| 00:00.02 | znoG | Hoondie: no increase in SIP traffic when you dial your DID? |
| 00:00.29 | Hoondie | znoG: i am behind NAT.. when they contact me, do they use port 5060? i have forwarded that to my * box.. |
| 00:00.44 | znoG | most likely |
| 00:00.54 | znoG | you'll probably want to run tcpdump on your FW then |
| 00:01.03 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 00:01.31 | Hoondie | my FW is a cisco 800 ADSL Router.. |
| 00:01.41 | Hoondie | 827 to be exact.. |
| 00:02.02 | znoG | are you sure port 5060 is being forwarded properly? |
| 00:02.06 | znoG | and your RTP ports? |
| 00:02.33 | Hoondie | RTP Ports no.. I'll recheck 5060.. TCP or UDP or both? |
| 00:02.38 | znoG | UDP |
| 00:02.51 | znoG | actually both |
| 00:03.03 | znoG | not sure why the TCP is needed, I thought it was all over UDP |
| 00:05.06 | [TK]D-Fender | znoG : SIP over TCP does exist, though not yet in * |
| 00:05.37 | znoG | what benefits does it bring over SIP->UDP ? |
| 00:05.52 | Hoondie | well, i only had TCP port forwarding.... this might fix it |
| 00:05.53 | znoG | i mean, SIP already does ACKs within the protocol even if its over UDP |
| 00:06.21 | [TK]D-Fender | znoG : Better firewall traversal and can ease setup of RTP. |
| 00:06.23 | znoG | just wondering what benefits you get from SIP over TCP |
| 00:06.27 | znoG | oh, yeah, that's true. |
| 00:06.58 | [TK]D-Fender | znoG : And it lets to protocol stack deal with retransmission instead of the SIP client. Far more efficient. |
| 00:07.06 | znoG | yeah, true! |
| 00:07.10 | Hoondie | still not working.. |
| 00:08.05 | [TK]D-Fender | Hoondie : You're setting up * behind NAT, right? |
| 00:08.32 | Hoondie | yep.. |
| 00:08.52 | [TK]D-Fender | Hoondie : Pastebin your sip.conf [general] section. |
| 00:08.52 | dongs | why do people still bother wiht nat |
| 00:10.05 | DrkShdw | nat? because ipv6 isn't widely supported yet |
| 00:10.22 | [TK]D-Fender | ~pb |
| 00:10.27 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 00:10.34 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 00:10.56 | Hoondie | http://pastebin.com/715693 |
| 00:13.29 | Helmchen | Hoondie: do you need the externhost? |
| 00:14.06 | Hoondie | Helmchen: not sure, grabbed it from the sample sip.conf |
| 00:14.29 | Helmchen | Hoondie: comment it out, try again |
| 00:14.42 | Helmchen | Hoondie: after registering your sip-provider should know your ip |
| 00:15.44 | Hoondie | nup.. |
| 00:17.10 | Helmchen | ok, was a try |
| 00:21.25 | [TK]D-Fender | Hoondie : Should have "nat=yes" in [general] as well |
| 00:21.58 | [TK]D-Fender | Hoondie : And you should have 5060,10000-20000 UDP forwarded to you * box |
| 00:22.46 | dongs | heh |
| 00:23.37 | Hoondie | [TK]D-Fender: should i recieve a call without the RTP ports forwarded? but just no voice? |
| 00:23.53 | talljon84 | Anyone have a suggestion for a gsm player for linux that works? |
| 00:24.02 | dongs | sox? |
| 00:24.05 | dongs | (lol) |
| 00:24.21 | [TK]D-Fender | Hoondie : Yes, thats about it... |
| 00:24.58 | SplasPood | Anyone happen to know why Asterisk Realtime would be ignoring my voicemail pin changes? |
| 00:25.15 | [TK]D-Fender | Hoondie : Outgoing calls may work. Keep in mind that some Cisco routers are COMPLETELY assy and are a death-sentence for your situation, and that you may need to change your firmware. |
| 00:25.35 | dongs | or you could just stop using nat |
| 00:25.43 | dongs | i mean whats the point of asterisk behind nat anyway |
| 00:25.43 | Hoondie | [TK]D-Fender: why do you say that? |
| 00:25.48 | dongs | you get one incoming call lol |
| 00:27.25 | file | [TK]D-Fender: go out and party |
| 00:27.42 | [TK]D-Fender | file : Getting dressed as we speak :) |
| 00:27.49 | file | exxxxxxcellent |
| 00:27.49 | [TK]D-Fender | 10 mins or so to go :) |
| 00:28.13 | dongs | [TK] more like trollkore amirite |
| 00:28.17 | SplasPood | dongs: come on man, don't you know... Nat is more secure! :P |
| 00:28.20 | Hoondie | [TK]D-Fender: nup, still nothing.. getting "unassigned number" on my phone when i call |
| 00:28.40 | [TK]D-Fender | dongs :... HUH?! |
| 00:29.35 | dongs | ~google trollkore |
| 00:32.07 | [TK]D-Fender | ok, I'm outta here, later peeps.... |
| 00:32.40 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 00:37.06 | bernardovieira | hi! has anyone out there tried the jitterbuffer for sip channels? (http://bugs.digium.com/view.php?id=3854) |
| 00:38.47 | file | bernardovieira: by the long list of bug notes, I'd say a few |
| 00:40.40 | bernardovieira | file: good point :P |
| 00:40.59 | file | why, what'cha wanna know |
| 00:42.07 | X-Rob | bernardovieira, it's in svn-trunk |
| 00:52.50 | *** join/#asterisk dacleric (n=dacleric@p54823EE6.dip0.t-ipconnect.de) |
| 01:06.19 | Hoondie | does anyone know how to port forward a range of ports on a cisco 800 series router? |
| 01:06.34 | jontow | SplasPood: for voicemail? |
| 01:06.37 | SplasPood | jontow: yes |
| 01:06.47 | jontow | it definitely changes the pass in the db |
| 01:06.54 | SplasPood | it doesn't seem to for me |
| 01:06.59 | jontow | correct GRANT ? |
| 01:07.03 | SplasPood | yup |
| 01:07.12 | jontow | i've been running it in production for >12 months |
| 01:07.19 | SplasPood | mysql? |
| 01:07.23 | *** join/#asterisk tlowe_ (n=tlowe@bgp.terrorist.net) |
| 01:07.31 | jontow | yes |
| 01:07.38 | jontow | now, its nonstandard so i can't say 100% |
| 01:07.44 | jontow | but it definitely works for me. |
| 01:07.44 | jontow | :) |
| 01:07.53 | SplasPood | non standard how? |
| 01:08.00 | jontow | i'm running pre-1.2 trunk |
| 01:08.03 | jontow | res_mysql, not res_odbc |
| 01:08.09 | SplasPood | yea |
| 01:08.13 | SplasPood | pre-1.2 trunk, why? |
| 01:08.13 | jontow | and its actually CVS HEAD, not trunk |
| 01:08.16 | jontow | its pre svn as well |
| 01:08.22 | jontow | because its stable and i have too much shit to do at work :P |
| 01:08.28 | SplasPood | I see |
| 01:08.40 | SplasPood | hrm, well runing any of the stable 1.2 releases from 1.2.8 on up |
| 01:08.44 | SplasPood | it doesn't give me any errors |
| 01:08.47 | SplasPood | it just doesn't change it |
| 01:08.51 | SplasPood | doesn't even issue the UPDATE |
| 01:08.59 | jontow | i'll tell ya what.. im probably gonna upgrade the damned thing sometime this coming week |
| 01:09.09 | jontow | since i've now got some issues with it that have accumulated |
| 01:09.19 | jontow | but theres a significant amount of custom crap i've had to throw in it |
| 01:09.30 | jontow | (voicemail for a medium sized ILEC and a small CLEC) |
| 01:10.03 | anonymouz666 | SIP jitterbuffer is interesting, I will test it tomorrow |
| 01:16.16 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 01:21.50 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.sprint-hsd.net) |
| 01:23.12 | *** join/#asterisk speedwagon (n=FreePBX8@dsl-20-177.cofs.net) |
| 01:23.18 | speedwagon | yello everyone |
| 01:23.28 | surfdue | hi! |
| 01:23.30 | surfdue | need help? |
| 01:23.56 | speedwagon | not really just stopping by to see how everyone is doing tonight. |
| 01:23.58 | surfdue | well I need a life so were even! |
| 01:24.03 | surfdue | lol |
| 01:25.02 | speedwagon | so what is going on tonight? |
| 01:25.24 | surfdue | well |
| 01:25.57 | surfdue | awkward sileces are in the forcast |
| 01:26.14 | littleball | hello, is it possible to disable the Proxy-Authorization for asterisk? |
| 01:26.23 | surfdue | there is a such thing? |
| 01:26.32 | speedwagon | proxy asterisk is not a proxy. |
| 01:26.42 | speedwagon | you need to explain please |
| 01:27.44 | littleball | speedwagon, ser is front of asterisk. Asterisk sends back Proxy-Authorization back to the client. I want to disable it |
| 01:28.18 | speedwagon | in the account put insecure=very |
| 01:28.58 | littleball | when the client send an invite request, the asterisk always send back proxy authroization responds. For some special requirement, i want to disable it |
| 01:31.01 | littleball | speedwagon, thanks. insecure works |
| 01:34.28 | speedwagon | littleball, np |
| 01:35.46 | Hoondie | anyone, can someone help me with inbound (from the pstn) SIP phone calls... Whenever i dial my DID number, i get "Unassigned Number" on my mobile (cell) phone |
| 01:36.09 | Hoondie | it seems to be registering fine |
| 01:36.42 | speedwagon | are you going out via a voip provider? |
| 01:38.16 | Hoondie | yea.. (nodephone.com) |
| 01:38.16 | sevard | with or via |
| 01:38.17 | sevard | :) |
| 01:38.18 | Hoondie | i can call outbound fine |
| 01:39.17 | *** join/#asterisk mrdigital (n=WildStyl@pool-72-81-33-28.phlapa.east.verizon.net) |
| 01:39.21 | speedwagon | set your outbound caller ID correctly |
| 01:39.35 | mrdigital | how many users can be conferenced using softphones + asterisk |
| 01:40.20 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 01:40.35 | Hoondie | speedwagon: it's the inbound call's i am having problems with |
| 01:41.31 | littleball | hi, i am looking for a scalable solution/design architect to relay SIP voice media. Any good suggestions? |
| 01:41.54 | littleball | speedwagon, any suggestion? |
| 01:41.54 | speedwagon | Hoondie, inbound to a Cell phone? |
| 01:42.15 | Hoondie | inbound from a cell phone to my * server |
| 01:42.19 | speedwagon | relay sip voice media??? |
| 01:42.25 | littleball | Hoondie, is it a call forward to your mobile phone? |
| 01:43.24 | Hoondie | littleball: i am having problems calling from the PSTN to my * server via sip |
| 01:43.37 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 01:43.58 | speedwagon | I don't understand |
| 01:44.02 | speedwagon | lost... |
| 01:44.11 | Hoondie | ok.. i'll explain my setup |
| 01:44.33 | speedwagon | littleball, you said you have a SER setup that does most of the hand off for sip accounts. But it stays out of the media path. |
| 01:45.32 | Hoondie | got an asterisk server with a VoIP phone on my lan.. got a cisco router.. my VoIP provider is also my isp.. they have assigned me a phone number and password... |
| 01:45.53 | Hoondie | i can call from my VoIP phone to my cell phone... but not from my cell to my VoIP phone |
| 01:46.05 | littleball | speedwagon, yes. It works. but i want to find out whether my solution is good or not |
| 01:46.47 | littleball | speedwagon, my SET setup only do routing in my design. not as registra even |
| 01:48.21 | littleball | Hoondie, you should ask your ISP whether their SIP incoming really works or not. It is not technical issue but service issue |
| 01:48.44 | littleball | especially, if your SIP phone is behind firewall |
| 01:52.17 | Hoondie | when i call from my land line, i get the phone number is not connected.. does that have anything to do with me at all? |
| 01:52.42 | sevard | <PROTECTED> |
| 01:52.56 | speedwagon | ok do sip debug make the call and see what it says. |
| 01:53.59 | littleball | Hoondie, i think your service provider doesn't provide SIP incoming call properly. |
| 01:54.44 | littleball | at least, your sip phone should ring if your servie provider provide the service. |
| 01:55.29 | Hoondie | speedwagon: nothing happens with i do a sip debug.. i done a tcpdump, and no extra sip data being sent/recieved.. althought sip show registry shows registered.. i do see the occational sip data being passed within tcpdump every time it registers with my provider |
