| 00:03.10 | Sweeper | snicker |
| 00:03.15 | Sweeper | he actually went and read the changelog |
| 00:03.19 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:06.28 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 00:07.14 | bryanfe2 | can someone please confirm for me: |
| 00:08.31 | bryanfe2 | is Asterisk *supposed* to be able to send SIP and RTP data back to a SIP client, if the SIP client is behind a NAT and no Stun server is in use? |
| 00:09.05 | bryanfe2 | because in my case, SIP client can register, SIP client can place call just fine. But when Asterisk sends it back it's audio, it's sending it to the client's internal IP (192.168.x.x) instead of it's external IP. |
| 00:09.16 | bryanfe2 | "nat=yes" and "host=dynamic" are correctly configured in sip.conf |
| 00:10.53 | Sweeper | bryanfe2: yea, it's SUPPOSED to play nice when you set nat=yes |
| 00:11.17 | bryanfe2 | thanks sweeper... I kinda knew that, but I have a new fresh install (asterisk 1.4) which isn't playing nice... |
| 00:11.26 | Sweeper | try putting it both in the general settings, and in the peer settings, and then restarting asterisk, and rebooting the phone |
| 00:11.46 | bryanfe2 | SIP commands are fine but RTP data is being sent to the wrong IP |
| 00:12.09 | bryanfe2 | trying a total reboot... |
| 00:12.19 | Sweeper | of the server? :/ |
| 00:12.26 | Sweeper | it's not windows mang :P |
| 00:12.35 | bryanfe2 | i know |
| 00:12.40 | bryanfe2 | I mean restart of AX and my sip client |
| 00:12.51 | bryanfe2 | no change :( |
| 00:13.19 | bryanfe2 | tethereal clearly showing me, all outbound SIP traffic is going to the external IP (correct), and all RTP going to the internal (incorrect) IP |
| 00:15.05 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 00:15.23 | Sweeper | very odd |
| 00:15.33 | bryanfe2 | aaahh! |
| 00:15.36 | bryanfe2 | there goes my last hair |
| 00:16.24 | Sweeper | mm |
| 00:16.35 | Sweeper | question, is your server behind NAT? |
| 00:16.48 | bryanfe2 | yeah |
| 00:16.51 | bryanfe2 | here is my ENTIRE sip.conf |
| 00:16.54 | bryanfe2 | [general] |
| 00:16.56 | bryanfe2 | context=default |
| 00:16.58 | bryanfe2 | disallow=all |
| 00:16.59 | bryanfe2 | allow=ulaw |
| 00:17.01 | bryanfe2 | nat=yes |
| 00:17.02 | bryanfe2 | externip=72.xx.xx.xx ; obfuscated from IRC |
| 00:17.03 | Sweeper | PASTEBIN DAMNIT |
| 00:17.04 | bryanfe2 | localnet=10.xx.xx.xx/24 ; obfuscated from IRC |
| 00:17.06 | bryanfe2 | canreinvite=no |
| 00:17.07 | bryanfe2 | [test3] |
| 00:17.08 | bryanfe2 | type=friend |
| 00:17.10 | bryanfe2 | disallow=all |
| 00:17.11 | bryanfe2 | allow=ulaw |
| 00:17.13 | bryanfe2 | context=default |
| 00:17.15 | bryanfe2 | callerid="Behind a NAT" <456> |
| 00:17.17 | bryanfe2 | nat=yes |
| 00:17.19 | bryanfe2 | qualify=yes |
| 00:17.20 | bryanfe2 | host=dynamic |
| 00:17.22 | bryanfe2 | canreinvite=no |
| 00:17.23 | bryanfe2 | sorry |
| 00:17.30 | Sweeper | http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html <-- check that out |
| 00:17.37 | Sweeper | there's a few different options |
| 00:17.40 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 00:17.55 | Sweeper | and some stuff specifically for asterisk behind a NAT |
| 00:18.10 | bryanfe2 | i looked at that earlier.. |
| 00:18.46 | Sweeper | ah |
| 00:19.10 | *** join/#asterisk DeepY0X (n=DeepY0X@200.121.197.5) |
| 00:23.38 | bryanfe2 | sweeper, I think we have problem "1.4.1" |
| 00:23.46 | Sweeper | ah |
| 00:23.50 | bryanfe2 | but it says 'this happens when nat=never or nat=no", and we most certainly have "nat=yes" |
| 00:24.02 | CCFL_Man2 | SwK: new survivorman soon |
| 00:24.05 | Sweeper | mayb you should go to 1.4.11 ;) |
| 00:24.22 | bryanfe2 | we're at 1.4.9 |
| 00:24.37 | Sweeper | eh, never hurts |
| 00:28.51 | *** join/#asterisk gardo (n=gardo@121.97.193.114) |
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| 00:45.34 | Braxus | Anyone have some ATAs (FXS - Ethernet) to recommend? |
| 00:46.16 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:47.49 | CCFL_Man2 | anyone here set up CAS signalling in a cisco? |
| 00:48.13 | *** join/#asterisk Chris_R (i=Deadly@cpe-66-27-154-43.socal.res.rr.com) |
| 00:51.40 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 01:04.09 | _ShrikE | I have just installed a TC400P on a server that already several g.729 licenses. Which will asterisk use by default? |
| 01:04.16 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:48ce:72ec:0:20e:a6ff:fec4:4e5d) |
| 01:04.32 | luke-jr_ | Where's the current best unlocking PAP2 instructions? :) |
| 01:04.40 | _ShrikE | err. TC400B |
| 01:06.59 | Braxus | http://www.voip-info.org/wiki/view/Linksys+PAP2+Unlocking+Methods < may start here if you haven't found anything yet... |
| 01:08.08 | luke-jr_ | I might also mention, it's a -NA |
| 01:09.37 | *** join/#asterisk nclx (n=nightcal@carnivore.scrapshells.com) |
| 01:12.00 | nclx | I'm having problems getting music on hold working. I have a default config in musiconhold.conf pointing to my /usr/share/asterisk/mohmp3 dir which contains several .mp3 files. I created an exten => 940,1,Answer() exten => 940,2,MusicOnHold() I watch it in the console and it says started music on hold, class default on channel 'SIP/blah blah, but I don't hear anything, any ideas? |
| 01:17.26 | nclx | I do get this when reloading moh |
| 01:17.29 | nclx | niki*CLI> moh reload |
| 01:17.29 | nclx | niki*CLI> |
| 01:17.29 | nclx | 1 class reloaded. |
| 01:17.30 | nclx | <PROTECTED> |
| 01:17.30 | nclx | <PROTECTED> |
| 01:17.30 | nclx | Aug 24 21:16:55 WARNING[26205]: res_musiconhold.c:852 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 01:26.13 | nclx | I read that I may need to ensure my kernel is compiled for a 1000Khz timing device, is there a userland interface to check that thing like cat /proc/cpuinfo or an equiv? |
| 01:28.20 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
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| 01:43.48 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 01:45.29 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 01:50.49 | *** join/#asterisk annonimous (n=ber@189.136.69.141) |
| 01:51.14 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 01:51.50 | annonimous | hiya! just a little doubt i have a sip trunk who in an ata like spa or pap2 expires session in 90 not in the 3600 how can i change the value if i want to set up the sip trunk as sip trunk without an ata? |
| 01:53.16 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 01:54.37 | *** join/#asterisk bkruse_home (n=kruz@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:54.55 | *** part/#asterisk bkruse_home (n=kruz@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:55.17 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:00.34 | CCFL_Man2 | SwK: new survivorman was good |
| 02:01.23 | *** join/#asterisk bkruse_home (n=kruz@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:01.53 | CCFL_Man2 | so, with setting the T1 controller in CAS mode, you need to specify a timeslot and a type for that timeslot, what timeslot do i specify for that channel and what do i chose for the type? |
| 02:02.04 | CCFL_Man2 | with cisco IOS |
| 02:02.48 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:02.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:02.58 | *** join/#asterisk bkruse_home (n=kruz@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:03.35 | *** part/#asterisk bkruse_home (n=kruz@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:05.37 | luke-jr_ | How about an example PAP2-NA .cfg file? "_ |
| 02:11.05 | *** join/#asterisk Strom_M (n=strom@static-68-236-161-53.ny325.east.verizon.net) |
| 02:12.46 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:19.29 | *** join/#asterisk rfernandez (n=ber@189.136.69.141) |
| 02:19.43 | rfernandez | hi anybody here have experience in sip trunking? |
| 02:32.59 | Strom_M | rfernandez: what kind of experience |
| 02:34.29 | rfernandez | Strom_M see i have a little trouble with a sip trunk, if i use my ata and set the register expires at 90 the sip line goes well but if i set it up in asterisk the channel gets active and then i cant make calls (i have 2 lines one in my ata one directly on my asterisk) |
| 02:35.09 | rfernandez | Strom_M and then i need to perform a reload if i want to call...... |
| 02:35.12 | Strom_M | what do you mean, exactly |
| 02:36.38 | rfernandez | Strom_M ok, i mean if i set up directly my sip lines into the asterisk i cant call cause i got an "all circuits are busy now", but if i set up the same lines in an spa2000 i can make calls... |
