| 00:08.12 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 00:10.35 | RonDutt | Anyone know of any Skype to SIP bridge services? Don't want the hardware. I can't remember the name but there was a company a while back that provided a service where they ran multiple instances of Skype and provided SIP information for users who signed up. |
| 00:12.08 | *** join/#asterisk Shizuo (i=shizuo@200-171-49-211.dsl.telesp.net.br) |
| 00:12.16 | Shizuo | Hey there, Digium slaves |
| 00:12.41 | snowboarder04 | cheers [TK]D-Fender, that looks like it's licked it |
| 00:12.56 | [TK]D-Fender | snowboarder04: You're welcome. |
| 00:13.11 | [TK]D-Fender | Shizuo: Come to spread more FUD? |
| 00:13.23 | Shizuo | Hi |
| 00:13.26 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 00:15.18 | snowboarder04 | Shizuo: check out http://www.chanskype.com/ or http://www.astricon.net/2008/glendale/web/skype.php |
| 00:15.26 | Shizuo | snowboarder04: Why? |
| 00:15.41 | [TK]D-Fender | snowboarder04: that'd be RonDutt you'd be referring to that |
| 00:15.49 | [TK]D-Fender | snowboarder04: aim failure :) |
| 00:16.04 | snowboarder04 | heh, np |
| 00:16.14 | snowboarder04 | Shizuo: because you asked |
| 00:16.26 | [TK]D-Fender | snowboarder04: no, HE didn't. |
| 00:16.43 | Shizuo | I am confused |
| 00:16.50 | Shizuo | Crappy IRC client? |
| 00:16.50 | RonDutt | Ah, thanks. |
| 00:16.54 | snowboarder04 | errm, yeah, brain failure :/ |
| 00:17.17 | snowboarder04 | RonDutt: check out http://www.chanskype.com/ or http://www.astricon.net/2008/glendale/web/skype.php |
| 00:17.27 | snowboarder04 | sorted |
| 00:17.45 | snowboarder04 | way too late for me to be up evidently |
| 00:18.04 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:19.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:32.21 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 00:41.39 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:50.17 | AndyML | i'm having problems with dahdi - it 'starts', dahdi_cfg -vvv shows the appropriate stuff, but asterisk won't even run commands starting with 'dahdi' (dahdi show status, etc) i've tried module load chan_dahdi.so etc - no luck. - http://pastebin.com/m2b028afc - I've tried recompiling asterisk to no avail. |
| 00:51.28 | *** join/#asterisk ingenius (n=alektro@host253.190-31-41.telecom.net.ar) |
| 00:51.38 | *** join/#asterisk smash- (n=smash@c-98-232-250-106.hsd1.or.comcast.net) |
| 00:52.17 | smash- | Hey, does anyone here have a fonality pbxtra box im wondering if there currently having outtage with there webui. as my box is fine but im unable to connect to webui.. |
| 00:56.19 | *** part/#asterisk Shizuo (i=shizuo@200-171-49-211.dsl.telesp.net.br) |
| 01:03.43 | [TK]D-Fender | AndyML: From what you've shown you have no channels defined for * use |
| 01:05.59 | axisys | anyone tell me how do I make a call to my home number and have the home phone to ring? I have a TDM400P (2 fxs, 2 fxo). my land line is going to FXO and my phone is connected to FXS port.. |
| 01:06.10 | axisys | so far it is failing like this http://pastebin.com/ffffdcfa |
| 01:08.17 | [TK]D-Fender | From: "Cell Phone VA" <sip:5719999999@192.168.1.106>;tag=as02ac2474 |
| 01:08.19 | [TK]D-Fender | To: <sip:phone@192.168.1.106> |
| 01:08.25 | [TK]D-Fender | axisys: wtf IS @PHONE? |
| 01:08.38 | [TK]D-Fender | axisys: and * is sending the call to ITSELF |
| 01:09.00 | [TK]D-Fender | axisys: and what is your "home phone" in this scenarin? |
| 01:09.06 | axisys | [TK]D-Fender: phone is one of the context in sip http://pastebin.com/f55f2103 |
| 01:10.01 | [TK]D-Fender | axisys: host=192.168.1.106 <- this appears to be your * IP |
| 01:10.04 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:10.15 | axisys | [TK]D-Fender: home phone is not defined.. but matchin with `s' in extension http://pastebin.com/f4f4b247f |
| 01:10.28 | [TK]D-Fender | axisys: what do you mean "not defined"? |
| 01:10.28 | axisys | [TK]D-Fender: yes that is my asterisk server IP |
| 01:10.42 | [TK]D-Fender | axisys: Your description is becoming increasingly broken... |
| 01:10.55 | [TK]D-Fender | axisys: WTF is your PHONE entry doing pointing to ASTERISK's IP? |
| 01:11.26 | snowboarder04 | anyone know if vonage provide SIP access? |
| 01:11.42 | [TK]D-Fender | snowboarder04: They do if you sign up for a soft-phone account |
| 01:11.53 | snowboarder04 | ah ha |
| 01:11.57 | snowboarder04 | cheers |
| 01:12.04 | [TK]D-Fender | axisys: exten => s,1,Dial(SIP/phone@phone,10) <- bad format for this. SIP/phone <- proper |
| 01:12.12 | [TK]D-Fender | axisys: no "@phone" |
| 01:12.26 | [TK]D-Fender | axisys: And where IS that device located? |
| 01:13.17 | axisys | [TK]D-Fender: [phone] is pointing to context => phone1 |
| 01:14.16 | [TK]D-Fender | axisys: You misunderstand how to use devices. You send calls to "SIP/phone" to call the device referred to by [phone] in sip.conf. |
| 01:14.45 | axisys | http://pastebin.com/f14235de <-- current extension and sip file |
| 01:15.27 | [TK]D-Fender | axisys: WHAT is [phone]? |
| 01:17.48 | axisys | [TK]D-Fender: http://pastebin.com/f9ab8f4f w/ the change in extension u suggested.. |
| 01:18.14 | [TK]D-Fender | axisys: WHAT is [phone]? <-------------------------- |
| 01:18.19 | axisys | [TK]D-Fender: [phone] is what I defined one of the fxs port |
| 01:18.26 | [TK]D-Fender | No, it ISN"T |
| 01:18.36 | [TK]D-Fender | axisys: that is in SIP>CONF. How the hell is that an FXS PORT? |
| 01:19.21 | [TK]D-Fender | axisys: and what is the device that uses [phone]? |
| 01:20.08 | axisys | [TK]D-Fender: i guess i can change in extention to DIAL(from-internal) ? which is the fxs port where my phone is attached |
| 01:20.21 | [TK]D-Fender | !??!?!! |
| 01:20.53 | [TK]D-Fender | axisys: What exactly is your phone attached to? |
| 01:21.41 | axisys | i have tdm400p like pci card w/ 2 fxs and 2 fxo ports.. my phone is connected to one of the fxs port and my phone line is on one of the fxo port |
| 01:21.54 | [TK]D-Fender | axisys: then this has NOTHING to do with SIP <- |
| 01:22.09 | [TK]D-Fender | axisys: and you are not dialing that other port in your dial statement |
| 01:22.28 | axisys | ok |
| 01:22.53 | [TK]D-Fender | axisys: and I still can't see where you defined any channels for use with * |
| 01:23.16 | jaytee | hmmmmmm |
| 01:23.30 | [TK]D-Fender | jaytee: http://pastebin.com/m2b028afc <-- you show me where.... |
| 01:23.42 | [TK]D-Fender | jaytee: I see a clearly missing file and no reference to one that he DID share. |
| 01:23.45 | axisys | [TK]D-Fender: http://pastebin.com/f39178dde |
| 01:24.09 | [TK]D-Fender | axisys: Yes, and there appears to be a file you DIDN'T show us |
| 01:24.20 | jaytee | [TK]D-Fender, what? you think I'm disagreeing? he's obviously not done his homework. |
| 01:24.27 | AndyML | [TK]D-Fender: http://pastebin.com/m14889097 - is this what I was missing? or is there more? |
| 01:25.02 | axisys | [TK]D-Fender: http://pastebin.com/f10cadf30 and http://pastebin.com/f5ac5b2de |
| 01:25.40 | [TK]D-Fender | AndyML: Ah, that was YOURS |
| 01:25.42 | [TK]D-Fender | blarg |
| 01:26.16 | [TK]D-Fender | axisys: Ok, you just need to fix your dialplan. |
| 01:26.33 | AndyML | [TK]D-Fender: sorry to add to the confusion |
| 01:26.53 | [TK]D-Fender | AndyML: Yeah, yours did not define any zap channels due to a bad INCLUDE |
| 01:27.04 | [TK]D-Fender | anylook at the names fo what's referred to. |
| 01:27.21 | axisys | my goal is to able to call my nmbr from my cell and my phone ring .. but not sure how to tell it in dialplan |
| 01:27.49 | [TK]D-Fender | axisys: what kind of CHANNEL are you calling? |
| 01:27.56 | [TK]D-Fender | ***HINT*** |
| 01:29.09 | axisys | Zap |
| 01:29.32 | [TK]D-Fender | axisys: Zap/[port#] |
| 01:29.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:29.51 | axisys | so Dial(Zap/1) since my phone is attached to channel 1 ? |
| 01:29.59 | [TK]D-Fender | axisys: Looks right. |
| 01:30.08 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:30.57 | AndyML | ok, i've fixed the bad include. here are the results after an asterisk and dahdi restart - http://pastebin.com/m39ef4053 |
| 01:32.27 | [TK]D-Fender | AndyML: Where is your attempt to load the module manually? |
| 01:33.01 | axisys | [TK]D-Fender: that worked |
| 01:33.10 | axisys | [TK]D-Fender: thank you so much man! |
| 01:33.34 | jaytee | [TK]D-Fender, you're just racking up the good karma points tonight! |
| 01:33.59 | AndyML | [TK]D-Fender: http://pastebin.com/m7c813646 |
| 01:34.02 | [TK]D-Fender | axisys: You're welcome. |
| 01:34.42 | [TK]D-Fender | AndyML: See if they got alias'd to Zap. |
| 01:34.50 | AndyML | http://pastebin.com/m7d87cd78 |
| 01:35.02 | AndyML | where would i find that? |
| 01:35.29 | [TK]D-Fender | andFirst fix your signaling mismatch. |
| 01:35.43 | [TK]D-Fender | AndyML: and then look for ZAP commands at CLI |
| 01:36.05 | axisys | hmm voicemail is failing |
| 01:36.08 | axisys | Executing [s@from-pstn:2] VoiceMail("Zap/4-1", "line1") in new stack |
| 01:36.14 | AndyML | [TK]D-Fender: k |
| 01:36.17 | axisys | [Mar 15 21:35:20] WARNING[7075]: app_voicemail.c:3896 leave_voicemail: No entry in voicemail config file for 'line1' |
| 01:36.19 | [TK]D-Fender | AndyML: And FreePBX has turned your configs into repetitive spaghetti garbage. |
| 01:36.33 | AndyML | beautiful |
| 01:36.47 | AndyML | no zap commands at the CLI |
| 01:36.53 | axisys | i do have a entry for line1 in voicemail |
| 01:37.09 | [TK]D-Fender | AndyML: I would recompile * from scratch from freshly extracted tarballs again |
| 01:37.09 | axisys | line1 => 1234,Asif Iqbal,vadud3@gmail.com |
| 01:37.18 | AndyML | k - will do. |
| 01:37.19 | [TK]D-Fender | axisys: needs to be a NUMBER |
| 01:38.43 | [TK]D-Fender | jaytee: My karma ran over your dogma :p |
| 01:39.17 | jaytee | <PROTECTED> |
| 01:41.44 | [TK]D-Fender | fires up his Infinitie Improbability Drive |
| 01:42.00 | jaytee | don't forget your towel! |
| 01:42.17 | axisys | [TK]D-Fender: still failing http://pastebin.com/f64cd91ae |
| 01:42.37 | axisys | root@improvise:/etc/asterisk# cat voicemail.conf | tail -1 |
| 01:42.37 | axisys | 8720 => 1234,Asif Iqbal,vadud3@gmail.com |
| 01:43.14 | axisys | i did the module reload app_voicemail.so and dialplan reload |
| 01:44.20 | [TK]D-Fender | axisys: pastebin the ENTIRE config |
| 01:44.38 | axisys | [TK]D-Fender: http://pastebin.com/f67600d80 |
| 01:44.56 | axisys | [TK]D-Fender: voicemail.conf ^ |
| 01:45.39 | [TK]D-Fender | axisys: [other] <- the problem. * looks in [default] unless you tell it otherwise |
| 01:45.58 | [TK]D-Fender | axisys: You seriously need to read the instructions for the apps you are using and the BOOK in general. |
| 01:46.00 | [TK]D-Fender | ~book |
| 01:46.01 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:46.53 | *** part/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
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| 01:50.39 | *** mode/#asterisk [+o Mog] by ChanServ |
| 01:51.19 | [TK]D-Fender | 1.5 months til distro upgrade! |
| 01:51.46 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 01:52.34 | axisys | [TK]D-Fender: i have been reading it.. it is just not easy to get it .. |
| 01:54.07 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 01:55.06 | phix | hey |
| 01:55.30 | phix | how does one print out the cadence numbers of an incoming phone call? |
| 01:56.00 | phix | I read somewhere that enabling verbose mode was enough to print it out in the asterisl console however it isn't doing it |
| 01:56.43 | phix | the reason I want to do this is not just for setting up distinctive rings but for making phones ring the same as if they were plugged directly into landline, asterisk's default ring sounds different |
| 01:56.49 | axisys | [TK]D-Fender: i am looking at extensions.conf.sample.. i see lot of voicemail examples there.. but not one that shows to pick a different context like [other] |
| 01:57.18 | phix | hmmm |
| 01:57.37 | phix | I also need to do some LDAP lookups too :\ but that is for something different |
| 01:59.45 | [TK]D-Fender | axisys: "core show application voicemail" <- |
| 01:59.51 | jaytee | good god! I'd forgotten how totally insulting to one's intellect the movie Hackers was to people that actually use and understand computers. |
| 02:00.26 | carrar | does some circles around jaytee on his roller blades and camo painted laptop |
| 02:01.49 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 02:02.46 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 02:03.48 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 02:04.53 | carrar | jaytee, the only thing still true of that movie is Angelina Jolie is still fricken hot |
| 02:07.01 | [TK]D-Fender | carrar: Sure... if you're into 2 dimensionaly people like her and Kate Moss :) |
| 02:07.29 | [TK]D-Fender | whips up a faux-French brittish accent |
| 02:07.38 | [TK]D-Fender | "Not even if it is wafer thin?" |
| 02:07.44 | carrar | I wouldn't push her out of my bead for being a hacker and eating crackers |