| 01:56.21 | littleball | Hoondie, the reasons could be either (1)your sip provider doesn't have sip incoming call service. (2)they don't solve NAT issue properly if your SIP phone behind NAT. |
| 01:57.16 | speedwagon | Hoondie, if your not getting anything in to your system Then it's either your FW blocking ports or there setup. |
| 01:57.29 | Hoondie | littleball: they say they do client-server communications not peer-peer to solve any NAT issues.. they advertise that they have the incomming calls sip service |
| 01:58.23 | speedwagon | Hoondie, your asterisk registering to them is Client-Server |
| 01:58.30 | littleball | Hoondie, i am trying to provide the same services to my customers. NAT is an issue. Even NAT solved just now discussed with speedwagon, the scalability could be another issue |
| 01:59.18 | speedwagon | Hoondie, make sure you start with your sip.conf with setting up your externip= and localnet= settings |
| 01:59.31 | speedwagon | also your settings for the account should have nat=yes in it's settings |
| 01:59.34 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 01:59.45 | speedwagon | ~nat |
| 01:59.52 | jbot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 01:59.52 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:00.00 | AlexCTI | Hi, everyone, I have a peer connected called enrique, but that users is not created on my system |
| 02:00.33 | speedwagon | AlexCTI, then you are allowing guess accounts into your setup |
| 02:01.26 | AlexCTI | where can i disable that? |
| 02:02.15 | speedwagon | AlexCTI, your setup in the sip.conf put at the top context=trap, make a context in the extensions.conf with [trap] then exten => s,1,hangup |
| 02:02.21 | littleball | Hoondie, is the service provider Australian company? |
| 02:02.57 | AlexCTI | ok, |
| 02:03.00 | speedwagon | or congestion or you can send them to an ivr hell.. your choice |
| 02:04.04 | littleball | any ser expert around here? |
| 02:05.06 | Hoondie | littleball: yep.. internode |
| 02:05.13 | AlexCTI | speedwagon, context=trap must be under [general] right? |
| 02:05.26 | speedwagon | AlexCTI, yes |
| 02:06.23 | Hoondie | can someone help me test my * server?? need someone with a soft phone (or hard phone) to connect to my asterisk server and call me.. need to test the port forward on my cisco.. |
| 02:07.43 | speedwagon | Hoondie, give me about 5 minutes or so. I need to change my baby's diaper.. |
| 02:08.01 | Hoondie | speedwagon: cheers.. |
| 02:08.52 | speedwagon | Hoondie, in the mean time pm me the info so I can put it into xlite |
| 02:15.39 | AlexCTI | speedwagon, Thanks |
| 02:16.07 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 02:16.25 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 02:21.17 | *** join/#asterisk Eigh (n=jason@207.179.67.67) |
| 02:35.36 | littleball | hello, anyone has experience of integration of ser and *? |
| 02:35.47 | littleball | and can explain how does it work? |
| 02:36.51 | *** join/#asterisk JoseBravo (n=JoseBrav@cm4126.uniweb.net.co) |
| 02:37.29 | JoseBravo | Where can I buy cheap phones online? |
| 02:37.46 | DrkShdw | <PROTECTED> |
| 02:37.46 | littleball | JoseBravo, how much is cheap? |
| 02:38.57 | JoseBravo | No idea, what do you recommends? |
| 02:40.15 | littleball | 30USD? 50 USD or 80USD? |
| 02:40.29 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 02:41.23 | Qwell | 12USD |
| 02:41.31 | coppice | US$1 |
| 02:41.42 | Druken | who is good for cdn termination ? |
| 02:41.49 | littleball | i can only get 30USD. which is cheapest i got |
| 02:44.11 | *** join/#asterisk copantl (n=galel@207.13.77.32) |
| 02:44.43 | copantl | hi |
| 02:44.48 | copantl | any body home? |
| 02:45.37 | copantl | hello? |
| 02:45.55 | surfdue | no |
| 02:46.27 | Hoondie | is it possible to intergrate our current phone system (LG) and the voicemail function of *? |
| 02:46.39 | surfdue | why not |
| 02:46.56 | surfdue | do you want to hook the phones up to the asterisk system? |
| 02:47.06 | surfdue | or you mean have them dial a number? |
| 02:47.06 | copantl | any body know what happen with the a2billing guys? |
| 02:47.09 | surfdue | that goes to the system |
| 02:47.18 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 02:47.24 | surfdue | copantl, whats a2billing ? |
| 02:47.53 | copantl | areski billing system for asterisk |
| 02:48.07 | Hoondie | surfdue: if a call is not answered it goes through to voicemail... our phone system at the moment is not voip.. |
| 02:48.39 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 02:48.42 | surfdue | Hoondie, ok so you want your psnt system to failover to an asterisk voicemail box |
| 02:48.55 | surfdue | copantl, do they have a channel possible? |
| 02:48.56 | Hoondie | yea |
| 02:49.17 | surfdue | Hoondie, I wouldnt be exactly sure directly I think you can handle the system by setting up an asterisk box with phone hardware |
| 02:49.23 | copantl | yes a2billing , but since a week ago nobody anwer |
| 02:49.37 | surfdue | Hoondie, im sure you can hack your system to redirect to a phone number that is on an asterisk box |
| 02:50.22 | JoseBravo | littleball can I found phones starting in 30? |
| 02:50.26 | Hoondie | surfdue: how would the asterisk box know what mailbox the call is ment for? |
| 02:51.22 | littleball | JoseBravo, how many you want? |
| 02:51.24 | Qwell | Hoondie: callerid |
| 02:51.35 | Qwell | erm |
| 02:51.42 | Qwell | no, that wouldn't work, heh |
| 02:51.57 | littleball | JoseBravo, give me your email address. i can send to you. |
| 02:51.59 | JoseBravo | I want to buy 3 |
| 02:52.12 | JoseBravo | For test |
| 02:52.24 | surfdue | littleball, you cell phones for $30? |
| 02:52.25 | surfdue | voip? |
| 02:52.33 | littleball | JoseBravo,u need to pay the post fee. :-) |
| 02:52.54 | littleball | k |
| 02:52.55 | copantl | surfdue: www.asteriska2billing.org |
| 02:53.15 | copantl | <PROTECTED> |
| 02:53.24 | surfdue | cool |
| 02:53.29 | surfdue | im making a biling system |
| 02:53.30 | surfdue | lol |
| 02:53.53 | littleball | surfdue, sip phones |
| 02:54.13 | surfdue | littleball, you mean hardware or software |
| 02:54.25 | littleball | i also have mobile sip/GSM phone, SIP phone. hardware of course |
| 02:54.47 | surfdue | littleball, have model numbers or other info we mainly buy only cisco |
| 02:55.05 | surfdue | littleball, we simply have several cisco pap2's |
| 02:55.17 | Qwell | linksys |
| 02:55.20 | asterboy | kick fucking ass! |
| 02:55.21 | Qwell | not cisco |
| 02:55.30 | Qwell | HUGE difference |
| 02:55.31 | asterboy | Oilers shut out the Americans |
| 02:55.33 | Qwell | file: omg, ignoring me? |
| 02:55.37 | file | Qwell: omg no |
| 02:55.41 | Qwell | omg lies |
| 02:55.47 | asterboy | Carolina 0 |
| 02:55.49 | Qwell | :D |
| 02:55.49 | file | omg messages from Qwell |
| 02:55.52 | Qwell | omg! |
| 02:55.58 | asterboy | big fat goose egg |
| 02:55.58 | JoseBravo | littleball did you send the info |
| 02:56.23 | asterboy | Edmonton is about to errupt into a ball of party fury |
| 02:56.28 | littleball | i will ask my collegue to send to you tomorrow. ok? I am at home |
| 02:56.29 | surfdue | Qwell, no its cisco says cisco right on it not linksys |
| 02:57.28 | file | it's Linksys |
| 02:57.33 | Sedorox | surfdue: they say Linksys, product of Cisco Systems |
| 02:57.36 | Sedorox | or something like that |
| 02:57.39 | Sedorox | the Pap2's are Linksyts |
| 02:57.42 | Sedorox | which is a cisco company |
| 02:57.50 | file | it's an SPA2000 derivative design, with a few parts different (cheaper parts) |
| 02:58.11 | littleball | anyway, my friend manufacture phones in China. So i can get cheap one. Anyway, i am using and tested it on both ser and asterisk |
| 02:58.20 | surfdue | Sedorox, mine says cisco sipura? |
| 02:58.26 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:58.37 | file | I have a PAP2-NA right beside me |
| 02:58.49 | file | CIsco Systems logo, but "Linksys Phone Adapter" :D |
| 02:58.50 | surfdue | littleball, do you have any pics |
| 02:59.01 | surfdue | file no its a sipura sorry |
| 02:59.06 | surfdue | file im not talking about the one in my house :P |
| 02:59.06 | Sedorox | ahh right.. cisco bought sipura too |
| 02:59.12 | Qwell | surfdue: then it isn't a pap2.. |
| 02:59.13 | littleball | of course, yes. i am using myself. |
| 02:59.17 | Qwell | Sedorox: no, linksys bought sipura ;) |
| 02:59.21 | surfdue | Qwell, yes I appologized sir. |
| 02:59.26 | Sedorox | so basically cisco :p |
| 02:59.34 | Qwell | Sedorox: no, it's run seperately, afaik |
| 02:59.40 | Sedorox | ah ok |
| 03:00.55 | alephcom | asterboy: Good for us. :-) |
| 03:02.34 | littleball | Qwell, can u give me some help on ser works with *? |
| 03:04.23 | JoseBravo | I need a simple billing system, I think astbill have many things.. |
| 03:04.43 | JoseBravo | littleball send me the info about yours phones |
| 03:05.34 | littleball | ok. send to you tomorrow. |
| 03:07.04 | JoseBravo | Anoye know a simple software for billing? |
| 03:17.14 | alephcom | JoseBravo: How simple. I have a fairly simple stuff at www.astpp.org |
| 03:27.30 | *** join/#asterisk DonX (i=don@gw.sparkhosting.net) |
| 03:27.56 | DonX | anyone have a BRI line they can help me test something with? |
| 03:28.05 | JoseBravo | alephcom the demo password was changed |
| 03:28.41 | alephcom | JoseBravo: abcde for now. I'll fix it right soon though |
| 03:33.13 | *** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 03:33.37 | Hoondie | anyone able to use a softphone to call me.. connecting to my * box externally to test my port forwards? |
| 03:35.01 | DrkShdw | hoondie, what protocal? |
| 03:35.06 | Hoondie | SIP |
| 03:35.48 | DrkShdw | PM me extension info. my mic is broken though. but if yer just checking port forwarding, shouldn't be a concern |
| 03:37.56 | *** join/#asterisk oneman (n=oneman@ip68-230-208-113.rd.hr.cox.net) |
| 03:38.04 | oneman | hello |
| 03:38.22 | oneman | app_voicemail.c:6269 load_config: Failed to load configuration file. Module not activated. |
| 03:38.34 | oneman | Whats going on? How do I activate the voicemail module? |
| 03:38.40 | JoseBravo | alephcom user admin password abcde? |
| 03:38.40 | Qwell | by giving it a valid config |
| 03:38.50 | oneman | hrm |
| 03:38.51 | oneman | ok |
| 03:38.53 | oneman | thanks |
| 03:40.26 | oneman | Im not sure whats wrong with the config.. but thanks for telling me that.. |
| 03:40.43 | CunningPike | oneman: pastebin your voicemail.conf |
| 03:40.45 | CunningPike | ~pb |
| 03:40.53 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 03:40.53 | alephcom | JoseBravo: I'll pm you |
| 03:41.39 | oneman | http://pastebin.com/715890 |
| 03:41.46 | oneman | its basicly stock cept the last few liens |
| 03:41.48 | oneman | lines.. |
| 03:42.27 | oneman | if i remove the last few lines it still bombs tho |
| 03:44.22 | oneman | hrm |
| 03:44.29 | oneman | I think it might have been a permissions issue |
| 03:44.45 | oneman | I am truly sorry for bothering if it was... |
| 03:45.01 | oneman | I do have a much more difficult problem though :P |
