00:00.34 | Dr-Linux | what port should be opened in firwall for the CISCO ip phones (7960/40) ? |
00:01.18 | [hC] | I should be able to use a TDM02B (TDM400P with 2 fxo modules) with two more X100P's right? |
00:01.20 | twisted[asteria] | well, if it's a fir wall, be sure to use the right drill bit, otherwise your firwall might split |
00:01.33 | RyanW | Dr-Linux: tcpdump for it. SIP uses udp 5060 for starters if thats any help. |
00:01.36 | Beirdo | hehehe |
00:02.27 | riddlebox | {zombie}: Thanks thats what it was, I had dtmf=inband in sip.conf |
00:02.47 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
00:02.53 | Dr-Linux | RyanW: 5060 port is already opened |
00:03.10 | *** join/#asterisk businesstsi (n=business@ppp-69-228-130-223.dsl.irvnca.pacbell.net) |
00:05.06 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
00:05.09 | Seldon1975 | hi guys |
00:06.05 | Seldon1975 | please help! what does "pbx_extension_helper.c: cannot find extension context 'default'" mean |
00:06.19 | Seldon1975 | the digium guy has screwed my config |
00:06.25 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
00:06.32 | Dr-Linux | twisted[asteria]: on the server i'm using >> -A INPUT -p udp -m udp --dport 5060 -j ACCEPT |
00:07.23 | Dr-Linux | its working with everything, but as i congiured a Cisco ip phone, remotely, its not premited, so i disable the firewall, and phone is working fine now |
00:07.38 | Dr-Linux | how can i do, that what port is being blocked? :S |
00:08.54 | RyanW | <RyanW> Dr-Linux: tcpdump for it. |
00:08.57 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
00:10.15 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
00:12.38 | riddlebox | Seldon1975:is your voicemail stuff looking for the context default? |
00:13.20 | Seldon1975 | riddle: yes |
00:14.13 | riddlebox | Seldon1975:do you have a default context for extensions.conf? |
00:14.20 | Seldon1975 | yes, at the top |
00:15.02 | riddlebox | can you go to pastebin and paste your extensions.conf and voicemail.conf? |
00:15.18 | SpaceBass | I'm having a problem with incoming zap contexts and distinctive ring... everything works fine until someone calls zap/2 then all calls (even to zap/1) are routed like they came from zap/2 |
00:15.20 | SpaceBass | http://pastebin.ca/34282 |
00:15.35 | Seldon1975 | riddle: my extensions.conf is: http://pastebin.com/470190 |
00:15.45 | *** join/#asterisk YaroMan (i=YaroMan@cpe-204-210-153-209.hvc.res.rr.com) |
00:15.49 | YaroMan | Hello |
00:15.54 | Seldon1975 | riddle: does it really have to do with voicemail.conf? Im ge4tting this issue when trying to dial out |
00:15.55 | SpaceBass | anyone ever heard of zap calls "jumping" context like that? |
00:16.18 | YaroMan | Does any one use here BroadVoice with Asterisk @ Home? |
00:16.21 | riddlebox | Seldon1975:just looking at all possiblities |
00:16.41 | Seldon1975 | riddle: ok thx; can I get back to you I have the digium guy on the phone |
00:16.49 | riddlebox | yeah |
00:16.53 | SpaceBass | YaroMan yes - three accounts |
00:17.23 | *** join/#asterisk ManxPower (n=ewieling@29.sub-70-219-77.myvzw.com) |
00:17.32 | riddlebox | Seldon1975:he is probably smarter than me lol |
00:17.43 | SpaceBass | YaroMan keep it here... I may get called away and others may have better advice/help then I |
00:17.51 | SpaceBass | but yeah, glad to help YaroMan |
00:18.23 | YaroMan | Ok thanks!!! What I'm trying to do it is to learn how it is works before go in to production |
00:18.44 | YaroMan | I download Asterisk @ Home 2.2 Iso and installed on my Old PC |
00:18.50 | SpaceBass | YaroMan gotcha |
00:18.52 | YaroMan | P3 1 Ghz with 1 GB Ram |
00:19.17 | YaroMan | I was following this instructions http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm |
00:19.34 | twisted[asteria] | yuck |
00:20.08 | SpaceBass | ok... |
00:20.41 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
00:21.42 | [hC] | Using the exten => something/cid expression, can 'cid' be a wildcard match? I thought it could. |
00:22.22 | SpaceBass | YaroMan still there? |
00:22.30 | YaroMan | yeah |
00:22.40 | SpaceBass | did you have a question? |
00:22.44 | YaroMan | hold on on a phone with a client |
00:22.49 | SpaceBass | oh |
00:26.13 | YaroMan | i'm back |
00:26.17 | SpaceBass | k |
00:26.23 | SpaceBass | anyone have a sec to take a look at my zapata-auto ? http://pastebin.ca/34282 |
00:26.27 | SpaceBass | once someone calls zap/2 all calls take that context, even calls for zap/1 |
00:26.38 | SpaceBass | YaroMan so you haven't actually set up the box yet? |
00:26.41 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
00:26.55 | YaroMan | so i setup box allredy |
00:27.04 | SpaceBass | ok |
00:28.35 | YaroMan | i install but i have a truble with a trunk |
00:28.54 | SpaceBass | so you got a broadvoice BYOD account? |
00:30.31 | SpaceBass | open up an SSH session into that box so you can see the CLI |
00:30.41 | SpaceBass | type sip show registery and see what it says |
00:32.40 | YaroMan | ok hold on |
00:32.51 | YaroMan | [root@asterisk1 ~]# sip |
00:32.51 | YaroMan | -bash: sip: command not found |
00:33.08 | SpaceBass | YaroMan you have to be in the asterisk CLI |
00:33.10 | SpaceBass | type asterisk -r |
00:33.12 | YaroMan | i got BYOD bussines unlimited |
00:33.35 | YaroMan | Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 2631) |
00:33.42 | YaroMan | asterisk1*CLI> |
00:33.53 | YaroMan | I'm in ;) |
00:33.53 | SpaceBass | just the output of the command: sip show registery |
00:34.09 | Dr-Linux | my Cisco phone doesn't work behind a firewall, but the softphone are working, what things need to be checked? |
00:34.34 | SpaceBass | Dr-Linux using the same SIP account or 2 different ones? |
00:34.57 | YaroMan | asterisk1*CLI> sip show registery |
00:34.57 | YaroMan | No such command 'sip show registery' (type 'help' for help) |
00:35.04 | Dr-Linux | SpaceBass: same |
00:35.16 | Alric | its sip show registry, right? |
00:35.17 | SpaceBass | YaroMan sip show registry |
00:35.21 | SpaceBass | sorry I cannot spell tonight |
00:35.30 | YaroMan | k |
00:35.36 | SpaceBass | Alric only if yous spellz good |
00:35.40 | Dr-Linux | SpaceBass: talking to me? |
00:35.59 | YaroMan | asterisk1*CLI> sip show registry |
00:35.59 | YaroMan | Host Username Refresh State |
00:35.59 | YaroMan | sip.broadvoice.com:5060 8458671927@s 120 Request Sent |
00:35.59 | YaroMan | asterisk1*CLI> |
00:36.01 | SpaceBass | Dr-Linux not just then |
00:36.08 | SpaceBass | YaroMan behind a firewall? |
00:36.12 | YaroMan | yes |
00:36.15 | YaroMan | Router |
00:36.21 | Dr-Linux | SpaceBass: so what should i do? |
00:36.30 | Dr-Linux | SpaceBass: should i change the SIP account? |
00:36.40 | SpaceBass | YaroMan did you forward ports 5060 UDP and 10,000-20,000 TCP to your asterisk box? |
00:36.50 | YaroMan | nope |
00:36.55 | SpaceBass | Dr-Linux I'm thinking... not sure why 1 phone would work and 1 would not |
00:37.00 | SpaceBass | YaroMan thats your problem |
00:37.00 | YaroMan | i'll do it now hold on please |
00:37.47 | Dr-Linux | SpaceBass: one is softphone and otherone is Cisco hard phone :S |
00:38.24 | SpaceBass | Dr-Linux so the phones are behind a firewall dialing out to an asterisk box ... is the asterisk box also behind a firewall? |
00:38.38 | SpaceBass | Dr-Linux Try forwarding those same SIP ports to the IP of the cisco phone |
00:38.46 | SpaceBass | Dr-Linux suspect that may be the problem |
00:39.14 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
00:39.22 | Dr-Linux | SpaceBass: i have disabled the firewall |
00:40.18 | Dr-Linux | SpaceBass: my ip phones work good on public IP |
00:41.59 | SpaceBass | Dr-Linux so the cisco is getting a public IP? |
00:42.51 | Dr-Linux | SpaceBass: well, no i just load SIP firmware on it using public ip and it was working dialingout/in everything .. but now i put it behind a firewall and now its not working |
00:43.55 | *** join/#asterisk YaroMan_ (i=YaroMan@cpe-204-210-153-209.hvc.res.rr.com) |
00:44.06 | YaroMan_ | I'm back just restarted my router |
00:44.17 | SpaceBass | ok |
00:44.33 | SpaceBass | Dr-Linux exactly... thats the issue |
00:44.50 | SpaceBass | the firewall is preventing communication to the SIP (asterisk) server |
00:45.13 | SpaceBass | Dr-Linux if you have access to the Asterisk server you can try adding nat=yes to the configureation |
00:45.22 | SpaceBass | or trying forwarding the SIP ports to the phone |
00:45.26 | YaroMan_ | ok should I restart now astrisk box? |
00:45.29 | SpaceBass | YaroMan any luck registering? |
00:45.33 | SpaceBass | YaroMan no need |
00:45.39 | SpaceBass | YaroMan in the asterisk CLI type: reload |
00:45.47 | Dr-Linux | SpaceBass: i did nat=yes as well |
00:46.01 | Dr-Linux | SpaceBass: on the PBX server firewall is disabled |
00:46.03 | SpaceBass | anyone using a Cisco phone with more than one Asterisk box ? |
00:46.11 | Dr-Linux | SpaceBass: do it still need to forward the ports? |
00:46.19 | YaroMan_ | Done! |
00:46.22 | SpaceBass | Dr-Linux when you say "disabled" what do you mean? is it still behind NAT? |
00:46.53 | Dr-Linux | SpaceBass: no, PBX box is not behind the firewall |
00:46.59 | SpaceBass | Dr-Linux if the phone and the asterisk box are getting private IPs (IE not public) then you have a nat firewall |
00:47.08 | SpaceBass | Dr-Linux so the asterisk box has a public IP ? |
00:47.10 | *** join/#asterisk nswint (n=nswint@c-24-98-129-84.hsd1.ga.comcast.net) |
00:47.13 | SpaceBass | YaroMan get it working? |
00:47.15 | Dr-Linux | SpaceBass: yes |
00:47.22 | YaroMan_ | i dont know how can I check it? |
00:47.27 | Dr-Linux | SpaceBass: asterisk box has public IP |
00:47.34 | YaroMan_ | let me try to login with my phone |
00:47.34 | SpaceBass | Dr-Linux but the cisco phone has a private IP, correct? |
00:47.39 | Dr-Linux | and the Phone has pvt IP address |
00:47.42 | Dr-Linux | SpaceBass: yes |