| 02:07.45 | [TK]D-Fender | jaytee: ;) |
| 02:14.41 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 02:16.02 | jaytee | "RISC architecture is the future." ???? WTF? when did Intel put out Pentiums that were RISC? hahahaha ROFL |
| 02:16.53 | [TK]D-Fender | jaytee: And since when were Pentiums anything but the past regurgitated ;) |
| 02:22.23 | apeiron | egad, my beautiful Ogg Vorbis music sounds terrible when I encode it for music on hold. :( |
| 02:23.09 | apeiron | Would I, um, break the internets if I tried to play 44.1khz music over a phone line? :) |
| 02:26.18 | apeiron | considers streaming WBUR instead |
| 02:35.16 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:36.06 | apeiron | Hmm. I'm following the directions from http://voip-info.org/wiki-Asterisk+config+musiconhold.conf, copy-pasting, but all I hear is silence when I place a call on hold. What am I missing? |
| 02:36.14 | axisys | [TK]D-Fender: thanks a lot |
| 02:36.21 | apeiron | (yes, I changed the stream URL to a valid stream) |
| 02:36.44 | apeiron | This is the icecast example I'm trying to get to work. |
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| 03:00.12 | AndyML | [TK]D-Fender: i had fxols in system.conf and fxoks in chan_dahdi.conf - fyi... |
| 03:00.17 | AndyML | g'night! |
| 03:03.03 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:05.39 | apeiron | whee, it works. |
| 03:06.01 | apeiron | The sound really, really, really *has* to be 8000 hz coming to *, it seems. |
| 03:08.14 | [TK]D-Fender | apeapeYou'd almost think it was documented and the waved in front of your face furiously.... |
| 03:08.21 | [TK]D-Fender | apeiron: You'd almost think it was documented and the waved in front of your face furiously.... |
| 03:09.17 | jaytee | hmmm, maybe that's why none of my Slim Whitman mp3's won't play on MOH? |
| 03:09.19 | apeiron | nods |
| 03:09.30 | apeiron | jaytee, Yeah. You *have* to transcode them to 8000 Hz. |
| 03:09.39 | apeiron | jaytee, sox is really handy for that. |
| 03:09.44 | keith4_ | shakes his fist at debian |
| 03:09.52 | jaytee | omg! someone took something I said as if it wasn't really sarcasm! |
| 03:10.23 | jaytee | yeah, sox works better than Audacity in most situations |
| 03:10.35 | apeiron | Especially on-the-fly as with a radio stream... |
| 03:10.50 | apeiron | Let's spawn an X11 app each time someone goes on hold! |
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| 03:16.02 | keith4_ | assaults the debian kernel with a shovel |
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| 03:32.44 | apeiron | whee, I have NPR MOH. :D |
| 03:37.47 | snowboarder04 | NPR? |
| 03:40.22 | apeiron | npr.org |
| 03:42.20 | snowboarder04 | cool, did you follow a HOWTO? |
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| 03:43.55 | apeiron | http://voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 03:44.13 | apeiron | Combination of the pipe method and the ogg123/sox combination for recoding to raw, 8000 Hz. |
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| 04:11.36 | AJayMN | Anyone use Asterisk for Video? |
| 04:11.46 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 04:13.04 | apeiron | has, but not too extensively |
| 04:13.25 | AJayMN | im trying to find a decent video conferencing program.. |
| 04:13.35 | apeiron | qutecom/wengophone works well. |
| 04:13.46 | AJayMN | tried Bria, X-Pro, etc... from CounterPAth but the video quality is horrid |
| 04:13.50 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 04:14.09 | AJayMN | a customer of mine wants to stream a radio station online with video from each guest... |
| 04:14.27 | apeiron | heh, exactly what I was looking at doing. |
| 04:15.02 | apeiron | As I said, qutecom/wengophone is decent, and fairly cross-platform (and is open source, so it's as cross-platform as your C++ skills can make it). |
| 04:15.23 | AJayMN | ill have to take a look |
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| 04:59.37 | Gopaul | how to get the hangup event with manager events? |
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| 05:33.23 | jplank | I have a peer trying to connect to asterisk on port 5061 and the user in sip.conf has port=5061, but * asterisk is ignoring the registers on port 5061 and still thinks its talking to it on 5060. Any ideas? |
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| 05:36.52 | Bryon_tmi_soluti | Hello there. |
| 05:37.12 | Bryon_tmi_soluti | Is anyone here knowledgable about STUN and recent * builds? |
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| 07:17.02 | Gopaul | how to get the hangup event via manager event? |
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| 07:42.27 | fcois93 | hello |
| 07:44.26 | phix | hi mate |
| 07:44.30 | phix | how are you? |
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| 07:54.46 | tlyng | Anyone got any idea on how to create something similar as WaitForSilence() and WaitForNoise() but instead waiting for silence or noise, it should wait for a particular sound. I know it's perhaps complex, the sound is a classic beep so it should be possible to identify without to heavy algorithms :-) Unfortunately I'm kinda inexperienced in using asterisks C-api, so I was wondering if anyone could point me in a direction. I've been studying t |
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| 07:59.53 | jblack | tlyng: Good luck. |
| 08:00.09 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 08:10.27 | tlyng | jblack: hehe, promising :-) Well I'll hack around and see what I come up with. |
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| 09:03.33 | *** join/#asterisk telephony (n=brad@fl-207-30-21-215.sta.embarqhsd.net) |
| 09:03.38 | telephony | howdy |
| 09:04.06 | telephony | anyone around? |
| 09:11.40 | *** join/#asterisk unk^ (n=unk@1-1-5-12a.gfa.gbg.bostream.se) |
| 09:12.11 | unk^ | What are the hardware requirements for asterisk? |
| 09:15.22 | telephony | it depends on what you want to do with it |
| 09:15.56 | unk^ | 2 IP-phones , just a laboration. |
| 09:16.13 | unk^ | Basic. Just to get it work. |
| 09:16.31 | unk^ | 2 network cards or can it go threw a switch? |
| 09:16.45 | telephony | you could run it in a virtual machine if you have a fast enough desktop |
| 09:17.06 | unk^ | im running kubuntu 8.10 |
| 09:17.10 | unk^ | its installed |
| 09:17.12 | telephony | i have it running on a 2.4 ghz amd with tons of room to spare |
| 09:17.16 | telephony | 1 gb of ram |
| 09:17.29 | unk^ | ok, nice. How many calls can it handle at the same time? |
| 09:17.48 | telephony | more than my internet connection is capable of |
| 09:17.56 | unk^ | kk |
| 09:17.56 | telephony | ive never set out to test it |
| 09:18.20 | telephony | ive run a dual core 2.4 with well over 20 calls before |
| 09:18.33 | telephony | and asterisk cant really take advantage of the second core |
| 09:18.36 | telephony | from what i understand |
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| 09:20.20 | bobsaccamano | hi..how do i configure 911 dialing in asterisk 1.6 for SIP Channels? |
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| 09:29.58 | bobsaccamano | i need help with the dial plan |
| 09:30.01 | bobsaccamano | anyone here? |
| 09:31.09 | WeazelON | just ask mate, asking anyone here will probably get you nowhere. |
| 09:32.41 | WeazelON | telephony, try installing "htop" you would be able to see your core levels realtime, and every mem or cpu usage is displayed there tied to its running process. |
| 09:33.31 | telephony | ill check it out i was usting top prior |
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| 11:22.44 | bobsaccamano | how do i get a cisco phone to register with asterisk? I have configured the server address and extension in sip config of the Ciso 7940 Phone. But when i make a SIP Call, i get 404 not found |
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| 11:23.49 | *** join/#asterisk maverick (n=maverick@83.240.229.38) |
| 11:24.12 | maverick | How can I find you is asteriks is running in real time mode ? |
| 11:24.17 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 11:24.49 | maverick | there is some issues in asterisk real time mode and changes in system time ? |
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| 11:29.48 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
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| 11:31.14 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 11:33.51 | neurosys | [TK]D-Fender: Do you use teh Manager in 1.6? |
| 11:34.17 | maverick | how can I find if asterisk is running in real time mode ? |
| 11:34.36 | [TK]D-Fender | neurosys: Not yet. |
| 11:34.46 | maverick | there is any issue regarding real time and changes in system clock time ? |
| 11:35.44 | neurosys | [TK]D-Fender: It seems the Originate command has changed or is gone. Or there is a new access level for manager.conf. I have it set to persmission all, but cants find the originate command. |
| 11:35.47 | WeazelON | maverick, I dont understand your question mate, |
| 11:36.12 | [TK]D-Fender | neurosys: Feel free to actually show something.... |
| 11:36.35 | WeazelON | maverick, are you searching for a certain bug or you have a problem which you are looking an answer for ? |
| 11:36.38 | maverick | weazelON: is there any way to find if asterisk is ruuning in real time mode ? |
| 11:36.55 | neurosys | [TK]D-Fender: Sorry TK. The Message response is : Permission Denied |
| 11:37.07 | WeazelON | when you say real time, you mean when you don't enter the CLI and check for yourself ? |
| 11:37.33 | *** join/#asterisk mattwj2002 (n=matt@c-71-63-163-89.hsd1.mn.comcast.net) |
| 11:37.37 | mattwj2002 | hi guys |
| 11:37.55 | mattwj2002 | I know this is off topic.... |
| 11:38.11 | *** join/#asterisk trymi1 (n=please@203.81.71.107) |
| 11:38.15 | maverick | weazelON: ? |
| 11:38.25 | mattwj2002 | but Boost Mobile is doing unlimited talk, text, web, etc for $50 per month |
| 11:38.26 | mattwj2002 | :) |
| 11:38.40 | WeazelON | maverick, are you asking "how can you know asterisk is running the background ? " |
| 11:38.59 | mattwj2002 | ps aux | grep asterisk |
| 11:39.22 | maverick | weazelON: no .... I asking if is running in real time |
| 11:39.58 | mattwj2002 | ooh sorry |
| 11:40.00 | mattwj2002 | :( |
| 11:41.23 | WeazelON | if you are running asterisk 1.2 and above, its probably realtime. |
| 11:41.39 | neurosys | [TK]D-Fender: i think i found it |
| 11:42.21 | neurosys | [TK]D-Fender: There are some added auth classes added into manager.conf |
| 11:43.21 | neurosys | [TK]D-Fender: The sample file for 1.6 articulates them. That's what i get for assuming I had read it before. in 1.4 :P |
| 11:43.34 | maverick | weazelON: but .... is there anyway to find out ? |
| 11:44.01 | maverick | weazelON: and ... there is any known issue regarding asterisk in real time and a change in system time ? |
| 11:44.18 | WeazelON | maverick, do you have a clock system time issue ? |
| 11:44.38 | *** part/#asterisk mattwj2002 (n=matt@c-71-63-163-89.hsd1.mn.comcast.net) |
| 11:45.37 | maverick | weazelON: yes ... my clock resync in ntp with an early hour and asterisk died |
| 11:46.41 | WeazelON | maverick, and how do you know those 2 are connected ? |
| 11:47.35 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 11:47.38 | maverick | weazelON: don't know ....... that is what I asking you |
| 11:47.52 | maverick | Is there any known issue |
| 11:48.07 | maverick | with asterisk and a change in system clock |
| 11:48.13 | maverick | I'm asking .... |
| 11:48.17 | maverick | do you know ? |
| 11:48.18 | WeazelON | maverick, then my answer would be " not likely " but anything is possible... |
| 11:48.24 | *** join/#asterisk criten (n=criten@216.75.233.220.exetel.com.au) |
| 11:48.40 | maverick | ok |
| 11:48.52 | maverick | weazelON: thanks for your help |
| 11:49.00 | WeazelON | is the ntp server you are synchronizing is a local one ? or via wan ? |
| 11:50.27 | maverick | wan |
| 11:50.45 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 11:50.54 | [TK]D-Fender | maverick: the words "wild conclusions" and "paranoia" come to mind |
| 11:51.02 | WeazelON | how did you do the settings for the ntp ? |
| 11:52.47 | *** join/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 11:52.48 | maverick | [TK]D-Fender : why ? |
| 11:52.54 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 11:53.22 | maverick | weazelON: just put a ntp address server on ntp configs |
| 11:54.43 | [TK]D-Fender | [07:43]<maverick>weazelON: and ... there is any known issue regarding asterisk in real time and a change in system time ? <- because you're throwing random guesses out in the air on partial word-matches. |
| 11:55.19 | *** join/#asterisk caliel (n=caliel@unaffiliated/caliel) |
| 11:55.33 | WeazelON | i'm only trying to fix his clock :D |
| 11:55.59 | caliel | hello. i'm totally new to asterisk. my question is : is it possible to send FaXes via VoIP / ADSL line ? |
| 11:56.02 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 11:56.20 | jstew | caliel: Not reliably. |