| 03:45.01 | sevard | there ought to be a channel to flood to and a bot pastebins it all for you |
| 03:45.15 | oneman | compiling zaptel module for ubuntu dapper.... |
| 03:45.31 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:49.12 | sparkleytone | oneman: works fine for me |
| 03:49.36 | oneman | sparkleytone: what works? |
| 03:49.44 | oneman | my voicemail is now working :D |
| 03:49.48 | sparkleytone | zaptel module |
| 03:50.03 | oneman | what steps did you follow to build it? |
| 03:50.11 | oneman | or is there a pre built package for dapper somewhere? |
| 03:50.17 | sparkleytone | no prebuilt package |
| 03:50.24 | sparkleytone | i just downloaded all the sources and installed them |
| 03:50.33 | sparkleytone | i did, however, also roll my own kernel |
| 03:50.50 | sparkleytone | i've heard that there are ubuntu zaptel issues |
| 03:51.02 | sparkleytone | but it sounds like its a problem with the prebuilt kernel |
| 03:52.09 | *** join/#asterisk bmg505 (n=leon@c1-235-14.rndf.isadsl.co.za) |
| 03:52.25 | oneman | aye |
| 03:52.41 | oneman | I have questions about doing text to speech |
| 03:52.53 | oneman | is festival usable or do I need to buy a voice from that company? |
| 03:53.16 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
| 03:55.51 | sevard | useable doesn't mean awesome quality. |
| 04:05.53 | speedwagon | Good night folks. It's bed time. |
| 04:17.09 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 04:26.46 | *** join/#asterisk aclark78 (n=aclark@cpe-70-116-103-202.houston.res.rr.com) |
| 04:27.25 | aclark78 | hello? anyone here? |
| 04:27.28 | Qwell | no |
| 04:27.33 | aclark78 | hehe |
| 04:27.53 | aclark78 | i'm planning an asterisk setup and am evaluating phones and had a question about polycom sip phones |
| 04:28.25 | coppice | "useable doesn't mean awesome quality" sounds like a slogan for much that is VoIP :-) |
| 04:28.33 | aclark78 | is it possible to control the phones (ie. hangup, answer, transfer, etc) from a computer |
| 04:28.47 | aclark78 | through a http connection or something else? |
| 04:29.17 | aclark78 | specifically the 501 |
| 04:30.03 | aclark78 | i know snom supports this but i haven't seen anything specific to the polycoms, have i missed it or does it not exist? |
| 04:31.52 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 04:32.34 | *** join/#asterisk chino (n=daquino@c-68-84-57-212.hsd1.nj.comcast.net) |
| 04:32.58 | chino | can i set the reregister attempt interval ? |
| 04:36.00 | *** part/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 04:36.39 | chino | i just want to make it reregister in smaller intervals |
| 04:44.22 | *** part/#asterisk aclark78 (n=aclark@cpe-70-116-103-202.houston.res.rr.com) |
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| 05:06.33 | *** part/#asterisk chino (n=daquino@c-68-84-57-212.hsd1.nj.comcast.net) |
| 05:09.09 | *** part/#asterisk dgilmore (n=dennis@fedora/dgilmore) |
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| 05:34.43 | stephane_ | jour |
| 05:42.47 | sevard | my upstairs neighbors listen to the shittiest rap |
| 05:49.50 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
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| 06:10.13 | P-NuT | Hi all. |
| 06:11.15 | P-NuT | If I'm trying to call an extension which is configured to dial VMB 201, but it keeps trying to dial 8500. Would you guys know of any reason? |
| 06:11.39 | P-NuT | I think it may have something to do with the contexts but I'm not sure.. |
| 06:13.02 | SheriF_WorK | i have a problem: asterisk is registered as a client to another SIP server... and when i try to call another device registering to that SIP server from asterisk i got Redirecting then asterisk try to dial LOCAL channel ... which is wrong .. any idea where the problem might be ? |
| 07:03.11 | *** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
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| 08:15.56 | gcarrillog | hi |
| 08:16.12 | gcarrillog | i was had asterisk on my fbsd 6.0 |
| 08:16.41 | gcarrillog | but today i was deinstalled them |
| 08:17.26 | gcarrillog | now i have a problem when i try install of port |
| 08:17.55 | gcarrillog | chan_zap.c: In function `pri_dchannel': |
| 08:17.55 | gcarrillog | chan_zap.c:9077: error: structure has no member named `call' |
| 08:19.45 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 08:22.39 | *** join/#asterisk Pkunk (n=Pkunkage@125.22.145.158) |
| 08:23.17 | Pkunk | whats a good and cheap sip phone ? one that won't break easy |
| 08:30.11 | SheriF_WorK | Pkunk: GrandStream is good BT100 but the LCD shows numbers only |
| 08:32.31 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 08:36.14 | *** join/#asterisk neilbags (n=neilbags@c211-31-32-206.randw1.nsw.optusnet.com.au) |
| 08:36.25 | Pkunk | SheriF_WorK: if i don't need LED , is the Linksys PAP2 a good option ? |
| 08:36.28 | neilbags | howdy |
| 08:37.10 | SheriF_WorK | Pkunk: never used Linksys i have also GipTell and it's wonderfull |
| 08:37.52 | Pkunk | i was thinking of something like pap2 or sipura spa-2002 since analog phones are cheaper to replace |
| 08:38.06 | neilbags | Pkunk: I have a PAP2, it works fine ... what functionality were you after (or worried about)? |
| 08:41.08 | Pkunk | neilbags: use it for outgoing calls thro asterisk gateway |
| 08:43.39 | neilbags | Pkunk: yeah it works fine, i also have 8 of them in an office going through an asterisk gateway. no problems with incoming or outgoing calls. The only thing if changed from the default config is the dial plan |
| 08:44.06 | neilbags | besides ip, login, password etc obviously |
| 08:44.28 | Pkunk | k so i guess itss better maintainence wise to use analog phone thro pap2 instead of a proper sip hardphone ? |
| 08:45.05 | neilbags | well they already had a few cordless phones around so they just bought one for each employee |
| 08:45.33 | SheriF_WorK | guys dose asterisk support redirecting ? |
| 08:46.10 | SheriF_WorK | http://lists.digium.com/pipermail/asterisk-users/2006-April/146983.html |
| 08:49.17 | neilbags | SheriF_WorK: well that message is from april 5 thi year, so if the changlog from then doesn't say its been implemented then it hasn't been |
| 08:49.56 | SheriF_WorK | neilbags: but thats very strange .. i mean asterisk as a client for another SIP server should support redirecting . |
| 08:50.59 | neilbags | <shrugs> |
| 08:54.57 | *** join/#asterisk LoneShadow (n=a@c-67-188-235-220.hsd1.ca.comcast.net) |
| 08:59.58 | SheriF_WorK | neilbags: .. |
| 09:01.44 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-178-54.38-151.net24.it) |
| 09:02.28 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 09:02.43 | BladeRunner05 | Hi all, I setup my voip account but what I try to call asterisk the service provider answer that the number not exist and asterisk report this erro: retrans_pkt maximum retries exceeded on transmission |
| 09:04.42 | neilbags | BladeRunner05: post your sip.conf to pastebin.com and blank out the passwords |
| 09:05.39 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 09:17.33 | BladeRunner05 | neilbags: this is my sip.conf http://pastebin.com/716095 |
| 09:21.09 | neilbags | i dunno, looks ok to me ... did you provider tell you what settings to use, or did you use a website or figure it out for yourself? |
| 09:22.00 | BladeRunner05 | neilbags: the setting are ok, I can call outside, its ok |
| 09:44.16 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 09:54.00 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 10:00.17 | nettie | det |
| 10:04.03 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 10:04.03 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:13.26 | geoffl | Hi. I'm an Asterisk newbie - so please excuse the basic question. Is there a way to ring all the extensions in a context without listing them (e.g. in Dial()) |
| 10:17.15 | SheriF_WorK | geoffl: i think that what ring group do ? |
| 10:17.33 | SheriF_WorK | u tell Dial( ring group ) .. but i'm not sure ;-) that what i think about . |
| 10:18.12 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 10:18.35 | geoffl | SheriF_WorK: I suspect that I'm missing something basic - because the only way that I know to create a ring group is to define the group in a variable: |
| 10:18.56 | geoffl | myringgroup=101&102&103 |
| 10:19.08 | geoffl | Dial(${myringgroup}) |
| 10:19.22 | geoffl | How do you define ring groups the proper way? |
| 10:24.28 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 10:25.10 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 10:27.32 | *** join/#asterisk JDofFED (n=jd@c-69-243-134-140.hsd1.in.comcast.net) |
| 10:47.01 | JDofFED | I have a question, how can I tell if I am using ztdummy or not./ |
| 10:47.16 | SheriF_WorK | JDofFED: do u have a Zaptel Card ? |
| 10:47.28 | JDofFED | no |
| 10:47.32 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 10:47.39 | SheriF_WorK | JDofFED: ok then lsmod | grep ztdummy |
| 10:47.41 | SheriF_WorK | as root |
| 10:47.58 | JDofFED | ztdummy 3796 0 |
| 10:47.59 | JDofFED | zaptel 205700 1 ztdummy |
| 10:48.09 | JDofFED | thats the output |
| 10:48.18 | SheriF_WorK | JDofFED: then the module is loaded |
| 10:48.32 | P-NuT | Hi all, Does anybody use Grandstream BT101's |
| 10:48.42 | P-NuT | ? |
| 10:48.42 | JDofFED | okay, next question, one a caller is put in the queue, the music plays for about a second and quits |
| 10:48.43 | SheriF_WorK | P-NuT: BT100 here |
| 10:48.52 | P-NuT | How do you find them> |
| 10:48.53 | P-NuT | ? |
| 10:48.58 | P-NuT | are they ok? |
| 10:49.19 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 10:49.30 | vivek | hello all |
| 10:49.39 | SheriF_WorK | JDofFED: don't know :-s check the configurations i don't use queue |
| 10:49.48 | JDofFED | well even on hold |
| 10:49.58 | SheriF_WorK | P-NuT: fine just what bothers me that the caller ID don't come with names the LCD only showes numbers :-s |
| 10:50.10 | P-NuT | really? |
| 10:50.21 | P-NuT | There's no way around that? |
| 10:50.22 | JDofFED | SheriF_WorK: it does that on both, should I complie mpg123 |
| 10:50.26 | SheriF_WorK | P-NuT: that was my problem and that what i have been told ;-) |
| 10:50.36 | SheriF_WorK | JDofFED: not sure .. |
| 10:50.37 | P-NuT | hmm.. |
| 10:50.43 | SheriF_WorK | P-NuT: no i don't think there is no way around.. |
| 10:50.55 | P-NuT | but the voice quality is ok, and it does have the message light? |
| 10:50.56 | SheriF_WorK | P-NuT: i'm using Giptel and it's fine not bad . |
| 10:51.09 | P-NuT | hmm.. |
| 10:51.10 | SheriF_WorK | P-NuT: yes the LCD is blinking Blue when u have Message |
| 10:51.12 | P-NuT | interesting.. |
| 10:51.16 | P-NuT | awesome. |
| 10:51.22 | P-NuT | thanks a bunch guys! |
| 10:51.33 | SheriF_WorK | P-NuT: the giptel has a small red led keep blinking |
| 10:51.58 | SheriF_WorK | JDofFED: but if u listen to the music even for one sec i think that the mpg123 is working fine. |
| 10:52.16 | JDofFED | SheriF_WorK: hmm, then I wonder why it cuts out like that |
