so good! |
| 17:35.46 | juuva | sure.. and cows fly |
| 17:36.26 | chiardon | really they fly??? |
| 17:36.30 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:36.35 | juuva | maybe some of them ;) |
| 17:36.49 | [TK]D-Fender | chiardon, on American Airlines, yes ;) |
| 17:37.04 | chiardon | I never seen that!!! so bad to me!! |
| 17:37.10 | juuva | of course there abe proper gsm-voip gateways in the market, but prices are quite high |
| 17:37.35 | juuva | 300+ euros/channel |
| 17:37.52 | monstertruck | juuva, thats one of the reasons i was looking at the dock and talk |
| 17:38.01 | monstertruck | although 300E per channel is not so bad |
| 17:38.19 | monstertruck | the one i saw was $3000 for 4 channels |
| 17:38.24 | chiardon | haha thnx Fender but I have a vertigo issue!! |
| 17:38.35 | juuva | with PSTN you could end up with poor sound quality and failures to detect remote hangup/answer |
| 17:39.16 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:39.40 | blitzrage | oej: !!! |
| 17:39.46 | juuva | those eurotech routers are about 300E/channel.. for 1300E you could get 4 channel VoiSmart PCI-card |
| 17:40.08 | oej | The one and only! |
| 17:40.11 | juuva | Would like to test one of those |
| 17:40.13 | blitzrage | schweeeet |
| 17:40.17 | blitzrage | how goes? |
| 17:41.10 | *** join/#asterisk Un1x_ (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 17:45.17 | *** join/#asterisk seele_ (n=seele@190.90.27.162) |
| 17:45.52 | monstertruck | juuva, will try one of the wurotech routers |
| 17:45.54 | monstertruck | thanks |
| 17:46.26 | seele_ | hello, how can I balance the calls between two PRI (60 lines) |
| 17:47.43 | [TK]D-Fender | seele_, on outbound you can look at channel occupancy, on inbound, you can't |
| 17:48.29 | juuva | monstertruck: just make sure that you can return it if you don't like it.. |
| 17:49.08 | monstertruck | yeah, thats why im buying one, if it works, then ill get a multichannel router |
| 17:49.15 | seele_ | [TK]D-Fender, yes for outbound, i need to make the same number of calls for both PRI channels channels |
| 17:49.51 | [TK]D-Fender | seele_, then before choosing which PRI to use go check how many channels are in use on each. |
| 18:06.52 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
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| 18:35.48 | yakkop | hi, anyone know if its possible to have the Dial app play a sound file instead of a ring? I see the music on hold option but that doesn't let you specify an exact sound. |
| 18:38.30 | Juggie | make a music on hold class with only one sound |
| 18:38.47 | *** join/#asterisk unixmaniac (n=datapuls@s-spanoudakis.ath.forthnet.gr) |
| 18:39.03 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 18:40.04 | yakkop | Juggie: is it possible to do that dynamicly? i.e. without restarting |
| 18:43.21 | Juggie | edit musiconhold.conf and reload moh yes |
| 18:44.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:45.50 | yakkop | thanks |
| 18:46.41 | *** join/#asterisk Strom_M (n=strom@206.166.206.34) |
| 18:49.33 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 18:50.37 | kiwoneka | may someone direct me to a place to find the latest software for the ip 650 |
| 18:50.45 | Strom_M | kiwoneka: your reseller |
| 18:50.58 | kiwoneka | no |
| 18:51.13 | Strom_M | yes |
| 18:51.17 | kiwoneka | just someone that has a bunch of phone that need |
| 18:51.52 | Strom_M | your sense makes sentence not !!! |
| 18:52.07 | wunderkin | !1111 |
| 18:52.16 | kiwoneka | i have been at this for a while |
| 18:52.48 | Strom_M | kiwoneka: contact the company which sold you the phone(s) and request the firmware |
| 18:52.59 | Strom_M | or you can find the second-latest firmware on polycom's site |