| 02:37.00 | Strom_M | you get a recording saying "all circuits are busy now"? |
| 02:37.06 | Strom_M | are you using a GUI on top of asterisk? |
| 02:37.22 | rfernandez | Strom_M yes why? |
| 02:37.26 | Strom_M | which GUI? |
| 02:37.35 | rfernandez | freepbx |
| 02:37.39 | *** join/#asterisk aris_g (n=manager@200.71.48.212) |
| 02:37.45 | Strom_M | you'll want to go to #freepbx |
| 02:38.11 | aris_g | hello to all |
| 02:38.29 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 02:38.29 | aris_g | i have a problem..... |
| 02:38.38 | rfernandez | Strom_M well they tell me (i have more of a month fighting with my box..) that if its about sip protocols i need to join here =S |
| 02:39.10 | Strom_M | it sounds like a configuration error, not a protocol problem |
| 02:39.15 | rfernandez | i see |
| 02:39.47 | rfernandez | Strom_M just the last question if you need to change a register timeout from a sip provider which command do you need to use "maxexpirey" r what? |
| 02:41.15 | Strom_M | that can work |
| 02:42.23 | rfernandez | Strom_M k, we are going in a good way then... in some tutorials on the net i found maxexpirey and maxexpiry its the same thing? |
| 02:43.02 | aris_g | i have a asterisk server with 1 e1 conneted to pstn and other e1 connected to a Pbx. I startup a recording service with monitor...I record all calls and users send fax at any time......when this happens the fax falls.... |
| 02:43.20 | coldsteal | i keep getting this in my apterisk -r http://rafb.net/p/whx9f860.html |
| 02:43.27 | coldsteal | could someone help me? |
| 02:43.50 | aris_g | what can i do to this???? |
| 02:43.55 | aris_g | please help me |
| 02:45.45 | aris_g | without Monitor() the fax received well |
| 02:46.40 | aris_g | the fax machine is connect in PBX plant...not in asterisk.... |
| 02:48.53 | aris_g | Can i stop monitor() when exist a fax call? |
| 02:49.03 | *** join/#asterisk d3wayne (n=dwayne@c-71-228-186-75.hsd1.al.comcast.net) |
| 02:49.03 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 02:54.36 | aris_g | aNY IDEA? |
| 02:54.43 | aris_g | sorry. |
| 02:54.48 | aris_g | any idea? |
| 02:56.27 | Strom_M | aris_g: is it an E1 PRI? |
| 02:56.34 | Strom_M | or just straight CAS E1? |
| 02:59.09 | aris_g | yes it's an Pri... |
| 02:59.44 | aris_g | It's an E1 pri |
| 03:00.08 | Strom_M | ok, so faxes are sent to a different DID number than regular phone calls, right? |
| 03:01.26 | *** join/#asterisk enmaca (n=enmaca@189.157.117.149) |
| 03:03.54 | aris_g | no....with the same DID number the PBX plant send call to a fax machine ... with an ivr .. |
| 03:04.11 | Strom_M | how do you detect the faxes? |
| 03:04.53 | aris_g | mm. i can't .... any call will be a faxcall... |
| 03:05.18 | CCFL_Man2 | tsurko: hey man, were you the one who uses the adit 600? |
| 03:05.18 | Strom_M | so wait, are they ALL faxes? |
| 03:05.32 | *** join/#asterisk Kenton (n=chatzill@h-66-167-237-216.mclnva23.dynamic.covad.net) |
| 03:06.07 | aris_g | no... |
| 03:06.13 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:06.30 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:07.06 | *** join/#asterisk umdstu_ (n=rfid@mobile-166-217-048-221.mycingular.net) |
| 03:07.28 | aris_g | they're normal calls but also faxcalls ,, |
| 03:08.15 | aris_g | this is the problem with monitor... |
| 03:08.42 | *** join/#asterisk troy- (n=Troy@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 03:08.59 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 03:09.47 | luke-jr_ | does phone wire go straight thru? |
| 03:14.33 | aris_g | luke-jr_: to me? |
| 03:14.53 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
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| 03:20.35 | luke-jr_ | ... |
| 03:20.59 | JT | luke-jr_: your question was not at all clear |
| 03:21.04 | luke-jr_ | oh well |
| 03:21.09 | luke-jr_ | my phone rings, so I think it does |
| 03:21.36 | luke-jr_ | any ideas on resetting a PAP2-NA I don't know the reset password for? |
| 03:22.34 | CCFL_Man2 | luke-jr_: factory reset |
| 03:23.42 | luke-jr_ | CCFL_Man2: yeah, how? |
| 03:24.15 | CCFL_Man2 | actually, you'll need to know the admit password for that |
| 03:24.20 | CCFL_Man2 | admin |
| 03:24.31 | CCFL_Man2 | it's a vonage adapter? |
| 03:27.38 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 03:28.10 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 03:31.09 | luke-jr_ | CCFL_Man2: no |
| 03:31.12 | luke-jr_ | CCFL_Man2: iConnectHere |
| 03:31.14 | luke-jr_ | so it's a -NA |
| 03:31.55 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 03:32.11 | variable_office | Is it possible to use the RealTime command to select multiple rows? |
| 03:33.02 | CCFL_Man2 | luke-jr_: you gotta call iConnectHere and get the password |
| 03:33.46 | CCFL_Man2 | maybe not |
| 03:33.55 | luke-jr_ | :/ |
| 03:34.17 | variable_office | or does anyone use the realtime command |
| 03:34.18 | CCFL_Man2 | google the factory reset procedure and see if it works without a password |
| 03:34.26 | luke-jr_ | it doesn't |
| 03:35.11 | luke-jr_ | and it won't accept my conifg |
| 03:37.54 | *** join/#asterisk umdstu_ (n=rfid@mobile-166-217-048-221.mycingular.net) |
| 03:55.27 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 03:55.57 | asterisknerds | <PROTECTED> |
| 03:56.20 | *** join/#asterisk OuterSpace (n=OuterSpa@200.121.192.239) |
| 03:56.21 | variable_office | does anyone know what happens if RealTime command gets multiple rows, does it just select the first row? |
| 03:56.36 | OuterSpace | hi, how to drop an stuck sip client ? |
| 03:59.01 | *** join/#asterisk dlynes_laptop (i=dlynes@63.83.93.104) |
| 04:02.36 | *** join/#asterisk bmg505 (n=leon@196.209.183.201) |
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| 04:06.56 | *** join/#asterisk nsphang1117 (n=nsphang@adsl152.dyn212.pacific.net.sg) |
| 04:07.09 | nsphang1117 | hi anyone here can help |
| 04:07.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:09.45 | nsphang1117 | ? |
| 04:13.00 | Kenton | help with what? |
| 04:13.03 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-075-182-081-167.nc.res.rr.com) |
| 04:14.57 | nsphang1117 | Asterisk issues |
| 04:15.08 | nsphang1117 | currently i am having a TDM400 card with 3 FXO modules on it |
| 04:15.47 | nsphang1117 | i have place the modules on port 1,2,4 with the config in the zaptel.conf as fxsks=1,2,4 |
| 04:15.54 | nsphang1117 | however the light on the port 4 does not light up |
| 04:16.03 | nsphang1117 | any way to check whether is the module spoilt? |
| 04:16.10 | J4k3 | swap modules |
| 04:16.16 | J4k3 | ie - swap 1 and 4 |
| 04:18.15 | variable_office | hey j4k3 whats happening? |
| 04:18.29 | J4k3 | nada, enjoying a rare day at 'home' |
| 04:18.43 | variable_office | wireless been keeping you out? |
| 04:18.48 | J4k3 | nah |
| 04:18.51 | J4k3 | female |
| 04:18.56 | J4k3 | ;) |
| 04:19.04 | variable_office | ah, thats far better than wireless |
| 04:19.13 | variable_office | wireless keeps me out, lol |
| 04:20.25 | variable_office | j4k3, you wouldnt happen to know if the realtime command can handle multiple rows? |
| 04:22.22 | OuterSpace | hi, im using asterisk 1.4, im trying to hangup an stuck sip client. sip show channels dont show channelnames ... |
| 04:23.03 | Corydon76-dig | "show channels" |
| 04:23.13 | Corydon76-dig | or "core show channels" |
| 04:24.49 | OuterSpace | 0 active channels |
| 04:25.15 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 04:25.18 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 04:26.04 | Corydon76-dig | Then it's not hung in Asterisk |
| 04:26.09 | Corydon76-dig | Try rebooting the phone |
| 04:26.18 | OuterSpace | on sip show channels it shows a list of ips, i know what ip needs hangup |
| 04:26.48 | Corydon76-dig | Which minor version of 1.4? |
| 04:27.09 | OuterSpace | 1.4.11 |
| 04:27.18 | Corydon76-dig | I remember something like that in early versions, but it shouldn't be like that anymore |
| 04:27.53 | OuterSpace | Call-ID: c9e6a1b5-13c4-46cfaeac-2ed34e5-2756 |
| 04:34.25 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 04:36.21 | *** join/#asterisk pbassel (n=chatzill@CPE00045a8370a4-CM00111ae50e86.cpe.net.cable.rogers.com) |
| 04:37.18 | variable_office | is there a way to search the callerid string in asterisk for a specific substr? |
| 04:38.57 | luke-jr_ | um, sure |
| 04:39.24 | variable_office | how so? |
| 04:40.09 | luke-jr_ | switch(${CALLERID(number)}) { pattern .1234.: <bla>; break; } |
| 04:40.54 | variable_office | luke-jr_ another quetion for you, do you know if you can make realtime retreive multilpe rows for use in the dialplan? |