00:47.48 | nswint | Hey you guys.. I blocked my first telemarketer with the blacklist and zapteller feature!!!! |
00:47.52 | *** join/#asterisk GXTi (i=realme@freenode/developer/GXTi) |
00:48.05 | SpaceBass | Dr-Linux try forwarding the SIP ports (udp 5060 tcp 10000-20000) to the phone |
00:48.17 | Dr-Linux | hhm.. |
00:48.20 | SpaceBass | (you can narrow the 10,000 - 20,000 in /etc/asterisk/rtp.conf) |
00:48.35 | Dr-Linux | SpaceBass: yeah these ports are there |
00:48.48 | YaroMan_ | nope does not work ;( |
00:48.50 | SpaceBass | SIP is notorious for not traversing firewalls well |
00:49.06 | SpaceBass | Dr-Linux they are there? are they forwarded to the phone or to your computer running the softphone? |
00:49.42 | Dr-Linux | SpaceBass: how can i verify, they are just in rtp.conf , but iptables firewall is stopped |
00:49.51 | SpaceBass | YaroMan what does the sip show registry |
00:50.06 | YaroMan_ | hold on |
00:50.19 | YaroMan_ | asterisk1*CLI> sip show registry |
00:50.19 | YaroMan_ | Host Username Refresh State |
00:50.19 | YaroMan_ | sip.broadvoice.com:5060 8458671927@s 120 Request Sent |
00:50.19 | YaroMan_ | asterisk1*CLI> |
00:51.10 | SpaceBass | Dr-Linux keep it in here... I may have to run soon |
00:51.48 | YaroMan_ | also wich password should I use the one I register with or the one under devices on broadvoice? |
00:52.30 | SpaceBass | the one from devices from BV |
00:53.11 | *** part/#asterisk nswint (n=nswint@c-24-98-129-84.hsd1.ga.comcast.net) |
00:53.24 | riddlebox | YaroMan_:it is a different password that you have to ask BroadVoice for |
00:53.25 | YaroMan_ | asterisk1*CLI> sip show registry |
00:53.25 | YaroMan_ | Host Username Refresh State |
00:53.25 | YaroMan_ | sip.broadvoice.com:5060 8458671927@s 120 Request Sent |
00:53.25 | YaroMan_ | asterisk1*CLI> |
00:53.28 | SpaceBass | Dr-Linux keep it in here... I may have to run soon |
00:53.52 | YaroMan_ | look like they support is not there ;( |
00:54.05 | YaroMan_ | I send them an email this mornign no replay |
00:54.13 | YaroMan_ | was trying to call same story |
00:54.16 | YaroMan_ | on hold for ever |
00:54.24 | ManxPower | ~docs |
00:54.25 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
00:54.28 | ManxPower | ~mailinglist |
00:54.29 | jbot | methinks mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
00:54.31 | SpaceBass | brb |
00:54.35 | riddlebox | YaroMan_:I have tried calling them too, but after 8 minutes it just hangs up on you |
00:54.47 | riddlebox | they usually respond to my emails within a day |
00:54.50 | YaroMan_ | no i was on a hold for good hr |
00:54.56 | riddlebox | ouch |
00:54.58 | YaroMan_ | I have at home Vonage ;) |
00:55.14 | riddlebox | I am using broadvoice |
00:55.28 | YaroMan_ | i just signup for broad voice last night |
00:55.33 | YaroMan_ | so i can try them |
00:55.50 | SpaceBass | YaroMan where were we? |
00:55.50 | riddlebox | I like them but they definetly need better customer service |
00:56.02 | YaroMan_ | SpassBoss |
00:56.03 | SpaceBass | you forwarded the ports right? and made sure 5060 was UDP ? |
00:56.15 | YaroMan_ | i ded reload ans |
00:56.15 | YaroMan_ | asterisk1*CLI> sip show registry |
00:56.15 | YaroMan_ | Host Username Refresh State |
00:56.15 | YaroMan_ | sip.broadvoice.com:5060 8458671927@s 120 Request Sent |
00:56.15 | YaroMan_ | asterisk1*CLI> |
00:56.27 | YaroMan_ | yes i did it too |
00:56.46 | Druken | ~pastebin |
00:56.47 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/ |
00:57.00 | SpaceBass | I think pasting less than 5 lines is ok... |
00:57.02 | *** join/#asterisk stormfr (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net) |
00:57.02 | SpaceBass | but thats just me |
00:57.20 | SpaceBass | YaroMan_ for some reason its not getting the registeration back |
00:57.37 | SpaceBass | what does your register string look like? (remember to use * for the password) |
00:58.01 | YaroMan_ | where can I find this? |
00:58.06 | YaroMan_ | I was using AMP |
00:58.21 | SpaceBass | Its the last field in APM |
00:58.23 | SpaceBass | at the very bottom |
00:58.51 | YaroMan_ | ok hold on |
00:59.07 | stormfr | hello, is there somebody here with experience with chan_bluetooth ? |
00:59.17 | YaroMan_ | 8458671927@sip.broadvoice.com:xxxxxxxxx :8458671927@sip.broadvoice.com |
00:59.47 | SpaceBass | make sure you don't really have a space after the password :845.... |
01:00.10 | YaroMan_ | i dont have any space ;) |
01:00.28 | SpaceBass | ok... b/c I saw that exact problem a week ago |
01:01.45 | Dr-Linux | awww |
01:01.50 | Dr-Linux | my phone is working now ;) |
01:02.39 | YaroMan_ | Dr-Linux what service do u use? |
01:03.57 | chris-fn | gentlemen, can anyone help me with an Asterisk::Manager (Manager.pm) problem? |
01:06.30 | YaroMan_ | peopel can some one help me solve a problem with broadvoice and sterisk |
01:07.41 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
01:09.59 | riddlebox | YaroMan_:did you sign up for the BYOD service with broadvoice? |
01:10.17 | YaroMan_ | yes I did |
01:10.26 | YaroMan_ | BYOD Unlimited Bussines |
01:10.32 | riddlebox | did they send you a registration email? |
01:10.35 | YaroMan_ | i got 2 phone numbers from thme |
01:10.40 | YaroMan_ | let me check |
01:11.02 | RyanW | I have a grandstream HT486 firmware 1.0.6.7 If i configure a secret in asterisk and also in the device. the device will not register. the error message is "SIP/2.0 401 Unauthorized" Any ideas? |
01:11.06 | YaroMan_ | BroadVoice Services Activation Confirmation |
01:11.53 | riddlebox | YaroMan_:so you have the sip registration info they gave you right? |
01:12.16 | YaroMan_ | no I find on a portal |
01:13.01 | riddlebox | YaroMan_:I was under the impression that they sent you the password and stuff that you need to connect asterisk to their network |
01:14.16 | YaroMan_ | if you login to account there and click on Account then My Devices |
01:14.24 | YaroMan_ | I find that inf there |
01:16.15 | riddlebox | YaroMan_:yeah I dont have anything in there except for my ATA device they sent me, I am going to mess with that in a few weeks, I just got asterisk working, now I want to test it then do that part of it |
01:16.56 | YaroMan_ | can you help me to configurate my asterisk please? |
01:19.20 | YaroMan_ | also one more quastion is it posible to run Asterisk on VPS server? |
01:21.00 | *** join/#asterisk jefrey (n=jefrey@202.190.203.200) |
01:27.34 | sbingner | wtf is UPS server? |
01:28.12 | denon | he said VPS |
01:31.44 | *** join/#asterisk infinity1 (n=brendon@solara.netcal.com) |
01:32.01 | infinity1 | cna you do elseif with ael? i've tried it with no luck |
01:33.51 | Qwell | infinity1: try else if? |
01:36.04 | *** part/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
01:36.29 | infinity1 | Qwell: it works. i'm on crack :) |
01:37.23 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
01:37.25 | *** part/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
01:40.25 | lilo | (whoops ;) |
01:40.27 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
01:40.43 | Qwell | lilo: What did you do this time? :p |
01:40.48 | mog_home | QWELL |
01:41.04 | Qwell | MOG!@ |
01:41.04 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
01:41.27 | mog_home | how goes it |
01:41.40 | Qwell | good, you? |
01:41.41 | YaroMan_ | Is it posible to run Asterisk on VPS? |
01:41.44 | mog_home | fantastic |
01:41.47 | mog_home | vps? |
01:41.58 | Qwell | sounds like an IBM server or something |
01:42.15 | mog_home | very pretty server? |
01:42.24 | mog_home | it runs on my cobalt which is quite sexy |
01:42.26 | Qwell | maybe |
01:42.57 | *** join/#asterisk colle (n=colle@c-232be155.27-1-64736c10.cust.bredbandsbolaget.se) |
01:42.59 | mog_home | its a tad slow |
01:43.03 | YaroMan_ | no on Virtual Private Server? |
01:43.05 | mog_home | but sense i added ram its a champ |
01:43.10 | mog_home | which is? |
01:43.24 | YaroMan_ | I have a server wich I use for VPS hosting |
01:43.28 | YaroMan_ | it is Dual Xeon |
01:43.30 | Qwell | ahh, A VPS |
01:43.31 | YaroMan_ | with 4 GB ram |
01:43.33 | YaroMan_ | and SCSI |
01:43.34 | Qwell | lame |
01:43.38 | Qwell | You do NOT want to do that. |
01:43.42 | mog_home | it will problably work just fine |
01:43.51 | mog_home | but qwell is probably right |
01:43.52 | Qwell | mog_home: it's like segmented servers |
01:43.56 | YaroMan_ | it has only 3 accounts there so far |
01:43.56 | Qwell | lame, lame, lame |
01:44.01 | mog_home | like uml? |
01:44.08 | mog_home | or jail? |
01:44.08 | YaroMan_ | no i use H-SPhere VPS servers |
01:44.15 | stormfr | hello, is there somebody here with experience with chan_bluetooth ? |
01:44.21 | YaroMan_ | they install CentOS |
01:44.53 | mog_home | probably not stormfr whats question thouhg |
01:46.08 | stormfr | mog_home : i have all working fine for make a call with AG, sip phone get the ring (as the mobile) but when the channels is answered, voice only go to mobile and not to sip phone |
01:50.39 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
01:52.02 | *** join/#asterisk Weezey (n=weezey@CPE001195cf5c03-CM0011e67c1f93.cpe.net.cable.rogers.com) |
01:52.30 | Weezey | realtime's telling me to check the debug, but /var/log/asterisk/debug isn't there |
01:52.42 | Weezey | what do I need to enable to generate debug stuffs? |
01:53.47 | stormfr | go to logging.conf and uncomment ;full to full |
01:53.54 | Weezey | thanks |
01:53.55 | stormfr | and type set debug 255 in CLI |
01:54.04 | stormfr | (as well set verbose 255) |
01:55.32 | *** join/#asterisk ronaldl79 (n=ILuv2Tra@c-24-8-54-203.hsd1.co.comcast.net) |
01:55.52 | ronaldl79 | Hello. |
01:55.55 | Weezey | hi |
01:56.04 | ronaldl79 | Hi, Weezey. How are you? |
01:56.20 | ronaldl79 | (Your name made me think of the 'Jeffersons' TV show) |
01:57.03 | Weezey | she spelled it Weezie |
01:57.12 | ronaldl79 | Anyone been following this Gizmo project? It's based on Jabber and Sip ... I think I'll sign up and connect it to * |