| 11:57.05 | jstew | You can if your provider supports T.38 |
| 11:57.26 | caliel | jstew: oh well, i have a VoIP number provided by my ISP, i would like to send / receive faxes without using paper and toner ('cause i don't have a fax machine) |
| 11:57.30 | WeazelON | or, if you use an iaxmodem soft u can make the faxes go through the print option |
| 11:57.52 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 11:58.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:58.23 | jstew | An analog line is the only way to reliably receive faxes from other people in my experience. |
| 11:58.48 | caliel | do i need a 52k modem, doesn't I jstew ? |
| 11:58.50 | jstew | IAXmodem works on a lan just fine though |
| 11:59.36 | caliel | mmm ok |
| 11:59.37 | jstew | Hmm.... Well you can get something like one of the TDM cards or you can set up hylafax and get a modem that's compatible with that software |
| 12:00.44 | caliel | that's cause I read about FoIP or something like this. next question is a step before that one : what can I read to configure asterisk on my Gentoo box ? I have this VoIP number, and I would like to call and receive calls, but i'm too noob for now in asterisk technology. can you help me ? |
| 12:01.05 | maverick | [TK]D-Fender : have you ever tried to do the test yourself ? |
| 12:01.31 | bobsaccamano | guys does a cisco phone require any special extensions or sip channels defined for asterisk? |
| 12:01.54 | bobsaccamano | im getting a 404 not found for a 7940 number |
| 12:04.44 | *** join/#asterisk tokozedg (n=slA@85.118.98.122) |
| 12:05.15 | maverick | [TK]D-Fender : have you ever tried to do the test yourself ? |
| 12:06.05 | tokozedg | hello. when i register for example X-lite , it has 2 lines, if one is busy asterisk calls in second, and i want to make so that every client has only one line, anyone can help me? |
| 12:07.48 | WeazelON | cancel "CallWaiting" via asterisk's extension or if you are using FreePBX you can do it from there |
| 12:08.08 | WeazelON | <--- tokozedg |
| 12:08.15 | [TK]D-Fender | maverick: "the test"? What test? |
| 12:08.34 | [TK]D-Fender | bobsaccamano: Please PB the complete failed call with SIP debug. |
| 12:08.35 | [TK]D-Fender | ~pb |
| 12:08.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 12:08.49 | *** join/#asterisk keith4 (n=keith@lust.cc.lehigh.edu) |
| 12:09.51 | tokozedg | WeazelON, no i`m not using freepbx, and how can i cancel call waiting? |
| 12:10.23 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 12:10.47 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:12.02 | WeazelON | tokozedg, in the asterisk CLI --- > database del CW <ext number> |
| 12:12.32 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:12.50 | tokozedg | linux-z93h*CLI> database del CW 151 |
| 12:12.50 | tokozedg | Database entry does not exist. |
| 12:13.17 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 12:13.18 | tokozedg | asterisk version is 1.4 |
| 12:13.36 | c0rnoTa | hello guys |
| 12:14.07 | c0rnoTa | i want to define russian tone data for answer detection on FXO |
| 12:14.26 | c0rnoTa | i know, i must write it in main/dsp.c |
| 12:14.40 | [TK]D-Fender | tokozedg: pastebin your complete attempt along with the database dump so we can see the key-pair you're aiming for. |
| 12:14.54 | maverick | [TK]D-Fender : run asterisk ... put the system time a hour behind and check what happens to asterisk ? |
| 12:14.58 | [TK]D-Fender | c0rnoTa: No, indications.conf |
| 12:15.37 | [TK]D-Fender | tokozedg: Hold on.. cancel call waiting? On what? |
| 12:15.59 | c0rnoTa | [TK]D-Fender: i read on asterisk knowlage base, that parameter progzone make answer detection, isn't it? |
| 12:16.03 | [TK]D-Fender | tokozedg: Ah, Scrolled up and saw. |
| 12:16.18 | tokozedg | i dont know, i just want to have only one line for each client |
| 12:16.51 | [TK]D-Fender | c0rnoTa: the zone tells * what entry to use for progress tones, etc. the setting in indications.conf should match the local signaling for the specified zone |
| 12:17.23 | c0rnoTa | [TK]D-Fender: i'll try |
| 12:17.24 | [TK]D-Fender | tokozedg: "call-limit=1", in sip.conf should do it. |
| 12:17.52 | tokozedg | [TK]D-Fender, ok, i`ll try |
| 12:19.35 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 12:20.49 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 12:22.55 | tokozedg | [TK]D-Fender, failed |
| 12:23.21 | tokozedg | client has incoming and outgoing lines at the same time |
| 12:23.22 | [TK]D-Fender | tokozedg: add "type=peer", "limitonpeer=yes" |
| 12:24.06 | [TK]D-Fender | tokozedg: If this fails, check the WIKI for the other in/out limit options for sip.conf, and if that fails you'll have to check in the dialplan with ChanIsAvail |
| 12:24.30 | tokozedg | [TK]D-Fender, ok thanks |
| 12:30.46 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:33.47 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 12:36.38 | *** part/#asterisk lbt (n=david@78.32.229.233) |
| 12:36.53 | c0rnoTa | [TK]D-Fender, could you help again? |
| 12:37.25 | [TK]D-Fender | c0I've answered about everything I can for this particular topic.... |
| 12:37.32 | [TK]D-Fender | c0rnoTa: I've answered about everything I can for this particular topic.... |
| 12:42.43 | WeazelON | tokozedg, if you type in asterisk cli -- > database show CW < is the extension you have problem with enlisted ? |
| 12:42.44 | c0rnoTa | [TK]D-Fender, i have set in indications.conf my tone parameters (frequency and delays), but when i dial to Busy (dial throu line4, receive to line9, line9 have exten => s,1,Busy), my dial line (line4) still have answer, but answered line (line9) generating busy tone. After that answered line gives hangup. But dialed line still up and i receive second ring on line9 |
| 12:42.58 | c0rnoTa | [TK]D-Fender, what is it? |
| 12:43.27 | c0rnoTa | [TK]D-Fender, why it appers? |
| 12:43.34 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 12:43.35 | c0rnoTa | sorry 4 my english plz |
| 12:44.12 | tokozedg | WeazelON, when i type database show CW , there is nothing |
| 12:45.27 | bobsaccamano | [TK]D-Fender, here is the pastebin.. http://pastebin.com/m58094529 |
| 12:46.21 | bobsaccamano | im getting a 404 for the Cisco phone no |
| 12:46.43 | WeazelON | that is very strange |
| 12:46.54 | WeazelON | what is the asterisk version ? |
| 12:47.02 | bobsaccamano | 1.6.0.6 |
| 12:47.06 | tokozedg | Asterisk 1.4.23 |
| 12:47.20 | bobsaccamano | [TK]D-Fender, you still there? |
| 12:47.51 | bobsaccamano | also sip show peers gives an unspecified host for the Cisco phone |
| 12:49.30 | [TK]D-Fender | bobsaccamano: Issue isn't the Cisco. |
| 12:49.32 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:49.45 | [TK]D-Fender | bobsaccamano: Looking for 8888 in motorola (domain 26.1.0.2) SIP/2.0 404 Not Found |
| 12:49.54 | [TK]D-Fender | bobsaccamano: Dialplan error, go fix it |
| 12:51.00 | tokozedg | how can i register DEVICE_STATE function in asterisk 1.4? |
| 12:51.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:53.40 | kaldemar | tokozedg: the function is not a part of 1.4. you'll have to backport it from 1.6 if you want it. |
| 12:55.41 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 12:59.41 | jjshoe | op up? |
| 13:00.58 | jjshoe | nice little onjoiner |
| 13:00.58 | jjshoe | [07:57] <c0ldk1ll3r> hi, you should take a look at freeswitch, it's a lot better than asterisk... join #freeswitch or visit us at http://www.freeswitch.org/ |
| 13:03.02 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:03.02 | *** mode/#asterisk [+b *!n=c0ldk1ll@*.com] by russellb |
| 13:03.02 | *** kick/#asterisk [c0ldk1ll3r!n=russellb@asterisk/digium-open-source-team-lead/russellb] by russellb (russellb) |
| 13:03.32 | snowboarder04 | can anyone recommend a decent UK SIP/VoIP provider? |
| 13:03.36 | russellb | tokozedg: there is a backport available |
| 13:03.41 | russellb | ~devstate |
| 13:03.42 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/, or http://www.asterisk.org/node/48325 |
| 13:03.59 | russellb | kaldemar: FYI ^^ |
| 13:04.13 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:04.14 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:04.18 | lmadsen | jjshoe: eh? |
| 13:04.57 | jjshoe | russellb <3 |
| 13:05.52 | russellb | lmadsen: already took care of it, but thanks |
| 13:06.37 | lmadsen | np! |
| 13:08.44 | dpryo | Is there any "STABLE" branch of Asterisk? |
| 13:08.59 | lmadsen | we don't tag things as "stable" |
| 13:09.07 | lmadsen | use the latest release |
| 13:09.21 | kaldemar | russellb: affirmative. :) |
| 13:09.23 | lmadsen | you have a choice between latest 1.4, and latest 1.6.0 at this time |
| 13:11.04 | tokozedg | russellb, DEVSTATE not registered, fuckk |
| 13:11.12 | russellb | eh? |
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| 13:52.27 | Chainsaw | lmadsen: I have two easy patches in the bug tracker that haven't been applied yet. |
| 13:52.47 | Chainsaw | lmadsen: Anything special I need to do please? The clearance from legal has been received, so the patches should be visible now. |
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| 13:56.41 | bobsaccamano | [TK]D-Fender, does the dial plan for sip nos change with the user entity? |
| 13:56.59 | [TK]D-Fender | bobsaccamano: huh? |
| 13:57.45 | [TK]D-Fender | bobsaccamano: the error tells you explicitly what extension it is looking for and that it does not exist. This is a complete non-issue. Go look in your dialplan why you don't have a match for 888 |
| 13:57.47 | [TK]D-Fender | 8888* |
| 13:58.13 | bobsaccamano | [TK]D-Fender, hmm..okay..thanks |
| 13:58.47 | *** part/#asterisk L84Supper (n=ly@c-24-12-193-200.hsd1.il.comcast.net) |
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| 14:06.31 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:06.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:08.38 | ruben23 | hi |
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| 14:10.38 | Chainsaw | putnopvut: Who would I need to speak to about getting a dahdi-tools build fix applied please? |
| 14:11.49 | putnopvut | Chainsaw: The best way would be to open a bug report, then someone will take care of it. |
| 14:11.56 | Chainsaw | putnopvut: I did that: http://bugs.digium.com/view.php?id=14638 |
| 14:12.00 | GeminiDomino | Strange issue, hopefully someone can help me out. When I feed a script to the manager via netcat, I get a "Connect attempt from x.x.x.x: unable to authenticate" It does not parse the manager.conf file like it would if the login/pass was bad. However, when I telnet from the exact same host, and cut/paste the exact same commands, I get successful authentication. Any ideas? |
| 14:12.20 | putnopvut | Chainsaw: Ah, I see...and six days ago, too. |
| 14:12.39 | putnopvut | I'll bug someone about taking a look at it. |
| 14:13.02 | Chainsaw | putnopvut: Thank you. There's a new one just like it (similar problem). 14671 |
| 14:14.19 | putnopvut | Chainsaw: thanks for the bug reports. I'm surprised no one has at least commented on 14638 |
| 14:15.00 | Chainsaw | putnopvut: I got a "so don't do that" reply out of Qwell, but it was on IRC, not the bug. |
| 14:15.14 | Chainsaw | (With regard to using LDFLAGS="-Wl,--as-needed") |
| 14:15.21 | *** join/#asterisk stoffell (n=kristof@fw.catsanddogs.com) |
| 14:15.53 | putnopvut | Chainsaw: oh I see, it's because you passed explicit LDFLAGS instead of appending them to the ones already there. |
| 14:16.00 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 14:16.01 | putnopvut | gotcha |
| 14:16.28 | Chainsaw | putnopvut: Not really, no. It's about the ordering of object files versus libraries in the Makefile. |
| 14:16.45 | telephony | i had a question about jabbersend is anyone familiar with it |
| 14:17.15 | putnopvut | Chainsaw: Right, I understand that the behavior is incorrect, I was just wondering why the LDFLAGS were being overwritten. I understand now. |
| 14:17.46 | Chainsaw | putnopvut: Well, they don't get overwritten as such. |
| 14:17.57 | Chainsaw | putnopvut: The library gets discarded because it isn't "used" by the object file. |
| 14:18.06 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 14:18.16 | putnopvut | Chainsaw: ah, it is now becoming more clear :) |
| 14:18.22 | Chainsaw | putnopvut: Ordering gets significant with that flag on. |
| 14:18.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:18.29 | telephony | as far as adding a line somewhere that sends a jabber message when the extension is answered i cant seem to find where the code is that passes the call from hold or the ring group to the individual caller |
| 14:19.14 | telephony | anyone have any insight on it |
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| 14:27.38 | mw-home | Having argument with coworkers. I have 1 GHz asterisk box I'm using to handle calls for < 5 people. I use top to watch the CPU state. If the CPU is 90% idle, is that a sign that I don't need a faster box? |