| 04:40.59 | luke-jr_ | where <bla> can be as simple as matched=1; |
| 04:41.13 | luke-jr_ | no idea on realtime |
| 04:41.25 | variable_office | what is the .1234.: ? |
| 04:41.37 | luke-jr_ | switch(${CALLERID(number)}) { pattern .1234.: matches=1234; break; default: matches=; } |
| 04:41.40 | luke-jr_ | the pattern |
| 04:41.51 | luke-jr_ | that would match '1234' somewhere in it |
| 04:42.40 | variable_office | what would matched=1 equal? it would set the variable ${matched} to one? |
| 04:42.48 | luke-jr_ | yes |
| 04:43.09 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 04:50.14 | CrazyTux[m] | Why would one not want to setup a STUN server? What issues may arise, pros/cons, etc? Would the benifits out way the cons? |
| 04:52.01 | jql | there are cons? :) |
| 04:52.13 | jql | I don't do it because I want to know when someone is natted |
| 04:52.21 | jql | in my logs |
| 04:52.31 | CrazyTux[m] | jql, why would you want to know that/ |
| 04:52.32 | jql | but, that's a rather selfish reason |
| 04:52.46 | jql | debugging purposes |
| 04:53.04 | CrazyTux[m] | jql, hmm, any other reasons you can think of? |
| 04:53.13 | jql | informational and voyeuristic purposes. :) |
| 04:53.28 | jql | I know you run on a 10. network. muahaha |
| 04:53.59 | jql | and, I don't trust stun behind a PIX firewall |
| 04:54.14 | jql | I had troubles, which I didn't feel like tracing down |
| 04:54.32 | CrazyTux[m] | jql, anything specific? |
| 04:54.55 | jql | PIX has a fixup sip setting, which actually rewrites the sip/rtp traffic per the firewall rules |
| 04:55.32 | jql | that fixup caused the packets to go wacky when the soft phone knew its supposed address already |
| 04:55.48 | jql | the PIX is too smart for its own good |
| 04:55.59 | jql | but, you can just turn off stun per UA |
| 04:56.11 | jql | so that's not an argument against the server itself |
| 04:56.24 | CrazyTux[m] | jql, thank you, be right back. |
| 04:57.03 | variable_office | luke-jr_ i cant find any documentation of the switch command? |
| 04:57.08 | variable_office | it doesnt seem to run |
| 04:57.42 | luke-jr_ | ... |
| 04:57.46 | luke-jr_ | it's a statement |
| 04:58.05 | variable_office | exten => _NXXNXXXXXX,101,Switch(${CALLERID}) { pattern .Block.: is_u=yes; break; } wont run |
| 04:58.26 | luke-jr_ | erm, no |
| 04:58.30 | luke-jr_ | it's not .conf |
| 04:58.34 | variable_office | it says -> No application 'Switch' for extension |
| 04:58.40 | luke-jr_ | extensions.ael |
| 04:59.40 | variable_office | does that slow things down? |
| 05:01.22 | variable_office | luke-jr_ ie, is it any slower than .conf ? |
| 05:01.48 | luke-jr_ | only to load it initially |
| 05:02.07 | luke-jr_ | so nothing significant |
| 05:02.33 | variable_office | humm i may have to play with that |
| 05:03.03 | variable_office | i assume you can still call AGI like normal |
| 05:03.21 | luke-jr_ | depends on what you mean by "like normal" |
| 05:03.53 | variable_office | AGI(xxxx) and then the agi outputs "EX |
| 05:04.08 | variable_office | welll it outputs its normal items |
| 05:04.09 | luke-jr_ | you need to add a semicolon at the end |
| 05:04.13 | variable_office | ah |
| 05:14.17 | J4k3 | thank god for sms |
| 05:14.36 | J4k3 | ... cuz thats all that works off any cellular network at my house, dammit :P |
| 05:17.08 | *** part/#asterisk aris_g (n=manager@200.71.48.212) |
| 05:25.57 | CCFL_Man2 | dammit what the hell is tip and ring on this connector |
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| 05:26.22 | CCFL_Man2 | dark blue and light blue, is light blue tip and dark blue ring? |
| 05:29.39 | Sweeper | CCFL_Man2: if it doesn't work, switch it |
| 05:29.46 | Sweeper | half the time they're not to code anyways |
| 05:30.20 | CCFL_Man2 | Sweeper: it'll work with any phone except one that doesn't have a polarity gaurd |
| 05:30.42 | CCFL_Man2 | such as an old WE touchtone |
| 05:31.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:31.18 | CCFL_Man2 | but i want to be correct anyway :P |
| 05:31.41 | asterisknerds | <PROTECTED> |
| 05:31.57 | L|NUX | hello |
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| 06:23.20 | CCFL_Man2 | can anyone explain the difference between a timeslot and channel in CAS? |
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| 06:33.04 | _x86_ | evening |
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| 06:57.25 | CCFL_Man2 | _x86_: i think a call center with a channel bank and fxs ports connected to cheap pots phones is a great idea |
| 06:59.05 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 07:03.58 | L|NUX | can some one help me with asterisk to quintum internconnection ? |
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| 08:03.12 | I-Bz | hi .. does asterisk requires sound card on the machine? |
| 08:06.27 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
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| 08:10.45 | pkunkra | asterisk doesn't really require hardware. |
| 08:11.04 | pkunkra | but if you want to play sound through speakers, you'll obviously need one. |
| 08:11.47 | Bladerunner05 | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 08:12.00 | Bladerunner05 | cpp is installed what it mean ? |
| 08:13.19 | antimoof | you need to look at config.log to see what exactly the error is. |
| 08:13.52 | Bladerunner05 | sure it look for g++ compiler, so resolved |
| 08:14.24 | Bladerunner05 | when I run configure I need to specify anythings usefull ^? |
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| 08:19.52 | tzafrir_laptop | Bladerunner05, what distro? |
| 08:20.24 | tzafrir_laptop | I-Bz, no. |
| 08:21.02 | tzafrir_laptop | I-Bz, It can use a local sound card for a sort-of phone. This is generlly useful for announcements and such, and for testing. |
| 08:21.43 | tzafrir_laptop | But usually you either use voip phones (hardware devices or soft phones) or dedicated hardware to connect to the PSTN |
| 08:22.01 | tzafrir_laptop | or to connect to local phones |
| 08:22.17 | I-Bz | thanks |
| 08:24.02 | I-Bz | does TDM400P has any IRQ conflict with LAN cards? |
| 08:24.36 | jql | that has more to do with your computer than the card itself |
| 08:24.40 | jql | "maybe" |
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| 08:31.52 | J4k3 | cat /proc/interrupts |
| 08:31.59 | J4k3 | see if theres any obvious conflicts |
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| 08:39.36 | Bladerunner05 | How I run asterisk in realtime ? |
| 08:43.58 | coppice | hum. non-real time VoIP. interesting idea :-) |
| 08:50.27 | tzafrir_laptop | Bladerunner05, what meaning of realtime? |
| 08:50.38 | tzafrir_laptop | realtime scheduling priority? the option -p ? |
| 08:54.56 | Bladerunner05 | so the ability to change config file without reloading |
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| 09:12.03 | Bladerunner05 | With latest * and tdm400p (the zapata.conf is here http://www.pastebin.ca/669986) when * hang up a call the caller line remain busy for 20 seconds about |
| 09:19.27 | tzafrir_laptop | Bladerunner05, do you use busydetect? |
| 09:19.43 | tzafrir_laptop | in zapata.conf busydetect=yes |
| 09:20.52 | Bladerunner05 | yes |
| 09:21.56 | Bladerunner05 | this is my zapata.conf http://www.pastebin.ca/669986 |
| 09:28.45 | tzafrir_laptop | Bladerunner05, so remove it if you don't want busydetect |
| 09:28.52 | tzafrir_laptop | This is a feature, not a bug... |
| 09:29.35 | tzafrir_laptop | But do you have a better way to detect a disconnection of a call at the PSTN side? |
| 09:29.54 | tzafrir_laptop | polarity reversal, power denial? |
| 09:32.58 | Bladerunner05 | I try polarity reversal but do the same |
| 09:33.29 | Bladerunner05 | I have to wait 20 seconds about before the line was free |
| 09:33.40 | Bladerunner05 | so I notice in cli> that * give hangup command |
| 09:38.10 | Bladerunner05 | indication.conf set to the right country is usefull in resolving this problem ? |
| 09:41.48 | Bladerunner05 | so it free the channel after 60 seconds past hangup command |
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| 09:55.46 | tzafrir_laptop | Bladerunner05, do you get polarity reversal from you provider? |
| 09:58.14 | Bladerunner05 | I'm in italy I don't know that also I try with and without it |
| 09:59.24 | JT | Bladerunner05: indications.conf only sets asterisk generated tones |
| 09:59.29 | JT | not zaptel generated tones |
| 09:59.34 | JT | that's zaptel.conf |
| 10:00.44 | Bladerunner05 | •JT_• ok |
| 10:01.00 | Bladerunner05 | So where I can found the right tone for italy telecom ? |