01:57.52 | mog_home | no its not based on jabber i thought |
01:58.02 | mog_home | but the jingle stuff is intresting as all hell |
01:58.37 | ronaldl79 | I just read that it was, who knows. |
01:58.52 | ronaldl79 | Google's Jingle, ahh, haven't digged into that yet, but I've read about it. |
01:59.21 | ronaldl79 | I'm glad Google chose to build their IM network on an open protocol -- it's the future of IM, imho. |
01:59.33 | mog_home | indeed |
02:00.00 | SkramX | Hiya |
02:00.02 | ronaldl79 | I was just thinking of someone implementing jingle into *. |
02:00.06 | ronaldl79 | Hi, Skram. |
02:00.28 | ronaldl79 | It would be nice to have the ability to speak with Google Talk users and clients built on Jingle. |
02:00.35 | mog_home | yup |
02:00.38 | ronaldl79 | Sweet, Mog. |
02:00.41 | mog_home | very |
02:00.42 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
02:00.53 | mog_home | it needs some tlc though |
02:00.56 | Nugget | google talk needs to hurry up and enable s2s |
02:00.59 | FuriousGeorge | hey all |
02:01.13 | mog_home | they arent nugget |
02:01.20 | mog_home | or at least i dont think they will |
02:01.24 | mog_home | but they should |
02:01.25 | Nugget | they say they are. |
02:01.30 | Nugget | but they're vague about it |
02:01.32 | mog_home | yeah thats what i hear |
02:01.32 | ronaldl79 | Hey Furious |
02:01.50 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
02:01.56 | SkramX | anyone heard of calpop.com ? Are they reliable (if you know of them( |
02:02.00 | ronaldl79 | Jabber is a neat protocol. |
02:02.06 | SkramX | My company is looking for a good place for a seconday server |
02:02.07 | mog_home | yup |
02:02.34 | ronaldl79 | How's your jabber project coming along, mog? |
02:02.40 | mog_home | quite well actually |
02:02.46 | ronaldl79 | Great. |
02:02.48 | mog_home | i hit amazing stuff just recently |
02:03.01 | mog_home | wish i had more time to work on it |
02:03.07 | mog_home | thankfully school is over for now |
02:03.14 | ronaldl79 | If I had a lot of money, I'd contribute to a lot of open source projects. |
02:03.34 | ronaldl79 | mog, have you checked out Gabcast? |
02:03.39 | mog_home | lol if i had more time id contribute more... |
02:03.41 | mog_home | yeah |
02:03.44 | mog_home | very creative |
02:03.45 | ronaldl79 | What do you think? |
02:03.55 | mog_home | will be much cooler when we hit wideband |
02:04.01 | ronaldl79 | I think so too -- I only got excited about it because of * being used on the backend. |
02:04.13 | ronaldl79 | Definitely -- I noticed the wideband branch. |
02:04.18 | FuriousGeorge | i got two wctdms one with 4 fxs and one woth 3 fxo. im loading the modules, and running ztcfg and nothing complains then when i start asterisk i get: chan_zap.c:920 zt_open: Unable to specify channel 5: No such device |
02:04.19 | mog_home | 8k mono doesnt sound amazing though |
02:04.30 | mog_home | but wideband might be goodenough |
02:04.33 | ronaldl79 | Wideband supports stereo, right? |
02:04.37 | mog_home | no |
02:04.37 | FuriousGeorge | but all the lights are on, on the back of the box |
02:04.42 | mog_home | 16khz audio samples |
02:04.45 | mog_home | as apposed to 8 |
02:04.58 | ronaldl79 | Yeah, I read that much about it...:P |
02:05.06 | ronaldl79 | I wonder if Digium will use Global IP Sound's architecture? |
02:05.12 | *** part/#asterisk loick (n=loick@APuteaux-151-1-26-46.w82-124.abo.wanadoo.fr) |
02:05.15 | mog_home | ? |
02:05.44 | ronaldl79 | Well, there seems to be a lot of companies using their technology for wideband voice. |
02:05.44 | FuriousGeorge | um, is the first bay on the second wctdm not bay 5? if not, what is it? |
02:06.15 | mog_home | indeed |
02:06.20 | mog_home | i dont know |
02:06.31 | mog_home | i doubt it though |
02:06.41 | ronaldl79 | I tell ya what, Asterisk is like a drug. |
02:06.44 | ronaldl79 | I'm addicted. |
02:06.47 | mog_home | lol |
02:06.50 | mog_home | very true |
02:07.11 | ronaldl79 | It's exciting to use such an open and flexible platform. |
02:07.30 | ronaldl79 | After almost two years, I'm still amazed... |
02:09.11 | mog_home | wow thats a long time |
02:11.05 | ronaldl79 | hehehe |
02:11.19 | ronaldl79 | Actually, maybe more like 1.5 years. |
02:11.35 | ronaldl79 | I'm still learning....got really serious about it the last year |
02:12.05 | mog_home | yup |
02:12.49 | ronaldl79 | But it's running everything voice now .... I was a Vonage and AT&T subscriber... and was spoiled by their 'pretty interfaces' ... which is why it took me sometime to adopt * for home use ... but, heck, I haven't looked back. |
02:13.07 | ronaldl79 | It's insane all you can do with * ... it's like a big toy to me that never goes out of style. |
02:13.21 | YaroMan_ | wow spent almost all day but cant make my asterisk to vork with BroadVoice |
02:13.35 | mog_home | i work all day long on pbx stuff but my pbx is so lame |
02:13.36 | ronaldl79 | YaroMan...I use BV on my * box. |
02:13.57 | ronaldl79 | Are you an * consultant, mog? |
02:14.07 | mog_home | in support/dev |
02:14.15 | ronaldl79 | Oh, cool. |
02:14.15 | mog_home | but my pbx does very little at home |
02:14.33 | mog_home | only cool code is the jabber stuff im playing with |
02:15.01 | ronaldl79 | I'm going to visit Digium next year ... |
02:15.08 | mog_home | heh |
02:15.12 | mog_home | we need to have tours |
02:15.17 | mog_home | its a pretty cool place |
02:15.29 | ronaldl79 | It would be a closer trip if I still lived in Nashville... |
02:15.38 | ronaldl79 | I'm in Denver now .... |
02:15.40 | mog_home | where you live ronaldl79 |
02:15.42 | mog_home | ahh |
02:16.02 | ronaldl79 | Yeah, I lived in Nashville for 3 1/2 years until this past April |
02:16.05 | mog_home | thats a nice place |
02:16.15 | ronaldl79 | I've accomplished more here for my business (Riverscape) than I did in Nashville |
02:16.19 | mog_home | should have stayed... |
02:16.31 | mog_home | why you comming to digium? |
02:16.32 | ronaldl79 | I love Nashville ... it's always a second home for me. |
02:16.55 | *** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
02:16.58 | ronaldl79 | To get a sense of the company's culture.... |
02:17.21 | Nugget | ronaldl79 just likes visiting dollywood and twitty city. |
02:17.21 | ronaldl79 | It's an acquisition interest |
02:17.48 | ronaldl79 | lol |
02:17.52 | ronaldl79 | Never been to dollywood |
02:21.04 | h3x0r | yes |
02:21.07 | h3x0r | oops |
02:21.27 | mog_home | wow that was cold... |
02:22.33 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
02:24.05 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
02:24.07 | PakiPenguin | morning |
02:24.33 | mog_home | morning |
02:24.39 | PakiPenguin | can anyone tell me whether i should go for channel bank or for ip phones? |
02:24.51 | [TK]D-Fender | IP Phones if at all possible. |
02:25.13 | [TK]D-Fender | Channel bank will cost you more and provide less functionality. |
02:25.13 | PakiPenguin | cool! and which ip phones do you suggest? |
02:25.22 | [TK]D-Fender | Depends ont he need. |
02:25.31 | [TK]D-Fender | What kind of setup? |
02:25.43 | mog_home | how many phones you setting upo |
02:25.47 | mog_home | are you already wired? |
02:25.53 | PakiPenguin | 20 |
02:25.54 | PakiPenguin | yeah |
02:25.56 | *** part/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net) |
02:26.10 | [TK]D-Fender | Yes, Will you be able to run a 2nd wire for the phone seperate from the computer's? |
02:26.12 | PakiPenguin | sorry 30 |
02:26.27 | [TK]D-Fender | What kind of call volume? |
02:26.39 | mog_home | i think most people are happier with ip |
02:26.40 | PakiPenguin | yes we can do that , the place is pretty neatly wired up , 2 ports on every desk for ethernet |
02:26.47 | mog_home | channel bank can be cheaper |
02:26.57 | [TK]D-Fender | Ok, I'd suggest Polycom 100% |
02:27.07 | PakiPenguin | call volume is ~100 calls a day per agent , incoming , outgoing |
02:27.14 | PakiPenguin | [TK]D-Fender, can you link me please? |
02:27.15 | h3x0r | i saw some funny shit the other day |
02:27.16 | [TK]D-Fender | Need speakerphone? |
02:27.21 | PakiPenguin | not really |
02:27.24 | *** join/#asterisk luisedo (n=luisedo@208.195.215.4) |
02:27.30 | h3x0r | a call center with a spa-1001 adapter and an analog phone on 30 desks. |
02:27.37 | h3x0r | lazy bastards |
02:27.42 | [TK]D-Fender | So a basic call center with say hard-wired headsets? |
02:27.49 | PakiPenguin | yeah |
02:27.54 | luisedo | hi every body |
02:28.17 | [TK]D-Fender | h3x0r : 1 thing to keep in mind : REN <- |
02:28.27 | luisedo | i'm looking for some kind of guru who can help this soul |
02:28.59 | [TK]D-Fender | PakiPenguin : Then I'd suggest Polycom IP 301's for your agents (the majority I suspect), and IP 601's for managers. |
02:29.16 | PakiPenguin | How much a IP301 cost? [TK]D-Fender |
02:29.27 | luisedo | does any one of you have worked with festival? |
02:29.34 | [TK]D-Fender | http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm |
02:29.37 | mog_home | i have luisedo |
02:29.42 | mog_home | whats the prob |
02:29.48 | *** join/#asterisk franx (n=Francisc@23-79-246-201.adsl.terra.cl) |
02:29.50 | [TK]D-Fender | $114 for the IP 301, $250 for the 601 |
02:29.56 | luisedo | hi mog |
02:30.03 | SkramX | 301 == okay? |
02:30.12 | Dr-Linux | my asterisk BOX is behind the firewall, canreinvite=no option in sip.conf should be set to no or yes? |
02:30.16 | luisedo | i'm trying to make an ivr wirh festival and asterisk |
02:30.17 | [TK]D-Fender | SkramX : for a minimal agent on headset, yes |
02:30.20 | mog_home | yeah |
02:30.24 | Dr-Linux | i can't hear ivr sounds etc |
02:30.27 | PakiPenguin | hmmms |
02:30.32 | luisedo | but when festival is loaded |
02:30.33 | [TK]D-Fender | All Polycom's are very solid phones. |
02:30.40 | mog_home | yeah |
02:30.43 | luisedo | loads the 100% of my processor |
02:30.51 | mog_home | you using punctuation? |
02:31.00 | luisedo | yep |