| 14:28.49 | florz | mw-home: what is a "faster box"? |
| 14:29.19 | tzafrir_laptop | mw-home, no. This is not a sign that you need a faster box |
| 14:29.25 | mvanbaak | telephony: I have this here at home: JabberSend(asterisk,my_jabber@account/BitlBee,Incoming call from [${CALLERID(num)} - ${CALLERID(name)}]); |
| 14:29.44 | poller | If the CPU is 90% idle, that means you're using 10% CPU. |
| 14:29.53 | poller | ie, no need to upgrade |
| 14:30.19 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 14:30.40 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
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| 14:31.03 | ayeso | Can I send faxes over the g729 codec? |
| 14:31.32 | telephony | probably not |
| 14:31.37 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) |
| 14:32.34 | [TK]D-Fender | ayeso: No |
| 14:32.43 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 14:32.59 | ayeso | [TK]D-Fender: Thats the answer i was looking for, thanks... and that is a no for sure correct? |
| 14:33.25 | JT | definitely not |
| 14:33.27 | [TK]D-Fender | ayeso: In the back of the head, mafioso-style. |
| 14:33.37 | JT | g.729 is a voice codec |
| 14:33.48 | ayeso | [TK]D-Fender: Excellent |
| 14:34.21 | JT | designed for the express purpose of transporting compressed typical human voice signals at telephone quality in ways that can be understood by other humans |
| 14:35.54 | lmadsen | fax needs an uncompressed codec, with no packet-loss or jitter |
| 14:36.27 | apeiron | hm. Let's say I want to use T.38 for sending faxes. Does my provider need to have an H.323 gateway setup, or can I do that through SIP? |
| 14:36.28 | coppice | fax needs G.711, which is a compressed codec, but the one modems were designed to live with |
| 14:36.45 | *** join/#asterisk soulclaimer (n=dpearson@mail.databalance.com) |
| 14:36.59 | ayeso | coppice: My understanding was that g711u is uncompressed... is that not the case? |
| 14:37.13 | coppice | G.711 is a lossy compression scheme |
| 14:37.50 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 14:39.25 | soulclaimer | Hello everyone, I have a .lock file issue with my voicemail was wondering if someone had seen this before. Everything I find online is either a cron job every minute or update asterisk |
| 14:40.52 | soulclaimer | Happens to random voicemails, .lock file prevents new voicemail from being received and keeps users from accessing voicemail. |
| 14:41.38 | Mr_BOnD_007 | can i use asterisk for inbound calls also ? |
| 14:42.08 | apeiron | Mr_BOnD_007, er, yes? |
| 14:42.20 | *** join/#asterisk hi365 (n=hi365@85.130.230.240) |
| 14:42.58 | Mr_BOnD_007 | okie :D |
| 14:43.32 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:43.34 | Mr_BOnD_007 | i want to setup VICIDIAL GUI i had allready installed the Asterisk but now what to do i dont know the manual configuration using SIP VOip Account |
| 14:43.41 | *** part/#asterisk telephony (n=brad@fl-207-30-21-215.sta.embarqhsd.net) |
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| 14:47.39 | bougyman | Mr_BOnD_007: which asterisk did you use, and which vici version? |
| 14:48.01 | bougyman | you have astguiclient working? |
| 14:48.36 | mw-home | poller: so, can I use the CPU idle state to tell me when I need a better box? In other words, when the CPU load is really high, then I should spend some $$$. |
| 14:49.10 | mw-home | Is a 1 GHz CPU acceptable for a small office (<5 simultaneous calls)? |
| 14:49.18 | [TK]D-Fender | mw-home: Sure |
| 14:49.21 | bougyman | maybe |
| 14:49.31 | poller | It's more then acceptable |
| 14:49.35 | bougyman | if you aren't recording the calls. |
| 14:49.44 | bougyman | or doing any other funky stuffs. |
| 14:53.16 | jjshoe | mw-home plenty of power |
| 14:53.35 | jjshoe | just make sure you're not transcoding where you don't need to be. |
| 14:55.40 | mw-home | When I do an asterisk call, I hear my own voice echoed back. How do I block that out? |
| 14:55.44 | Katty | guys, i have some sad news. |
| 14:56.02 | jjshoe | mw-home depends on lots of things. pri? analog? voip? |
| 14:56.04 | Katty | apparently. i have been labeled as a terrorist. |
| 14:56.05 | Katty | http://www.kansascity.com/news/breaking_news/story/1086524.html |
| 14:56.17 | Katty | well, not directly labeled. |
| 14:58.25 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 14:58.29 | mw-home | jjshoe: I have a T1 plugged into a switch and the asterisk box on the other side of the switch. Using voip through a company called flowroute. |
| 14:58.56 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 15:00.34 | jjshoe | mw-home and on which one is the echo? |
| 15:00.46 | apeiron | "Troopers have been shot by members of groups," |
| 15:00.48 | apeiron | ^ such fail |
| 15:00.49 | jjshoe | oh sorry, data t1 mw-home? |
| 15:01.06 | jjshoe | ~echo |
| 15:01.06 | jbot | hmm... echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of course mentioned in this bad summary: ... |
| 15:01.29 | mw-home | jjshoe: yeah, it is a data T1. |
| 15:02.19 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:04.08 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 15:13.04 | lmadsen | woh... that link is severely out of date from astdocs |
| 15:13.24 | lmadsen | I wonder what page it was referencing... |
| 15:16.41 | lmadsen | ~echo |
| 15:16.42 | jbot | i heard echo is an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 15:16.49 | lmadsen | better |
| 15:18.52 | soulclaimer | anyone know how to fix .lock files poping up in voicemail directories> |
| 15:24.25 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
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| 15:25.21 | *** part/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 15:26.19 | freeportway | ok, I'm a little confused. If I have a phone number that is on a provider, how do I get that phone number routed to my asterisk box? |
| 15:26.39 | freeportway | or better, how can i get a phone number routed... this part really confuses me |
| 15:28.49 | styelz | register your asterisk server with the provider |
| 15:31.30 | freeportway | well, that would be interesing. Say I have Packet8... I doubt they'll allow for that. How would i get some phone numbers and do that myself? |
| 15:31.55 | freeportway | maybe thats the answer? I find a provider, port those numbers to them? |
| 15:32.00 | jjshoe | freeportway ? |
| 15:32.09 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:32.21 | freeportway | yes? |
| 15:32.24 | freeportway | jjshoe? |
| 15:32.24 | jjshoe | freeportway Packet8 is an itsp no? why wouldn't they point the numbers for your account at where the account is registerd? |
| 15:32.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:32.51 | freeportway | packet8 is a provider like Vonage, for example |
| 15:33.00 | freeportway | im not sure their business is pointing numbers... |
| 15:33.04 | jjshoe | freeportway oh, you're looing to violate tos? |
| 15:33.06 | jjshoe | looking* |
| 15:33.16 | freeportway | Heh, well. No |
| 15:33.26 | jjshoe | best of luck! |
| 15:33.38 | freeportway | You misunderstood |
| 15:33.49 | freeportway | I want to replace packet8 |
| 15:33.56 | freeportway | they have our phone numbers |
| 15:34.04 | *** join/#asterisk strathisla8282 (n=merlin82@AStrasbourg-753-1-12-127.w90-56.abo.wanadoo.fr) |
| 15:34.11 | strathisla8282 | Hi people |
| 15:34.18 | freeportway | Im confused as to how a phone number is pointed to an astrisks box |
| 15:34.24 | areay | is there an IAX2 client for symbian? for some reason, when I'm outside of my LAN I get one-way audio with SIP, but IAX works fine... I've tried the port forwarding for SIP but it still doesn't work, so I'm pretty much limited to IAX2... |
| 15:34.59 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:35.01 | jjshoe | freeportway it never is. |
| 15:35.08 | jjshoe | freeportway a phone number is pointed at service. |
| 15:35.14 | freeportway | maybe i just need a "provider" that will host the phone number, you called them a itsp provider? |
| 15:35.23 | jjshoe | ~istp |
| 15:35.29 | freeportway | istp |
| 15:35.31 | jjshoe | ~itsp |
| 15:35.32 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 15:35.33 | jjshoe | yes, typoe. |
| 15:35.42 | strathisla8282 | Anybody knows how to set up call deflection ? |
| 15:36.01 | freeportway | excellent |
| 15:36.02 | freeportway | i'll look |
| 15:36.07 | jjshoe | strathisla8282 what's call deflection? :P |
| 15:36.15 | freeportway | ~itsplist |
| 15:36.56 | strathisla8282 | jjshoe: when for example you have an incoming call on a gsm card and you want to reroute it without taking it. |
| 15:37.17 | strathisla8282 | it's like transfer, but without answering |
| 15:39.10 | apeiron | Usually that sort of thing results in voicemail. |
| 15:39.38 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:40.16 | jjshoe | strathisla8282 are you asking how to use asterisk dialplans? I'm confused on what you need help with. |
| 15:41.06 | strathisla8282 | take a look at this, maybe it helps you : http://www.junghanns.net/en/calldeflection.html |
| 15:41.17 | strathisla8282 | sorry for my poor english... |
| 15:42.12 | apeiron | So your question is "can you translate this page into a dialplan for me" |
| 15:42.29 | strathisla8282 | lol, no. |
| 15:42.43 | strathisla8282 | I already have a complete dialplan that works ;) |
| 15:42.46 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:42.46 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:42.52 | strathisla8282 | but i'm trying to implement this feature |
| 15:43.16 | strathisla8282 | the capiCD command doesn't seem to exist on my server |
| 15:43.27 | strathisla8282 | wheter capiCD nor ZapCD |
| 15:44.15 | strathisla8282 | For the moment, I simply do a Dial() to the destination, but that implies placing a new call on a second line. |
| 15:44.35 | strathisla8282 | With call deflection, the incoming call is re-routed and there's no need for a second line. |
| 15:45.17 | apeiron[imploded | How is this different than "if dialed extension doesn't pick up in x seconds, fall-through to next item in dialplan"? |
| 15:45.28 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 15:46.11 | strathisla8282 | the difference is that I dial an external number (a GSM in fact), that implies costs. |
| 15:46.21 | strathisla8282 | If it's rerouted, no new call needed. |
| 15:46.52 | strathisla8282 | Now, I don't even know if it's supported on a GSM line. |
| 15:47.03 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:47.31 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:48.17 | fcois93 | in asterisk 1.6 Set(${CDR(accountcode)}=${SIP_HEADER(x-accountcode)}) dont work !!! the debug display Set("SIP/5067-b6d23ae8", "=Acro") in new stack |
| 15:48.18 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 15:48.18 | strathisla8282 | Said differently, the incoming call doesn't need to be answered, and thus doesn't imply to place a new call. |
| 15:48.19 | fcois93 | why? |
| 15:51.53 | strathisla8282 | apeiron: do you understand the difference ? |
| 15:52.08 | apeiron | strathisla8282, Yes, but not sure how to do it, sorry. |
| 15:52.17 | strathisla8282 | ok, no matter. |
| 15:52.41 | [TK]D-Fender | fcois93: What is in the accountcode? |
| 15:52.50 | [TK]D-Fender | fcois93: So far looks like nothing at all |
| 15:54.31 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:54.31 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:54.59 | Gido-E | documenteur-extraordinaire :-) |
| 15:55.02 | strathisla8282 | :) |
| 15:55.51 | fcois93 | [TK]D-Fender: I want to use accountcode to insert the user account (like Acro here) |
| 15:56.47 | fcois93 | [TK]D-Fender: asterisk can read =Acro but dont read that I want to write on accoundcode var! the debug display "=Acro"! |
| 15:57.28 | *** part/#asterisk freeportway (n=freeport@74.92.4.54) |
| 15:57.44 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-156-35.lns10.mel4.internode.on.net) |
| 15:57.57 | [TK]D-Fender | fcois93: Stop putting ${} around the function you want to WRITE TO |
| 15:58.55 | [TK]D-Fender | Gido-E: documenteur = book that lies? ;) |
| 15:59.22 | Gido-E | [TK]D-Fender see whois of blitzrage |
| 15:59.39 | fcois93 | [TK]D-Fender: yes! I just found it 10sec ago :) |
| 15:59.45 | fcois93 | [TK]D-Fender: thank you |
| 15:59.54 | *** join/#asterisk RobH (n=RobH@120-118.186-72.tampabay.res.rr.com) |
| 16:00.02 | [TK]D-Fender | Gido-E: We'd met a few times, I know full-well who he is :p |
| 16:01.04 | [TK]D-Fender | ok, lunch time, BBIAB |
| 16:01.06 | *** join/#asterisk mazpe| (n=mazpe@c-71-196-32-43.hsd1.fl.comcast.net) |
| 16:01.26 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:01.57 | blitzrage | [TK]D-Fender: book that lies? wtf? |
| 16:02.22 | strathisla8282 | blitzrage: docu-menteur |
| 16:02.28 | strathisla8282 | menteur => liar |
| 16:03.05 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:04.21 | *** join/#asterisk dni (n=f00kj00@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 16:04.57 | mazpe| | is there another command in 1.6.0.x for g729 show? |