| 10:01.37 | Wafa | Bladerunner05, as what tzafrir just asked you .. it is something done by your provider |
| 10:01.55 | Wafa | i had the same issue, the provider did the fix for me |
| 10:02.13 | Wafa | except if you trunk is to a PBX that you own |
| 10:02.39 | Wafa | I also had the problem earlier, and we solved it together with ericsson people |
| 10:03.41 | Bladerunner05 | now I chek |
| 10:05.04 | Wafa | as far as I know it is a polarity issue or what so-called "disconnect supervision" |
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| 10:36.40 | Bladerunner05 | there is a way to see my pstn line parameters with a tool ? |
| 10:39.58 | JT | not ones set by the telco, no |
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| 10:51.35 | Bladerunner05 | c'e' qualche italiano qui ? |
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| 12:18.51 | asterisknerds | <PROTECTED> |
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| 12:43.51 | masus | hiii all |
| 12:44.54 | comatose | hi, i just recognized a problem with my asterisk setup that i cant explain (also havent found anything through google). so i wonder if someone here can point me on the problem. the actual problem is that blind call xfer only works correct if i dial out.. using 't' for incoming call as dial argument, hitting the blind xfer key and dialing an extension gives me no error, in fact it shows the correct extension in the right con |
| 12:44.55 | comatose | text but nothing happens except that the whole call is dropped |
| 12:46.05 | comatose | oh and i should mention that it worked with an older asterisk version |
| 12:46.43 | masus | need an softphone for linux with g729 codec.. |
| 12:46.51 | masus | free ;) |
| 12:48.26 | comatose | anyone? |
| 12:48.55 | masus | nat ? |
| 12:49.06 | masus | nat problems maybe .. |
| 12:49.11 | tzafrir_laptop | comatose, what version? |
| 12:49.20 | comatose | ver 1.4.10.1 |
| 12:49.24 | comatose | and no nat problem.. its ISDN |
| 12:49.31 | masus | hmm |
| 12:49.40 | comatose | (and the problem isnt regarded at all to nat) |
| 12:50.32 | tzafrir_laptop | comatose, maybe you need to set T as well? |
| 12:50.41 | tzafrir_laptop | also: what ISDN? zaptel? |
| 12:50.47 | comatose | no, misdn |
| 12:50.58 | comatose | hmm.. but i wont allow the caller to transfer ... |
| 12:52.01 | comatose | this is what asterisk tells me after i entered an extension |
| 12:52.12 | comatose | <PROTECTED> |
| 12:52.12 | comatose | <PROTECTED> |
| 12:53.46 | comatose | but 250 isnt executed and the call is dropped on both ends |
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| 12:55.52 | asterisknerds | <PROTECTED> |
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| 13:55.30 | asterisknerds | <PROTECTED> |
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| 14:02.23 | _ShrikE | morning gents |
| 14:02.25 | _ShrikE | and ladies |
| 14:02.44 | mvanbaak | mornin |
| 14:03.10 | robl^ | mornin! |
| 14:03.22 | *** mode/#asterisk [+b asterisknerds!*@*] by Qwell |
| 14:03.28 | Qwell | Whoever owns that bot - fix it |
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| 14:09.59 | puzzled | hi |
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| 14:11.23 | kwame | Hi, yesterday I was having an issue with one of my ekiga clients trying to connect to the asterisk server |
| 14:11.44 | kwame | the client is on a vpn (the ip is 10.1.4.x) and the server is in 192.168.98.x |
| 14:11.59 | kwame | there is already a tunnel stablished and the tunnel works correctly |
| 14:12.33 | kwame | I added a line like this in my sip.conf localnet=10.1.4.0/255.255.255.0 |
| 14:12.41 | kwame | and I get registration failed on my ekiga client |
| 14:12.55 | kwame | If I'm connected on the same network the client works with no problems |
| 14:15.43 | kwame | ad |
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| 14:20.53 | Airwolf- | asterisk is periodically call my ringgroup ... i don't know what's happening ... anyone know the reason possibilities for such event ? |
| 14:22.15 | Airwolf- | i suspect it's because of some missed call |
| 14:28.46 | *** join/#asterisk comatose (n=comatose@dslb-084-056-075-251.pools.arcor-ip.net) |
| 14:29.10 | comatose | hi, again |
| 14:29.13 | comatose | anyone here using misdn? |
| 14:31.54 | comatose | i guess i found a bug in asterisk (1.4) / misdn |
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| 15:16.49 | Necromancer78 | hello |
| 15:21.57 | *** join/#asterisk So3kris (n=jan-will@ids.netland.nl) |
| 15:22.12 | So3kris | hello |
| 15:22.32 | So3kris | is someone using pap2 on 1.4.11 |
| 15:23.24 | So3kris | I had update my asterisk from 14.2 to 1.4.11 but my pap2 defices doesn't work |
| 15:25.21 | enex | and another question...has anyone ever had the following problem: outgoing voice is only transmitted after the sentence spoken by the sending party (using a softphone, connected to an Asterisk PBX on the local network) has been finished being spoken (i.e. a period of silence starts). incoming voice (over analogue phone, being converted into VoIP elsewhere/remotely) is fine running without a problem. No dropped packets, latency is fine, CPU und |
| 15:25.21 | enex | RAM on both client and server are below 10%, too ... |
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| 15:39.53 | rickross | anyone out there who could give my system a quick ring to test inbound direct SIP calling? |
| 15:40.08 | rickross | my address is rick@dzone.com |
| 15:40.49 | rickross | We're not sure if we have handling of direct SIP calls going correctly into the IVR |
| 15:41.12 | rickross | or, for that matter, whether a call in that form should go straight to the extension? |
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| 16:39.01 | didge | hello. I am having difficulties using SIP to call my trixbox with ekiga. Ekiga reports that I have a registered account, however it also reports "Security Check Failed." what can I do ? |
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| 16:54.50 | rbd | is it possible to execute an application command through asterisk (e.g. meetmeadmin)? |
| 16:54.57 | rbd | err through the AMI I mean |
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| 17:12.10 | kiwoneka | good afternoon to all |
| 17:13.22 | kiwoneka | i need some help finding a resource that is detailed in setting up callerid for each extension (outbound) i have a did for each extension |
| 17:13.25 | kiwoneka | please help |
| 17:15.29 | sheldonh | kiwoneka: see http://www.voip-info.org/wiki/view/CallerID and http://www.voip-info.org/wiki/view/Setting+Callerid |
| 17:16.14 | *** join/#asterisk decobbb (n=bbradley@66.255.125.146) |
| 17:16.18 | decobbb | hello all |
| 17:16.36 | decobbb | i really could use some help getting my analog card up and going |
| 17:16.54 | decobbb | anyone out there have a few min ? |
| 17:16.59 | decobbb | pls? |
| 17:18.28 | kiwoneka | sheldonh: I thank you, i will start there |
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| 17:37.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:38.20 | variable_office | would it be possible to do pattern matching in extensions.conf like "exten => _NXXNXXXXXX,100,GoToIf($["${CALLERIDNAME}" = ."Blocked". | "${CALLERIDNAME}" = "Unavailable"]?105)" ? |
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| 18:00.37 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 18:01.41 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:02.05 | kiwoneka | sheldonh: that did the trick, thanks very much |
| 18:03.07 | Nivex | How do I make a Cisco 7960 display a name on an outbound call? That is, if I dial 6123 I'd like the display to say "To Fred Flinstone 6123" |
| 18:03.40 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:04.30 | Sweeper | Nivex: probably need to have the number in the phone's directory |
| 18:05.05 | *** join/#asterisk miknix (n=miknix@bl8-81-16.dsl.telepac.pt) |
| 18:05.27 | miknix | hello |
| 18:06.17 | Nivex | Sweeper: no joy |
| 18:06.47 | Nivex | I figured there'd be some SIP message that could get sent back to the phone to indicate the called ID |
| 18:07.05 | Nivex | since there is no way a single phone could store a large company's entire phonebook |
| 18:07.10 | Sweeper | uh |
| 18:07.13 | miknix | does asterisk sip protocol support PUBLISH and NOTIFY events for SIMPLE/SIP IM messaging? |
| 18:07.13 | Sweeper | why not? |
| 18:07.18 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 18:07.22 | Sweeper | it holds a 15mb firmware image |
| 18:07.39 | Sweeper | a "large phonebook" would be what, 50kb? |