02:31.03 | mog_home | are you using the 8khz voices? |
02:31.10 | luisedo | festival ('test to say') |
02:31.22 | SkramX | festival sucks bro.. |
02:31.23 | mog_home | try taking out punctuations there is an old bug involved with punctuation |
02:31.25 | SkramX | seriously. |
02:31.27 | mog_home | but its not the best |
02:31.31 | luisedo | i duno, where cain i check if i'm using 8k voices? |
02:31.31 | mog_home | it works though |
02:31.35 | mog_home | and is funny if its not your biz |
02:31.37 | mog_home | if you dont knw |
02:31.41 | mog_home | you probably arent |
02:31.46 | mog_home | and thats your problem |
02:31.53 | mog_home | voip-info has a guide on it |
02:31.59 | mog_home | im sorry i dont know off of top head |
02:32.10 | luisedo | yesp ive read voip-info already |
02:32.37 | luisedo | i ve other problem |
02:32.40 | PakiPenguin | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43410781952.htm <-- is this phone any good? |
02:32.51 | PakiPenguin | the gxp - 2000 |
02:33.22 | [TK]D-Fender | PakiPenguin : I've heard is sounds poor, and well... it feels like cheap junk. Since you are talking about what seems to be a call center, get the Polycom's..... |
02:33.35 | PakiPenguin | oh okay |
02:33.48 | myke420247 | i have a bunch of gxp2000's |
02:33.49 | luisedo | as you can c (buecause of my very poor english :D) the ivr that i want to make must be in spanish |
02:33.49 | myke420247 | they're ok |
02:33.52 | myke420247 | no sidetone tho |
02:34.01 | luisedo | so i want to set spanish as the default language on festival |
02:34.58 | [TK]D-Fender | I don't think you'll haer a bad thing about Poycom. |
02:35.19 | [hC] | how to i REMOVE my temporary greeting if iv'e set one? |
02:35.31 | franx | hi, could you tell me where to find a guide for sip hardphones installation? |
02:35.37 | myke420247 | polycom is $$$ |
02:35.46 | luisedo | i'm using ubuntu 5.10 and the only scm file is the init.scm... where i tried adding the line (laguage_spanish) as it's said on the festival maual... but it's not working, when i load festival an error message stops the load procces |
02:36.13 | [TK]D-Fender | myke420247 : You do get what you pay for.... The feel, operate, and manage very well. |
02:36.14 | Katty | [TK]D-Fender: i'm making happy bread. |
02:36.29 | [TK]D-Fender | Katty : mew? |
02:36.41 | Katty | [TK]D-Fender: it's when you make a double batch. |
02:36.50 | Katty | [TK]D-Fender: and then, when it's done baking, take it to someone else |
02:37.09 | [TK]D-Fender | Katty : Strange to see you on at night too.... |
02:37.29 | Katty | [TK]D-Fender: not really. i'm just usually talking in other channels on other irc servers. |
02:39.36 | [TK]D-Fender | Ah.... |
02:39.43 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
02:39.52 | [TK]D-Fender | fine... stranger to see you HERE at night :) |
02:42.10 | Katty | (= |
02:43.31 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-89.cybersurf.com) |
02:44.13 | PakiPenguin | hmms |
02:44.13 | myke420247 | kitty katty |
02:44.16 | myke420247 | how many cats do you have? |
02:44.21 | myke420247 | <- 6 |
02:44.24 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
02:46.12 | Druken | AHHHHHH!!!!!!!!!!!!! |
02:46.42 | Katty | ... |
02:46.49 | shido6 | ahh the good old budgetone |
02:47.17 | luisedo | does anyone knows how to change the default language of festival? |
02:47.19 | Druken | how is Katty tonight? |
02:47.53 | Katty | mopey and depressed. |
02:47.55 | myke420247 | gxp != budgetone |
02:47.58 | Katty | hence the happy bread. |
02:48.13 | Druken | why depressed? |
02:48.23 | Katty | don't ask |
02:48.32 | Katty | and i don't want fixed either, so don't bother trying |
02:49.09 | shido6 | I was depressed |
02:49.14 | shido6 | then I sold my cisco 7960s |
02:49.25 | shido6 | and bought a trip to see the family |
02:49.45 | Druken | Katty: aight.. |
02:50.05 | Katty | kthx |
02:50.22 | PakiPenguin | myke420247, do you have a gxp? |
02:50.31 | myke420247 | paki, about a dozen |
02:50.41 | Druken | shido6: why you get rid of the cisco's? |
02:50.51 | Dandan | i have a gxp |
02:50.54 | shido6 | because I dont need them anymore |
02:50.57 | shido6 | Im selling 941's |
02:51.00 | Dandan | about 70 :) |
02:51.14 | myke420247 | damn |
02:51.25 | [TK]D-Fender | shido6 : and why are you selling 941's? |
02:51.26 | Dandan | yeah running on .13 beta :) |
02:51.27 | shido6 | I only used 1 line on the cisco's really |
02:51.30 | myke420247 | dandan, any way to get sidetone on them? that's the biggest complaint so far |
02:51.40 | shido6 | 941's are cheaper |
02:51.49 | Dandan | myke420247: no, same situation here |
02:51.51 | shido6 | and u can configure up to 4 line appearances |
02:51.52 | PakiPenguin | Dandan, are they nice phones? i mean quality / voice quality? |
02:51.54 | Dandan | i am happy with blf |
02:51.56 | shido6 | 1/2 the price of s 7960 |
02:52.04 | myke420247 | blf? |
02:52.07 | [TK]D-Fender | shido6 : when you say "selling", do you mean your personal ones, or as a business? |
02:52.13 | Dandan | PakiPenguin: well, I do not have any complains from my users |
02:52.16 | shido6 | business |
02:52.17 | Dandan | busy light |
02:52.27 | Dandan | and auto answer |
02:52.58 | [TK]D-Fender | shido6 : the 941 is a great little phone, but with what you can get a Polycom IP 501 for these days, its a hard arguement. |
02:53.07 | shido6 | my first sip phone here the budgetone works quite well with g729 on the NuFone net |
02:53.11 | PakiPenguin | thanks a lot [TK]D-Fender :) |
02:53.18 | shido6 | how much can you get polycom 501's for |
02:53.19 | shido6 | ? |
02:53.22 | myke420247 | yeah i have a barbietone at home |
02:53.32 | Dandan | i have evaluated a snom 190, poly 300 and those gxp's |
02:53.34 | PakiPenguin | barbitone :p |
02:53.40 | Dandan | and i liked gxpes the most :) |
02:53.58 | shido6 | yeah jerjer coined that one when he and I were screwing with the elcheapo phones |
02:54.10 | [TK]D-Fender | 170$USD. typically 941's =$150USD. For 20$ you get a better speakerphone, Polycom quality and pixl display, PoE possible, a 2nd eth port..... |
02:54.15 | shido6 | then more people started using the term and giggling even more |
02:54.34 | shido6 | now we're waiting for Mattel or Grndstream to really make one |
02:54.36 | Dandan | good one shido6 |
02:54.42 | [TK]D-Fender | I run 26 Polycom IP 600's and 1 x 601 at work. |
02:54.42 | luisedo | PLEASE people... you look you are experienced enough to helpme, i'm desesperate, and i've to give'em this work finished in less than 12 hours... |
02:54.59 | luisedo | does any one know how to change the default voice on festival? |
02:55.01 | Dandan | luisedo: we are all ears |
02:55.12 | luisedo | thanks Dandan |
02:55.21 | Nivex | [TK]D-Fender: where? The cheapest I've seen 501's new is $199 |
02:55.24 | Dandan | lemmie see :) |
02:55.29 | [TK]D-Fender | www.atacomm.com |
02:56.26 | luisedo | i've looked around everywhere but i can't find how to change the defoul language on festival, does anyone know how to make it? |
02:56.31 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
02:56.43 | Nivex | [TK]D-Fender: thanks! |
02:57.09 | *** join/#asterisk Zach^^ (n=Zachary@65.121.244.130) |
02:57.15 | Zach^^ | how do i seup incoming calls to load holiday greeting then load mainmen options? |
02:57.26 | Nivex | it's a bummer they don't let their firmware upgrades out... or is it difficult to convince them you're a reseller and get access? |
02:57.31 | Dandan | edit main menu :) |
02:57.54 | *** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
02:57.54 | Dandan | luisedo: did you get it? |
02:58.20 | Druken | rtp is udp right? |
02:58.31 | shido6 | http://festvox.org/docs/manual-1.4.3/festival_7.html |
02:58.32 | Nivex | Druken: yes |
02:58.36 | Zach^^ | Dandan i setup holiday greeting.. to play a short msg then i want it to auto move to the next menu |
02:58.39 | shido6 | --language LANG |
02:58.39 | luisedo | yes Dandan |
02:58.41 | Druken | Nivex: thought so :) |
02:58.46 | luisedo | i'm starting to read couse ma connection is too bad |
02:58.51 | shido6 | currently LANG may be one of english spanish or welsh |
02:58.59 | shido6 | depnding on what voices are actuallya vailable in your installation |
02:59.19 | Druken | welsh.. isn't that a wanabe english with a UK accent? |
02:59.25 | luisedo | yes i've tried --language spanish when i start festval server |
02:59.40 | luisedo | but it doesn't work when i use the command test2wave |
02:59.45 | Dandan | Zach^^: either edit the menu you are using now, or just point the context from to the new mnu and the GoTo to the old menu |
02:59.45 | luisedo | text2wave |
03:00.23 | Zach^^ | Dandan... i have it working but there is a long delay |
03:00.46 | Dandan | can you paste it somewhere? |
03:00.49 | Dandan | ~pastebin |
03:00.51 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.com/ |
03:01.04 | luisedo | off course |
03:01.08 | luisedo | give me one sec |
03:01.20 | Ateboy | hi, I'm about to install asterisk@home for testing, but I'm not sure since I heard some bad comments about that on this channel... |
03:01.30 | Dandan | lol that was for zach and luisedo :) |
03:01.45 | SkramX | Yes! I just got a small holiday bonus from the boss! |
03:01.49 | Ateboy | I must admit I don't know much about asterisk but that is why I'm experimenting... |
03:01.49 | Dandan | Ateboy: i can't tell you anything, i used 0.7 :) |
03:01.55 | [TK]D-Fender | Nivex : Its easy to get, just ask around. I have about 1/2 dozen versions myself |
03:01.59 | Dandan | SkramX: <= jealous! |
03:02.23 | Dandan | ~docs |
03:02.24 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
03:02.24 | SkramX | Dandan: It wasnt much.. and its not even a real job, just a small support job for a small web hosting company |
03:02.36 | Dandan | Ateboy: http://www.oreilly.com/catalog/asterisk |