| 16:06.45 | *** join/#asterisk Imo (n=Imo@brln-4db83a50.pool.einsundeins.de) |
| 16:07.04 | *** join/#asterisk blargman_ (n=blargman@12.234.16.130) |
| 16:09.20 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 16:12.04 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 16:13.53 | *** join/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com) |
| 16:14.28 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 16:15.44 | blitzrage | strathisla8282: oh.... damn french :) |
| 16:16.40 | *** join/#asterisk sack (n=sack@249.Red-83-55-223.dynamicIP.rima-tde.net) |
| 16:16.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:16.48 | strathisla8282 | blitzrage: yes, this sort of people is not completely dead :D |
| 16:17.16 | jjshoe | strathisla8282 are you thinking of centrex transfer and clear? |
| 16:17.59 | jjshoe | strathisla8282 this is the closest thing I know of in asterisk that might be able to accomplish what you're asking for, although it might still cost you. |
| 16:18.25 | [TK]D-Fender | blitzrage: Salut mon-ostie!... I mean "hi!" :) |
| 16:19.52 | strathisla8282 | jjshoe: if I have correctly understood what a centrex is, yes, it's something like this. |
| 16:26.13 | *** join/#asterisk Kyosh (i=48595da5@gateway/web/ajax/mibbit.com/x-90b040b7f46776ca) |
| 16:27.10 | jjshoe | merlin8282 you're simply hoping you can tell an incoming call 1) wrong number 2) try this number |
| 16:27.15 | Kyosh | is there an open-g729 available for asterisk? |
| 16:27.38 | coppice | naughty-g729 |
| 16:27.55 | Kyosh | i know i know but its been requested so much for use |
| 16:28.01 | Kyosh | id rather use ulaw or gsm |
| 16:28.36 | jjshoe | g729 is so cheap |
| 16:29.11 | Gido-E | Kyosh check: http://asterisk.hosting.lv/ |
| 16:29.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:30.53 | *** join/#asterisk Badrobot- (i=Xnomar@cpe-76-173-233-75.socal.res.rr.com) |
| 16:34.43 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 16:35.14 | merlin8282 | jjshoe: No, already did that for 2 weeks ^^ |
| 16:35.25 | PTorres | hi everyone |
| 16:35.30 | jjshoe | merlin8282 no, I'm saying that's a much easier way to explain it. |
| 16:35.32 | merlin8282 | I mean, I played a message saying this |
| 16:35.39 | merlin8282 | oh, yes |
| 16:35.53 | merlin8282 | Not exactly, in fact |
| 16:36.05 | merlin8282 | not a wrong number. It has to be transparent to the user |
| 16:36.09 | jjshoe | "bugger off and try 123-456-7890" |
| 16:36.12 | jjshoe | yes, I know. |
| 16:37.27 | jjshoe | instead of mom->pbx->cell you want it to do mom->pbx | pbx->mom try cell | mom->cell |
| 16:37.47 | PTorres | I have a isdn question , any takers ? |
| 16:37.58 | jjshoe | merlin8282 is this a feature in the legacy world? |
| 16:38.01 | jjshoe | PTorres 42. |
| 16:38.34 | merlin8282 | jjshoe: functions exist for call deflection, yes, if it's your question. |
| 16:38.38 | jjshoe | PTorres right now you have 0 takers, since you haven't asked a question, your chances are none. If you asked, there's 305 people, I bet your chances go up. |
| 16:38.45 | jjshoe | merlin8282 "functions"? on which systems? |
| 16:39.05 | merlin8282 | In Asterisk I mean, especially in Bristuff, appearantly |
| 16:39.08 | PTorres | lol, I guess I will go ahead |
| 16:39.12 | merlin8282 | Like capiCD or ZapCD |
| 16:39.22 | jjshoe | merlin8282 problem solved, get a bri then. |
| 16:39.25 | merlin8282 | But I don't know the internals |
| 16:40.03 | merlin8282 | Sorry I've to go, i'll try to understand this better than today, tomorrow... |
| 16:40.15 | merlin8282 | I already sent an email to junghanns.net for support |
| 16:40.17 | kb3ien | Is mysql for cdr not in the basic tree, but in something called addons? |
| 16:40.26 | PTorres | I have this e1 isdn trunk, it starts to link , but then the telco does not respond to the RR messages, all standard configuration, we have many e1 already working fine |
| 16:40.41 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-5-154-128.oc.oc.cox.net) |
| 16:41.35 | merlin8282 | thanks all, bye ;) |
| 16:41.35 | PTorres | it goes like >>SABME <<UA >> RR >> RR >> RR .... timeout |
| 16:41.35 | fcois93 | [TK]D-Fender: I need to do a gotoif with 2 condition. I have to do like that: GotoIf($["$var_1 == "yes" && $var_2 == "yes"]?1,2) ? is the && is right? |
| 16:41.35 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-12-127.w90-56.abo.wanadoo.fr) |
| 16:41.35 | Shaun2222 | any problem with routing a fax call through asterisk over VOIP? I want a call in from my PRI T1 to go through asterisk over SIP to a converter to the fax machine... |
| 16:41.35 | PTorres | it works time to time, but then it gets kinda looked in that cycle |
| 16:41.35 | PTorres | locked* |
| 16:41.50 | [TK]D-Fender | fcois93: You have other errors in there. |
| 16:42.22 | Chainsaw | Shaun2222: If your Asterisk server and the converter device both support T.38 fax, you should be fine. |
| 16:42.42 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 16:42.46 | Shaun2222 | uhh, does asterisk support that by default? |
| 16:42.48 | PTorres | I dont think I have a problem on my end... what should I do, or ask the telco guys to do? |
| 16:43.07 | fcois93 | GotoIf($[$var_1 == "yes" && $var_2 == "yes"]?1,2), and like that? |
| 16:44.13 | Chainsaw | Shaun2222: 1.6 should, provided you compile in spandsp. |
| 16:44.21 | Chainsaw | Shaun2222: Not sure about 1.4 or lower though. |
| 16:47.17 | PTorres | still no takers ? |
| 16:47.36 | Shaun2222 | i'm on 1.6 |
| 16:47.51 | Shaun2222 | but if it's somthing that new i'm wondering if my devices will support it. |
| 16:48.01 | *** join/#asterisk mmlj4-play (n=jkelly@209.16.86.78) |
| 16:48.02 | Chainsaw | Shaun2222: Provided you have spandsp compiled into Asterisk you should be fine. |
| 16:48.12 | Chainsaw | Shaun2222: Unless the conversion device isn't T.38 compliant, that is. But you'll see soon enough. |
| 16:50.02 | Shaun2222 | Chainsaw: whats the module for 1.6 called |
| 16:50.09 | Chainsaw | Shaun2222: spandsp |
| 16:50.28 | Shaun2222 | ya nothing in the modules dir with *sand* |
| 16:50.37 | Shaun2222 | is it a addon or i just need to tell it to build |
| 16:50.38 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-75-169.w86-215.abo.wanadoo.fr) |
| 16:50.39 | Chainsaw | Try span. |
| 16:50.43 | Chainsaw | Instead of sand. |
| 16:51.00 | Chainsaw | You need the Asterisk configure script you want it. |
| 16:51.02 | Shaun2222 | no, nothing of span either |
| 16:52.36 | Shaun2222 | what do you mean i need th config script |
| 16:53.16 | Shaun2222 | i dont see it in a make menuselect |
| 16:55.37 | PTorres | so... nobody saw nothing like that before ? _sniff_ |
| 16:56.00 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-75-169.w86-215.abo.wanadoo.fr) |
| 16:56.01 | kb3ien | anyway to build only the mysql parts of asterisk-addons, quickly? |
| 16:56.25 | Shaun2222 | Chainsaw: why is this needed, i just want the call to route directly to the fax machine/device.. i dont need asterisk to answer it at all. |
| 16:56.49 | Chainsaw | Shaun2222: Because you want Asterisk to select the correct codec for this "call". |
| 16:57.07 | Shaun2222 | what codec is correct? |
| 16:57.20 | Chainsaw | G711 I believe. |
| 16:57.22 | Shaun2222 | cant i just force that codec if a call comes in on the fax number. |
| 16:57.34 | Chainsaw | Do feel free to ask others. |
| 16:57.52 | Shaun2222 | or force the converter device to use G711 |
| 16:58.14 | kb3ien | the typos in ooh323 are messing up my build. |
| 16:58.45 | Shaun2222 | bah, i dont even have g711 |
| 16:58.50 | Shaun2222 | g722 :) |
| 16:58.51 | [TK]D-Fender | fcois93>GotoIf($[$var_1 == "yes" && $var_2 == "yes"]?1,2), and like that? <- you apparently still have no clue how to properly use variables & functions. |
| 16:59.29 | PTorres | <PROTECTED> |
| 16:59.32 | fcois93 | [TK]D-Fender: how have I to do ? |
| 16:59.53 | Shaun2222 | my linksys VOIP router has G711u support. |
| 17:00.03 | Shaun2222 | and g711a |
| 17:00.13 | [TK]D-Fender | fcois93: Go read the CHANNELVARIABLES docs in your tarball and the WIKI page of function usage |
| 17:00.47 | Shaun2222 | [TK]D-Fender: your the man, i just want to route a call from my pri t1 to a SIP converter box so the fax machine can answer it... what needs to be done? |
| 17:01.18 | [TK]D-Fender | Shaun2222: "core show application dial" |
| 17:01.34 | Shaun2222 | noo... i mean any special codecs or modules i need to use? |
| 17:01.38 | Shaun2222 | i know how to route the call to the device |
| 17:01.41 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 17:01.49 | [TK]D-Fender | Shaun2222: g.711 of some sort. |
| 17:02.38 | Shaun2222 | where do i get that from, asterisk doesnt look to ship with it. |
| 17:02.43 | fcois93 | [TK]D-Fender: the vars are examples. I test headers instead of $var_1 and $var_2 |
| 17:03.34 | [TK]D-Fender | fcois93: I only comment on actual code, not pseudo-code junk |
| 17:03.55 | [TK]D-Fender | Shaun2222: ULAW & ALAW |
| 17:04.56 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 17:05.08 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:05.09 | Shaun2222 | ah, it's just not called g711 they call it ulaw and alaw. makes sense after reading a blurb on it. |
| 17:05.47 | *** join/#asterisk nima0102 (n=nima@91.98.220.77) |
| 17:06.05 | [TK]D-Fender | Shaun2222: G.711u(law) , G.711a(law) |
| 17:06.14 | coppice | and in the SDP they call then PCMA and PCMU when they aren't really PCM at all :-\ |
| 17:07.18 | [TK]D-Fender | coppice: SHHHH!!! thats a seek-rat! |
| 17:09.24 | dni | how do u turn off sip debug ?? i thought ti was sip debug off from the console |
| 17:09.44 | henk | sip set debug off |
| 17:10.02 | dni | localhost*CLI> sip set debug off |
| 17:10.02 | dni | No such command 'sip set' (type 'help' for help) |
| 17:10.46 | henk | dni: works in my 1.4.21 |
| 17:11.14 | dni | im using Asterisk SVN-branch-1.2-r170580 |
| 17:11.14 | dni | . |
| 17:11.19 | henk | dni: and to 'sip debug off' it says The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. |
| 17:15.35 | *** join/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 17:16.03 | mintee | what's a decent webgui monitoring and cdr billing platform? |
| 17:16.20 | mintee | anything good about a2billing? |
| 17:16.40 | mintee | or am i just better off writing my own? |
| 17:17.36 | UQlev | mintee: why not if you have skills |
| 17:17.45 | mintee | time |
| 17:19.16 | nima0102 | <<<<<<<<<<<<<<<<<<<<<please note>>>>>>>>>>>>>>>>>>> |
| 17:19.45 | nima0102 | d6d3 are braodcatsting this message |
| 17:19.47 | nima0102 | " hi, you should take a look at freeswitch, it's a lot better than asterisk... join #freeswitch or visit us at http://www.freeswitch.org/" |
| 17:20.08 | mintee | oh snap |
| 17:20.47 | EmleyMoor | Trying to get an mplayer stream as a moh option - followed method on voip-info.org and it fails... running the script alone gives ls: cannot access /tmp/asterisk-moh-pipe.*: No such file or directory ; mplayer: could not connect to socket ; mplayer: No such file or directory and then needs ^C to stop |
| 17:20.51 | jaytee | any software that needs people to broadcast that it's better than brand X is obviously crap |
| 17:20.51 | [TK]D-Fender | nima0102: and who is "d6d3"? |
| 17:21.09 | mintee | 13:30 [FreeNode] -!- d6d3 [n=c0ldk1ll@zonarails.com] |
| 17:21.09 | mintee | 13:30 [FreeNode] -!- ircname : c0ldk1ll3r |
| 17:21.09 | mintee | 13:30 [FreeNode] -!- server : irc.freenode.net [http://freenode.net/] |
| 17:21.09 | mintee | 13:30 [FreeNode] -!- : is signed on as account c0ldk1ll3r |
| 17:21.09 | mintee | 13:30 [FreeNode] -!- End of WHOIS |
| 17:21.43 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:21.46 | [TK]D-Fender | nima0102: to who? |
| 17:21.48 | nima0102 | [TK]D-Fender: I do not know him |
| 17:21.59 | [TK]D-Fender | nima0102: I do, he's been banned here before. |
| 17:22.06 | dni | i jsut downloaded a more recent version of asterisk 1.4) and im getting this compile error: configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 17:22.06 | dni | <PROTECTED> |
| 17:22.08 | [TK]D-Fender | nima0102: and earning it more all the time. |
| 17:23.02 | jaytee | I should talk to one of the freenode ops I know and have him klined |
| 17:23.33 | [TK]D-Fender | jaytee: I never got the message, it doesn't appear to have made its way in here and he's not in any channel. I'll loeave well enough alone for now. |
| 17:23.57 | jaytee | yeah, until it becomes a nuisance |
| 17:24.10 | [TK]D-Fender | jaytee: I'm giving him some rope ;) |
| 17:24.32 | jaytee | that's being fair |
| 17:25.32 | nima0102 | [TK]D-Fender: thanks for your attention |
| 17:25.32 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:27.38 | dni | did anyone get this error when compiling 1.4 or 1.6 ? configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 17:27.38 | dni | <PROTECTED> |
| 17:28.37 | *** join/#asterisk delphus_ (n=delphus@unaffiliated/delphus) |