| 10:52.19 | vivek | I have got some basic questions to ask you people. I am located in a different city and my mom is located in a different city. My mom is not too happy around a computer. Can i run asterisk on my server in college and get a voip wifi hardware and wifi router connected to the internet and talk over to her ? I can't place any hardware at her place if something goes wrong i can't fix it. The isp at her place can't give her a fixed ip either. |
| 10:53.01 | JDofFED | SheriF_WorK: the CLI doesn't show anything werid too, it just says starting to play music on hold (default) |
| 10:53.54 | SheriF_WorK | vivek: yes u can get a SIP wifi phone to ur mom and a wireless connection .. in ur asterisk configruations u create a SIP ext. to ur mom and host = dynamic so when the ISP change the ip it still able to register to ur asterisk and u can use softphone to log into asterisk " in ur city " so u can call ur mom in any place has an open wifi connection . |
| 10:54.09 | SheriF_WorK | JDofFED: and after taht ? what the other hand listen ? |
| 10:54.18 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 10:54.39 | JDofFED | SheriF_WorK: i don't understand your question |
| 10:54.40 | SheriF_WorK | JDofFED: and turn on the debug at /etc/asterisk/logger.conf |
| 10:54.51 | SheriF_WorK | JDofFED: check /var/log/asterisk/full |
| 10:55.01 | JDofFED | SheriF_WorK: ok |
| 10:55.12 | SheriF_WorK | JDofFED: i mean the other end listen for the MOH for 1 sec . then what he listens too ? |
| 10:55.25 | JDofFED | SheriF_WorK: nothing, it goes blank |
| 10:55.43 | SheriF_WorK | JDofFED: then check the debugging . |
| 10:56.03 | vivek | Can you suggest some reasonably priced wifi phone hardware ? |
| 10:56.43 | JDofFED | SheriF_WorK: Jun 18 06:56:21 VERBOSE[3771] logger.c: -- Started music on hold, class 'default', on SIP/800-a5cc |
| 10:56.57 | JDofFED | SheriF_WorK: that is the last line in the log |
| 10:57.42 | SheriF_WorK | vivek: don't know i liked a linksys with colored screen were around 250 USD :P but u can fight alot cheaper as i think . |
| 10:57.59 | SheriF_WorK | JDofFED: what music do u use ? |
| 10:58.11 | SheriF_WorK | JDofFED: try to use the music that comes with asterisk |
| 10:58.19 | vivek | SheriF_WorK: I need to be on a fixed ip address though right ? |
| 10:58.36 | JDofFED | SheriF_WorK: i use mp3's that have been changed to the proper type |
| 10:59.12 | SheriF_WorK | vivek: didn't work with it .. but sure it can support DHCP " the phone " .. the idea is how u'll configuer asterisk to hand the dynamic public IP " host = dynamic " |
| 10:59.27 | SheriF_WorK | JDofFED: can't tell :-s i'm out of ideas |
| 11:00.07 | JDofFED | SheriF_WorK: thanks for your help thougt, I will try after I get up later today |
| 11:00.21 | vivek | SheriF_WorK: er i meant one end has to be static right ? or that's also not a problem with something like a dyndns service ? |
| 11:01.45 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 11:03.12 | Mw3 | hi. i have a problem: extension 'i' is not working on our pri. i haven exten => i,1,Answer etc, etc ... in my pri-incoming context but asterisk logs this rather than executing extension 'i': -- Extension '218' in context 'pri-incoming' from '' does not exist. Rejecting call on channel 0/27, span 1 |
| 11:03.22 | Mw3 | does anybody know the reason? |
| 11:07.07 | X-Rob | i doesn't work that way |
| 11:07.13 | X-Rob | you want _X. |
| 11:08.45 | Mw3 | hm, since when ? i found the example on voip-info.org about setting PRI_CAUSE and hanging up for invalid extensions |
| 11:11.22 | Mw3 | _X. really works. thank you. but why i does not? |
| 11:14.09 | SheriF_WorK | vivek: don't know really |
| 11:14.34 | SheriF_WorK | vivek: i think asterisk should be static |
| 11:14.50 | SheriF_WorK | vivek: and u'll need to fix the NAT problems if u have NAT or firewall |
| 11:20.11 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 11:22.26 | SheriF_WorK | what do i need to make asterisk works with h323 ? |
| 11:23.28 | P-NuT | Hi all, this may be a bit off topic, but does anyone know where to get spare parts for cisco phones? I need a backstand ffor my 7905 phone... |
| 11:40.42 | SheriF_WorK | what do i need to make asterisk works with h323 ? can someone point me to the headlines ? |
| 11:40.43 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 11:41.51 | RoyK | SheriF_WorK: chen_ooh323 from -addons or chan_h323 |
| 11:42.16 | RoyK | and a good amount of luck, nice gods, nice weather etc |
| 11:43.57 | SheriF_WorK | RoyK: yes exactly i need all this cuz my asterisk 1.0.X !!! |
| 11:44.15 | SheriF_WorK | i'll upgrade now ! it's over i'm sick of this CVS version that been more than 1 year old :'( |
| 11:44.48 | RoyK | 1.2.9.1 is nice |
| 11:45.07 | SheriF_WorK | RoyK: i don't need OpenH323 to make chan_h323.so works ? |
| 11:45.40 | RoyK | for chan_h323, you need openh323 and pwlib |
| 11:45.49 | RoyK | i don't think you need that for chan_ooh323 from -addons |
| 11:46.00 | SheriF_WorK | RoyK: one more thing i'm really haveing problem .. asterisk is a client for another SIP server . ... asterisk dosn't support 302 redirecting ??? i got errors and asterisk try to call Local/exten@context which is not exists as local |
| 11:46.02 | RoyK | chan_ooh323 doesn't support video, though |
| 11:46.40 | SheriF_WorK | RoyK: hum may be i'll try to install binaries ;-) if binaries i'll use 1.2.7.1 as i think that shiped with mandrvia 2006... |
| 11:47.54 | RoyK | erm |
| 11:47.55 | SheriF_WorK | hum no i think i'll compile :-D |
| 11:47.55 | RoyK | don't |
| 11:47.57 | SheriF_WorK | it's 1.1 |
| 11:48.04 | RoyK | get the 1.2.9.1 source |
| 11:48.09 | RoyK | get -addons |
| 11:48.09 | RoyK | compile and install |
| 11:48.15 | SheriF_WorK | RoyK: any idea bout this redericting issue ? |
| 11:48.21 | RoyK | no |
| 11:48.23 | SheriF_WorK | http://lists.digium.com/pipermail/asterisk-users/2006-April/146983.html |
| 11:48.38 | SheriF_WorK | i'm soo depressed ! how come asterisk as a client doesn't support redireciting :( |
| 11:58.05 | SheriF_WorK | RoyK: chan_ooh323 is much better than chan_h323 ? |
| 12:00.44 | dlynes_home | SheriF_WorK: chan_ooh323c was commissioned by Digium |
| 12:01.08 | dlynes_home | SheriF_WorK: it was actually paid for |
| 12:01.33 | dlynes_home | SheriF_WorK: also chan_ooh323c doesn't put you through pwlib/openh323 dependency hell |
| 12:03.41 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 12:03.45 | l-fy | ehlo #asterisk |
| 12:03.53 | l-fy | i still need some help with iax |
| 12:03.56 | l-fy | please |
| 12:12.01 | SheriF_WorK | dlynes_home: ok then i'll try ooh323 |
| 12:15.37 | l-fy | SheriF_WorK > what are you trying to do? |
| 12:15.55 | SheriF_WorK | l-fy: using h323 with asterisk |
| 12:16.06 | l-fy | a ok |
| 12:16.17 | l-fy | i was thinking that you do like all the others |
| 12:16.25 | SheriF_WorK | what the others do ? |
| 12:16.28 | l-fy | using asterisk with sip and using yate for sip-h323 transaction |
| 12:16.35 | l-fy | translation |
| 12:17.18 | SheriF_WorK | l-fy: may be i'll use yate too but i don't know anything about h323 at the moment still reading in voip-info |
| 12:20.36 | *** join/#asterisk Flosoft (n=admin@213.219.186.227.adslpower.by.edpnet.be) |
| 12:21.55 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 12:29.39 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 12:30.35 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 12:34.09 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 12:35.24 | l-fy | hey robin_sz |
| 12:35.27 | l-fy | hey RoyK |
| 12:38.05 | RoyK | hey, fd |
| 12:41.07 | l-fy | why my asterisk can't carry more than 200 calls from sip to iax? |
| 12:41.58 | *** join/#asterisk Wildy (n=tsolias@c220-239-121-25.dandn1.vic.optusnet.com.au) |
| 12:42.05 | Wildy | hello |
| 12:42.28 | Wildy | i need some advice can anyone help regarding 7970 on asterisk (SIP) |
| 12:42.30 | l-fy | hi Wildy |
| 12:42.34 | l-fy | shooy |
| 12:42.35 | l-fy | shoot |
| 12:42.37 | Wildy | hi l-fy |
| 12:42.40 | Wildy | okat |
| 12:42.44 | Wildy | i have a 7970 |
| 12:42.50 | Wildy | i havent installed SIP yet |
| 12:43.06 | Wildy | i have asterisk.. |
| 12:43.31 | Wildy | alot of people are saying they havent got SIP working on 7970 authenticating to asterisk? |
| 12:43.43 | l-fy | really? |
| 12:43.45 | l-fy | i have no idea |
| 12:43.54 | l-fy | probably yet another sip cisco stack |
| 12:44.01 | Wildy | ahh okay. |
| 12:44.16 | l-fy | cisco is chaining the sip versions quite often, and asterisk is not the best example of sip implementation either :) |
| 12:44.52 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 12:44.54 | Wildy | hehe |
| 12:44.57 | Wildy | yeah i know what ya mean |
| 12:45.11 | ghenry | anyone used or have opinions on pound-key distro? |
| 12:46.10 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 12:48.14 | Greek-Boy | l-fy; i thought asterisk was pretty good when it came to SIP? |
| 12:48.30 | l-fy | Greek-Boy > define sip |
| 12:48.32 | l-fy | define good |
| 12:48.43 | Greek-Boy | stable |
| 12:48.44 | l-fy | and i'm not the best person to talk about how good asterisk is with sip |
| 12:48.45 | Greek-Boy | consistent |
| 12:48.49 | l-fy | right |
| 12:48.54 | l-fy | whatever you say is true |
| 12:49.14 | [Airwolf] | l-fy, what do you then prefer to have as a sip server ? |
| 12:49.59 | Mw3 | is it possible that caller id is not available on PRI by default (i mean i should order from the telco and maybe pay for it). i'm runnging pri debug span 1 on my pri, but i can't see the calling number information only the called number. |
| 12:50.52 | *** join/#asterisk ]Xman[ (n=ciccio@host4-176-static.113-81-b.business.telecomitalia.it) |
| 12:50.56 | ]Xman[ | hi |
| 12:50.58 | l-fy | [Airwolf] > i can't talk about that, since i did the sip stack design of the software i use :)) |
| 12:51.27 | ]Xman[ | can i use any command for read the sip response? |
| 12:51.35 | ]Xman[ | in extension.conf? |
| 12:53.11 | ]Xman[ | i need us a if condition for do not disturb response |
| 12:53.18 | ]Xman[ | any1 can help me? |
| 12:53.26 | RoyK | l-fy: see my notes on asterisk scalability at http://sitsotd.org |
| 12:55.37 | l-fy | RoyK > how can i grow iax scalability |
| 12:57.01 | RoyK | benchmark the system first..... |
| 12:57.12 | l-fy | RoyK > some of the infos from that document are not entire acurate |
| 12:57.17 | l-fy | i did |
| 12:57.19 | RoyK | i don't know if the iax problem is the same as with rtp, which was my testing |
| 12:57.23 | l-fy | i can't go futher than 200 calls |
| 12:57.31 | RoyK | what part is "not entierly accurate"? |
| 12:58.32 | l-fy | i'm trying to figure out now |
| 12:58.54 | RoyK | well, please help me to fill in if you have some additions |
| 12:59.11 | l-fy | i 'm reading now and trying to put things together |
| 12:59.12 | l-fy | anyway |
| 12:59.17 | RoyK | the testing was done on several different computers, distros etc, and the results were the same |
| 12:59.22 | l-fy | how can i convince asterisk to handle more iax calls without trunking? |
| 12:59.24 | RoyK | this was sip/sip bridging |
| 12:59.39 | RoyK | why not trunking? |
| 12:59.48 | l-fy | btw |
| 12:59.51 | l-fy | about sip nat |
| 12:59.53 | RoyK | trunking will allow for far less user/kernel context switches |
| 12:59.58 | l-fy | did you saw my sip article on freshmeat? |
| 13:00.07 | l-fy | i don't want that |