| 18:53.08 | kiwoneka | i bought them on ebay |
| 18:53.49 | kiwoneka | the one on the site does not work |
| 18:53.55 | kiwoneka | i have been there |
| 18:54.12 | Strom_M | the one on polycom.com? |
| 18:54.12 | kiwoneka | but thanks for you assistance |
| 18:54.18 | kiwoneka | yes |
| 18:54.21 | Strom_M | that's odd |
| 18:54.30 | Strom_M | because i've used that exact version |
| 18:54.36 | Strom_M | and it worked for me a hundred times |
| 18:54.44 | kiwoneka | at this link http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip650.html |
| 18:54.59 | *** join/#asterisk RomDump (i=romdump@faeroes.freeshell.ORG) |
| 18:55.10 | kiwoneka | could be how i am downloading the zip file |
| 18:55.11 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:55.33 | kiwoneka | i am on my mac |
| 18:55.45 | Strom_M | use wget on your asterisk server :) |
| 18:55.53 | kiwoneka | ok |
| 18:55.57 | kiwoneka | i have tried that too |
| 18:55.57 | _DAW | r u click the link for 2.0.3 Rev B? |
| 18:56.03 | kiwoneka | yes |
| 18:56.21 | Strom_M | how exactly does it "not work"? |
| 18:56.34 | kiwoneka | it will not extract the zip |
| 18:56.42 | *** join/#asterisk oej_ (n=olle@apollo.webway.se) |
| 18:57.15 | Strom_M | just for shits and giggles, let me try |
| 18:57.33 | kiwoneka | your a riot! |
| 18:57.46 | kiwoneka | ls |
| 18:57.59 | Strom_M | ERROR: THIS IS IRC |
| 18:58.35 | Strom_M | -rwxr-xr-x omgwtfbbq.tar.gz |
| 18:58.40 | *** part/#asterisk RomDump (i=romdump@faeroes.freeshell.ORG) |
| 18:58.49 | *** join/#asterisk syneus (n=syneus@host213-98-dynamic.8-87-r.retail.telecomitalia.it) |
| 18:59.15 | kiwoneka | [root@tunes config]# unzip spip_ssip_sip_2_0_3b_sig.zip |
| 18:59.15 | kiwoneka | Archive: spip_ssip_sip_2_0_3b_sig.zip |
| 18:59.15 | kiwoneka | file #1: bad zipfile offset (local header sig): 0 |
| 18:59.16 | kiwoneka | <PROTECTED> |
| 18:59.16 | wunderkin | mm bbq |
| 18:59.52 | Strom_M | well, hmm, it does the same on mine |
| 19:01.23 | kiwoneka | omg, your kidding |
| 19:04.18 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 19:04.20 | _Raptor_ | P[ 1] stack_init: Cannot add layer 4 to this port. << mISDN gurus please tell me whats wrong |
| 19:04.40 | Strom_M | your/you're homophone error |
| 19:04.46 | *** join/#asterisk oej_ (n=olle@apollo.webway.se) |
| 19:05.31 | kiwoneka | so, you have a working sip that you can share Strom_M |
| 19:05.49 | Strom_M | if you promise to learn the difference between "your" and "you're" |
| 19:06.11 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:06.11 | kiwoneka | sure |
| 19:06.33 | kiwoneka | you're the one to teach |
| 19:06.48 | *** join/#asterisk dennisharrison (n=dennisha@71-81-37-101.dhcp.slid.la.charter.com) |
| 19:12.07 | *** join/#asterisk oej_ (n=olle@apollo.webway.se) |
| 19:20.53 | rob0 | ok, need to figure out my zap trouble ... http://pastebin.slackadelic.com/289 |
| 19:22.30 | Strom_M | rob0: did you swap the modules like I suggested earlier? |
| 19:23.30 | rob0 | Oh I missed that. Put them where? |
| 19:23.36 | Strom_M | switch their places |
| 19:23.50 | rob0 | hmmm ok, I'll try |
| 19:23.52 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:23.53 | Strom_M | so put the one FXS where the other is, and vice versa |
| 19:24.06 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:24.09 | rob0 | what about the empty module 2 slot? |
| 19:24.15 | Strom_M | if the problem follows the FXS module, then it is can be RMA time |
| 19:24.23 | rob0 | ah |
| 19:24.35 | *** part/#asterisk griels (i=user@60.67-246-213.ippool.namesco.net) |
| 19:24.43 | rob0 | ok, will have to shut down for this, bbiab |
| 19:24.53 | Strom_M | you're IRCing from your asterisk box? |
| 19:24.57 | rob0 | nono |
| 19:25.16 | rob0 | but it's my Internet connection to the place where I'm in irssi. |