| 17:29.14 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 17:29.55 | delphus_ | is there any app command that do exactly what waitexten do when someone dials an exten, it resets the context and start looking for exten paterns again |
| 17:30.51 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 17:31.55 | KavanS | is there any way to ring an extension (cell phone) and use the feature to "press 1 to accept call"....but NOT use followme? |
| 17:33.06 | delphus_ | I mean, it should get the variable like an exten and go back to the context and match the default patterns... |
| 17:33.11 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 17:33.12 | *** join/#asterisk Math` (n=mrene@64.254.252.146) |
| 17:34.20 | [intra]lanman | [TK]D-Fender: he's been told about doing that... he's not "affiliated" with us, just FYI |
| 17:35.46 | [intra]lanman | it's just unfortunate... the fact that he's speaking the truth and noone will believe it cus he's being a retard about it ;-) |
| 17:35.49 | *** join/#asterisk ttyS1 (n=julian@adsl-074-246-089-066.sip.bct.bellsouth.net) |
| 17:36.45 | apeiron | dni, Do you have a C++ preprocessor at /lib/cpp? |
| 17:36.50 | mercutioviz | [intra]lanman: are you *trying* to get kickban'd from this channel? :P |
| 17:37.16 | dni | apeiron, [root@localhost src]# ls -al /lib/cpp |
| 17:37.16 | dni | lrwxrwxrwx 1 root root 14 Mar 12 10:12 /lib/cpp -> ../usr/bin/cpp |
| 17:37.19 | ttyS1 | How can I make the RTP media to be peer to peer instead of being handled by Asterisk ? |
| 17:37.21 | [intra]lanman | mercutioviz: lol, no... i don't think they'd do that for something so petty... just poking a little fun |
| 17:37.30 | [intra]lanman | but i'm done now |
| 17:37.35 | apeiron | dni, And is /usr/bin/cpp valid? |
| 17:37.39 | mercutioviz | :) |
| 17:37.54 | dni | apeiron, yes |
| 17:37.56 | *** join/#asterisk bob_slacker (n=vncsnvs_@189.114.27.40) |
| 17:38.09 | apeiron | dni, Not sure then. Sure you've got all your compiler tools installed? |
| 17:38.21 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 17:38.40 | dni | apeiron, yea man, everything should be proper.. i compiled and ran 1.2 with no issues |
| 17:39.06 | apeiron | dni, Really unsure, then. Ask your vendor about it, perhaps? |
| 17:39.24 | dni | im running centOS |
| 17:39.30 | apeiron | I'm sorry. |
| 17:39.32 | dni | 5.2 |
| 17:39.37 | dni | ok thanks for the feedback tho |
| 17:39.49 | *** join/#asterisk rnst (n=Ernzt@teisa.netvision.com.py) |
| 17:39.49 | apeiron | You might want to see a doctor about the CentOS thing. |
| 17:39.59 | *** join/#asterisk shaun2222 (n=Shaun222@ip68-5-154-128.oc.oc.cox.net) |
| 17:40.48 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 17:44.12 | kb3ien | nice Freepbx just reset my mysql root password! |
| 17:44.18 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 17:46.06 | *** part/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com) |
| 17:49.14 | shaun2222 | dni: i've gotten those before, not with asterisk but with other peices of software. Those ones are always fun to figure out wtf went wrong. |
| 17:49.19 | shaun2222 | dni: check the config.log |
| 17:51.58 | *** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 17:52.12 | dni | shaun2222, this si where it failed |
| 17:52.12 | dni | conftest.c:9:28: error: ac_nonexistent.h: No such file or directory |
| 17:55.25 | *** join/#asterisk m3F (n=m3F@190.43.32.109) |
| 17:55.48 | *** join/#asterisk Eduardo_Assis (n=Eduardo_@200-207-61-133.dsl.telesp.net.br) |
| 17:56.37 | m3F | hi! |
| 17:56.55 | m3F | i am an openSUSE user, and want to install Asterisk |
| 17:56.59 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 17:56.59 | *** mode/#asterisk [+o Mog] by ChanServ |
| 17:57.08 | bob_slacker | Eduardo_Assis, hello, this is Linus Torvalds, and i pronounce Linux as Linux. |
| 17:57.13 | m3F | but my question is, is it include a GUI? |
| 17:57.30 | Eduardo_Assis | bob_slacker, wathever ... rs |
| 17:58.00 | dni | shaun2222, i also see this error regarding g++ ... ./configure: line 5324: g++: command not found |
| 17:58.00 | dni | <PROTECTED> |
| 17:58.13 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 17:58.22 | m3F | which packages do i need to install to have a GUI to manage a Call Center of about 15 phones |
| 17:58.24 | dni | im running gcc 4.1.2 |
| 17:58.25 | m3F | ? |
| 17:59.12 | Eduardo_Assis | m3F, install Trixbox |
| 17:59.31 | *** join/#asterisk iure_da_luz (n=t7DS@201.18.239.194) |
| 17:59.31 | apeiron | dni, um, no... cpp and g++ are different. |
| 17:59.36 | iure_da_luz | Eduardo_Assis hello |
| 17:59.39 | apeiron | dni, It seems you do not, in fact, have a full toolchain installed. |
| 17:59.44 | iure_da_luz | Eduardo_Assis you no speak english! |
| 17:59.51 | apeiron | Doesn't anyone know how to build things from source these days? |
| 18:00.01 | *** part/#asterisk Math` (n=mrene@64.254.252.146) |
| 18:00.02 | Eduardo_Assis | iure_da_luz, more less |
| 18:00.27 | dni | apeiron, i built 1.2 with no issues. Im just getting these odd errors in 1.4 and 1.6 |
| 18:00.41 | iure_da_luz | Eduardo_Assis sorry |
| 18:00.42 | apeiron | dni, Something changed between then and now, obviously. |
| 18:00.53 | Eduardo_Assis | iure_da_luz, no problem |
| 18:01.13 | m3F | Eduardo_Assis, but i do not want to install the whole Trixbox distro, i would want to install it in my openSUSE |
| 18:01.36 | iure_da_luz | Eduardo_Assis yes, i have a problem! |
| 18:01.55 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.ip.grandenetworks.net) |
| 18:02.01 | Eduardo_Assis | m3F, Run your own gui |
| 18:02.15 | m3F | Eduardo_Assis, if i add the Trixbox repository: http://yum.trixbox.org/centos/ , which packages do i have to install? |
| 18:03.10 | Eduardo_Assis | m3F, download packages freepbx |
| 18:03.12 | iure_da_luz | Eduardo_Assis I do not understand why you left the asterisk-us, why? |
| 18:03.31 | Eduardo_Assis | iure_da_luz, I lesft asterisk-br |
| 18:03.41 | Eduardo_Assis | left |
| 18:03.43 | iure_da_luz | lesft? |
| 18:03.45 | kb3ien | im trying to install asterisk-addons 1.4.7 my mysql options are XXX out in make menuconfig. all that is missing in ./configure is mysql_config what is that file supposed to do? |
| 18:04.05 | iure_da_luz | because you is not here and there? |
| 18:04.14 | Eduardo_Assis | m3F, http://www.freepbx.org/download-freepbx |
| 18:04.15 | ddickenson | Is this where I'd go to find some information about sharing extensions on multiple phones? I don't necessarily need the "SLA" key system type effect, just to be able to see if an extension is off hook and or pick up that extension from another phone |
| 18:04.51 | dni | finally got it working |
| 18:04.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:05.07 | dni | w000000000t! |
| 18:05.14 | kb3ien | offhook is a littel hard to detect. its easy to use presence to see if a call in in progress. |
| 18:05.29 | ddickenson | call in progress would work |
| 18:05.38 | iure_da_luz | Eduardo_Assis how can I create a dialplan to dial the extensions that are with "username" and not with "3001"? |
| 18:05.41 | Eduardo_Assis | m3F, You have experience with php? |
| 18:05.51 | kb3ien | what kinds of phones? |
| 18:06.17 | m3F | no, Eduardo_Assis |
| 18:06.32 | Eduardo_Assis | iure_da_luz, [3001] to [name] |
| 18:06.53 | ddickenson | The basic setup is going to be 2 pots lines coming in that ring a ring group of 4 main "common area" phones then goes to an IVR that can transfer to one of 8 rooms. Problem is I want to be able to put the main phones on the "room" phones so they can know if one is in use. |
| 18:07.01 | ddickenson | cisco 7960's with sip firmware |
| 18:07.10 | Eduardo_Assis | m3F, then downloads freepbx and use |
| 18:07.12 | KavanS | is there any way to ring an extension (cell phone) and use the feature to "press 1 to accept call"....but NOT use followme? |
| 18:07.24 | iure_da_luz | Eduardo_Assis I have to assign a username to the branch? |
| 18:07.39 | kb3ien | hrm, not my forte, but 'presence' is what you seek i think. |
| 18:07.50 | Eduardo_Assis | iure_da_luz, yes |
| 18:07.52 | [TK]D-Fender | KavanS: I've already answered this before... "core show application dial" <- M() |
| 18:07.55 | ddickenson | excellent, I'll check into it |
| 18:08.02 | iure_da_luz | Eduardo_Assis exten => _XXXX,1,Dial(SIP/${EXTEN},20,tTr) |
| 18:08.10 | KavanS | [TK]D-Fender: ok, sorry I did not see reply....looking into now |
| 18:08.13 | Eduardo_Assis | (SIP/name) |
| 18:08.43 | iure_da_luz | Eduardo_Assis have no variable available for usernames such as this? |
| 18:08.44 | [TK]D-Fender | ddickenson: On many common SIP phones you can assign a BLF+SD key and make the SD exten check the status of the device its meant to refer to and act accordingly. |
| 18:08.50 | iure_da_luz | exten => _XXXX,1,Dial(SIP/${EXTEN},20,tTr) Eduardo_Assis |
| 18:08.52 | ddickenson | just out of curiosity has anyone been able to make the SLA (Shared line appearance) work in trixbox 2.6.2...? |
| 18:09.03 | Eduardo_Assis | iure_da_luz, Calls of entry or exit? |
| 18:09.14 | iure_da_luz | exit and entry |
| 18:09.15 | KavanS | [TK]D-Fender: sweet, that's what I was looking for |
| 18:09.24 | iure_da_luz | or entry and exit |
| 18:09.35 | ddickenson | forgive my igrorance but what do those acronyms stand for? BLF+SD |
| 18:09.45 | [TK]D-Fender | ddickenson: AFAIK, trixbox's FreePBX fork doesn't do this, nor does the oridignal. |
| 18:09.50 | [TK]D-Fender | ~blf |
| 18:09.51 | jbot | [blf] Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 18:09.56 | [TK]D-Fender | ddickenson: + Speed Dial |
| 18:10.12 | [TK]D-Fender | ddickenson: basic presence. |
| 18:10.12 | Eduardo_Assis | iure_da_luz, _XXXX,1,Dial(SIP/username,20,tTr) |
| 18:10.12 | ddickenson | ahh, yes. that would be good |
| 18:10.18 | iure_da_luz | Eduardo_Assis nops |
| 18:10.35 | ddickenson | any pointers to pubs that explain how to implement this? |
| 18:10.39 | iure_da_luz | Eduardo_Assis I want to dial using the username |
| 18:10.51 | iure_da_luz | exten => _iure,1,Dial(SIP/${EXTEN},20,r) |
| 18:11.10 | [TK]D-Fender | ddickenson: Look at the pickup apps. use the HINT priority. thats it |
| 18:11.21 | iure_da_luz | Eduardo_Assis dialplan seek a rule for all usernames |
| 18:11.42 | ddickenson | [TK]D-Fender: thanks |
| 18:12.05 | Eduardo_Assis | iure_da_luz, Never used so _name,1,Dial() sorry |
| 18:12.31 | Eduardo_Assis | iure_da_luz, only Dial(SIP/name|120|tT) |
| 18:12.51 | iure_da_luz | what? |
| 18:13.08 | Eduardo_Assis | iure_da_luz> exten => _iure,1,Dial(SIP/${EXTEN},20,r) |
| 18:13.15 | Eduardo_Assis | ------------------------------^ |
| 18:13.19 | Eduardo_Assis | --------------------------^ |
| 18:13.41 | iure_da_luz | hmmm |
| 18:13.53 | [TK]D-Fender | Eduardo_Assis: No point in ${EXTEN} there |
| 18:14.14 | iure_da_luz | Eduardo_Assis guax we be spying |
| 18:14.30 | ddickenson | It's been quite a while since I've done anything IRC is there a PM function somewhere? Also I wanted to talk to anyone willing that has any experience hooking up Nortel Option 81c with lineside t1 cards to an asterisk system. |
| 18:15.07 | kb3ien | okay getting closer: cdr_addon_mysql.c:314: error: �ast_config_load� undeclared (first use in this function) |
| 18:15.17 | PTorres | hey guys... me again :) ... can anyone take a look at this isdn trace please http://pastebin.com/m4246fccb |
| 18:17.03 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:17.21 | kb3ien | well its right. that function isnt declared anywhere the complier was told to look. |
| 18:18.06 | apeiron | ddickenson, What was the issue with your toolchain setup? |
| 18:18.08 | apeiron | er |
| 18:18.09 | apeiron | dni, ^^ |
| 18:19.10 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:20.14 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:20.16 | ddickenson | apeiron: I am trying to figure out the programming on both nortel side and asterisk side to hook up my 81c to my asterisk box via lineside t1 card. Problem that I've seen in all the step by steps online is that they assume that your pbx is licensed for pri and dchannel use which mine is not. I can only do basic 24 channel t1's, no dchannel |
| 18:20.37 | apeiron | ddickenson, Sorry, I meant dni. Not familiar with your setup. |
| 18:20.46 | ddickenson | ah, np |
| 18:20.59 | apeiron | is quite the * nub |
| 18:21.11 | ddickenson | you're not alone |
| 18:21.19 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:21.22 | dni | apeiron, g++ was the issue |
| 18:21.30 | apeiron | dni, Not installed? |
| 18:21.34 | dni | its in a diff package on centos |
| 18:21.53 | dni | i needed to install gcc-c++ |
| 18:21.59 | dni | thanks for the feedback and help |
| 18:22.13 | apeiron | I would hope that they have a meta-package for the toolchain. |
| 18:23.19 | *** part/#asterisk iure_da_luz (n=t7DS@201.18.239.194) |
| 18:26.22 | kb3ien | progress: --with-asterisk=/a/sane/path fixed on problem. uncovered another. |
| 18:26.25 | kb3ien | cdr_addon_mysql.c:314:30: error: macro "ast_config_load" requires 2 arguments, but only 1 given |