03:02.45 | SkramX | I have my own hosting company as well.. but it is targeted to asterisk and linux guru's/intermediates. |
03:02.45 | Ateboy | already ordered |
03:02.50 | Dandan | SkramX: <= still jealous |
03:02.54 | SkramX | Dandan: :| |
03:02.59 | Dandan | :)))) |
03:03.00 | Ateboy | with "switching to VoIP" |
03:03.01 | Dandan | j/k :) |
03:03.06 | Dandan | good luck :) |
03:03.10 | SkramX | It was enough to buy a 941!! |
03:03.13 | Dandan | Ateboy: i have 3 books so far :) |
03:03.29 | SkramX | lol |
03:03.44 | luisedo | ready |
03:03.47 | luisedo | http://pastebin.com/471210 |
03:03.58 | luisedo | that's my whole extensions.conf |
03:04.00 | Dandan | Ateboy: http://geekgazette.com/index.php?option=com_content&task=view&id=40&Itemid=2 |
03:04.04 | Zach^^ | Dandan what file is it? |
03:04.11 | luisedo | extensions.conf |
03:04.17 | Dandan | Zach^^: extensions.conf? i do not know |
03:04.21 | Dandan | maybe you have some includes |
03:04.33 | Dandan | Ateboy: that book is free for download |
03:04.39 | Dandan | Zach^^: you should DL it too |
03:04.59 | Ateboy | I know, but since it was reasonably priced, I bought it to encourage the project or the author |
03:05.14 | Ateboy | why download? for easy search? I d/l it as well... |
03:05.17 | Dandan | luisedo: instead of background use playback |
03:05.33 | Ateboy | was waiting for a working asterisk setup to really go through it |
03:05.39 | Dandan | Ateboy: yeah, i strongly DISCOURAGE you from the YELLOW book |
03:06.06 | Ateboy | I read a few how-tos, the wiki... so many stuff... |
03:06.06 | Dandan | Ateboy: switching to voip is 50% theory |
03:06.16 | Ateboy | dandan: which yellow book? |
03:06.17 | Dandan | then they get down to business |
03:06.22 | Dandan | Ateboy: hold on :) |
03:06.27 | luisedo | Danda, i use background buecause i can press a number while i'm hearing the file |
03:06.29 | Ateboy | dandan: I should cancel my order? |
03:06.31 | Zach^^ | Dandan http://pastebin.com/471216 |
03:06.35 | Dandan | Ateboy: no |
03:06.40 | Dandan | those books from oreilly are good |
03:06.48 | Dandan | luisedo: to do what? |
03:06.53 | luisedo | and i dont have to wait to finish te audio file to press a number |
03:06.59 | Dandan | Zach^^: one sec |
03:07.14 | luisedo | an ivr |
03:07.27 | Ateboy | dandan: I don't follow you... I see you're busy though... I'll wait |
03:08.39 | [TK]D-Fender | I like my 941... |
03:08.44 | *** join/#asterisk ManxPower (n=ewieling@29.sub-70-219-77.myvzw.com) |
03:08.49 | SkramX | I should get one.. |
03:09.29 | Dandan | Ateboy: that Piece Of Shit: http://www.amazon.com/gp/product/0975999206/ref=pd_sim_b_4/102-5837222-8861720?%5Fencoding=UTF8&v=glance&n=283155 |
03:09.41 | Dandan | Zach^^: one sec, looking |
03:10.13 | Ateboy | dandan: Ok, first time I see this one. I'll stay far from it. |
03:10.44 | Dandan | Ateboy: i can recommend the oreilly ones i read them both |
03:10.54 | Ateboy | I should get them soon |
03:11.00 | Dandan | Zach^^: boy, your extensions.conf is messy |
03:11.11 | Zach^^ | Dandan i got it.... |
03:11.12 | Dandan | why is background as priority 9? |
03:11.14 | Zach^^ | next thing |
03:11.15 | Dandan | :) |
03:11.33 | Zach^^ | when i press # it says no directory entires match your search |
03:11.53 | Ateboy | what kind of setup could I do just to practice with only a dedicated asterisk computer, w/o buying hardware for now? install a softphone on 2 other computers on the network? |
03:12.05 | Dandan | include => app-directory |
03:12.20 | Dandan | Ateboy: i use sipps on two pcs |
03:12.25 | Dandan | and got a x100p clone from ebay |
03:12.29 | Dandan | fro 9.95 |
03:12.30 | Dandan | afair |
03:12.41 | Dandan | Zach^^: look in app-directory |
03:12.45 | Dandan | what is going on there |
03:12.55 | Dandan | Zach^^: is that *@home? |
03:13.00 | Zach^^ | yep |
03:13.06 | Defraz | In talking to my LD termination provider they want me to prefix the phone number with 2342432# can I use the # in there. It doesn't seem to work. |
03:13.08 | Ateboy | dandan: what is sipps ? |
03:13.24 | Defraz | Can I escape it with anything? |
03:13.30 | Dandan | do you know what nero is? |
03:13.44 | raeth` | lol AMP must hate my guts. |
03:13.48 | alephcom | Defraz: You shouldn't need to escape it with anything. I do something similar to that. |
03:13.51 | Dandan | Ateboy: http://ww2.nero.com/sippstar/enu/index.html |
03:13.57 | Zach^^ | Dandan http://pastebin.com/471225 |
03:14.37 | Defraz | hmmm yea amp hates it but I was curious if I could do it right in the extentions file instead. |
03:14.41 | Dandan | Zach^^: exten => #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS}o) |
03:14.47 | Dandan | check your contexts |
03:14.55 | myke420247 | is sippstar based on asterisk? |
03:14.56 | Dandan | (sorry I am no an *@home expert) |
03:14.57 | Zach^^ | ? |
03:15.04 | Dandan | myke420247: do not think so |
03:15.10 | Dandan | it is windows based |
03:15.17 | Defraz | I am going to prepend it with 6 digits then # then the number dialed (10 digits) |
03:15.22 | myke420247 | what's good about it, for $400? |
03:15.40 | Dandan | myke420247: hm, lets see :) |
03:16.22 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
03:16.29 | Dandan | gotta find a place to DL it |
03:16.29 | Dandan | :) |
03:16.32 | Zach^^ | Dandan if i call *411 it send me to the list and works |
03:17.01 | Dandan | Zach^^: what does your asterisk -rvvvvvv say? |
03:17.31 | Dandan | shmaltz: and...? |
03:17.40 | Zach^^ | Dandan when i hit # from the main menu? |
03:17.45 | shmaltz | Dndan, 3 work, 2 more to go |
03:17.49 | Dandan | Zach^^: yup |
03:18.10 | *** join/#asterisk jahani2 (n=k@adsl-19-47-192-81.adsl.iam.net.ma) |
03:18.10 | Zach^^ | <PROTECTED> |
03:18.10 | Zach^^ | <PROTECTED> |
03:18.10 | Zach^^ | <PROTECTED> |
03:18.18 | Dandan | good |
03:18.28 | Dandan | Zach^^: and...? |
03:18.33 | Dandan | (remember about pastebin) |
03:18.38 | Zach^^ | yea |
03:18.54 | shmaltz | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5838739062&rd=1&sspagename=STRK%3AMEWN%3AIT&rd=1 |
03:19.06 | Zach^^ | http://pastebin.com/471231 |
03:19.21 | shmaltz | well, the other 2 work as well, not a bad deal for $30.00 |
03:20.16 | Dandan | History:1 bid (US $19.99 starting bid) |
03:20.23 | Dandan | lol the bid war was fierce! |
03:20.36 | Dandan | Zach^^: you gotta ask *@home gurus |
03:20.47 | Dandan | i see that the agi directory had problems |
03:20.49 | Dandan | and exited |
03:20.58 | mog_home | asterisk@home guru... lol |
03:21.20 | Dandan | shmaltz: Very Irresponsible , Not worth doing bussiness with !!! NOT A GOOD EBAYER !! :))) LOL |
03:21.27 | Dandan | mog_home: oxymoron :) |
03:21.48 | Zach^^ | mog_home u can help ??? :D:D |
03:22.00 | shmaltz | Dandan, you talking about that stupid fb that guy left me? |
03:22.05 | mog_home | with? |
03:22.23 | Dandan | shmaltz: yah :) |
03:23.03 | Zach^^ | when i press # for the company dir... it say no directory entires match your search |
03:23.45 | shmaltz | Dandan, that idiot didn't answer my email for 5 days, and when it came to get the model number refused to do so. |
03:24.15 | Dandan | shmaltz: I know, eB is full of ppl like him :) |
03:24.20 | mog_home | whats your dialplan look like |
03:24.28 | Dandan | i got some fb from a$$h-les :) |
03:24.43 | Zach^^ | mog_home have to explain.. i am neb to asterisk |
03:24.50 | luisedo | this is my dialplan http://pastebin.com/471210 |
03:25.36 | shmaltz | Dandan, it doens't look realy good for me, because it's still on the first page, but since I'm only a buyer on eBay I guess ppl don't realy care. |
03:25.38 | Zach^^ | neb=newb |
03:25.46 | Dandan | luisedo: exten=s,3,Festival('Para escuchar un poema marque 4') LOL :) |
03:25.58 | shmaltz | the only time I ever wanted to sell something on eBay, they yanked it as an illegal auction |
03:26.00 | Dandan | shmaltz: don't worry pretty much all sellers disregard :) |
03:26.12 | shmaltz | Dandan, I realized that |
03:26.13 | Dandan | shmaltz: were you selling your kidneys? |
03:26.21 | luisedo | Dandan, what's wrong with that line? |
03:26.24 | shmaltz | Dandan, not realy |
03:26.38 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
03:26.41 | franx | i'i have a file called SIP000E08DAEA1E.cnf, should i upload it to the phone? |
03:26.43 | Dandan | luisedo: nothing, i am in a good mood really :) |
03:26.53 | Dandan | never encountered a press 4 for a poem |
03:26.56 | luisedo | ooooooooooooooooo ic ;) |
03:27.04 | shmaltz | I was trying to sell a hand held keymaker/cutter, they told me it's illegal because you need a locksmith license for that |
03:27.06 | luisedo | hahaha |
03:27.19 | shmaltz | in fact I'm still trying to get rid of it |
03:27.41 | luisedo | it's an example... what i must do is to give medical dates using voip |
03:27.47 | Dandan | shmaltz: I think if you are apprehended by pol1ce with thaose tools in your pocket you would get arrested :) |
03:28.04 | shmaltz | Dnadan, not the key cutter |
03:28.13 | luisedo | but i'll use festival to read the hour of the date or the name of the doctor or the pacient... |
03:28.25 | Dr-Linux | i faced this error "[root@i2c-pbx root]# Ouch ... error while writing audio data: : Broken pipe" |
03:28.32 | Dr-Linux | what could be happend :S |
03:28.38 | Dandan | i do not know how it works but i am pretty sure it is included in the paranoidal paradign of the police departments |
03:28.40 | shmaltz | Dandan, it also depends which state (for the picking set I have in my trunk), in my home state of NJ you could get arrested just for the pcking set |
03:28.45 | Dr-Linux | i'm unable to start asterisk :S |
03:28.48 | ManxPower | Dr-Linux, thats usually a mpg123 error |
03:29.02 | luisedo | of course when the db is ready... |
03:29.13 | Dandan | luisedo: nice :) |
03:29.24 | luisedo | it's suposed that we well joint the whole proyect in one week |
03:29.42 | Dandan | well good luck :) |
03:29.55 | Dandan | i just bought an usb sound card to use with asterisk |