| 13:00.15 | l-fy | i want to run tests against y |
| 13:00.27 | l-fy | and since y can handle, * should also be able to do it |
| 13:00.30 | RoyK | didn't see the article.... |
| 13:00.38 | RoyK | well, obviously |
| 13:00.49 | RoyK | but asterisk isn't really built to scale very well |
| 13:00.59 | RoyK | just a sec |
| 13:01.04 | l-fy | http://freshmeat.net/articles/view/2079/ |
| 13:01.15 | l-fy | that's not for me to judge |
| 13:01.18 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
| 13:01.27 | RoyK | http://lists.digium.com/pipermail/asterisk-dev/2006-March/019361.html |
| 13:02.43 | l-fy | RoyK > heavy switching in * if i remeber corectly is in * engine |
| 13:02.48 | l-fy | not in the modules |
| 13:02.59 | l-fy | modules are pretty similar more or less with y modules |
| 13:03.16 | l-fy | just that y has a few tricks more to handle more calls |
| 13:03.35 | ]Xman[ | any italian? |
| 13:03.46 | RoyK | with sip/rtp bridging, we saw a very high system load |
| 13:03.50 | RoyK | system time, not user |
| 13:04.23 | l-fy | is still in the core |
| 13:04.41 | RoyK | well, it was kernel, not asterisk spending time |
| 13:04.42 | l-fy | you have to trust me in this one since is too difficult to explain, but in y was avoided by design |
| 13:04.49 | RoyK | anyway |
| 13:04.53 | l-fy | but * is giving the comand |
| 13:04.59 | RoyK | what sort of load are you seing? |
| 13:05.00 | RoyK | seeing |
| 13:05.09 | l-fy | bassically * is switching the context too ofeten |
| 13:05.11 | RoyK | system or user? i/o? |
| 13:05.21 | l-fy | well, i can't handle more than 200 calls from sip to IAX |
| 13:05.28 | RoyK | what sort of load happens then? |
| 13:05.33 | RoyK | system or user? i/o? |
| 13:05.44 | l-fy | after that the iax looses syncronization |
| 13:05.47 | tzafrir | "from sip to iax" is also sip |
| 13:06.22 | l-fy | bassically is not able to send full frames, and is loosing sync, and mini frames are droped of course |
| 13:06.47 | l-fy | tzafrir > i'm just trying to get some help |
| 13:06.58 | RoyK | well |
| 13:07.02 | RoyK | what sort of load happens then? |
| 13:07.20 | l-fy | is not really loaded but is doing this |
| 13:07.37 | RoyK | ok.. |
| 13:07.52 | RoyK | so a top/sar/sysstat/vmstat shows what sort of load? |
| 13:08.09 | RoyK | 200 bridged calls should generate quite a bit cpu load |
| 13:08.25 | l-fy | i don't want to bridge them |
| 13:08.29 | l-fy | are from sip to iax |
| 13:08.29 | RoyK | l-fy: check the load anyway |
| 13:08.37 | l-fy | i don't see a reason to bridge them |
| 13:08.44 | RoyK | sip/iax is a call bridge by design |
| 13:09.04 | RoyK | receive rtp, strip rtp header, pack in iax, send |
| 13:09.23 | RoyK | anyway - use vmstat or sar or something and check system load |
| 13:10.57 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:11.03 | RoyK | l-fy: when you have the true system load data, you can start debugging. also, make sure to turn on oprofile if this is linux |
| 13:12.03 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 13:13.27 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:13.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:32.32 | SplasPood | hrm are there any embedded boxes I can stick at a client's location to handle local SIP routing (ie, calls between extens don't leave the location)? |
| 13:32.47 | SplasPood | And don't tell me a linksys wrt54 running ser :P |
| 13:33.18 | l-fy | SplasPood> you intend to use a small router with software on it? |
| 13:33.58 | SplasPood | I don't intend anything, yet :) |
| 13:34.10 | l-fy | well, you can use something else with uclibc |
| 13:34.12 | SplasPood | I just don't wanna be hacking up linksys gear for clients |
| 13:34.20 | l-fy | just use any embeded box |
| 13:34.26 | l-fy | with ser or anything else |
| 13:34.53 | SplasPood | I'd rather get something all integrated already rather than doing my own custom stuff |
| 13:35.26 | SplasPood | custom stuff is fine for me... but when I'm deploying these things to random customers I don't really have the team to handle development, support, etc.. |
| 13:35.29 | l-fy | how many calls you intend to do? |
| 13:36.03 | SplasPood | these are smaller offices.. maybe 20 handsets... I just don't want paging between handsets and stuff to suck up their bandwidth, rather have the phones invite between em |
| 13:48.05 | ]Xman[ | any1 can help me to recognize in extension a do not disturb status on sip client? |
| 14:02.26 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 14:12.32 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 14:21.42 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:23.48 | *** join/#asterisk teknoprep (n=teknopre@c-68-83-86-17.hsd1.pa.comcast.net) |
| 14:23.49 | teknoprep | hey all |
| 14:24.16 | teknoprep | anyone run asterisk on OpenBSD? i am looking for an easy to install web config panel for it |
| 14:24.27 | teknoprep | free web config of course |
| 14:24.41 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 14:24.50 | *** part/#asterisk Flosoft (n=admin@213.219.186.227.adslpower.by.edpnet.be) |
| 14:30.52 | teknoprep | does asterisk have a builtin web interface? |
| 14:33.34 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 14:34.25 | nextime | teknoprep : i don't use it, and i don't use obsd, but you can try freepbx ( /join #freepbx, www.freepbx.org ) |
| 14:36.12 | l-fy | hey neilbags |
| 14:36.14 | l-fy | nextime |
| 14:36.19 | l-fy | i need your help again :)) |
| 14:36.59 | nextime | hello l-fy i have only few minutes, but try to ask |
| 14:37.00 | teknoprep | is there a web front end built into asterisk? |
| 14:37.23 | nextime | teknoprep : not yet |
| 14:37.26 | l-fy | my asterisk while sending phone calls over iax starts to not send any more full frames |
| 14:37.27 | teknoprep | ok |
| 14:37.29 | l-fy | any idea why? |
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| 15:12.27 | *** join/#asterisk yacyac (n=yac@202.159.249.169) |
| 15:14.08 | yacyac | hiiiii |
| 15:14.14 | yacyac | anyone around |
| 15:17.03 | tzafrir | no |
| 15:17.18 | yacyac | hey tzafrir |
| 15:17.39 | tzafrir | However you should still ask your question... |
| 15:18.17 | yacyac | just a silly question |
| 15:18.31 | yacyac | if you getting late you should move ahead |
| 15:18.50 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 15:20.59 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 15:21.20 | vivek | hello all |
| 15:21.32 | yacyac | hey vivek |
| 15:21.43 | yacyac | tintin |
| 15:21.53 | vivek | hello yacyac what are you upto ? |
| 15:22.04 | yacyac | nm man |
| 15:22.16 | yacyac | where the hell where you all these days |
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| 17:42.48 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 17:44.12 | asterboy | Can't seem to get my Polycom 600 to boot. |
| 17:44.32 | asterboy | Was working fine for months, then I moved the power and now it won't boot. |
| 17:45.09 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 17:45.13 | asterboy | It says its trying the existing config, but it just keeps recycling |
| 17:45.46 | drray | re tftp the firmware? |
| 17:46.04 | asterboy | trying to get it to even find my boot server. |
| 17:46.10 | asterboy | tftp failed. |
| 17:46.19 | asterboy | ftp failed. |
| 17:46.54 | asterboy | I know the net conncetion is good cause I'm on here from the LAN port on the phone. |
| 17:47.04 | drray | you might have to move your tftp server to match what it thinks |
| 17:47.11 | drray | it should be |
| 17:47.27 | asterboy | checking if it even talks to the ftp server |
| 17:50.01 | asterboy | it must be that the onboard .cfg file gets corrupt and it won't boot until it can properly talk to an ftp server |
| 17:50.11 | asterboy | to reload the file |
| 17:50.29 | asterboy | progress... |
| 17:51.04 | asterboy | ...seems it did not like the ftp server for whatever reason...but the Tftp server is now pushing the application on the phone. |
| 17:51.29 | asterboy | dam...rebooted in the middle of loading. |
| 17:51.39 | asterboy | 25% |
| 17:51.51 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 17:51.51 | asterboy | 50% |
| 17:51.57 | drray | huzzah! |
| 17:52.02 | drray | my budgetone did that |
| 17:52.10 | asterboy | skipped to Running... |
| 17:52.19 | drray | congrats |
| 17:52.31 | asterboy | yippie |
| 17:52.34 | asterboy | working now. |
| 17:52.40 | asterboy | real annoying though. |
| 17:52.43 | drray | yeah |
| 17:52.53 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:53.24 | asterboy | being picky about the ftp server and corrupting the onboard config doesn't help. |
| 17:54.12 | asterboy | interesting the budgetone does the same thing...wonder if any other Polycom users get the same. |
| 17:55.38 | asterboy | anyway thanks for the input drray |
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| 18:16.40 | [TK]D-Fender | asterboy : Just because the switch is not dead doesn't me the phones PHY isn't |
| 18:16.59 | [TK]D-Fender | asterboy : Did you upgrade it recently? |
| 18:28.43 | drray | it could have lost its link during an upgrade |
| 18:28.54 | drray | that's how I messed my budgetone up |
| 18:29.15 | drray | they are fragile |
| 18:31.04 | [TK]D-Fender | drray : Less likey with Polycom as it has a seperate boot block. |
| 18:31.29 | drray | yes |
| 18:31.34 | drray | but still fragile |
| 18:31.40 | drray | you just need a perfect storm |
| 18:31.55 | [TK]D-Fender | drray : I *have* seen 2 Polycom IP 600's just completely *die* however |
| 18:32.18 | [TK]D-Fender | One that was totally inoperable, and the other whose networking fell on deaf ears. |
| 18:32.24 | [TK]D-Fender | asterboy : RMA that sucker |
| 18:32.30 | l-fy | [TK]D-Fender > most of the voip phones are like that |
| 18:34.05 | l-fy | but i have to admit i've never ever had any problem with snom |
| 18:34.14 | drray | or my cisco |
| 18:34.26 | drray | cisco has other issues |
| 18:35.26 | l-fy | cisco is some of the worse sip implementations |
| 18:39.16 | *** join/#asterisk CYPRESS_A (n=nate@c-24-99-153-133.hsd1.ga.comcast.net) |
| 18:40.18 | Dr-Linux | l-fy: ever use cisco 7920? |
| 18:40.20 | l-fy | nope |
| 18:40.25 | l-fy | only 7905 |
| 18:45.57 | Dr-Linux | l-fy: actually i got a cisco 7920 phone, not sure if i can load sip firmware on it .. |
| 18:46.59 | l-fy | i don't even bother |
| 18:47.07 | l-fy | i just buy snom |
| 18:47.11 | l-fy | is far cheaper and better |
| 18:47.49 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 18:47.57 | Dr-Linux | hhm.. |
| 18:48.33 | l-fy | hi gr0mit_home |
| 18:48.44 | gr0mit_home | hi i-fy! |
| 18:48.52 | gr0mit_home | how are you these days? |
| 18:48.59 | l-fy | very well thank you |
| 18:49.04 | l-fy | fighting with * |
| 18:49.28 | gr0mit_home | i thought you were still working on YATE? |
| 18:49.32 | l-fy | i do |
| 18:49.52 | l-fy | we just finished our iax stack and we do compatility and performance tests with * |
| 18:50.04 | gr0mit_home | ah i see |
| 18:50.07 | l-fy | and also some hacking into * to figure out what is not written in the iax draft |
| 18:50.15 | gr0mit_home | probably a lot! |
| 18:50.21 | l-fy | well, iax is a good protocol and it deserves a good implementation |