| 19:25.29 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:25.30 | kiwoneka | Strom_M: you find that sip |
| 19:25.40 | *** join/#asterisk cayorde (n=flexable@host245-111-dynamic.17-87-r.retail.telecomitalia.it) |
| 19:25.47 | Strom_M | kiwoneka: I already sent you a privmsg |
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| 20:05.13 | rob0 | Strom_M was right, RMA time. :( |
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| 21:07.30 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 21:08.00 | supjigator | Anyone mind helping me debug a IAX2 connection between two machine? I have it working in on direction but no the other. |
| 21:09.06 | sfb1 | how doesn't it work in the other direction? |
| 21:09.47 | supjigator | It won't qualify and looks like an auth problem. |
| 21:10.01 | supjigator | I see POKE PONG INVAL |
| 21:11.07 | supjigator | I have a Box A which has two interfaces. One public and one private. Phones are on private and I'm using IAX to a public server I have setup with PRI PSTN connection. |
| 21:11.07 | supjigator | BoxB can send calls to BoxA. BoxA cannont send any calls to Box B |
| 21:11.07 | kiwoneka | good evening to all does any one have a working zip file of http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip650.html sip 2.0.3 revb? |
| 21:11.51 | kiwoneka | they zip off of polycom does not work |
| 21:13.15 | sfb1 | is the public server box b? |
| 21:14.05 | kiwoneka | yes |
| 21:14.09 | supjigator | Yes. |
| 21:14.40 | supjigator | A also has a public address as well. Two nic cards. |
| 21:15.02 | sfb1 | so the poke pong invalid is showing on box b? |
| 21:16.04 | supjigator | I see the error on both sides but it looks like mirror |
| 21:16.44 | sfb1 | is boxA firewalled on the public interface? any firewall log messages happening? |
| 21:17.39 | supjigator | It would appear that iax2 debug is telling me that A to b is failing and b to a is succeding becuase there are ACK's as well just not on box A. Always POKE PONG INVAL on box a and iax2 show peers shows unreachable |
| 21:18.04 | supjigator | ON box B iax2 show peers shows reachable and can make calls to box A |
| 21:18.43 | sfb1 | oops, thinking the wrong way around then - is B firewalled? |
| 21:19.00 | supjigator | A iptables has defaults to accept |
| 21:19.29 | supjigator | B as well. |
| 21:20.24 | supjigator | It looks like they are able to talk. Does INVAL mean packets aren't making it to box b? |
| 21:20.30 | supjigator | Cause it looks like they are. |
| 21:21.19 | supjigator | Um can't be a firewall issue cause B and A are recieving IAX packets from each other. |
| 21:21.57 | supjigator | I've spent days on this including upgrading the older box to the same version |
| 21:22.17 | sfb1 | http://tools.ietf.org/id/draft-guy-iax-03.txt suggests a POKE message must be replied to with a PONG |
| 21:22.37 | supjigator | Yep I see POKE and then a PONG then an INVAL |
| 21:22.56 | sfb1 | but it doesn't suggest that PONG can be invalid - it's just like an acknowledgement |
| 21:23.14 | supjigator | On a working side I see POKE PONG ACK |
| 21:23.31 | supjigator | so the RFC doesn't really tell us much as to what the actual problem is. |
| 21:33.15 | justdave | are there any known issues with having a TE120P and a TDM400P in the same machine? |
| 21:33.27 | justdave | getting lots of "NMI for unknown reason XX on CPU X" messages after booting the server with both in it |
| 21:38.47 | justdave | ok, just booted with only the TDM and they didn't go away. must be RAM or something :( |
| 21:42.17 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 21:44.10 | sfb1 | supjigator: well, if it's not firewall or routing (both are talking and poke/pong are being transferred), and not asterisk itself as it's apparently working on both |