03:30.01 | Dandan | anyone has any experience? |
03:30.48 | ManxPower | danalien, is it supported by Linux? |
03:30.53 | ManxPower | if not, send itback |
03:31.03 | Dandan | actually anyone used alsa for an overhead announcer? |
03:31.12 | mog_home | why you need a sound card |
03:31.13 | Dandan | ManxPower: yah, i checked that beforehand |
03:31.14 | Dandan | :) |
03:31.17 | Dr-Linux | i kill the mpg123 process, but still same happend:S |
03:31.31 | Dandan | mog_home: ^^ |
03:31.46 | mog_home | Dandan: |
03:31.47 | Dandan | Dr-Linux: which version of mpg do you have? |
03:32.17 | Dandan | mog_home: ? |
03:32.23 | mog_home | ??? |
03:32.31 | Dandan | mog_home: actually anyone used alsa for an overhead announcer? |
03:33.55 | mog_home | yeah |
03:34.04 | mog_home | its pretty simple |
03:34.11 | Dandan | do you have any scripts to share? :) |
03:34.30 | mog_home | no need for scripts |
03:35.07 | Dandan | i am just afraid that it doesn't work well with alsa |
03:35.13 | Dandan | i read that * prefers oss |
03:35.21 | mog_home | nah |
03:35.27 | mog_home | its the same |
03:35.31 | Zach^^ | mog_home any idea about the company directory problem? |
03:35.44 | Dandan | mog_home: ty |
03:37.42 | Zach^^ | mog u around? |
03:37.54 | mog_home | no clue its got to be a dial plan issue |
03:38.07 | Zach^^ | what file is the dialplan in? |
03:38.15 | Qwell | ~wikis |
03:38.16 | jbot | rumour has it, wikis is http://www.voip-info.org |
03:38.22 | Qwell | Zach^^: You have some reading to do... |
03:38.25 | mog_home | extensions.conf |
03:38.40 | luisedo | Zach, extensions.conf |
03:40.15 | Zach^^ | what do i need to look for? |
03:40.59 | shmaltz | Zach^^, the wiki |
03:41.32 | luisedo | you need to read a little... |
03:41.48 | shmaltz | Funny Quote of the Day - George Burns - "When I was a boy the Dead Sea was only sick." |
03:42.07 | franx | in order to get an ipphone working is it necesary to upload something there? |
03:42.11 | *** join/#asterisk darwin35 (n=kvirc@c-24-8-199-118.hsd1.co.comcast.net) |
03:42.13 | shmaltz | Funny Quote of the Day - Jay London - "It all started when my dog began getting free roll over minutes." |
03:42.20 | Dandan | shmaltz: yeah, indeed :) |
03:42.22 | shmaltz | franx, define there |
03:42.57 | franx | shmaltz: i mean to the phone |
03:43.09 | shmaltz | franx, what phone? |
03:43.14 | shmaltz | what phone you using? |
03:43.15 | franx | the ip phone |
03:43.33 | Dandan | franx: which one? |
03:43.36 | franx | the ip phone im trying to configure |
03:43.37 | Dandan | any pictures? |
03:43.46 | franx | spa 841 |
03:44.06 | franx | sipura spa 841 (sorry for the dumb answers, im kinda slow) |
03:44.10 | darwin35 | the 841 is easy |
03:44.20 | *** join/#asterisk andu (n=andu@S0106000476ee2cfe.cg.shawcable.net) |
03:44.23 | darwin35 | go to admin mode and advanced |
03:44.29 | darwin35 | goto line 1 |
03:44.29 | SwK | chan_zap.c:62:2: #error "You need newer libpri" |
03:44.31 | SwK | damn it |
03:44.40 | SwK | i just checked this shit outta svn |
03:44.40 | shmaltz | franx, log into the web page of the phone, by pointing your browser to http://x.x.x.x/ where x.x.x.x is the ip address of the phone |
03:44.53 | mog_home | did you install? |
03:44.58 | darwin35 | adn put in the proxy you connect to adn the usename adn password |
03:45.03 | shmaltz | darwin35, they never existed |
03:45.03 | SwK | yes |
03:45.18 | SwK | libpri # svn up |
03:45.18 | SwK | At revision 282. |
03:45.30 | Qwell | SwK: that's hot |
03:45.35 | mog_home | its probably nubbed somewhere swk |
03:45.40 | SwK | yeah |
03:45.43 | mog_home | unless cresl1n is jokin on ya |
03:45.45 | SwK | its just pissing me off |
03:46.24 | shmaltz | http://www.wired.com/news/culture/0,1284,69861,00.html?tw=rss.TOP |
03:46.34 | shmaltz | I think we should send the 841 for contest |
03:46.54 | Qwell | 841 shipped |
03:47.25 | franx | thanks a lot !!1 |
03:47.37 | Dandan | shmaltz: Duke Nukem Forever :) |
03:47.37 | shmaltz | Qwell, but it's discontinued, |
03:47.44 | andu | Hello does anyone know where I find some step by step docs about setting up the 'hint' priority ? I tried everything returned by google with no luck; show hints conctantly shows 0 watchers and state is Idle for anything but my polycom 600 |
03:47.46 | Qwell | doesn't mean it's not vaporware |
03:47.50 | Qwell | rather, doesn't mean it is |
03:48.46 | mog_home | ? |
03:48.51 | mog_home | 841? |
03:48.58 | shmaltz | I hope Zach^^ left because he was too busy reading :) |
03:49.09 | shmaltz | mog, yeah, whats wrong |
03:49.41 | mog_home | whats 841 Qwell |
03:49.56 | shmaltz | ~841 |
03:50.06 | shmaltz | jbot is ignorant |
03:50.11 | franx | hehe |
03:50.18 | franx | i have a login problem :/ |
03:50.32 | darwin35 | jbot sex |
03:50.34 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
03:50.37 | franx | what is the default user/password |
03:50.41 | mog_home | wow |
03:50.46 | mog_home | gross |
03:51.10 | darwin35 | for the 841 its blank |
03:51.20 | *** join/#asterisk bmg505 (n=leon@dsl-146-15-60.telkomadsl.co.za) |
03:51.31 | darwin35 | factory default |
03:51.44 | mog_home | whats 841 |
03:52.05 | franx | thanks |
03:52.12 | Dandan | ~sippura |
03:52.20 | rob0 | 741xx is Tulsa, OK; 641xx is KC, MO ... 841 must be further west |
03:52.21 | darwin35 | what ever the nmbr on the sipura hard phone is |
03:52.25 | Dandan | ~grandstream |
03:52.26 | jbot | 65USD /phone email to voipsales@xvoip.com |
03:52.42 | darwin35 | dont buy grandstream |
03:52.47 | darwin35 | not worth it |
03:52.47 | Dandan | whoa which one? |
03:52.58 | Dandan | darwin35: too late, already have 70 of them |
03:53.06 | darwin35 | for the same money you can get a open src pa168 based phoone |
03:53.10 | shmaltz | mog, whats gross? |
03:53.25 | mog_home | jbot sex |
03:53.26 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
03:53.28 | Dandan | darwin35: which one? |
03:53.37 | shmaltz | mog, I think it's funny |
03:53.43 | mog_home | it is |
03:53.50 | mog_home | the first million times i read it |
03:53.54 | darwin35 | I have the yughen ywh10 |
03:54.11 | andu | is there anything wlse required in sip.conf besides subscribecontext = ... to make use of the hint extension? |
03:54.16 | darwin35 | you can see them on iareaphone.net |
03:54.23 | andu | darwin35: I found those to be very slow |
03:54.45 | darwin35 | no not with frimware upgrades |
03:54.56 | shmaltz | well, I got this one once: |
03:54.58 | shmaltz | sex is like math, add the bed, subtract the cloths, devide the legs, leave your solution, and hope they don't multiply |
03:55.09 | mog_home | uh hu |
03:55.17 | andu | darwin35: for a phone I mean there's a lot of lag refreshing the lcd etc, the ATA are far better IMO |
03:55.51 | darwin35 | you have to turn off the verbosity in the software |
03:55.58 | darwin35 | it boots faster |
03:56.17 | darwin35 | but you have to know how . in the nnext ver I release it will be in fast boot mode |
03:56.25 | darwin35 | 1/3 the time to boot |
03:57.33 | andu | next ver like in the redfox firmware or something else ? |
03:58.07 | darwin35 | I use redfox src but modify it for my needs and turn off some things they leave on |
03:58.51 | darwin35 | thus far I like how it has turned out |
03:59.48 | darwin35 | but the only one I have is for the ywh10 |
03:59.58 | andu | do you have a d/l link ? I'd like to give revive my ywh10 |
03:59.59 | darwin35 | but the only one I have is for the ywh10/12 |
04:00.10 | darwin35 | I have the passthrew port |
04:02.04 | darwin35 | I will in a few let me finish my last patch |
04:02.09 | Ateboy | quick question: for a beginner, should I go with centos 3 (kernel 2.4) or 4 (2.6)? |
04:02.15 | andu | thanks darwin35 |
04:02.23 | darwin35 | I am working on now to get the mwi to work right |
04:02.45 | andu | lcd message ? |
04:02.57 | andu | or how did you go about it ? |
04:03.04 | darwin35 | yeah the count is wrong |
04:03.10 | darwin35 | and it should flash |
04:06.43 | darwin35 | right now it comes up with message:255 |
04:06.49 | darwin35 | it should count |
04:09.11 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
04:13.09 | *** join/#asterisk kimosabe (n=kimosabe@201.153.61.211) |
04:21.53 | Dandan | Ateboy: 3 |
04:21.53 | kimosabe | hi lol |
04:31.46 | Ateboy | 3 |
04:31.50 | Ateboy | 3? |
04:32.26 | Dandan | yeah |
04:32.26 | Dandan | 2.4 |
04:33.26 | Ateboy | dandan: rationale? |
04:33.58 | Dandan | i still consider 2.6 beta |
04:34.12 | *** join/#asterisk franx (n=Francisc@185-76-246-201.adsl.terra.cl) |
04:34.22 | Dandan | do you want to recompile all those modules every single time they release new kernel? |
04:34.22 | *** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
04:34.41 | Ateboy | well, I will use the CentOS stock kernel |
04:34.45 | Ateboy | *would |
04:34.49 | Dandan | oh |
04:35.00 | Dandan | i still have 2.4 on all my serious servers |
04:35.00 | Dandan | :) |
04:35.13 | Dandan | it just suits me better YMMV |
04:35.17 | Ateboy | I have 2.6 on my prod server... |
04:35.18 | Ateboy | eh |
04:35.27 | Dandan | :) |
04:35.37 | Dandan | i have 2.4 with grsec and I am happy :) |
04:36.01 | Ateboy | I see |
04:36.30 | Ateboy | I'm very insecure compared to you... |
04:36.38 | darwin35 | www.digitalgunfire.com |
04:41.00 | franx | if i have two x-lites (a and b) running behind the same subnet and i call an external x-lite c, how does my server resolve whether to send responses from c to a or b? |
04:41.26 | Dandan | franx: that depends on canreinvite= |
04:41.30 | Dandan | and your routing table |
04:42.03 | franx | does c connect directly to a or b? |
04:42.17 | *** part/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
04:42.19 | Dandan | franx: it all depends |
04:42.27 | Dandan | if you have canreinvite=no |
04:42.31 | Dandan | then your * box |
04:42.51 | Dandan | will route calls through the asterisk |
04:43.03 | Dandan | if your canreinvite is set to yes then |