| 18:08.35 | Sweeper | that'd be really big, actually |
| 18:08.50 | Sweeper | since an entry should only be 50B, tops |
| 18:09.01 | Sweeper | that's 1000 entries mang :P |
| 18:09.46 | Nivex | ok, so it's theoretically possible |
| 18:09.51 | Nivex | now how do I *do* it? :-P |
| 18:10.13 | Sweeper | look up the provisioning docs for your phon |
| 18:10.15 | Sweeper | phone |
| 18:20.22 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 18:20.41 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:24.16 | Yourname` | How good is it to do svnupdate? |
| 18:24.22 | Yourname` | on 1.4 |
| 18:25.32 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 18:27.47 | *** join/#asterisk logicwrath (n=some@c-68-41-24-98.hsd1.mi.comcast.net) |
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| 18:36.24 | russellb | Yourname`: svn up |
| 18:36.35 | russellb | Yourname`: but you must be running from a svn checkout to do that, not a tarball |
| 18:37.25 | Yourname` | russellb: I recently grabbed the svn checkout, and have been using it due to a bug file had closed. Now, I'm just wondering how good is it for be to update it.. (and if I can see the changes that have been made in recent commits as old as 5-6 days ago) |
| 18:37.50 | russellb | should be fine :) |
| 18:38.00 | russellb | besides, you can always go back |
| 18:38.04 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:38.06 | russellb | "svn info" to see what revision you're at now |
| 18:38.11 | russellb | "svn up" to get up to date |
| 18:38.25 | russellb | "svn up -r 80000" to update to a specific revision |
| 18:39.22 | Yourname` | Shit, did the info later, lol |
| 18:39.38 | Yourname` | And then I look up the revision number to see the changes on svn.digium.com? |
| 18:40.20 | Yourname` | (well, just did the svn up and now making) |
| 18:40.42 | luke-jr_ | anyone know how to unlock a PAP2-NA? |
| 18:43.50 | russellb | Yourname`: yeah, you can look at http://svn.digium.com to have a web view of changes |
| 18:44.02 | russellb | or you can look at logs from the command line |
| 18:44.18 | russellb | svn log -r 81000:HEAD gives you the log of changes sine revision 81000 |
| 18:44.47 | miknix | sorry for this stupid question but could asterisk act as sip registrar? I'm looking to the configuration files and it doesn't seem to.. |
| 18:44.51 | Yourname` | Gotcha. No biggie.. all I will need to do when I do an svn update is make sure of http://lists.digium.com/pipermail/asterisk-dev/2006-May/020838.html idx <= issue |
| 18:45.38 | russellb | miknix: yeah |
| 18:45.56 | russellb | miknix: a peer defined in sip.conf with "host=dynamic" is a peer that will register with asterisk. |
| 18:46.35 | miknix | russellb> thanks! www.asteriskdocs is down. I really had not a clue about that |
| 18:48.18 | Yourname` | Oh, umm, russellb.. sorry to ask this, after svn update, is configure needed? Or can I do a make? |
| 18:48.45 | Yourname` | I have a feeling after the update I should've make clean; configure and then make |
| 18:48.49 | Yourname` | :S |
| 18:48.51 | russellb | Yourname`: configure is only needed if 1) the configure script changed, or 2) you have installed new libs that you want it to find |
| 18:49.08 | russellb | #1 is automatically detected and the makefile will tell you to run it |
| 18:49.11 | russellb | #2 is up to you :) |
| 18:49.35 | Yourname` | ah, because I restarted asterisk and the version still says Asterisk SVN-branch-1.4-r80088M built by root @ nasa on a i686 running Linux on 2007-08-25 18:41:39 UTC |
| 18:49.48 | Yourname` | I was expecting to see 80895 |
| 18:49.50 | russellb | Yourname`: ah, that's odd. |
| 18:50.00 | russellb | make clean would certainly fix that |
| 18:50.05 | Yourname` | Gotcha, thanks! |
| 18:50.07 | russellb | should have been done automatically for you though ... |
| 18:50.12 | russellb | oh well. |
| 18:50.25 | Yourname` | heh |
| 18:51.57 | Yourname` | Another little question I have is, can I set queue-priority for an inbound number? For example, exten=>5125552222,1,goto(testq,100,1) .. and then there's testq, but can I see the queue priority right on the inbound exten so when it goes into the queue it already knows what priority the calls are on? |
| 18:53.50 | russellb | i'm sure there is a way to do what you want, but i rarely touch the queue code. |
| 18:53.53 | russellb | so i don't remember |
| 18:54.52 | Yourname` | ah, All good. |
| 18:54.58 | Yourname` | Thanks a lot for this help, russellb . |
| 18:55.51 | russellb | np |
| 18:59.18 | miknix | russellb> and now how do I make it listen for REGISTER messages? I have bind addr and port set, but after starting asterisk the port is closed. Am I missing something? |
| 18:59.27 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584025.dsl.bell.ca) |
| 19:00.03 | russellb | as long as you hvae the right bindport/bindaddr settings, it should just do it when you start asterisk (and have chan_sip loaded) |
| 19:00.17 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
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| 19:04.40 | miknix | loader.c: /usr/lib64/asterisk/modules/chan_sip.so: undefined symbol: ast_park_call |
| 19:04.40 | miknix | Aug 25 20:03:55 WARNING[5523] loader.c: Loading module chan_sip.so failed |
| 19:05.57 | miknix | wtf? who called mpg123? It's burning my cpu |
| 19:07.14 | *** join/#asterisk Blue_Ice (n=Blue_Ice@195-130-159-121.iFiber.telenet-ops.be) |
| 19:10.05 | miknix | done.. ast_park_call was in res_features.so |
| 19:10.35 | *** join/#asterisk Strom_M (n=strom@65.42.208.133) |
| 19:12.08 | miknix | russellb> damn.. chan_sip is being loaded. but sip port keeps closed. |
| 19:12.16 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:14.18 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 19:14.21 | Lucky7 | Hey everyone |
| 19:14.26 | Lucky7 | i got a pretty neat issue |
| 19:14.31 | Lucky7 | i've got a PBX setup with a T1 |
| 19:14.33 | Yourname` | Is there a way I can slow down sip re-registrations via CLI? There's this one agent whose phone keeps re-registering.. |
| 19:14.37 | Lucky7 | em winkstart |
| 19:14.42 | Lucky7 | I'm able to call in |
| 19:14.44 | Lucky7 | perfectly |
| 19:14.58 | Lucky7 | but if I call OUT, the system doesn't recognise that the calls been picked up on the other end. |
| 19:15.09 | Lucky7 | it just rings and rings and rings, even though i've picked up the phone |
| 19:16.08 | russellb | miknix: another application using that port? |
| 19:16.34 | miknix | russellb> no.. |
| 19:16.46 | luke-jr_ | anyone know how to unlock a PAP2-NA? :/ |
| 19:17.03 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 19:17.17 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 19:17.28 | russellb | miknix: try "module unload chan_sip.so" and "module load chan_sip.so" and see if you get any error messages |
| 19:19.01 | miknix | russellb> no problem.. |
| 19:19.33 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:20.37 | miknix | russellb> restarting the daemon only shows > Aug 25 20:11:02 NOTICE[7467] cdr.c: CDR simple logging enabled. on the log |
| 19:23.03 | Lucky7 | can anyone point me in the right direction for looking for my problem |
| 19:23.10 | Lucky7 | I know all the T1 stuff is right. |
| 19:23.11 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 19:23.22 | Lucky7 | i walked through that with the T1 provider |
| 19:28.06 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 19:28.52 | miknix | russellb> I'm currently dispending too much time with a sip registrar, I have to continue developing a sip client for a university class. time is running out.. thanks for your help |
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| 19:29.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:29.29 | Lucky7 | I'm perplexed. |
| 19:29.43 | Lucky7 | if the call can come in perfectly fine |
| 19:29.49 | Lucky7 | but can't dial out |
| 19:29.52 | Lucky7 | over a T1 |
| 19:30.29 | elixer | Lucky7: what provider? |
| 19:30.45 | Lucky7 | XO Communications |
| 19:30.50 | Lucky7 | using em_w start |
| 19:30.59 | elixer | hmmm |
| 19:31.13 | elixer | had the same problem here a few days ago with at&t |
| 19:31.17 | Lucky7 | I can call in, and everything works perfect, sounds beautiful. |
| 19:31.31 | elixer | mucking around with the nsf value in zapata.conf seemed to fix it |
| 19:31.38 | elixer | might be specific to at&t though |
| 19:31.41 | Lucky7 | but if I call out, my cell rings, I answer it, but the asterisk box never notices i picked up |
| 19:31.45 | elixer | ohhh |
| 19:32.15 | elixer | yeah i dunno |
| 19:32.21 | elixer | :) |