| 21:44.58 | sfb1 | have you set verbose logging? |
| 21:45.14 | sfb1 | for more detailed debugs (I think that's an option) |
| 21:48.52 | justdave | booted without the TDM and it booted fine. doesn't like the TDM card :( |
| 21:49.16 | sfb1 | what's a TDM card? |
| 21:49.32 | justdave | hook up POTS phone lines or phones to the server |
| 21:50.23 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:51.23 | sfb1 | ah |
| 21:58.48 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 22:00.12 | sfb1 | my sip trunk establishes the connection, but wont pass any audio. |
| 22:00.22 | sfb1 | I bet it's a NAT issue. :/ |
| 22:07.05 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 22:07.50 | hi365 | how can i get the voice mail time stamp to play in 24hr format? do i have to add the option to every line in voicemail.conf? |
| 22:10.45 | snuff-work | should be a 'default' near the top of the voicemail.conf.. |
| 22:10.51 | snuff-work | just change that one |
| 22:13.49 | *** join/#asterisk Stc884 (n=Roberto@host55-198-dynamic.9-79-r.retail.telecomitalia.it) |
| 22:13.51 | Stc884 | hello |
| 22:14.00 | Stc884 | italiani cè qualcuno ? |
| 22:14.50 | *** join/#asterisk shinao1 (n=shinao1@41.205.187.85) |
| 22:17.42 | tzafrir_home | Stc884, try English, I guess |
| 22:18.29 | Stc884 | no problem tzafrir_home thanks |
| 22:18.47 | tzafrir_home | Stc884, there's #asterisk-it , but it is empty right now |
| 22:19.01 | Stc884 | yes, thanks |
| 22:19.35 | tzafrir_home | justdave, it should work. Just make sure that the modules load in the right order |
| 22:20.51 | justdave | what order should they be? |
| 22:21.10 | justdave | I saw a post on a forum saying the TDM cards don't like 64 bit slots |
| 22:21.21 | justdave | the new machine is all 64 bit slots |
| 22:21.28 | justdave | it was indeed in a 32-bit slot on the old machine |
| 22:21.32 | [TK]D-Fender | justdave, guess you should have looked first |
| 22:22.13 | justdave | how would I have known to look for that? :) it's an odd thing |
| 22:22.44 | justdave | swapping machines was a convenience to keep the downtime minimal (we were staging the upgrade on the new box, then swapping everything at the maintenance window) |
| 22:22.55 | justdave | already blown that screwing with the new box now. :) |
| 22:23.04 | justdave | gonna keep the old box and just swap the hard drives now |
| 22:23.24 | [TK]D-Fender | justdave, Sure, There's a new E10 gas out... go see if your old deisel VW Rabbit will run on it... |
| 22:23.38 | justdave | heh |
| 22:23.49 | [TK]D-Fender | When the slot changes... find an IQ |
| 22:24.30 | *** join/#asterisk Ebola (n=Ebola@host86-143-168-67.range86-143.btcentralplus.com) |
| 22:25.56 | Stc884 | whereis asteriskNow livecd ? |
| 22:27.03 | wunderkin | justdave, that can't be true |
| 22:27.30 | justdave | sounded funny to me, too, but it does fit the situation |
| 22:28.01 | justdave | the two machines are supposedly the same model, but apparently the replacement one is a newer revision and the bus doesn't quite match |
| 22:28.39 | justdave | both machines are about the same age, theoretically, we thought we were recycling an old box :) |
| 22:29.03 | justdave | they're HP DL385s |
| 22:29.58 | [TK]D-Fender | Stc884, http://www.asterisknow.org/ |
| 22:30.15 | [TK]D-Fender | Stc884, And this is NOT AsteriskNOW's support channel. |
| 22:30.23 | [TK]D-Fender | Stc884, please read the channel topic |
| 22:33.23 | wunderkin | justdave, more like a motherboard incompatibility probably, contact digium support |
| 22:40.27 | hi365 | tzafrir_home ping |
| 22:41.56 | hi365 | snuff-work: got it. 1 question: does the Q option always default to ABdY or can it play just the yesterday/today file if apllicable? |