| 18:50.34 | gr0mit_home | agreed! |
| 18:50.54 | gr0mit_home | when are you rewriting the Asterisk implementation?! |
| 18:51.08 | l-fy | we did that already |
| 18:51.11 | l-fy | is called yate |
| 18:51.21 | gr0mit_home | actually we have seen some issues with Asterisk |
| 18:51.23 | dlynes_home | yate has pbx features? |
| 18:51.26 | l-fy | just joking |
| 18:51.29 | l-fy | of course |
| 18:51.34 | l-fy | which kind of girl do you think i am? |
| 18:51.37 | drray | yate is neat |
| 18:51.39 | dlynes_home | i thought it was just a softswitch engine |
| 18:51.42 | gr0mit_home | we are seeing audio gaps when we record to disk |
| 18:51.52 | drray | yate works with tor2 drivers |
| 18:52.05 | gr0mit_home | there are chunks of silence being introduced into the audio |
| 18:52.05 | l-fy | tor2 you mean zaptel, than the answer is yes |
| 18:52.06 | dlynes_home | and sangoma wanpipe |
| 18:52.21 | gr0mit_home | of between 1 and 300 ms |
| 18:52.23 | l-fy | dlynes_home > of course not |
| 18:52.33 | drray | gr0mit - irq sharing? |
| 18:52.35 | dlynes_home | of course not what? |
| 18:52.41 | drray | or DMA not turned on |
| 18:52.41 | gr0mit_home | drray - nope |
| 18:52.44 | l-fy | dlynes_home > did you saw pbxassist? |
| 18:52.46 | gr0mit_home | no irq shared |
| 18:52.50 | dlynes_home | never heard of it |
| 18:52.55 | gr0mit_home | dma is on |
| 18:53.09 | dlynes_home | l-fy: it's a pbx module for yate? |
| 18:53.18 | gr0mit_home | it only happens with over about 8-10 calls |
| 18:53.53 | dlynes_home | l-fy: I was talking to someone on yate about the new pbx module, and he asked me to come on at about 1am, PDT...I came on at that time several times, but nobody's ever on at that time |
| 18:53.59 | l-fy | dlynes_home > yeap, it does conference and transfer in the way you want |
| 18:54.06 | l-fy | well, i am now :) |
| 18:54.24 | dlynes_home | ah..ok, i'll revisit it again then |
| 18:54.26 | l-fy | i had a busy week preparing for the release :) |
| 18:54.28 | dlynes_home | but i'm late...I've gotta run |
| 18:54.34 | l-fy | and i did tried to reach you :) |
| 18:54.41 | l-fy | damn |
| 18:54.46 | l-fy | my firefox just died :( |
| 18:54.47 | dlynes_home | oh yeah...that was you i was talking to |
| 18:54.53 | l-fy | in the middle of my pool game |
| 19:02.17 | l-fy | guys i need help with oseq and iseq numbers in iax |
| 19:05.11 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 19:15.05 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:31.50 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:34.56 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 19:35.15 | l-fy | hi riddlebox |
| 19:35.27 | riddlebox | hey whats up |
| 19:39.29 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 19:44.21 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:52.06 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 19:53.01 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:54.03 | l-fy | riddlebox > just wondering how are you? |
| 19:54.14 | l-fy | trying to get reconnected with old friends :)) |
| 19:54.44 | riddlebox | good, working on learning to program better u? |
| 19:55.27 | dlynes_office | Anyone come across this error before? "wanconfig: Failed to configure EC device wanpipe1 (err=22)" |
| 19:55.35 | *** join/#asterisk Twister (n=bob@71-213-209-229.sxcy.qwest.net) |
| 19:56.10 | dlynes_office | I get that error pop during bootup process |
| 19:56.19 | dlynes_office | However, I see no mention of an error in dmesg |
| 19:56.44 | Twister | hey all, could someone point me to a good tutorial on how to set up a cisco 7960 with sip, ive read through voip-info, through cisco and for some reason i cant make heads or tails of it, kinda lookin for sample config files and what should be in the tftp |
| 19:57.22 | dlynes_office | Twister: you could always use chan_skinny instead |
| 19:57.33 | dlynes_office | Twister: and then you don't have to deal with the buggy sip firmware |
| 19:57.47 | *** join/#asterisk Strom_C (n=strom@12.150.239.132) |
| 19:58.11 | dlynes_office | Twister: the author of chan_skinny is on here regularly, too (qwell, qwell[], ...) |
| 19:58.33 | Twister | ok |
| 19:58.36 | dlynes_office | Twister: so if you do have problems with it, and you talk to him nice enough, he might be kind enough to write some patches for it |
| 19:58.43 | Twister | that'll work then |
| 19:58.45 | Twister | :) |
| 19:58.53 | l-fy | dlynes_office > what did you setup, just TDMA? |
| 19:58.55 | Twister | well im very nice to everyone |
| 19:59.04 | dlynes_office | Twister: financial compensation usually goes a long way, too :) |
| 19:59.12 | dlynes_office | l-fy: yeah...a200d |
| 19:59.17 | Twister | thx dlynes_office and ya hehe |
| 19:59.23 | l-fy | ooooo, that one |
| 19:59.31 | l-fy | sorry, i'm far far away from those stuff |
| 19:59.45 | dlynes_office | Twister: but, the biggest thing he's looking for right now I think is donations of cisco phones that he can test with |
| 20:00.01 | dlynes_office | Twister: he doesn't have a lot of the different cisco models to test on |
| 20:00.40 | l-fy | dlynes_office > can we finish that disscusion, when you have complained about y? |
| 20:01.03 | dlynes_office | l-fy: complained about y? huh? |
| 20:01.15 | l-fy | pbxassist |
| 20:01.18 | dlynes_office | which discussion? and I don't remember complaining about anything |
| 20:01.46 | dlynes_office | and I don't remember complaining about yate |
| 20:01.50 | l-fy | <dlynes_home> l-fy: I was talking to someone on yate about the new pbx module, and he asked me to come on at about 1am, PDT...I came on at that time several times, but nobody's ever on at that time |
| 20:01.54 | dlynes_office | I haven't even used it yet |
| 20:02.06 | dlynes_office | ah...that's what you're talking about :) |
| 20:02.10 | l-fy | well, not is the time |
| 20:02.30 | l-fy | not = now |
| 20:02.40 | drray | is yate #yate? |
| 20:02.46 | l-fy | yes |
| 20:03.39 | *** part/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 20:09.24 | Twister | ya know..this may be a little sad..my company has a tekelec 6000 and a tekelec 7000 (the 6000 is supposed to be providing the voip to the consumer side and the 7000 is kinda like the gateway from what i can gather)...together they cost about 500K |
| 20:09.41 | Twister | and id still prefer a 3000 linux box with asterisk on it lol |
| 20:09.55 | Twister | $3000 |
| 20:12.01 | dlynes_office | tekelec? sounds like some kinda native language :p |
| 20:18.47 | Twister | lol |
| 20:19.15 | Twister | its this high end voice applications company..thing /shrug |
| 20:21.01 | Twister | im supposed to learn all about them and i have noo idea where to begin lol |
| 20:24.06 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:37.42 | af_ | I have a funny thing. if I put bristuffed asterisk (it happens with visdn as well) on a pc with an adsl modem (pci or usb) when isdn comes up, adsl freeeze and vice versa. any idea? |
| 20:43.30 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 20:44.49 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 20:49.03 | *** join/#asterisk saftsack (n=saftsack@p54A7FCEB.dip.t-dialin.net) |
| 20:52.20 | l-fy | af_ > hardware related issue |
| 20:52.29 | l-fy | try to change the mainboard |
| 20:52.34 | l-fy | or the power supply |
| 20:53.39 | af_ | well, that's funny because it happens with three mother board, of differente brand |
| 20:53.59 | af_ | and psu are different too |
| 20:54.09 | l-fy | i see |
| 20:54.14 | af_ | the best is with an eagle usb |
| 20:54.34 | l-fy | try another adsl |
| 20:54.38 | af_ | I could hardly believe with a pci card, but with an usb modem it's really strange by me |
| 20:54.54 | af_ | I tried two |
| 20:55.00 | af_ | usb eagle, pci conezant |
| 20:55.04 | af_ | so.... |
| 20:55.19 | af_ | it happens the same, anytime |
| 20:55.31 | l-fy | with any adsl modem? |
| 20:55.48 | af_ | I din't know, any is abig world for me |
| 20:56.09 | af_ | two is a good number to any |
| 21:03.53 | *** join/#asterisk onixx (i=1000@Quebec-HSE-ppp3620919.sympatico.ca) |
| 21:05.03 | onixx | hi all... I'm wondering if my ht286 sends an FSK Message Waiting packet to the analog phone. Anyone ? |
| 21:05.33 | onixx | I have the shutter dial tone but it won't light up my MWI light on the analog phone |
| 21:07.06 | SplasPood | Is there any way to convert the asterisk db DB() to utilize mysql rather than the db file? |
| 21:16.18 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 21:16.39 | *** join/#asterisk MedozasSVR (n=MedozasS@p549BB054.dip0.t-ipconnect.de) |
| 21:17.11 | MedozasSVR | hello - does anyone here have experience with asterisk realtime? |
| 21:19.53 | MedozasSVR | I have some questions regarding Asterisk 1.2.9.1 and realtime asterisk. Mentioned on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions the extensions "s" and "i" are availiable on SVN - as Asterisk has been released just shortly - is there now the support for "s" and "i" in 1.2.9.1, and when (if ever) will there be support for includes in extensions table? |
| 21:20.27 | MedozasSVR | I haven't found anything in changelog |
| 21:26.06 | l-fy | MedozasSVR > what's asterisk realtime? |
| 21:26.17 | MedozasSVR | asterisk with mysql backend |
| 21:26.58 | MedozasSVR | (or other databases like pgsql or connection via unixodbc) |
| 21:27.07 | l-fy | o god |
| 21:27.13 | l-fy | isin't that my default in * ? |
| 21:27.58 | MedozasSVR | sorry, i dont understand what you mean |
| 21:32.38 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 21:34.24 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 21:40.10 | MedozasSVR | i have asked before, but no answer yet - I have some questions regarding Asterisk 1.2.9.1 and realtime asterisk. Mentioned on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions the extensions "s" and "i" are availiable on SVN - as Asterisk has been released just shortly - is there now the support for "s" and "i" in 1.2.9.1, and when (if ever) will there be support for includes in extensions table? I haven't found anything |
| 21:48.56 | *** join/#asterisk frenzy (n=frenzy@196.46.104.246) |
| 21:49.03 | frenzy | hi |
| 21:49.17 | frenzy | need help.. |
| 21:49.19 | frenzy | Jun 19 00:36:03 VERBOSE[10766] logger.c: [app_rxfax.so]Jun 19 00:36:03 WARNING[10766] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_get_far_ident |
| 21:49.25 | frenzy | cannot start asterisk |
| 21:49.32 | frenzy | Jun 19 00:36:03 WARNING[10766] loader.c: Loading module app_rxfax.so failed! |
| 21:54.50 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 21:56.14 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 21:57.15 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 21:58.13 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 22:03.27 | *** join/#asterisk JoseBravo (n=JoseBrav@cm4126.uniweb.net.co) |
| 22:03.50 | JoseBravo | How can I delete all astbill example info? |
| 22:15.52 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 22:18.42 | dlynes_office | swears at his smc piece of crap @$@#%@#$%#$$@##$#@ router!!! |
| 22:19.27 | *** join/#asterisk Strom_C (n=strom@12.150.239.132) |
| 22:19.33 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:20.31 | JoseBravo | How I go out of CLI without stop Asterisk, the command quit don't work. |