| 18:26.58 | kb3ien | is there a chart of compatability between asterisk and asterisk-addon ? |
| 18:27.49 | kb3ien | functioncalls get away with that sort of thing, macros dont. |
| 18:40.56 | kb3ien | well addons 1.4.7 and 1.4.6 are both broken in this regard. |
| 18:41.58 | ddickenson | from what I hear cisco call manager phones are now supported in asterisk, is that correct? can you now use some of the features that were lost in the sip firmwares for the cisco phones? |
| 18:48.56 | pdmmm | ddickenson: chan_sccp-b or chan_skinny |
| 18:56.50 | ddickenson | I guess sccp... I thought those two were interchangeable |
| 18:59.40 | ddickenson | how do you reply to a certain person |
| 18:59.56 | [TK]D-Fender | ddickenson: So don't set for PRI. Use FXOLS for your signaling to your nortel if * is acting like the telco to it |
| 19:01.27 | ddickenson | do you need to set it up as a digital loop? actually currently the nortel would be acting as the telco to asterisk. all my t1's are coming in through there |
| 19:02.15 | soulclaimer | Anyone seen .lock files getting hung up in voicemail folders in asterisk 1.4.21.2 |
| 19:05.20 | *** join/#asterisk exsync (n=mjohnson@pdpc/supporter/active/exsync) |
| 19:12.50 | jblack | Who wants to hear some off topic insanity? |
| 19:13.30 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:18.16 | *** join/#asterisk Ritzerisk (n=Ritztech@65.105.209.226.ptr.us.xo.net) |
| 19:18.35 | Ritzerisk | anyone in here know hylafax pretty well |
| 19:18.47 | jblack | I've never managed to get it to work. |
| 19:19.04 | Ritzerisk | haha |
| 19:19.37 | mosty | i've used hylafax for years |
| 19:20.11 | Ritzerisk | have you ever gotten a msg MODEM TIMEOUT: waiting for v.21 carrier |
| 19:20.24 | Ritzerisk | ive seen this around the net quite a bit so i think im in the same boat |
| 19:20.46 | Ritzerisk | inbound is FINE i can get 8 - 14 page faxes 100% all the time its just issues outbound |
| 19:21.20 | Ritzerisk | No answer (T.30 T1 timeout) |
| 19:21.46 | mosty | from memory the t.30 t1 timeout means there was no fax machine at the other end |
| 19:22.56 | mosty | try the hylafax-users mailing list- they're pretty good |
| 19:23.06 | Ritzerisk | hmmm trippy of course ive tested dozens of differnt landlines and other boxes |
| 19:23.29 | Ritzerisk | i saw taht lee guy on there is pretty good but cant seem to find an answer to an issue that ive seen come up |
| 19:24.46 | mosty | can you test it with a standalone fax machine? |
| 19:26.23 | Ritzerisk | yea ive tested it with that too i might put a butt set on the end of it to see what i hear and if i lost audio somewhere |
| 19:26.36 | Ritzerisk | but what gets me is inbound is fine |
| 19:27.26 | EmleyMoor | Ritzerisk: Your nickname makes me think of an open source hotel telephone system <g> |
| 19:27.33 | Ritzerisk | but when i called my cell phone once with the sendfax -n -d i heard beep beep |
| 19:27.40 | Ritzerisk | but that could be different |
| 19:27.48 | Ritzerisk | haha ;) |
| 19:31.14 | *** join/#asterisk hawk (n=hawk@l.qw.se) |
| 19:33.00 | mchou | anyone have experience with Zulty ip hones? |
| 19:33.05 | mchou | phones* |
| 19:34.36 | SuPrSluG | anyone using AMD? my problem is when the call is answered it still sees it as a machine, using pretty much the standard amd. conf file for configuration. |
| 19:40.04 | *** join/#asterisk chrisbdaemon (n=chris@unaffiliated/chrisbdaemon) |
| 19:40.15 | EmleyMoor | Anyone installed AstyCrapper yet? I've been contemplating it. |
| 19:42.14 | jaytee | what the hell is AstyCrapper? |
| 19:42.28 | *** join/#asterisk aenaus_ (n=hdgfghf@91.140.102.149) |
| 19:42.39 | *** join/#asterisk reallost1 (n=reallost@12-208-208-159.client.mchsi.com) |
| 19:42.40 | EmleyMoor | It attempts to engage telemarketers in conversation |
| 19:43.16 | mchou | I was tempted but just decided to send telemarketers to /dev/null |
| 19:43.58 | mchou | astycrapper doesnt work well over voip |
| 19:44.10 | mchou | silence detection borked |
| 19:44.15 | EmleyMoor | Hmmm... fair enough |
| 19:44.26 | mchou | kinda defeats the whole purposed |
| 19:44.30 | reallost1 | I'm having a weird problem writing to a db from asterisk. I can select from the db just fine, but it won't seem to write to the db. |
| 19:45.02 | [TK]D-Fender | reallost1: If you pastebin something substantial for us to look at maybe we can tell you why |
| 19:45.10 | [TK]D-Fender | ~pb |
| 19:45.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 19:45.15 | reallost1 | Well, it doesn't give any errors. |
| 19:45.27 | [TK]D-Fender | reallost1: Doesn't mean its being done right |
| 19:45.42 | reallost1 | It displays the query with the debug set and I can take it and perform the insert manually. |
| 19:45.53 | mosty | reallost: what about the database server logs? |
| 19:45.57 | [TK]D-Fender | reallost1: If you don't show us, we can't help you |
| 19:46.20 | mosty | reallost: and if you try it manually using the same username/password as asterisk, does it still work? |
| 19:46.26 | reallost1 | I am not getting any errors in the db server logs also. |
| 19:46.37 | reallost1 | mosty, yeah I checked permissions. |
| 19:47.01 | reallost1 | let me see if I can pastebin anything... |
| 19:48.19 | PTorres | there were also 'ip' permissions (at least in postgresql) |
| 19:50.15 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 19:52.32 | reallost1 | PTorres, yeah I have the IP permissions set. This is postgres. |
| 19:52.40 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 19:53.37 | Ritzerisk | i wonder if i could use hylafax in the future as like setup for like virtual modems and terminals to virtual modems only because we have a big mitel customer database and we have alot of Remotes we do |
| 19:54.16 | Ritzerisk | like a vt100 ;) |
| 19:56.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
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| 20:01.09 | *** join/#asterisk cbullock81 (n=cbullock@adsl-251-32-49.jan.bellsouth.net) |
| 20:02.16 | cbullock81 | is there someone available that might be able to help me with a strange problem with a very simple dialplan? |
| 20:03.40 | *** join/#asterisk ingenius (n=alektro@247.175.73.200-static.serversur.net) |
| 20:04.46 | [TK]D-Fender | cbullock81: Show us what you've got, and what its not doing that it should |
| 20:04.50 | [TK]D-Fender | ~pb |
| 20:04.50 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 20:04.52 | [TK]D-Fender | ^^^^^^^^^ |
| 20:05.06 | cbullock81 | will do. thanks! |
| 20:05.36 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 20:10.17 | cbullock81 | http://pastebin.com/d7c6b8f4b ok. this is totally basic, but I'm a newbie and just trying to get the basics going. i can make internal calls between extensions w/o any problems, but when I get an incoming call on one of my dahdi channels (x100p card), it rings for the incoming caller, and it rings the internal extension that it's supposed to, but after 2 rings the incoming caller gets a fast busy. |
| 20:10.33 | cbullock81 | if you try to answer the call from the internal phone, you just get a dialtone |
| 20:11.34 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 20:11.47 | [TK]D-Fender | cbullock81: Huge oversight : you have not defined your codecs for your phones |
| 20:12.09 | [TK]D-Fender | cbullock81: Set ONE explicitly |
| 20:12.38 | [TK]D-Fender | cbullock81: Correct this, then enable SIP debug and pastebin a failed call if it doesn't solve it |
| 20:12.48 | cbullock81 | ok. will do that. thanks so much! |
| 20:13.16 | jplank | besides setting port=5061 is there anything else I have to do to * to listen on that port? |
| 20:13.34 | jplank | setting port=5061 in the endpoint in sip.conf** |
| 20:13.54 | jplank | assuming I'm not using bindport in [general] of sip.conf |
| 20:18.20 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:21.44 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 20:24.59 | *** part/#asterisk delphus_ (n=delphus@unaffiliated/delphus) |
| 20:26.52 | [TK]D-Fender | checkout time, bbiab |
| 20:26.59 | cbullock81 | [TK]D-Fender: http://pastebin.com/db38af5a Here is my sip debug. Still same thing. |
| 20:27.31 | cbullock81 | ah... just missed him.... anyone else want to take a stab at this problem |
| 20:28.53 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:29.14 | *** join/#asterisk bijit (n=benji@201.198.72.142) |
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| 20:32.47 | *** join/#asterisk af_ (n=getsmart@88-149-240-204.dynamic.ngi.it) |
| 20:42.06 | jaytee | anyone good with shell scripting in here? I'm using a script to run a mysqldump for the current date and using a variable called present that's set as present=`date +%b%d%y` but I'd like to have a variable called yesterday set the previous day's date value in the same format. |
| 20:47.24 | pdmmm | jaytee: date +&b&d&y -24H |
| 20:47.40 | pdmmm | er |
| 20:47.42 | pdmmm | %'s |
| 20:48.20 | pdmmm | er |
| 20:48.27 | pdmmm | date -v1d +%b%d%y |
| 20:48.28 | pdmmm | that |
| 20:49.03 | jaytee | pdmmm, ok, now I'm totally confused. are those two different methods? |
| 20:49.10 | pdmmm | hm |
| 20:49.12 | pdmmm | holdon |
| 20:49.13 | pdmmm | i'm off |
| 20:50.58 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:50.58 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 20:51.20 | pdmmm | on linux |
| 20:51.22 | pdmmm | <PROTECTED> |
| 20:52.14 | cbullock81 | anyone available to help me with a strange problem: incoming calls on dahdi channel go to fast busy, but ring the appropriate internal sip channel |
| 20:53.13 | rbd | any opinions on running asterisk reniced at -15 or so? I don't want to run asterisk -p (the system does some other things), but I would like to reduce the chances of voip jitters as much as possible |
| 20:57.47 | Qwell | rbd: jitter is a network latency thing |
| 20:58.09 | Qwell | also the phrase "voip jitters" is kinda funny |
| 20:58.11 | rbd | Qwell, sorry, I meant choppyness due to CPU-related issues |
| 20:58.11 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:03.59 | cbullock81 | [TK]D-Fender: I added the codec, but I've still got the same thing going on. I know nothing about the inner workings of SIP, but I've been studying the sip debug (http://pastebin.com/db38af5a) and it looks like there is a SIP Cancel: that goes out right after the call starts ringing |
| 21:06.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:08.37 | [TK]D-Fender | cbullock81: Answer the line first, wait (2), then move on to your dial. See what heppens |
| 21:09.02 | cbullock81 | ok. will try it now |
| 21:10.36 | cbullock81 | D-Fender: that worked! so, what could be the explanation for that? |
| 21:12.55 | [TK]D-Fender | cbullock81: Not 100% certain just yet |
| 21:14.10 | cbullock81 | D-Fender: well you have just saved me so much grief!!! thank you so much! do you think its something within asterisk, or within dahdi, or still unknown? |
| 21:18.34 | henk | somehow i got asterisk to display something like 'incoming call to FOO from BAR'. can anyone tell me what to do to get that? |
| 21:22.46 | lesouvage | Henk: You can use something like exten => s,n,NoOp(incoming call to ${EXTEN} from ${CALLERID(num)}) and the inf will show up in the cli |
| 21:26.01 | henk | lesouvage: good you mention it, but i thought my asterisk already did that somehow... i probably confuse the systems... |
| 21:27.40 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 21:27.56 | watchy | anyone ever had issues with a Mediatrix not responding to DTMF tones? |
| 21:28.58 | lesouvage | watchy: perhaps dtmf=inbound in combination with a compressed codec like gsm or g729 |
| 21:29.23 | bijit | watchy: had problems with mediatrix not connecting to asterisk...had to use realm = asterisk.. |
| 21:29.27 | watchy | hmm |
| 21:29.38 | watchy | problem is. i can pickup and dialout fine. |
| 21:29.51 | watchy | then hang up, pick up again. get dialtone. dial, but the dialtone never foes away |
| 21:29.52 | watchy | goes |
| 21:30.02 | watchy | and i'd paypal someone $100 right now to fix my issue |
| 21:34.11 | watchy | bijit: can you paste me one of your auth statements in your sip.conf? |
| 21:35.20 | bijit | watchy: what model you using? |
| 21:35.57 | watchy | i got 3 4124s |
| 21:38.27 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 21:38.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:38.58 | jplank | watchy: seems pretty easy to configure http://www.voip-info.org/wiki/view/Settings+Mediatrix+APAs+with+FreePBX |
| 21:40.28 | bijit | watchy: you got this dtmfmode=rfc2833 ? |
| 21:41.12 | watchy | yea actually they are easy jplank and it works fine |
| 21:41.30 | watchy | except when you hang a call up. you pick back up, you get a dialtone |
| 21:41.36 | watchy | but when you hit digits it doent work |
| 21:42.05 | jplank | that happens on all three of your boxes? |
| 21:42.12 | watchy | yea unfortunately |
| 21:42.29 | watchy | you pickup and get nice sounding dialtone, but no digits do anything |