| 19:32.44 | Strom_M | Lucky7: paste your Dial() line |
| 19:33.02 | Lucky7 | in my dialplan, or the output in asterisk? |
| 19:33.08 | Strom_M | in your dialplan |
| 19:33.09 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 19:33.22 | linagee | what's the best way to set up a SIP phone behind a NAT firewall? |
| 19:33.28 | Lucky7 | lemme grab it |
| 19:33.37 | Lucky7 | linagee > thats a nasty thing to attempt |
| 19:33.44 | Strom_M | linagee: open ports and enable some sort of keepalive |
| 19:33.51 | Strom_M | and set nat=yes on asterisk |
| 19:34.00 | elixer | ~sipnat |
| 19:34.00 | 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 |
| 19:34.08 | miknix | linagee> iptables has some sip connection helpers |
| 19:37.32 | russellb | you shouldn't have to use such hacks as "connection helpers" to get things to work ... SIP is so terrible ... |
| 19:38.50 | file | russellb: O.O |
| 19:39.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:41.05 | Strom_M | Lucky7: find that Dial() statement yet? |
| 19:41.20 | russellb | file: ! hey ! |
| 19:41.29 | elixer | dialplan stalls on a Playback() and you don't actually hear anything on your SIP phone, does this sound familiar to anyone? |
| 19:41.31 | Lucky7 | yea, its after a macro tho, so it doesn't make sense to me. one sec |
| 19:41.37 | file | russellb: hi! |
| 19:41.53 | Strom_M | Lucky7: paste it anyway |
| 19:42.31 | russellb | Yourname`: there's your hero now. |
| 19:42.47 | russellb | file: you rock. |
| 19:42.50 | miknix | man.. why did I accepted to develop a sip client! openser has partial SIMPLE/SIP implementation. ser SIMPLE/SIP |
| 19:42.52 | file | russellb: you roll. |
| 19:43.01 | Lucky7 | http://rafb.net/p/hkaHwf30.html |
| 19:43.07 | Lucky7 | dial is on line 28 |
| 19:43.22 | Yourname` | lol |
| 19:43.25 | miknix | is under heavy development. asterisk has 20000 configuration files. |
| 19:43.28 | Lucky7 | thats the crazy ass context though. |
| 19:43.28 | Strom_M | Lucky7: holy jesus |
| 19:43.32 | Yourname` | file: Another little question I have is, can I set queue-priority for an inbound number? For example, exten=>5125552222,1,goto(testq,100,1) .. and then there's testq, but can I see the queue priority right on the inbound exten so when it goes into the queue it already knows what priority the calls are on? |
| 19:43.36 | Strom_M | Lucky7: pastebin some CLI output :) |
| 19:43.40 | russellb | miknix: 20k? i disagree. |
| 19:43.49 | russellb | miknix: your problem began with the word "SIP" :) |
| 19:44.07 | russellb | o.O |
| 19:44.09 | file | so russellb, what are you doing here? |
| 19:44.17 | russellb | file: having a relaxing weekend? |
| 19:44.20 | Lucky7 | http://rafb.net/p/bKqtv389.html |
| 19:44.26 | file | russellb: usually relaxing weekends do not include #asterisk |
| 19:44.28 | Lucky7 | insane shit. |
| 19:44.52 | russellb | miknix: i'm sorry ... |
| 19:45.00 | russellb | miknix: it's certainly a projet within reah |
| 19:45.13 | russellb | miknix: but you need to define exatly what you want to accomplish |
| 19:45.25 | russellb | "SIP client" isn't enough, as there are thousands of pages of speifications |
| 19:45.37 | russellb | you have to define some subset of the protocol you are aiming for |
| 19:45.38 | elixer | russellb: is your 'c' key on the fritz? |
| 19:45.44 | russellb | elixer: yes, it is. |
| 19:45.48 | russellb | it's messed up on this laptop. |
| 19:46.04 | miknix | russellb> I currently have my client working.. It's making the REGISTER spec with digest realm. |
| 19:46.12 | elixer | well, i spilled a 32 ounce glass of water on my laptop last week. so it could be worse. |
| 19:46.13 | russellb | cool :) |
| 19:46.14 | Strom_M | Lucky7: alright...i suspected something which isnt the case |
| 19:47.19 | miknix | russellb> I was using SER.. although ser isn't supportinng very well the PUBLISH and NOTIFY messages |
| 19:47.26 | Strom_M | Lucky7: but I really dont know enough about freepbx to help you further |
| 19:47.45 | Lucky7 | whats the standard layout for the dial |
| 19:47.47 | russellb | miknix: ah. well, Asterisk has limited support in that area, as well. |
| 19:47.52 | Lucky7 | dial(sometihng,30,tr) |
| 19:47.58 | miknix | russellb> so I stoped here.. I'm currently looking for something that understands the SIMPLE/SIP spec |
| 19:48.01 | russellb | miknix: I don't think Asterisk supports receiving PUBLISH |
| 19:48.06 | miknix | damn.. |
| 19:48.10 | file | it does not. |
| 19:48.16 | russellb | you can SUBSCRIBE, and * will send NOTIFY |
| 19:48.33 | russellb | furthermore, we have limited, almost no support for SIMPLE |
| 19:48.52 | russellb | well, I think we can support it during a call? |
| 19:48.53 | miknix | damn |
| 19:49.05 | miknix | now I'm totally f**cked |
| 19:49.06 | russellb | but not messages outside of a call ... |
| 19:49.35 | miknix | russellb> My simple client wont understant calling functions.. only IM related |
| 19:49.43 | russellb | ah. |
| 19:49.49 | russellb | then Asterisk would not work for testing that. |
| 19:50.00 | miknix | any idea though? |
| 19:50.11 | russellb | tried openser? |
| 19:50.28 | russellb | and there is another open source SIP server called sipx |
| 19:50.37 | russellb | i don't know if either one of them support it or not |
| 19:51.04 | miknix | the PA module (presence one) of openser has partial implmentation only of SIMPLE/SIP |
| 19:51.06 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 19:51.17 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 19:51.20 | miknix | well.. it looks like I'll give a shot to sipx |
| 19:51.44 | variable_office | is there a appliction to tell what voicemailbox a given sip user has? |
| 19:51.53 | variable_office | ie the default mailbox |
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| 19:51.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:52.07 | miknix | thank you all |
| 19:52.12 | russellb | you're welcome |
| 19:52.27 | Strom_M | variable_office: "sip show user xxxx" |
| 19:52.46 | miknix | damn! sipX is java based.. now I'm totally f**ked |
| 19:52.54 | variable_office | does that work with realtime as well? |
| 19:53.39 | pacneil | can someone tell me what IAXINFO variable in extensions.conf is for? Should I just comment it out to start? |
| 19:54.12 | pacneil | I have been reading the book, but I don't see that information |
| 19:54.30 | russellb | pacneil: it's probably just an example of how to define a variable |
| 19:55.27 | pacneil | OK, most stuff is commented out, but that wasn't IAXINFO=guest |
| 19:55.35 | elixer | here is the problem i am having, i have two SIP phones, but on the internal 10.* network. when i try to dial to the other, i get no ringing at all. if i put a Playback(goodbye) before the Dial() in my extensions.conf, asterisk stalls on the Playback and i don't hear anything on the calling phone. here is the pastebin of an example (not my actual extensions.conf): http://pastebin.com/d36a81b7f |
| 19:55.48 | elixer | s/but on/both on/ |
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| 19:56.33 | elixer | this is asterisk 1.4.8 |
| 19:56.55 | elixer | nothing of interest in the log files either, even with core set verbose 1000/core set debug 1000 |
| 19:58.03 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 19:58.30 | elixer | no? no lovin? |
| 19:58.32 | elixer | heh |
| 19:59.41 | variable_office | Strom_M i meant more of; can i do this in extensions.conf to make sure that i send the caller into the correct voicemailbox for the user |
| 19:59.41 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 20:00.39 | Blue_Ice | is it possible do display the destination nr on the telephone? |
| 20:00.47 | Blue_Ice | next to the originating number of the caller |
| 20:05.09 | elixer | ok, it looks like Playback() /always/ stalls |
| 20:06.07 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:07.55 | luke-jr_ | anyone know how to unlock a PAP2-NA? :/ |
| 20:08.50 | russellb | google? |
| 20:09.04 | russellb | i have never done it, but i know there are articles out there for it .. |
| 20:15.27 | elixer | ok, now this is weird. if i `service zaptel stop` then everything works perfectly |
| 20:17.10 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 20:19.21 | luke-jr_ | russellb: I googled all yesterday, and haven't found a solution |
| 20:19.55 | luke-jr_ | everything is geared at unlocking the PAP2-VA (Vonage) |
| 20:20.32 | russellb | ahh |
| 20:20.34 | CoaxD | remove chip. apply 9v battery to all pins of chip. replace chip. voila. |
| 20:20.48 | luke-jr_ | CoaxD: seriously? :x |