| 22:20.41 | shmaltz | JoseBravo, exit |
| 22:20.56 | JoseBravo | No such command 'exit' (type 'help' for help) |
| 22:21.15 | X-Rob | you have started asterisk incorectly |
| 22:21.16 | shmaltz | JoseBravo, start asterisk from outside by not using c or r |
| 22:21.28 | shmaltz | so for now, type |
| 22:21.28 | shmaltz | stop now |
| 22:21.29 | shmaltz | then type |
| 22:21.36 | shmaltz | safe_asterisk |
| 22:21.37 | JoseBravo | I typed: asterisk -vvvvgc |
| 22:21.38 | shmaltz | then type |
| 22:21.42 | shmaltz | asterisk -cr |
| 22:22.15 | X-Rob | JoseBravo, you use 'safe_asterisk' to start asterisk. then 'asterisk -r' to connect to it |
| 22:22.21 | JoseBravo | Thanks |
| 22:26.00 | Strom_C | JoseBravo, did you connect to asterisk by typing "asterisk -r" or by typing "asterisk -c"? |
| 22:26.04 | Strom_C | safe_asterisk is only really a good idea when you know your asterisk configs aren't going to make asterisk die on startup |
| 22:26.07 | Strom_C | start it as a daemon by just typing asterisk |
| 22:27.55 | MedozasSVR | i have asked before, but no answer yet - I have some questions regarding Asterisk 1.2.9.1 and realtime asterisk. Mentioned on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions the extensions "s" and "i" are availiable on SVN - as Asterisk has been released just shortly - is there now the support for "s" and "i" in 1.2.9.1, and when (if ever) will there be support for includes in extensions table? I haven't found anything |
| 22:29.08 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:30.18 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:34.00 | Strom_C | then |
| 22:34.00 | Strom_C | asterisk -r |
| 22:34.36 | Strom_C | MedozasSVR, 's' and 'i' have been part of stable asterisk for a long long long time |
| 22:34.38 | Strom_C | MedozasSVR, so I imagine they'd work in realtime |
| 22:36.17 | Strom_C | what a quiet day in here |
| 22:36.31 | sevard | HOW ARE YOU DOING |
| 22:37.31 | *** join/#asterisk zagaya972 (n=d2s-comp@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
| 22:37.48 | Strom_C | Strom |
| 22:37.52 | Strom_C | not Storm |
| 22:37.54 | shmaltz | I know |
| 22:37.57 | file | Strommy Boy! |
| 22:38.04 | Strom_C | woot |
| 22:38.11 | Qwell | LOOK |
| 22:38.13 | shmaltz | I am a bit disleksic (or what ever its' spelled) |
| 22:38.14 | Strom_C | LQQK |
| 22:38.18 | file | hey Qwell! |
| 22:38.21 | Qwell | LOOK! |
| 22:38.22 | Strom_C | dyslexic |
| 22:38.35 | Qwell | r34696 :D |
| 22:39.10 | sevard | I'm very dyslexic, yesterday a friend of mine called and i thought it was my dad so I answered "hi dad" |
| 22:39.24 | shmaltz | sevard, that just means a bad memory |
| 22:39.43 | Qwell | pfft |
| 22:39.49 | sevard | it was some thing like 598-4495 where my dads is something like 954-4985 |
| 22:39.58 | *** join/#asterisk xnon (n=xnon@200.8.30.47) |
| 22:40.07 | xnon | hello |
| 22:40.11 | sevard | i wanted to talk to my dad :| not my stupid pal. |
| 22:40.21 | Qwell | hmm...maybe I should call mine |
| 22:40.22 | file | I tried to call my dad today, but alas... nothing |
| 22:40.22 | shmaltz | whats the /list command for IRC to list channels that have a patteren in it, like ux for linux and sux |
| 22:40.22 | xnon | anybody can help me? |
| 22:40.23 | sevard | Qwell: those numbers are made up for an example |
| 22:40.28 | Qwell | sevard: sure, sure ;) |
| 22:40.37 | xnon | anybody speak spanish |
| 22:40.38 | xnon | ? |
| 22:40.44 | sevard | Qwell: go a head and call those numbers, record them. |
| 22:40.51 | sevard | xnon: No. |
| 22:40.55 | sevard | ^^^ that's a joke. |
| 22:41.01 | sevard | :/ drunk on a sunday. |
| 22:41.08 | xnon | well ill try speak in english ok? |
| 22:41.18 | xnon | i have a problem with my calls |
| 22:41.21 | Qwell | So... my chan_skinny fixes have been merged to trunk :D |
| 22:41.35 | xnon | i have to do calls and recive |
| 22:41.58 | xnon | but the people from the extranet cant hereme |
| 22:42.47 | sevard | oh oh |
| 22:43.11 | xnon | sorry sorry i cant here peoples that i call |
| 22:43.57 | *** join/#asterisk SmittyHalibut (n=msmith@24-180-0-138.static.snlo.ca.charter.com) |
| 22:44.49 | SmittyHalibut | Anyone here have any experience getting a TE110P configured with a Channelized T1 (not PRI)? |
| 22:45.56 | *** join/#asterisk uRSOBEAUTIFL (n=fdsa@modemcable237.246-83-70.mc.videotron.ca) |
| 22:46.00 | uRSOBEAUTIFL | Hi |
| 22:46.12 | uRSOBEAUTIFL | I'm looking for info about FreePBX |
| 22:46.25 | Qwell | uRSOBEAUTIFL: see topic |
| 22:46.49 | *** join/#asterisk sevard (n=sev@24-179-179-47.dhcp.dlth.mn.charter.com) |
| 22:47.38 | uRSOBEAUTIFL | in fact I'd like to know how to get data from mysql or OLE DB and deliver to user by IVR |
| 22:48.09 | SmittyHalibut | He doesn't listen, does he? |
| 22:48.25 | *** join/#asterisk Strom_C (n=strom@12.150.239.132) |
| 22:48.41 | uRSOBEAUTIFL | sthg like: enter your customer number...we are looking for your data...your account total is 200$ |
| 22:49.08 | sevard | uRSOBEAUTIFL: write a perl script to be used with AGI |
| 22:49.36 | uRSOBEAUTIFL | Hi Sevard, do u have an example of this kind of script ? |
| 22:50.00 | sevard | yeah |
| 22:50.01 | sevard | google perl. |
| 22:50.39 | uRSOBEAUTIFL | is there a way to use PHP instead of perl ? |
| 22:50.55 | SmittyHalibut | yeah. |
| 22:50.59 | SmittyHalibut | google PHP |
| 22:53.01 | JoseBravo | I have created 2 users and 2 accounts in Astbill, I have my clients registered. But how can I call from one to other? Dialing what? |
| 22:54.30 | dlynes_office | you guys are so helpful :) |
| 22:58.40 | SmittyHalibut | He comes in here asking about FreePBX. Wrong forum. |
| 23:00.06 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:02.54 | SmittyHalibut | g'night, kids. |
| 23:08.05 | JoseBravo | I have created 2 users and 2 accounts in Astbill, I have my clients registered. But how can I call from one to other? Dialing what? |
| 23:11.39 | MedozasSVR | @strom: nope - of course the standard contexts like "s" and "i" are availiable since a long time, same as includes - but FOR SURE includes dont work yet - thats why i want to ask also same with "s" and "i" extensions as they definitly not STANDARD in 1.2 branch since asterwisk realtime is alive only since 1.2 - thats why im asking if anybody KNOWS if there is support for these contexts in 1.2.9.1 |
| 23:11.50 | dlynes_office | piece of #@%t f@%#@% smc router :((( |
| 23:12.23 | Strom_C | MedozasSVR, ok fine, but dont talk down to me like I'm an idiot |
| 23:12.34 | MedozasSVR | i sure dont mean it that way |
| 23:12.50 | Strom_C | well it comes across like that when you capitalize every sixth phrase |
| 23:13.45 | dlynes_office | Strom_C: that's what i would infer, too |
| 23:14.00 | Strom_C | :) |
| 23:14.40 | MedozasSVR | as i said, i dont mean it that way, sorry - but there are so many people thinking of and telling you something, but they didnt use asterisk realtime once at all - ive tried a lot what people have said only about 10% really helped -- really sorry, didnt mean it that way |
| 23:14.45 | dlynes_office | why do all these companies make such crappy pos routers, anyways? |
| 23:15.18 | Strom_C | dlynes_office, race to the bottom? |
| 23:15.42 | dlynes_office | Strom_C: no idea, but I have yet to find a decently priced router that works consistently, 100% of the time |
| 23:16.19 | MedozasSVR | did you ever use asterisk realtime, strom_c, and can you help in this manner? |
| 23:16.32 | dlynes_office | i'm tempted to just say screw the routers altogether and buy boxes for asterisk that have more pci slots so I can throw shorewall onto them |
| 23:17.03 | Strom_C | MedozasSVR, no, I have not used realtime. Why do you feel you need to use it? |
| 23:17.41 | dlynes_office | Strom_C: prolly so he doesn't need some monkey at the controls to adjust the config files to add customers and that kinda thing |
| 23:18.04 | dlynes_office | Strom_C: he can just write a slick web configuration gui instead os the switch monkeys can set up new routes and new extensions and htat kinda thing |
| 23:18.32 | *** join/#asterisk ali3nx (n=compiled@S010600107b3fb041.wp.shawcable.net) |
| 23:18.35 | Strom_C | dlynes_home, well sure, but I want to hear his reasons |
| 23:18.59 | dlynes_office | Strom_C: cheaper to higher idiots that can add extensions in a web gui, than asterisk admins :) |
| 23:19.06 | MedozasSVR | well - the elegance of having a database in the backend is one nice thing - for use of a second box (failover with backend-clustered mysql), then a much better possiblity for cdr-reading, and the better possibility for making an own small frontend specifically to my needs is better with mysql as with flat files |
| 23:19.25 | dlynes_office | erm |
| 23:19.27 | dlynes_office | s/higher/hire/ |
| 23:19.32 | ali3nx | hi i'm just doing some research on hardware for an asterisk office deployment and wondered if cisco 7912g voip phones would work fine? |
| 23:19.41 | [TK]D-Fender | hukt on fonix werkt 4 U! |
| 23:19.50 | dlynes_office | [TK]D-Fender: heh |
| 23:20.00 | *** join/#asterisk acid-storm (n=acid@adsl-69-231-91-144.dsl.irvnca.pacbell.net) |
| 23:20.05 | dlynes_office | [TK]D-Fender: sometimes my brain is thinking one thing, and my fingers are typing another :) |
| 23:20.18 | acid-storm | heh |
| 23:20.19 | ali3nx | :) |
| 23:20.20 | Qwell | dlynes_home: r34696 |
| 23:20.24 | Qwell | svn trunk |
| 23:20.27 | dlynes_office | ? |
| 23:20.33 | Qwell | ;) |
| 23:20.53 | dlynes_office | Qwell: asterisk has hooked on phonics support now?!?! |
| 23:21.02 | acid-storm | heh |
| 23:21.26 | *** join/#asterisk zagaya972 (n=d2s-comp@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
| 23:21.28 | Qwell | something like that |
| 23:21.40 | Qwell | ~seen [av]bani |
| 23:21.54 | jbot | [av]bani <n=[av]bani@washuu.anime.net> was last seen on IRC in channel #asterisk, 63d 56m 19s ago, saying: 'robin_sz: how are they?'. |
| 23:21.54 | *** part/#asterisk zagaya972 (n=d2s-comp@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
| 23:21.54 | MedozasSVR | anybody who can help me with asterisk realtime= |
| 23:21.55 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:22.36 | Qwell | I think he died |
| 23:22.38 | Strom_C | MedozasSVR, dude, if no one knows the answer, they won't generally respond. It's considered bad form to ask the same question in here EVERY FIVE MINUTES |
| 23:23.28 | MedozasSVR | ? |
| 23:23.28 | MedozasSVR | sry strom_c |
| 23:23.47 | Strom_C | we need one of those "you must be this tall to ride the ride" disclaimers for this channel |
| 23:24.13 | acid-storm | onjoin notice heh |
| 23:24.16 | acid-storm | works usually |
| 23:24.19 | Strom_C | except it should be "you must be this familair with general IRC etiquette" :) |
| 23:24.33 | MedozasSVR | sure, come on strom_c the same should be on your side not talking with or about idiots like they were idiots ;) |
| 23:24.46 | acid-storm | anyone have a recomendation on a decent voip phone when used in a biz setting? |
| 23:24.58 | MedozasSVR | Grandstream GXP-2000 |
| 23:25.04 | Strom_C | acid-storm, I'm very fond of the cisco 7940/7960 |