| 21:42.37 | *** join/#asterisk dni (n=f00kj00@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 21:42.40 | watchy | hang ity up wait a minute+ then it worksd |
| 21:43.11 | jplank | that sounds like either a flashhook problem, or the call isn't properly ending (or a little bit of both) |
| 21:43.26 | jplank | I've seen issues like that with other devices |
| 21:43.27 | watchy | would it be on the * side or mediatrix side? |
| 21:43.41 | jplank | I would guess the mediatrix side |
| 21:43.46 | jplank | do you have flashhook enabled? |
| 21:43.56 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 21:43.56 | jplank | (assuming theres a setting for it) |
| 21:43.59 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 21:44.06 | watchy | hmm. |
| 21:44.14 | watchy | lemme search for the setting in the mediatrix pdf |
| 21:44.15 | watchy | hold |
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| 21:45.06 | watchy | fxsFlashHookDetectionDelayMin R/W Minimum time in ms the hook switch must remain pressed to perform a flash |
| 21:45.06 | watchy | hook. |
| 21:45.07 | watchy | Default Value: 100 |
| 21:45.13 | watchy | thats a setting in the mediatrix |
| 21:45.27 | jplank | are you going to be using flashhook? |
| 21:45.46 | MainMax | Hi, does anyone know s SIP Trunk provider that have free demo accounts without credit card registration... |
| 21:45.48 | jplank | I'd def move it higher then 100ms, I'd set it closer to 750 to 1000ms |
| 21:46.08 | jplank | MainMax: a stupid one? |
| 21:47.20 | MainMax | any one :) just to try out dailout |
| 21:47.25 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 21:47.30 | bijit | watchy: when you hung up does CLI says hunup? |
| 21:48.32 | watchy | i tried to debug just 1 line it floods the console to much to see anything readable |
| 21:49.14 | lesouvage | mainmax: you can setup 2 asterisk boxes and give one of them the role of provider. |
| 21:49.59 | watchy | i'm gonna try to change the flash hook setting |
| 21:51.12 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
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| 21:52.04 | MainMax | I need to show my boss that all this could work before making a decision about getting into all this VOip stuff |
| 21:53.19 | bijit | ~book |
| 21:53.20 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:53.25 | MainMax | And how can you try a certain provider out before signing s contract with it... |
| 21:53.48 | stabler | MainMax: flowroute.com gives you a free $.25 credit to test out there service |
| 21:53.49 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 21:53.54 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 21:54.19 | stabler | .25 gives you plenty of test calls |
| 21:54.38 | watchy | i changed it jplank: but i cant test it till tommorow |
| 21:54.49 | MainMax | stabler: thanks. i need to make about 3 call 2 mins each :) |
| 21:55.05 | joako | MainMax: If you never used asterisk I would suggest you play with it a bit, buy 2-3 phones, use it for a month |
| 21:55.18 | stabler | MainMax: the $.25 credit will be plenty to do that |
| 21:55.43 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 21:56.01 | jplank | watchy: I'm not saying thats def the issue, I've just had similar problems with similar devices (Adtran, Verilink) and it always turns out to be a flashhook problem. Hope it helps |
| 21:57.49 | MainMax | i need kinda callcenter for 10 users. planning on using softphones with stereo headsets. AsteriskNOW on a linuxbox. Is that a good idea? |
| 21:59.10 | jplank | sure |
| 21:59.39 | jplank | anyone in here roll their own ISO (kind of like trixbox or PIAF)? |
| 21:59.44 | *** part/#asterisk Khratos (n=khratos@190.166.103.111) |
| 21:59.59 | jblack | Instead, you could announce to the world that you're instead hiring the deaf, and all support will now happen via im. |
| 22:00.13 | jblack | disabled people need jobs too. :P |
| 22:00.27 | jplank | wrong channel jblack? |
| 22:00.39 | jplank | oh, I get it |
| 22:01.01 | jplank | you'd def get brownie points for that from someone |
| 22:01.22 | watchy | hopefully this fixes it |
| 22:01.32 | watchy | i've been trying to get in touch with support at mediatrix |
| 22:01.35 | watchy | but jesus do they suck |
| 22:02.27 | MainMax | can anyone advise on a SIP provider with unlimited plans for outgoing calls (or a cheap ones if you buy in a bulk) and i dont need DID... |
| 22:03.02 | UQlev | MainMax: checkbox is good enough |
| 22:03.15 | bijit | ~voip |
| 22:03.16 | jbot | extra, extra, read all about it, voip is Voice over IP |
| 22:03.35 | bijit | jbot: lol |
| 22:03.35 | jbot | extra, extra, read all about it, lol is stands for Laughing Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
| 22:03.42 | jameswf | totaly digging druid... quite sexc |
| 22:04.01 | jameswf | ~botabuse |
| 22:04.02 | jbot | extra, extra, read all about it, botabuse is fun |
| 22:04.23 | bijit | ~provider |
| 22:04.46 | jplank | ~[TK]D-Fender |
| 22:04.47 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 22:04.53 | *** join/#asterisk neurosys (n=vinix@c-67-191-94-122.hsd1.fl.comcast.net) |
| 22:04.56 | jplank | heh |
| 22:05.03 | tzafrir_laptop | hmm... who edited ~botabuse? |
| 22:05.05 | bijit | had one that list good Voip provider :-) |
| 22:05.12 | MainMax | thanks ill check it out. |
| 22:05.21 | tzafrir_laptop | ~itsp |
| 22:05.21 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:05.52 | jplank | ~itsplist-us |
| 22:05.53 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 22:06.01 | jplank | there you go |
| 22:06.06 | jplank | :)\ |
| 22:07.25 | lesouvage | ~42 |
| 22:07.26 | jbot | rumour has it, 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
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| 22:33.46 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:33.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:34.03 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.24 (2009/03/16), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
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| 22:34.17 | lmadsen | Asterisk 1.4.24 is now available via http://downloads.digium.com/. Thanks! |
| 22:35.01 | apeiron | er, really? |
| 22:35.10 | apeiron | I'm getting "Index of /" |
| 22:36.43 | apeiron | ah, it's a few levels down. |
| 22:36.53 | apeiron | was expecting a pretty download page |
| 22:38.24 | lmadsen | we don't directly link because the main page where they are linked changes and only contains the latest releases |
| 22:38.31 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:38.37 | lmadsen | so every time we created a release, the link would break :) |
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| 22:39.49 | apeiron | ah. Okie. |
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| 23:34.15 | phix | hmmmm, debian package of asterisk doesn't seem to have NVFax, NVBackgroundDetect, etc. |
| 23:34.53 | phix | my zaptel drivers pick up a fax and complains if I don't have a fax extension, but when I do have a fax extension it tells me the call is UNKNOWN and fails |
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| 23:37.14 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-48.dyn.centurytel.net) |
| 23:37.30 | jeff_phillips | hello |
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| 23:37.42 | phix | hi |
| 23:37.47 | buzkashi | hi |
| 23:37.55 | phix | how's it going? |
| 23:38.00 | jeff_phillips | good, yourself? |
| 23:38.14 | phix | great except for my issues with asterisk |
| 23:38.26 | phix | well in particular the asterisk package that comes with debian lenny |
| 23:38.29 | [TK]D-Fender | phix: those are 3rd party apps, and not maintained, either |
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| 23:38.49 | phix | [TK]D-Fender: oh, what what app do I use for fax detection instead then? |
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| 23:38.57 | phix | s/what/so/ |
| 23:39.43 | [TK]D-Fender | phix: Only the "fax" standard extension for Zap IIRC |
| 23:39.44 | phix | thank you jbot |
| 23:40.15 | phix | [TK]D-Fender: so I can only fax detect on ZAP? |
| 23:40.23 | [TK]D-Fender | phix: last I checked |
| 23:40.27 | phix | (that is fine though, that is what I want it for) |
| 23:41.04 | jeff_phillips | i'm debating what hardware to use for the next project I'm working on |
| 23:41.07 | phix | s/(that) is/\1\'s/ |
| 23:41.13 | buzkashi | i have a tdm400p card installed and the caller can barley hear the enter the extension of the person you are trying to reach, how can I increase the volume? |
| 23:41.14 | phix | come on jbot :) |
| 23:41.29 | phix | buzkashi: easy |
| 23:41.33 | lesouvage | Is it correct/as intended that an outbound call innitiated by a callfile isn't traceble in the cdr? |
| 23:41.43 | jeff_phillips | I'm wondering how asterisk handles on 512 mb of ram -- I'd throw more in, but it's on a hosted server & they use ram as a big reason to upsell you to a much bigger monthly bill |
| 23:42.00 | jeff_phillips | I'm just doing straight sip-to-sip with an IVR in the middle |
| 23:42.03 | [TK]D-Fender | buzkashi: "rxgain" / "txgain" in zapata.conf/chan_dahdi.conf |
| 23:42.35 | buzkashi | Is there a document for dahdi.conf to tweak this out on the net somewhere? |
| 23:42.40 | phix | ;rxgain=0.0 |
| 23:42.40 | phix | ;txgain=0.0 |
| 23:43.03 | buzkashi | phix this increases the volume? |
| 23:43.04 | phix | uncomment them of corse and change the 0's to something higher, for txgain that is |
| 23:43.24 | phix | yes, tx == transmit, rx == recieve? |
| 23:44.36 | phix | [TK]D-Fender: ok, well it is putting it in the wrong context :\ |
| 23:45.25 | phix | [TK]D-Fender: I have faxdetect = on in zapata.conf |
| 23:45.50 | jeff_phillips | phix: You should probably tune the gain levels by calling your telco's milliwatt test # & watching it in ztmonitor |
| 23:45.52 | phix | I also tried = both as well, it seems to pick it up and complain if I dont have a fax exten for it, but when I do it fails |
| 23:46.04 | jeff_phillips | i had instructions bookmarked somewhere, one sec |
| 23:46.09 | phix | jeff_phillips: I see |
| 23:46.48 | [TK]D-Fender | phix: then its in the wrong place. |
| 23:46.51 | [TK]D-Fender | BBL |
| 23:46.56 | phix | :) |
| 23:47.00 | phix | [TK]D-Fender: <3 |
| 23:47.06 | jeff_phillips | http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment |
| 23:48.13 | jeff_phillips | i had a better link somewhere -- i think it's about time I organize my bookmarks |
| 23:48.52 | buzkashi | What number do I begin with ? |
| 23:49.49 | freddyk | hi all |
| 23:49.51 | jeff_phillips | buzkashi: you mean in ztmonitor? |
| 23:50.05 | freddyk | any expert of sip here ? i have a problem with sip registration on 1.6.1 trunk |
| 23:50.47 | jeff_phillips | buzkashi/phix: call your telco's 1004hz tone test generator #, run ztmonitor, and adjust rxgain until you get numbers around 14800 if my memory is right |
| 23:53.10 | buzkashi | the is being tested using cable telephone access can I still do this? |
| 23:54.02 | jeff_phillips | how are you connected to the cable service? |
| 23:54.21 | buzkashi | through a cable modem |
| 23:54.38 | buzkashi | cable modem is for internet, cable and telephone service |
| 23:55.02 | jeff_phillips | so the cable company gives you a box with a regular phone jack on it, and you plug your zap card into that? |
| 23:55.15 | buzkashi | yes |
| 23:55.47 | jeff_phillips | then treat it like a regular phone line... except you might have a hard time getting the cable company to give you a 1004 hz tone test # to call so you might have to dial another telco's test # |
| 23:56.17 | buzkashi | this can be done? dialing another telco test? |
| 23:56.18 | jeff_phillips | My company is getting a cable modem installed in our warehouse and the cable company intends on giving us a bundle deal with unlimited phone to the US & canada. |
| 23:56.37 | jeff_phillips | sure but it's preferred to dial the local one |
| 23:57.01 | buzkashi | Yes I worked with a client that did the same but there was no SLA offered and it was not asked for |
| 23:57.04 | jeff_phillips | I was wondering if there is a way of getting a cable phone service into asterisk digitally without doing the analog conversion |
| 23:57.23 | buzkashi | SIP |
| 23:58.08 | UQlev | jeff: sip/iax origination/termination services |
| 23:58.10 | jeff_phillips | Well I asked our cable company (Charter Communications) if their phone service was VoIP and they insist it isn't and doesn't use the connection's IP bandwdith at all but just uses their cable lines in some other way |
| 23:58.29 | jeff_phillips | but the cable company generally has lied to me about most things so far |
| 23:59.05 | UQlev | jeff: cable company may resell you others voip providers with their commission |
| 23:59.10 | buzkashi | correct man of the clec's while provide hardware that will support voip but what you wrote above is correct |
| 23:59.50 | buzkashi | The main issue is no SLA and I know of companies and organization who go down and it takes 3-4 days to get service restored |