| 22:42.51 | snuff-work | not sure.. i haven't played too much with it :) |
| 22:42.59 | hi365 | thansk |
| 22:43.20 | hi365 | in the custome time zones of voicemail.conf, does the Q option always default to ABdY or can it play just the yesterday/today file if apllicable? |
| 22:45.19 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 22:46.02 | snuff-work | i'm sure it does play the today/yesterday when applicable |
| 22:47.13 | *** join/#asterisk [ViAjErO] (i=ViAjErO@201.103.53.73) |
| 22:48.28 | tzafrir_home | hi365, pong |
| 22:49.23 | [ViAjErO] | someone has analog lines to PSTN doing outgoing calls with SIP ? i have incoming calls to my sip Xlite from pstn .. but i can't make calls to PSTN trought Xlite :( |
| 22:49.31 | hi365 | tzafrir_home: shavuah tov! i noticed that you had a couple of patches lined up for asterisk to enable "proper" hebrew numbering. what is the status of hebrew numbers as of now? |
| 22:50.10 | tzafrir_home | NirS promismed me he had good patches. But then disappeared. |
| 22:50.27 | [TK]D-Fender | [ViAjErO], that makes NO sense. Please be specific about the hardware and service providers involved. |
| 22:50.45 | tzafrir_home | I'll guess I'll take a look at say.c again |
| 22:50.50 | hi365 | snuff-work: it sure does, but you cant have a custom string defined and use Q/q PLUS what ever else you want. apperently Q defaults to ABdY |
| 22:51.21 | tzafrir_home | hi365, is this in say.c or in the voicemail? |
| 22:51.25 | hi365 | tzafrir_home: im sorry, my memmory must of faild (it sd-ram...) i though they where your patches |
| 22:51.50 | *** join/#asterisk kusznir (n=kusznir@66-233-138-60.lew.clearwire-dns.net) |
| 22:51.53 | tzafrir_home | they weren't good enough |
| 22:52.04 | hi365 | tzafrir_home: currently im working with voicemail. it actualy seems ok to me (IF you have the correct recordings) |
| 22:53.34 | *** join/#asterisk crayz_ (n=pmeserve@65.36.160.5) |
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| 22:54.34 | tzafrir_home | rob0, isn't morse code good enough? |
| 22:54.50 | rob0 | hmmm smoke signals! |
| 22:55.06 | rob0 | Does * support smoke signals? |
| 22:55.13 | [ViAjErO] | [TK]D-Fender: I have one TDM22B 2 FXS and 2 FXO, I have my FXS ports connected to PSTN, I have Xlite SIP client on my computer, right now i'm receiving calls from FXS-PSTN lines to my Xlite client (SIP extension) . But when I want to make an outgoing call dialing 9+8numbres as DialRule establish it rings in analog extension instead to call out PSTN |
| 22:56.00 | tzafrir_home | [ViAjErO], can you make calls to analog phones connected to the FXS ports? |
| 22:56.07 | sweeper | rob0: sure, just hook it up to a webcam and a usb fog machine |
| 22:56.18 | [TK]D-Fender | [ViAjErO], go to www.pastebin.ca and paste your zapata.conf and your relevent sections of extensions.conf |
| 22:56.32 | tzafrir_home | hmmm.... FXS ports connected to the PSTN? |
| 22:56.48 | tzafrir_home | can you accept incoming calls? |
| 22:56.48 | [ViAjErO] | yes ... |
| 22:57.04 | rob0 | Dial DLV-MCCXII |
| 22:57.07 | [ViAjErO] | yep |
| 22:57.17 | [ViAjErO] | i have made som calls from public lines ... |
| 22:57.26 | [ViAjErO] | then asterisk answers |
| 22:57.35 | kusznir | Hi all: I've got an asterisk system set up for my home. Incoming DID calls are directed toward a few extensions in the house. From my perspective, things work fine. However, the caller is presented with silence until someone answers or voicemail takes it. Config: http://www.pastebin.ca/619679 |
| 22:57.36 | crayz_ | is there any way to add/remove calls from a "queue" which simply holds calls in asterisk without trying to connect them? I want to use AGI/AMI to place calls into such a queue and then use AMI to connect them to an operator. I don't want to be tied into the asterisk distribution strategies in the queue though |