| 23:25.25 | acid-storm | whats the typical price break strom? |
| 23:25.28 | dlynes_office | Qwell: congrats |
| 23:25.31 | [TK]D-Fender | acid-storm : Polycom and Cisco are the only real ones for business... the rest are jsut too flakey |
| 23:25.53 | Strom_C | acid-storm, I've gotten the 7940 for $200 each in quantity |
| 23:26.01 | acid-storm | thats not too bad |
| 23:26.20 | dlynes_office | [TK]D-Fender: not true...I've had relatively good experiences with Aastra 9133i's |
| 23:26.29 | acid-storm | the 7940 takes poe |
| 23:26.30 | acid-storm | right? |
| 23:26.35 | MedozasSVR | well grandstream has made a great way heading businesses and reliable in any way (have about 150 running now for about 1 year) |
| 23:26.36 | acid-storm | so i dont have to do all those damn cubes |
| 23:26.45 | [TK]D-Fender | acid-storm : If you have a great discount on Cisco, they are great phones, but SIP is a little less than perfect on the platform. For every other scenario, there's Polycom. |
| 23:26.54 | MedozasSVR | and the cost is half of cisco (about 100$) |
| 23:26.55 | acid-storm | yea |
| 23:26.57 | Strom_C | yes, although I think you need a real cisco switch |
| 23:27.09 | acid-storm | yea hp procurves work as well |
| 23:27.10 | Strom_C | my issue with the polycom phones is that they are a major pain in the ass to set up and configure |
| 23:27.12 | acid-storm | Im a shoretel dealer |
| 23:27.15 | [TK]D-Fender | dlynes_home : Sorry, yes, the Aastra's aren't too bad. Just in most areas simply not worth the $ |
| 23:27.22 | acid-storm | the hp procurves do "okay" lol |
| 23:27.31 | acid-storm | but ya the polycom hmmm |
| 23:27.38 | dlynes_office | [TK]D-Fender: most areas, they're almost as expensive as the polycoms? |
| 23:27.58 | *** join/#asterisk CerealVore (n=denis@static-112.217.240.220.dsl.comindico.com.au) |
| 23:28.02 | Strom_C | polycom is great if you're into self-torture :) |
| 23:28.04 | [TK]D-Fender | dlynes_home : Yeah, pretty close... TOO close. IP430 cuts right into the 9133.... |
| 23:28.19 | CerealVore | hi guys |
| 23:28.29 | dlynes_office | [TK]D-Fender: so why do i hear about so many problems with the polycoms on here, then? |
| 23:28.40 | Strom_C | and, FWIW, I have yet to have issues with SIP on my cisco 7960s, and I've run them for a year and a half |
| 23:28.46 | [TK]D-Fender | dlynes_home : Echoy room? ;) |
| 23:28.51 | Qwell | Strom_C: chan_skinny :p |
| 23:29.02 | Strom_C | Qwell, is that in Trunk now> |
| 23:29.04 | Strom_C | ? |
| 23:29.05 | Qwell | now that it's working better in trunk, you have no excuse :p |
| 23:29.06 | Qwell | yep |
| 23:29.09 | Strom_C | ooooooooh |
| 23:29.16 | Strom_C | I will do that as soon as I get home then :) |
| 23:29.36 | dlynes_office | Strom_C: damn..you can afford to have a cisco for your home phone? |
| 23:29.37 | JoseBravo | Anoye help me please, I have installed Astbill but I don't know how to call from one user to another user. |
| 23:29.40 | dlynes_office | Strom_C: you must be filthy rich |
| 23:29.59 | CerealVore | hahaha |
| 23:30.03 | Strom_C | dlynes_home, my office is in my apartment |
| 23:30.04 | [TK]D-Fender | JoseBravo : This is not the place to ask for support for 3rd party GUI modules for *.... |
| 23:30.11 | Qwell | What's wrong with having a cisco at home? |
| 23:30.15 | CerealVore | 7960s? the ones with the colour screens that cost around $900 each? |
| 23:30.16 | dlynes_office | nothing |
| 23:30.18 | dlynes_office | if you're rich |
| 23:30.22 | Strom_C | no, thats 7970 |
| 23:30.23 | Qwell | CerealVore: 7970, and more like $400 |
| 23:30.36 | CerealVore | i was just looking at a whole bunch of them |
| 23:30.57 | CerealVore | a company i work for just outfitted their entire office with them |
| 23:31.08 | dlynes_office | damn |
| 23:31.10 | CerealVore | spent like $300k on their phone system |
| 23:31.17 | Strom_C | dlynes_home, besides, I'm a phone phreak. I've got perhaps twenty-five different telephone sets at home. |
| 23:31.18 | Strom_C | two of which are cisco 7960s :) |
| 23:31.28 | CerealVore | and still haven't programmed their callmanager correctly ;) |
| 23:31.33 | dlynes_office | I have trouble trying to wrangle enough cash out of my boss for aastras, never mind ciscos :) |
| 23:31.54 | CerealVore | 90% of the time you call them, all you get is "Cisco unity messaging service..." |
| 23:32.39 | *** join/#asterisk uRSOBEAUTIFL (n=fdsa@modemcable237.246-83-70.mc.videotron.ca) |
| 23:32.53 | Strom_C | dlynes_office, besides, being able to afford the cool toys is one of the great side effects of being a consultant |
| 23:32.57 | dlynes_office | uRSOBEAUTIFL is going to start asking freepbx questions again :p |
| 23:33.00 | acid-storm | anyone have a recommendation for a pri? (southern cali)? |
| 23:33.05 | Strom_C | acid-storm, SBC |
| 23:33.25 | acid-storm | they changed the unlimited usage rules i thought? |
| 23:33.54 | dlynes_office | unlimited usage rules on pris? |
| 23:33.56 | uRSOBEAUTIFL | do u know sthg "graphical" to build IVR ? |
| 23:34.00 | Strom_C | well, from a pure stability standpoint, I trust the ILEC more than I trust a CLEC |
| 23:34.02 | dlynes_office | there's such a thing as a limited usage rule? |
| 23:34.28 | Strom_C | uRSOBEAUTIFL, learn to use vim or go to #freepbx |
| 23:34.43 | dlynes_office | Strom_C: wtf is sthg? |
| 23:34.52 | Strom_C | I imagine "something" |
| 23:35.03 | dlynes_office | ah |
| 23:35.09 | dlynes_office | I never did learn how to speak aol |
| 23:35.12 | Strom_C | though it could probably just as easily mean "sausage, tongue, and half a goat" |
| 23:35.36 | uRSOBEAUTIFL | why are you so stupid guys. You seems to know everything and be unable to share you knowledges. Very useless |
| 23:35.42 | CerealVore | hahaha |
| 23:35.49 | dlynes_office | uRSOBEAUTIFL: dood....speak in English |
| 23:36.03 | dlynes_office | uRSOBEAUTIFL: People do speak English in Quebec |
| 23:36.12 | CerealVore | and thus the n00b feels the need to share his wisdom concerning those who know more than him |
| 23:36.58 | uRSOBEAUTIFL | allez vous faire enculer bande de cons |
| 23:37.02 | dlynes_office | uRSOBEAUTIFL: my graphical vim (gvim) works just fine to build ivrs |
| 23:37.09 | *** part/#asterisk uRSOBEAUTIFL (n=fdsa@modemcable237.246-83-70.mc.videotron.ca) |
| 23:37.15 | dlynes_office | hahahaha |
| 23:38.12 | dlynes_office | nice language asshole :p |
| 23:38.12 | pjchilds | thats pretty funny ;) |
| 23:38.32 | dlynes_office | pjchilds: you know french too, i take it? |
| 23:38.34 | Strom_C | "have it up the ass load of cretins"? |
| 23:38.50 | dlynes_office | Strom_C: yeah...sorta |
| 23:38.56 | pjchilds | bablefish... for when a vendors docs are all in french :) |
| 23:39.05 | dlynes_office | ah |
| 23:39.06 | Strom_C | the only french I know is "mauvais numero" |
| 23:39.25 | CerealVore | (excuse the poor use of terms) but with asterisk, can i create a group of trunks, so that when an internal user wants to dial out, i can use Dial(TrunkGroup/${EXTEN}) to dial out through whichever line is free? |
| 23:39.33 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 23:39.39 | Strom_C | CerealVore, yes |
| 23:39.43 | Strom_C | for Zap trunks, right? |
| 23:39.48 | dlynes_office | CerealVore: yeah use the 'group=' directive |
| 23:40.01 | CerealVore | 4 zap trunks and potentially limitless sip trunks |
| 23:40.25 | Qwell | Strom_C: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=9741358798&ssPageName=MERC_VI_RSCC_Pr4_PcY_BID_Stores_IT |
| 23:40.27 | Qwell | heh |
| 23:40.53 | Strom_C | oh boy, ROLM! |
| 23:40.58 | Qwell | My dad used to work for them |
| 23:41.12 | Qwell | like, forever ago |
| 23:41.29 | Strom_C | I have no idea what the switching equipment is like, but I did always think the telephone sets were particularly ugly |
| 23:41.40 | dlynes_office | yeah...one of our customers wanted us to airlift a rolm system out and replace it with a bcm |
| 23:42.05 | dlynes_office | those things are huge beasts |
| 23:42.17 | file | I was a bad girl, talkin' about a sad girl |
| 23:42.25 | file | Mr. Bigstuff, who do you think you are? |
| 23:42.53 | Strom_C | file, put away the disco records |
| 23:43.01 | file | I have the Sister Act 2 soundtrack |
| 23:43.02 | Strom_C | before you hurt someone :) |
| 23:44.43 | CerealVore | hey, can i use the group syntax on internal sip trunks too? |
| 23:45.13 | Strom_C | CerealVore, the group syntax is really only for Zap trunks, if I recall correctly |
| 23:45.27 | file | if you mean in the configuration file, it's for zap |
| 23:45.33 | Strom_C | with a sip trunk, you define it once and then just set up multiple calls |
| 23:45.35 | CerealVore | how would i create ring grounds for internal sip handsets then? |
| 23:45.45 | CerealVore | s/grounds/groups |
| 23:46.00 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:47.58 | *** join/#asterisk drjones1 (n=drjones@216.167.187.13) |
| 23:47.59 | drjones1 | hey guys |
| 23:48.05 | drjones1 | where is a good place to get some cisco 7960s |
| 23:48.12 | drjones1 | i used to come here and get advice |
| 23:48.18 | drjones1 | it was like voipsolutions.com or something |
| 23:48.20 | drjones1 | good guys |
| 23:50.21 | drjones1 | eh, ill just get em from infinity micro |
| 23:50.46 | *** join/#asterisk Iamtheman (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca) |
| 23:50.50 | Strom_C | its a lazy sunday afternoon in #asterisk |
| 23:50.59 | drjones1 | dang |
| 23:51.04 | drjones1 | i dont know if these guys sell the powerbrick with it |
| 23:52.59 | drray | ebay |
| 23:53.02 | drjones1 | does a retail 7960 come with the powerbrick/ |
| 23:53.32 | Qwell | drjones1: no |
| 23:53.45 | drray | atacomm is ok |
| 23:53.46 | CerealVore | Strom_C: is there an easier way to create a ring group, as opposed to say, SIP/Handset1&SIP/Handset2&SIP/HandsetN? |
| 23:54.00 | Strom_C | CerealVore, not as far as I'm aware |
| 23:54.17 | Strom_C | though there may very well be |
| 23:54.51 | drjones1 | Dwamnit |
| 23:54.55 | drjones1 | where should i get the power brick then |
| 23:54.59 | drjones1 | god damnit, why would you sell a product |
| 23:55.02 | drjones1 | with no powersupply |
| 23:55.11 | orlock | PoE |
| 23:55.15 | Strom_C | drjones1, because of PoE |
| 23:55.20 | drjones1 | PoE? |
| 23:55.23 | drjones1 | power over ethernet? |
| 23:55.24 | Strom_C | power over ethernet |
| 23:55.26 | orlock | yeah |
| 23:55.41 | drjones1 | anyone know where i can get some powerbricks? |
| 23:55.49 | drjones1 | or that company i used to get the cisco 7960s for |
| 23:55.55 | drjones1 | voipsolutions or somethin |
| 23:55.56 | drray | ebay |
| 23:57.31 | drjones1 | voiplink? |
| 23:57.33 | drjones1 | was that it? |
| 23:57.55 | Iamtheman | Hi guys, is there a Cisco phone that works better with Ast. ? |
| 23:58.17 | Strom_C | better than...what? |
| 23:58.33 | Strom_C | all cisco phones are more compatible with asterisk than cheesecake or rocks |
| 23:58.41 | hads | haha |
| 23:59.05 | *** join/#asterisk rene- (n=rene@201.152.34.100) |
| 23:59.24 | Iamtheman | a Cisco model in particular |
| 23:59.42 | drray | 7960 |
| 23:59.45 | Strom_C | 7940 / 7960 are the gold standard |
| 23:59.46 | drray | or 40 |
| 23:59.57 | drray | the 70's are lowering in price |