04:43.17 | Dandan | your clients can establish independents call pathx |
04:43.27 | Dandan | that is without any * middleman :) |
04:43.49 | franx | im having troubles doing that, i can send audio from a or b to c but from c to one of them not |
04:44.37 | Dandan | is C behind NAT (that is on the other side of NAT)? |
04:44.49 | franx | yes |
04:45.09 | franx | a and b are behind NAT |
04:45.13 | franx | c is outside |
04:45.40 | Dandan | hm |
04:45.51 | Dandan | then read about: STUN, RTP UDP and problems with NAT |
04:46.00 | Dandan | voip-info.org is your friend |
04:46.08 | Dandan | ~stun |
04:46.09 | jbot | somebody said stun was that feeling you get when you realise your SIP call actually got through! |
04:46.18 | Dandan | :> |
04:46.25 | Dandan | ~nat |
04:46.26 | jbot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
04:46.39 | franx | thanks a lot for your time |
04:46.43 | Dandan | YEAH: See docs. |
04:46.46 | Dandan | no problem :) |
04:48.17 | franx | is there an official asterisk reference? |
04:48.56 | Dandan | official? |
04:48.56 | Dandan | well |
04:49.19 | franx | well something close |
04:49.26 | Dandan | I find the books useful as a general introduction to asterisk, but voip-info.org is indespensible |
04:49.26 | Dandan | :) |
04:49.38 | Dandan | ~books |
04:49.42 | franx | ill take that into note |
04:49.45 | Dandan | ~book |
04:49.46 | jbot | it has been said that book is on the table |
04:49.48 | franx | hee |
04:49.57 | Dandan | http://www.oreilly.com/catalog/asterisk/ |
04:50.03 | Dandan | that one and Switching to VOIP |
04:50.17 | Dandan | this one you can DL for free |
04:50.22 | franx | i see |
04:50.34 | Dandan | http://geekgazette.com/index.php?option=com_content&task=view&id=40&Itemid=2 |
04:50.46 | Dandan | but I encourage you to buy it :) |
04:51.14 | franx | ill do it |
04:51.30 | franx | oreily owns |
04:51.37 | Dandan | yes they do |
04:52.29 | franx | must go, once again thanks a lot and ill rember buying those books :D |
04:53.20 | mog_home | or dl em |
04:55.24 | Dandan | well |
04:55.26 | Dandan | if you buy |
04:55.34 | Dandan | that encourages writers to write |
04:55.48 | Dandan | the more they write the better the material they produce (hopefully) |
04:55.49 | Dandan | :) |
04:57.14 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
04:57.51 | Ayano | I have an asterisk server set up, and it wont dial from extension to extension. I don't even know were to begin |
05:00.06 | Dandan | hm does your context include both extensions? |
05:06.28 | Ayano | its an aah install. So it should. where do I check? |
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05:09.41 | Dandan | no idea aah does not apply here |
05:09.43 | Dandan | sorry |
05:09.43 | Dandan | :/ |
05:10.36 | Dandan | time to go |
05:10.37 | Dandan | [d] |
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05:25.00 | lehel | hey |
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05:46.31 | andu | Any have any experience setting up asterisk presence with a polycom 600 ? I went over every post I could find without any luck; any pointers would be greatly appreciated even if it's to more docs |
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05:54.32 | tecnico | Hi. Any hints on what the problem may be when I keep getting "No one is available to answer at this time (1:0/0/0)" . I thought at first that the problem was at my peer's, but I get the same error with all my peers when this happens. Teliax gives me a different error, it says "Call Rejected.... No such context/extension" . The strange thing is that this happens on the same system that runs OK for days, then I get these errors and few hrs |
05:55.02 | shido6 | contact your service provider |
05:55.18 | tecnico | ISP you mean ? |
05:55.34 | shido6 | voip service provider |
05:55.59 | tecnico | like I said, the problem is simulatenous with multiple VOIP providers |
05:56.11 | shido6 | same versions? |
05:56.25 | tecnico | Teliax, voxee, syxtel |
05:56.32 | tecnico | all three at the same time |
05:56.40 | shido6 | are they running asterisk? |
05:56.56 | tecnico | My guess is yes. |
05:57.10 | tecnico | I'm using IAX for all of them |
05:57.41 | shido6 | you're gonna end up at Shit Creek if you keep travelling up Assumption Rd. |
05:58.34 | tecnico | well, my point is that the problem seems to be on my side.. I wouldn't expect all three to fail at the same time with the same error |
05:59.41 | tecnico | ass soon as I dial.. the call gets accepted by the provider and not even a second later I get the error |
06:00.01 | benjk | shido6: how do I dial international on NuFone, 011 prefix doesn't seem to work |
06:00.17 | Qwell | benjk: They need to enable it on your account |
06:00.20 | Qwell | speaking of which |
06:00.28 | Qwell | shido6: Got a rates.csv which lists everything? |
06:00.55 | benjk | shido6: how do I get international dialing enabled? |
06:03.41 | shido6 | 30 days |
06:03.52 | benjk | shido6: how come I am getting charged for calls on a DID that JerJer says is not yet activated |
06:03.57 | shido6 | and a verification process |
06:04.23 | benjk | ok |
06:15.42 | shido6 | heh |
06:15.49 | shido6 | are you making test calls to this number? :) |
06:15.50 | *** join/#asterisk Igbothom_III (n=HiltonT@203-206-170-99.perm.iinet.net.au) |
06:16.06 | shido6 | call it from the PSTN for testing |
06:16.10 | shido6 | not through the network |
06:16.19 | shido6 | we provision our end immediately |
06:16.20 | benjk | I make test calls yes, but from a different provider |
06:16.37 | benjk | in fact from several different providers |
06:16.47 | shido6 | so whenever the legacy telco decides to get their act together we're set to go |
06:17.02 | benjk | I don't blame you |
06:17.25 | benjk | but it says 10 days and its been 12 days now |
06:17.56 | benjk | or is that business days? |
06:19.03 | benjk | if saturdays and sundays don't count, then it will be one more day |
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06:19.29 | shido6 | sat sun, or holidays |
06:19.47 | benjk | they don't count for the 10 days period? |
06:20.23 | shido6 | no |
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06:23.26 | benjk | ok, I didnt recall what the website page said, only remember "10 days" and now I can't see that page anymore |
06:23.40 | benjk | but thanks for the info |
06:23.53 | benjk | should come online tomorrow then |
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06:24.33 | michael123 | guys what is the best echo canceller |
06:24.38 | michael123 | software |
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06:32.23 | *** join/#asterisk AFK1 (n=itsme@203.81.209.175) |
06:32.55 | AFK1 | hii guys, need help???????? can anyone help me with a recording solution in which digium can be used for just tapping the lines? |
06:38.07 | mog_home | what AFK1 |
06:38.17 | mog_home | you can just do record |
06:38.21 | mog_home | before any dial |
06:38.43 | *** join/#asterisk Primer (n=vi@sh.nu) |
06:39.05 | AFK1 | mog_home: let me explain u the scenario, |
06:39.21 | AFK1 | i have a customer with 24 analog lines running ,, theere is no PBX |
06:39.31 | AFK1 | now they want to record these 24 analog lines |
06:39.39 | Qwell | ouch |
06:39.44 | Qwell | can't get a PRI? |
06:39.46 | mog_home | record |
06:39.48 | AFK1 | one option is i put 2 x 24 cards, use one for in and one for out, |
06:39.50 | Qwell | 24 analog lines must cost a ton |
06:40.01 | AFK1 | but it increases the cost alot by putting 2 x 24 card |
06:40.02 | mog_home | yeah |
06:40.10 | Primer | Anyone here have a cisco 7920? I've got this working with asterisk. At work our AP uses 40 bit WEP (I know it sucks but the boss doesn't care, and he set it up), but at home I use WPA. This thing supposedly supports WPA, but I simply can't get it to work using WPA. Anyone have one working with WPA? |
06:40.15 | mog_home | quite a pretty penny i imagine |
06:40.27 | mog_home | man i need a jabber expert |
06:40.34 | AFK1 | i want to know, if i can use 1 x 24 port card and just tap in the 24 lines? |
06:40.56 | drray | afk1 - through a punchdownd block |
06:41.01 | mog_home | this plain sasl stuff is pissin me off |
06:41.03 | Qwell | "tap in"? |
06:41.08 | mog_home | get a break out box |
06:41.13 | mog_home | they are like 50 |
06:41.20 | Qwell | I think he's saying like... |
06:41.28 | drray | get a punchdown block, they are like $24 |
06:41.29 | Qwell | use the fxo and a splitter |
06:41.42 | AFK1 | ahan, Qwell |
06:41.49 | Qwell | AFK1: That would be horrible to do |
06:42.36 | AFK1 | Qwell: sooooo :) any tap in solution that woul dbe able to work :) |
06:42.46 | mog_home | breakout box |
06:42.51 | Qwell | No, I'm saying...don't do that. heh |
06:43.19 | Qwell | AFK1: You have 24 analog lines, and 24 analog phones? |
06:43.34 | AFK1 | yeah thats true? |
06:43.41 | Qwell | I don't know. Is it? |
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06:44.05 | AFK1 | yeah yeah 24 anolog lines connected to 24 analog phones:) |
06:44.18 | Qwell | then you'll want 2 x 24 port cards |
06:44.51 | Qwell | or tell your telco you want a PRI (which SHOULD save you quite a bit of money per month), and get 1 x 24 port card, and like a TE110P |
06:44.53 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
06:45.10 | Qwell | I can't even imagine what 24 analog lines would cost |
06:45.13 | AFK1 | no :) i want to use 1 x 24 card to do this rather using 2 x 24 caerds :) |
06:45.16 | h3x0r | im gonna print some t-shirts that says "TDM SUCKS" |
06:45.26 | Qwell | AFK1: You can do that, but you won't be able to hook up your phones |
06:45.31 | Qwell | You need one port per device. |
06:45.34 | Qwell | device being a line, or a phone |
06:45.35 | drray | I heart my TDM |
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06:45.48 | h3x0r | on the front it will say "But ATM sucks more" |