| 20:21.20 | CoaxD | yeah it'll make it go real good |
| 20:21.55 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 20:22.00 | luke-jr_ | ... |
| 20:23.07 | pacneil | I have a bridge for sale. |
| 20:23.31 | luke-jr_ | sounds expensive |
| 20:23.33 | pacneil | It's in Brooklyn and I take paypal. :-) |
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| 20:24.34 | elixer | russellb: you're the menuselect guy, yeah? or is that kevin? |
| 20:26.21 | elixer | russellb: there is like a two second pause when i press ESC to get back to the previous menu, and i am really impatient. so when you get around to it ;-) |
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| 20:49.51 | Strom_M | elixer: there's no delay when I use menuselect |
| 20:57.19 | russellb | elixer: ha ... haven't seen that before. |
| 20:58.35 | russellb | elixer: oh, just when press escape? neat... |
| 20:58.48 | russellb | no delay with the left arrow ... |
| 21:03.11 | russellb | ah. that's just how ncurses works apparently... |
| 21:10.20 | variable_office | what is a good method to implement *67 for a user? |
| 21:16.52 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 21:17.08 | elixer | russellb: no way. that's weak. i guess its waiting on a escape sequence. |
| 21:17.40 | elixer | could you just roll your own text-ui implementation? kthx |
| 21:17.41 | elixer | heh |
| 21:23.50 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584025.dsl.bell.ca) |
| 21:27.27 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:34.09 | *** join/#asterisk tomcontr3 (n=tomcontr@1-79-246-201.adsl.terra.cl) |
| 21:41.23 | *** join/#asterisk Weezey (n=ohno@206.210.111.32) |
| 21:42.06 | Weezey | anyone run sangoma and digium cards in the same box? I need to figure out which start order works (zaptel, wanrouter, etc..) best. |
| 21:44.11 | *** join/#asterisk JoseBravo (n=jbravo@190.157.26.81) |
| 21:46.01 | *** join/#asterisk bkruse_home (n=kruz@user-24-214-44-217.knology.net) |
| 21:46.56 | russellb | elixer: you're correct, it's waiting for an escape sequence and then times out |
| 21:47.14 | bkruse_home | russellb: ! |
| 21:47.20 | russellb | bkruse_home: greetings kind sir |
| 21:47.27 | bkruse_home | russellb: what you up to today |
| 21:47.40 | russellb | um ... playing video games, pulling a few weeds ... that's about it. :) |
| 21:47.47 | russellb | a little bit of coding this mornin' |
| 21:48.03 | bkruse_home | russellb: nice as always |
| 21:48.03 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:48.10 | russellb | bkruse_home: yourself? |
| 21:48.19 | bkruse_home | At my aunts house, some fam is in town. |
| 21:48.28 | russellb | fun times |
| 21:48.31 | bkruse_home | You should come and throw easter eggs with the cousins... |
| 21:48.34 | bkruse_home | lol |
| 21:48.35 | JoseBravo | I have a Zaptel Trunk, but when the caller hang up astersik didn't detected. |
| 21:51.37 | tzafrir_laptop | JoseBravo, analog? |
| 21:52.26 | tzafrir_laptop | Weezey, from what I remember: zaptel first, wanrouter later. But I'm not really sure |
| 21:52.44 | bkruse_home | tzafrir_laptop: I saw that sysfs commit last night |
| 21:52.53 | bkruse_home | tzafrir_laptop++ I am going to look into using it |
| 21:55.55 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 22:06.19 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:11.56 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584025.dsl.bell.ca) |
| 22:19.48 | *** join/#asterisk jkimball4 (n=jerrid@wsip-70-165-105-40.om.om.cox.net) |
| 22:20.00 | jkimball4 | Is there some sort of definitive reference for AMI output? |
| 22:24.27 | *** join/#asterisk pc500 (n=fwea@75-92-50-241.boi.clearwire-dns.net) |
| 22:24.33 | pc500 | How do you set the MEdia/RTP port on X-lite? |
| 22:27.02 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:31.24 | JoseBravo | tzafrir_laptop yes |
| 22:33.51 | *** join/#asterisk wacker (n=wacker@wb2flw.octothorp.org) |
| 22:37.41 | tzafrir_laptop | can you pastebin your zapata.conf? |
| 22:37.47 | wacker | Has anyone had success with ztdummy and kernel-2.6.22? It seeams that rtc_register, rtc_unregister and rtc_control have disappeared from the kernel. |
| 22:38.01 | tzafrir_laptop | hmmm... he left |
| 22:38.42 | tzafrir_laptop | wacker, http://bugs.digium.com/view.php?id=10314 |
| 22:38.53 | wacker | Thanks. |
| 22:40.11 | *** join/#asterisk ManxPower (n=manxpowe@015-815-281.area5.spcsdns.net) |
| 22:41.24 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 22:41.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:42.09 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 22:42.09 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:45.17 | *** join/#asterisk pc600 (n=fwea@75-92-50-241.boi.clearwire-dns.net) |
| 22:45.19 | pc600 | Hoes anyone know how to set the MEdia/RTP port on X-lite that it conencts to the server with? How is that figure negotiated? I set 10000-10040 in rtp.conf on asterisk, but the x-lite clienet still tried to connect to some random port in the 30,000s outside that range. |
| 22:48.58 | *** part/#asterisk miknix (n=miknix@bl8-81-16.dsl.telepac.pt) |
| 22:49.49 | pc600 | anyone? |
| 22:51.21 | lirakis | pc600: not sure .. i use the linux x-lite which is different than the windows version |
| 22:51.23 | ManxPower | pc600: you are confused. |
| 22:51.42 | ManxPower | every packet has 4 pieces of header info. SOURCE IP and PORT and DESTINATION IP and PORT. |
| 22:51.58 | pc600 | ManxPower - Yes, how does a sip client decide what destination ports to use? |
| 22:51.58 | ManxPower | You don't care about the source port on the X-lite machine |
| 22:52.16 | pc600 | I don't, I care about the destination on the asterisk server (from the client). |
| 22:52.19 | ManxPower | pc600: "sip show peers" shows the SOURCE port on the client side. |
| 22:52.33 | ManxPower | YES, you DO care about the destination port on the Asterisk |
| 22:52.56 | ManxPower | the port number is decided based on the RTP setup info exchanged. |
| 22:53.05 | pc600 | Yes, it connects via SIP to 5060. How do I set the RTP/media, only opened when a call session starts, to 10000-10038? |
| 22:53.29 | pc600 | How is that negotiated? Does the asterisk server tell the client what MEDIA port to connect on when it replies to it's request for a call on 5060? |
| 22:53.35 | pc600 | So I specify that on the client? |
| 22:53.39 | ManxPower | pc600: the part you care about is i rtp.conf |
| 22:53.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 22:53.52 | pc600 | on my Cisco phone, I can set "RTP MEDIA PORTS", and it works great. |
| 22:54.00 | pc600 | I also set rtp.con to 10000-10038 |
| 22:54.16 | pc600 | But x-lite tried to connect to * on 32,000 or something, which gets dropped. |
| 22:54.17 | ManxPower | pc600: There is NO NEED to set it on the client unless your NAT box is a piece of shit AND you want to run multpile phones behind the same NAT |
| 22:54.29 | pc600 | ManxPower - there is a firewall at the server which limits me to 40 udp ports. |
| 22:54.43 | pc600 | ManxPower - FIrewall, not NAT box, keep in mind. |
| 22:54.48 | ManxPower | pc600: So your firewall is a piece of shit then |
| 22:54.55 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 22:55.05 | pc600 | no, it works fine. I just need to know what are the standard RTP media ports, how it is negotiated, and how I set them. |
| 22:55.30 | ManxPower | pc600: since nobody ever needs to set the source port in X-lite, so I guess you'll have to contact the vendor for help with that. |
| 22:55.37 | ManxPower | THERE IS NO STANDARD RTP PORT RANGE. |
| 22:55.51 | pc600 | I need my x-lite client to connect on a specified RTP range at the server. Is this a client side option? Is this something the server tells the client to connect on during call setup negotiations via SIP/5060? |
| 22:55.52 | ManxPower | Now say it with me! There is no standard RTP port range. |
| 22:56.20 | pc600 | The servr clearly has to listen on a range. |
| 22:56.21 | ManxPower | pc600: the server tells the client what ports it is listening on. |
| 22:56.35 | ManxPower | Not the range, the port. |
| 22:56.43 | pc600 | ManxPower - correct, it tells it what to use. |
| 22:56.53 | pc600 | So how do I tell asterisk what to tell the client? |
| 22:57.15 | pc600 | because if it's rtp.conf, it's broken, or x-lite is ignoring it. |
| 22:57.29 | ManxPower | pc600: neither is the case. |
| 22:57.48 | lirakis | pc600: .. the client has to send to the port * is listening to |
| 22:57.53 | ManxPower | do a packet dump. Tell me what the source ip/port is and what the dest ip/port is. |