| 22:57.45 | [ViAjErO] | i dial extension and my Xlite rings and stablish the call |
| 22:57.56 | tzafrir_home | rob0, apart from XXX, what is this good for? |
| 22:58.20 | rob0 | toga parties? |
| 22:58.57 | [TK]D-Fender | tzafrir : he's obviously poorly grouped his channels |
| 22:59.08 | [ViAjErO] | hmmm |
| 22:59.16 | [ViAjErO] | ports are inverted ? |
| 22:59.20 | [TK]D-Fender | [ViAjErO], please provider the pastebin requested |
| 22:59.31 | [ViAjErO] | ok thank you |
| 22:59.33 | [TK]D-Fender | [ViAjErO], No, you have grouped your FXS & FXO togther wrong |
| 22:59.42 | [TK]D-Fender | [ViAjErO], in your configs |
| 23:00.05 | [ViAjErO] | i see in the card tdm22b this ... 1,4 are FXS 2,3 are FXO |
| 23:00.22 | [ViAjErO] | I think so |
| 23:00.54 | [ViAjErO] | j1,j4 are fxs right? and j2,j3 are fxo |
| 23:00.55 | rob0 | The card is physically Green-Red-Red-Green ? |
| 23:01.40 | [ViAjErO] | i'm going to pastebin |
| 23:02.57 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 23:03.39 | *** join/#asterisk [reed] (n=reed@firefox/gnu.webmaster.reed) |
| 23:07.23 | [ViAjErO] | pastebin 1 -> zapata.conf http://www.pastebin.ca/619687 |
| 23:07.39 | [ViAjErO] | pastebin 2 -> extensions.conf http://www.pastebin.ca/619690 |
| 23:09.18 | [TK]D-Fender | [ViAjErO], You are using the GUI and I do not see enough to tell what lines(s) need ot be fixed |
| 23:09.43 | [ViAjErO] | then ? |
| 23:09.54 | [ViAjErO] | what do I need ? |
| 23:09.58 | [ViAjErO] | ok |
| 23:10.05 | [ViAjErO] | i'll review the groping |
| 23:10.07 | [ViAjErO] | :) |
| 23:10.10 | [ViAjErO] | thank you |
| 23:10.14 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 23:11.03 | *** join/#asterisk ZeroPing (n=none@adsl-217-188-253.owb.bellsouth.net) |
| 23:12.08 | *** join/#asterisk juice (n=juice@mo-76-2-162-204.dhcp.embarqhsd.net) |
| 23:13.09 | [TK]D-Fender | [ViAjErO], Right no I belive you have set all your channels to be in 1 group. That is what is mixing up your channel selection |
| 23:19.13 | kiwoneka | good evening to all does any one have a working zip file of http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip650.html sip 2.0.3 revb? |
| 23:19.19 | kusznir | In the dialplan, does one need to do anything special after Answer to provide ringing to callers? |
| 23:19.26 | kiwoneka | the zip off of polycom does not work |
| 23:19.52 | _DAW | Speaking of polycom, is it possible to change the sip timers? |
| 23:21.34 | Sci_05 | _DAW: what do you mean sip timers? |
| 23:22.03 | _DAW | the Sip T1 timer. |
| 23:22.39 | Sci_05 | _DAW: what is it used for? |
| 23:23.46 | kiwoneka | i am kinda stuck on this one |
| 23:24.12 | kiwoneka | i just got a bunch of 650s and i need to upgrade |
| 23:24.32 | _DAW | Sci_05: T1 timer is round trip time estimate. I am using satellite so the 500ms default is too low. |
| 23:24.43 | kiwoneka | any one have a copy of 2.0.3 revb that works? |
| 23:27.15 | Sci_05 | hmmm thats one I am not sure about, did you look thru the provisioning scripts? Are you using a provising script? |
| 23:31.31 | kiwoneka | thanks wunderkin |
| 23:32.23 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:32.25 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 23:32.29 | kiwoneka | awsome, its comin wunderkin |
| 23:32.35 | kiwoneka | i really appreciate it |
| 23:36.08 | _DAW | Sci_05: I really dont think they support it. Surprising since it is not an uncommon option on other phones. |
| 23:36.50 | Sci_05 | what phone is it? 501, 301? |
| 23:37.27 | crayz_ | if I set MusicOnHold and then answer a call in AGI(without dialing through to an operator), will the music play for the call or is there something special I need to do to start the music? |