| 22:58.15 | lirakis | s/port/ports |
| 22:58.20 | pc600 | ManxPower - It negotiated the call setup on 5060/sip, then began RTP to 32,xxx something, which of course fails. |
| 22:58.41 | pc600 | DId asterisk tell the client, via SIP, to conenct on 32xxx? |
| 22:58.47 | pc600 | for the media? |
| 22:58.57 | ManxPower | pc600: what is the OTHER port of the packet |
| 22:59.10 | ManxPower | you are talking like each packet has only one port. |
| 22:59.26 | ManxPower | so what is the OTHER port. one of them is 32,xxx. |
| 22:59.30 | pc600 | I don't care much about the local port for a UDP packet. but none the less: |
| 22:59.35 | lirakis | pc600: ... |
| 22:59.38 | ManxPower | YES YOU DO!!!!!!!!!!!!!!!! |
| 22:59.45 | lirakis | pc600: src->dest ... |
| 22:59.50 | pc600 | client ip: x.x.x.x:10001 ---> server ip x.x.x.x:32434 |
| 22:59.53 | lirakis | pc600: dest->src |
| 23:00.00 | pc600 | lirakisr - MY problem is with SRC>DEST |
| 23:00.02 | pc600 | It ain't getting there |
| 23:00.07 | ManxPower | pc600: what are you using to find this information out. |
| 23:00.08 | lirakis | you can send from port 8800235923 to port 10000 if you want |
| 23:00.12 | pc600 | So DEST>SRC won't be an issue (yet) |
| 23:00.16 | pc600 | ManxPower - tethereal |
| 23:00.25 | pc600 | and wireshark on the client side. |
| 23:00.26 | lirakis | sigh... |
| 23:00.39 | ManxPower | pc600: well post your rtp.conf on pastebin |
| 23:00.55 | ManxPower | Asterisk defaults to 10000-20000 if there is no rtp.conf |
| 23:00.59 | pc600 | So yes, the return port matters to the client, but in my case I'm not getting that far that it does. |
| 23:01.18 | ManxPower | I think you have them reversed. |
| 23:01.31 | ManxPower | if your rtp.conf is not being read then the SERVER would be using port 10001 |
| 23:01.35 | ManxPower | that would be expected. |
| 23:02.01 | ManxPower | also remember that there is an RTP port for each direction of the audio. |
| 23:02.07 | ManxPower | well port pair at least. |
| 23:02.41 | pc600 | yes, but udp is stateless and my intial traffic isn't even getting there :( |
| 23:02.45 | pc600 | one sec, getting rtp.conf |
| 23:03.06 | ManxPower | why not just tell your firewall to allow 10000-10039 |
| 23:03.28 | ManxPower | that will give you 20 simul calls before you run out of ports. |
| 23:03.40 | pc600 | http://pastebin.com/dfbb6724 |
| 23:03.52 | pc600 | That's what I did (well, to 38) :) |
| 23:05.01 | ManxPower | If you reverse your port numbers that would be correct then |
| 23:05.48 | pc600 | ? Is that rtp.conf wrong? |
| 23:06.16 | ManxPower | No, I think your reading your packet dumps wrong. It would be too much of a cooncidence. |
| 23:06.56 | ManxPower | What ERROR are you getting when you try to make a call andway? |
| 23:07.15 | pc600 | times out |
| 23:07.17 | pc600 | during setup |
| 23:07.25 | pc600 | rtp traffic outbound from client |
| 23:07.30 | ManxPower | And the actual error message is..... |
| 23:07.32 | pc600 | on invalid port |
| 23:07.43 | pc600 | never arrives according to a sniff at the other side, firewalled. |
| 23:07.48 | pc600 | client dump shows invalid destination port from client |
| 23:07.58 | pc600 | ManxPower - fast busy, locally generated error on x-lite |
| 23:08.08 | ManxPower | Is the IP correct? |
| 23:08.21 | pc600 | Setup works fine if I VPN in past the firewall. |
| 23:08.26 | ManxPower | And you never answered my question about special SIP support in the firewall. |
| 23:08.28 | pc600 | Yes, it registers just fine. |
| 23:08.37 | pc600 | yes, cisco pix with "inspect sip" enabled. |
| 23:08.51 | bkruse | russellb: |
| 23:08.52 | ManxPower | TURN THAT OFF!!!!!!!!!! It screws up all the IP and port infor |
| 23:09.16 | pc600 | rtp doesn't pass right without it, period. |
| 23:09.24 | ManxPower | It is only useful for NAT and dynamic RTP port assignments. |
| 23:09.34 | pc600 | on the client side, there is nat. |
| 23:09.36 | ManxPower | I give up. |
| 23:09.54 | ManxPower | and you have nat=no on the asterisk side for that client? |
| 23:10.02 | pc600 | looking |
| 23:10.17 | ManxPower | You CANNOT do SIP fixup in the router and nat=yes. It won't work. |
| 23:10.19 | pc600 | sip show peers shows nat as empty, which usually means yes. |
| 23:10.45 | pc600 | Do you not need nat with sip fixup? |
| 23:10.56 | ManxPower | sip fixup does nat translation for you. |
| 23:11.05 | ManxPower | turn off one or the other. |
| 23:11.16 | pc600 | Ok, I'll turn it off. It's inspect sip and inspect rtsp in pix 7 (6.3 and less was fixup) |
| 23:11.25 | ManxPower | turn them both off |
| 23:11.35 | pc600 | done |
| 23:11.40 | ManxPower | You are manually allowing the RTP ports anyway, right? |
| 23:11.41 | pc600 | now I got to do a clear xlate |
| 23:11.42 | pc600 | might get disco |
| 23:11.50 | pc600 | I allow all on the client side |
| 23:11.55 | pc600 | permit ip any any |
| 23:12.26 | *** join/#asterisk pc500 (n=fwea@75-92-50-241.boi.clearwire-dns.net) |
| 23:12.27 | pc500 | ok back |
| 23:12.34 | pc500 | all that crap is off now |
| 23:13.06 | ManxPower | BTW, Cisco has a history of screwing up SIP if you enable it's SIP support. |
| 23:13.08 | pc500 | And if it still doens't work, I'm going to do a damn capture on the outside interface of the pix and see where it's going (compared to where the client is sending it) |
| 23:13.41 | ManxPower | SIP debug will tell you the ports the clients decide on |
| 23:13.54 | ManxPower | "sip debug" and "sip no debug" |
| 23:14.00 | ManxPower | in the asterisk cli |
| 23:14.14 | pc500 | SIP::Timeout, deleting session |
| 23:14.14 | pc500 | SIP::Deleting session for 172.16.0.81 to 66.225.32.67, 0 total |
| 23:14.14 | pc500 | <PROTECTED> |
| 23:14.14 | pc500 | <PROTECTED> |
| 23:14.14 | pc500 | <PROTECTED> |
| 23:14.40 | pc500 | oops, that's sip debug in the cisco cli :) |
| 23:14.50 | ManxPower | PASTEBIN.CA or die! |
| 23:15.18 | ManxPower | ~pb |
| 23:15.19 | jbot | A Pastebin is a place to paste your stuff 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 |
| 23:16.37 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
| 23:18.51 | *** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au) |
| 23:19.11 | Penggu | hi all... i've got a fundamental dialplan qustion: |
| 23:19.26 | Penggu | say there's an extensions '6600' that waits for user input for options, 1,2, 3, 4 |
| 23:19.42 | Penggu | each of which are extensions, located *after* 6600 |
| 23:19.48 | Penggu | (waitexten?) |
| 23:19.52 | Penggu | then there's 6601 |
| 23:19.55 | Penggu | which also has a menu |
| 23:19.58 | Penggu | 1,2,3,4 |
| 23:20.06 | Penggu | and then you have the extensions, 1,2,3,4 after 6601 |
| 23:20.08 | Penggu | now... |
| 23:20.13 | Penggu | will these conflict? |
| 23:20.19 | *** part/#asterisk jkimball4 (n=jerrid@wsip-70-165-105-40.om.om.cox.net) |
| 23:20.19 | ManxPower | if you want duplicate options you need them in different contexts |
| 23:20.28 | Penggu | would i need to rather do 6600+1, 6600+2 for extensions? |
| 23:20.35 | ManxPower | the ORDER does not matter within a context. |
| 23:20.40 | Penggu | hmm i c |
| 23:20.56 | Penggu | so it doesnt kind of 'fall through' |
| 23:21.27 | Penggu | so if i have 2 extensions, with the same name, one gets overwritten/replaced ? |
| 23:21.33 | Penggu | (under the same context) |
| 23:22.09 | ManxPower | it will issue an error message. |
| 23:22.33 | ManxPower | why don't you look for IVR examples on the Wiki or The Book |
| 23:22.38 | ManxPower | or the mailing list. |
| 23:22.41 | ManxPower | ~mailinglist |
| 23:22.42 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 23:22.43 | ManxPower | ~wiki |
| 23:22.48 | ManxPower | ~book |
| 23:22.49 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:24.01 | Penggu | thanks. |
| 23:24.05 | ManxPower | ~wiki ivr |
| 23:24.30 | ManxPower | well that was useless |
| 23:25.02 | Penggu | i guess i have to re-read up on extensions |
| 23:29.08 | pc500 | ManxPower - Do many off the shelf routers (linksys/dlink garbage) handle nat to the point where nat=yes in sip.conf is no longer required? |
| 23:29.24 | pc500 | ManxPower - From a service provider perspective, how do they set their equipment so it works anywhere? |
| 23:29.44 | pc500 | ManxPower - Or is that "problem" a special cisco traffic inspect problem. |
| 23:37.49 | Penggu | hmm, * sorts extensions |
| 23:38.02 | Penggu | so i guess that gives me an implied answer |
| 23:38.16 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584025.dsl.bell.ca) |
| 23:41.43 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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