| 23:37.54 | [TK]D-Fender | Sci_05, Didn't you see him repeat himself like a broken record a doze times like the rest of us? |
| 23:38.25 | Sci_05 | na its the weekend, I don't look at irc that much on the weekend....lol |
| 23:38.40 | [TK]D-Fender | crayz_, No, MoH will not play jsut because you answered. |
| 23:39.00 | crayz_ | how would I start it playing from AGI? |
| 23:40.28 | [TK]D-Fender | crayz_, There is no practical way to leave background music while doing other things. |
| 23:40.43 | [TK]D-Fender | crayz_, dialplan processing is entirely linear. |
| 23:42.12 | crayz_ | could I just play the music indefinitely and use AMI to interrupt & connect a call? I'm somewhat basing this on instructions from here: |
| 23:42.13 | crayz_ | http://www.orderlyq.com/asteriskqueues.html?winId=6382#orderlyq |
| 23:42.41 | crayz_ | which seem to indicate it's possible to setup MOH, transfer the call to AGI, and then later connect the call to an operator |
| 23:47.30 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 23:48.07 | [TK]D-Fender | crayz_, What are you doing in this AGI? |
| 23:49.44 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 23:50.44 | crayz_ | basically I want to be able to use a very customized call queuing system where a call comes into AGI, checks for operators in a database & connects if available, and otherwise queues the calls up and connects them to operators based on rules in the AGI script. also needs ability to "dequeue" calls and push them to an operator via AMI |
| 23:52.02 | crayz_ | from what I've read about the built-in queuing features in asterisk they're not going to work, or would be a huge pain to make work.... all the other dialplan decisions are in AGI, but doing any kind of queuing in AGI seems impossible or at least undocumented |
| 23:53.27 | crayz_ | ideally AGI would just shove the calls into a queue where they'd hear music, and then the background AMI script would decide which calls to connect to which operators - but I can't see any way of doing this either |
| 23:53.59 | [TK]D-Fender | crayz_, Oh boy..... this is looking like you should make your own Queue app |
| 23:54.24 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:54.52 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 23:55.37 | crayz_ | yea, there's about zero chance of me being able to write this in C.... that's why I was hoping to use some combination of AGI/AMI to do it :/ |
| 23:57.09 | [TK]D-Fender | crayz_, Sounds like an ugly hack (more like large series of...) if possible |
| 23:57.21 | Juggie | crayz_, write what |
| 23:58.07 | [TK]D-Fender | <crayz_> basically I want to be able to use a very customized call queuing system where a call comes into AGI, checks for operators in a database & connects if available, and otherwise queues the calls up and connects them to operators based on rules in the AGI script. also needs ability to "dequeue" calls and push them to an operator via AMI |
| 23:58.07 | [TK]D-Fender | <crayz_> from what I've read about the built-in queuing features in asterisk they're not going to work, or would be a huge pain to make work.... all the other dialplan decisions are in AGI, but doing any kind of queuing in AGI seems impossible or at least undocumented |
| 23:58.07 | [TK]D-Fender | <crayz_> ideally AGI would just shove the calls into a queue where they'd hear music, and then the background AMI script would decide which calls to connect to which operators - but I can't see any way of doing this either |
| 23:58.48 | crayz_ | the one thing I was looking at is WaitMusicOnHold indefinitely and then using some AMI command to steal the call and ring to an operator |