06:45.49 | Qwell | So, in your case, you have 48 devices, which needs 48 ports |
06:45.55 | Assid | this is really funny, i cant get moh working on 1 box.. but works perfect on my home box |
06:45.57 | drray | swallows |
06:46.06 | Qwell | Assid: got the right version of mpg123 on said box? |
06:46.08 | AFK1 | okay Qwell thanks :) |
06:46.17 | Qwell | AFK1: So, in other words...you need two cards |
06:46.29 | Assid | yeah.. but im tyring to use format_mp3 |
06:46.58 | Assid | im actually gonna try using the same mp3 files on both.. to see why it hates me |
06:47.10 | Qwell | Assid: format_mp3 hates some mp3s |
06:47.19 | Qwell | make them 8khz mono, and they should work fine |
06:47.34 | Qwell | AFK1: is there any reason you guys aren't using a PRI? |
06:47.36 | Assid | i did.. it is @ 8khz mono |
06:47.44 | Qwell | Assid: no id3 tags? |
06:47.50 | Assid | nope.. all removed |
06:47.52 | Qwell | supposedly that matters |
06:48.48 | Assid | winamp shows the file at 8kbps/8khz/mono |
06:49.36 | YaroMan_ | Hello everyone |
06:49.52 | YaroMan_ | I need some help to setup my Asterisk @ Home with BroadVoice |
06:50.37 | drray | I've never used asterisk @ home |
06:51.39 | YaroMan_ | is it not the same as regular asterisk? |
06:52.33 | mog_home | not quite |
06:52.40 | mog_home | its a little lame |
06:52.51 | mog_home | like it was shot in the leg |
06:53.35 | YaroMan_ | mog_home what do you use> |
06:53.46 | mog_home | svn trunk |
06:53.53 | mog_home | actually not even that |
06:53.54 | mog_home | my trunk |
06:55.15 | YaroMan_ | nice |
06:55.17 | mog_home | that and a debian box, and a gentoo box |
06:55.20 | mog_home | some other junk |
06:55.25 | YaroMan_ | so you have your own numbers? |
06:55.41 | mog_home | i have a did yeah |
06:55.48 | mog_home | but i just run asterisk for my home |
06:56.02 | mog_home | otherwise i work on asterisk code and help other people get it installed and going |
06:56.42 | mog_home | qwell what are you still doing up so late |
06:57.33 | Assid | Qwell: yeah looks like that particular set of mp3 files |
06:57.36 | YaroMan_ | i went to the bar |
06:57.41 | YaroMan_ | hook up with few girls |
06:57.45 | YaroMan_ | and came back home |
06:57.52 | Assid | oh wait |
06:59.26 | newl | If you came home alone, you didn't "hook" anything. :D |
06:59.30 | Qwell | mog_home: it's only 11 |
07:00.06 | mog_home | its 1 am out here |
07:00.13 | mog_home | man you go to bed early |
07:00.18 | Qwell | heh |
07:00.19 | drray | only 11 here |
07:00.21 | Qwell | not usually |
07:00.29 | mog_home | i didnt realize there was a 2 hour difference |
07:00.40 | mog_home | well you always go to bed when its around 4 here |
07:05.07 | Assid | Qwell: if i have to increase the volume do i add -v2.5 to the input options or to the output? |
07:13.16 | Assid | Qwell: remade the mp3 files.. apparently they werent ready to run on my home box either |
07:14.15 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
07:18.29 | Assid | but i get this quite often : Dec 20 02:18:11 WARNING[14964]: layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 5 bits! |
07:24.06 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
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07:27.48 | uchman | Is there a way to redirect a call with my ZyXEL Prestige 2000W? |
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08:16.05 | YaroMan_ | ok now i can recive a calls but I can't make calls ;( |
08:17.12 | *** join/#asterisk eivindtr (n=eivindtr@062016241059.customer.alfanett.no) |
08:18.43 | eivindtr | Hi all. Does anyone know about a softphone that allows you to specify autoanswer for a specific set of CallerIDs, or other filtering mechanisms? |
08:30.26 | shido6 | brb |
08:30.31 | infinity1 | no |
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08:47.35 | dioptre | can anyone here explain whether i could use asterisk as a hole punching rendezvous server? or is it easier and better to just make one? using iax? |
08:49.35 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
08:49.48 | Chotaire | morning.. anyone ever played with meetme talker detection? or even heard of it? |
08:55.14 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
09:03.30 | *** join/#asterisk __a (n=__a@85.105.12.111) |
09:04.31 | __a | guys, any clues as to why a timeout parameter to Queue command wouldn't work? A Queue just doesn't timeout and keeps ringing... |
09:05.26 | *** join/#asterisk [Wiebel] (i=wiebel@cp883187-a.tilbu1.nb.home.nl) |
09:05.36 | [Wiebel] | hmm |
09:05.45 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
09:06.24 | zoa | heya oej |
09:06.32 | dioptre | can anyone here explain whether i could use asterisk as a hole punching rendezvous server? or is it easier and better to just make one? using iax? |
09:06.34 | zoa | hows berlin ? |
09:06.45 | oej_ | Berlin? |
09:06.51 | [Wiebel] | Hi |
09:06.52 | zoa | dioptre, what do you mean with hole punching ? |
09:07.00 | zoa | you are not at berlin with kpf |
09:07.00 | [Wiebel] | anyone here ever tried playing with a cisco 7970 + video? |
09:07.01 | zoa | ? |
09:07.08 | dioptre | using iax thru firewalls? |
09:07.12 | __a | zoa: any idea why wouldn't a queue timeout? |
09:07.13 | zoa | aaah |
09:07.22 | zoa | __a, i'd say a bug |
09:07.46 | zoa | i don't trust the queue things, they always proved buggy to me in the past |
09:07.47 | *** join/#asterisk ADRnLn (n=Will@cpe-024-028-251-234.triad.res.rr.com) |
09:07.59 | *** part/#asterisk ADRnLn (n=Will@cpe-024-028-251-234.triad.res.rr.com) |
09:08.53 | dioptre | can anyone here explain whether i could use asterisk as a hole punching rendezvous server? or is it easier and better to just make one? using iax through firewalls? should i just use a stun server? |
09:10.25 | zoa | dioptre, i still dont get it |
09:10.31 | zoa | if you want to go through a firewall use iax |
09:10.42 | zoa | if you want to have meetme (rendezvous ?) then use asterisk |
09:10.57 | dioptre | but does iax penetrate fws? |
09:11.06 | zoa | yes |
09:11.54 | dioptre | cool, so how do they interchange info ? like with stun? to get each others ip addresses? |
09:12.05 | dioptre | thanks heaps for responding btw |
09:13.05 | dioptre | does asterisk proxy all traffic for an iax conversation? |
09:13.42 | zoa | im sorry but i dont really get your questions |
09:13.43 | zoa | aaaah |
09:13.46 | zoa | i guess i get it now |
09:13.52 | zoa | iax works in a different way |
09:13.58 | zoa | it uses only 1 ports |
09:14.05 | zoa | making it really nat friendly |
09:14.10 | zoa | nat is probably what you mean |
09:14.29 | zoa | so asterisk will try to transfer the two ends to each other |
09:14.36 | zoa | and if that doesnt work, it will proxy the call |
09:15.04 | dioptre | omg great! |
09:15.23 | dioptre | thanks! |
09:15.23 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
09:15.30 | dioptre | ive read like a thousand pages to find that out :) ta |
09:15.57 | Assid | actually.. if im not mistaken atleast 1 box should be globally/directly accessible |
09:16.00 | *** join/#asterisk apardo (n=_apardo@54.Red-83-50-238.dynamicIP.rima-tde.net) |
09:16.02 | zoa | yes true |
09:16.04 | zoa | but thats normal |
09:16.08 | zoa | there's no other way |
09:16.21 | dioptre | o so it does need hole punching? |
09:16.34 | dioptre | like in sip? |
09:16.53 | Assid | similar.. |
09:16.56 | dioptre | or does asterisk do that? |
09:17.03 | dioptre | or can it do that? |
09:17.07 | Assid | but sip uses RTP.. can cause certain issues for some firewalls |
09:17.30 | dioptre | but if i used stun would it work u reckon? |
09:17.53 | Assid | it should. yes. if the stun is public |
09:18.15 | Assid | BUT.. iax is a much safer bet if 1 box is globally available |
09:18.26 | Assid | besides.. its iax.... effectively native to * |
09:18.42 | dioptre | ok - but if its p2p |
09:19.03 | dioptre | ...have to use stun eh? |
09:19.14 | Assid | the main question is.. can any of those peers be available directly |
09:19.26 | dioptre | probably not |
09:19.32 | Assid | stun wouldnt really do much then |
09:19.38 | Assid | atleast i dont think so |
09:19.47 | zoa | stun wouldnt do anything for that |
09:19.49 | dioptre | but thats what hole punching is for eh? |
09:20.00 | zoa | dioptre, no |
09:20.04 | Assid | only for the data connection |
09:20.06 | zoa | that wouldnt help you |
09:20.07 | Assid | nothing else |
09:20.17 | zoa | if your servers are both behind nat, they cannot find each other |
09:20.22 | zoa | whatever you try |
09:20.26 | zoa | you need something to proxy |
09:20.36 | zoa | and stun is not a proxy |
09:20.43 | zoa | stun is just telling a client its own ip |
09:20.48 | zoa | public ip |
09:21.03 | zoa | you don't need that with iax |
09:21.12 | dioptre | ok - so what if i had clients behind nats/fw and a asterisk server public? |
09:21.29 | dioptre | would that work like stun? |
09:22.03 | Assid | okay lets put it this way |
09:22.22 | Assid | stun is a tunnel.. but the car doesnt know where to go because it doesnt have a global destination |
09:23.25 | Assid | dioptre: why cant you just DMZ/port fw to the asterisk box? |
09:24.47 | dioptre | well it may not be possible for p2p |
09:25.02 | *** join/#asterisk heroine (n=heroine@lit75-2-82-225-244-34.fbx.proxad.net) |
09:25.06 | Assid | well.. even p2p needs a server where peers connect to |
09:25.12 | dioptre | true |
09:25.12 | heroine | hi |
09:25.26 | dioptre | but i was hoping that could talk with asterisk |
09:25.32 | dioptre | which would be public |
09:25.47 | dioptre | (and i hoped i didnt have to use stun) |
09:25.47 | Assid | if any asterisk box is public.. yes, you can do it |
09:25.54 | dioptre | and both clients could be behind fw |
09:25.56 | zoa | they can without using stun |
09:25.56 | Assid | without using stun. |
09:26.05 | Assid | iax |
09:26.24 | dioptre | would the asterisk proxy the conv? |
09:26.26 | dioptre | or tunnel? |
09:26.27 | zoa | yes |
09:26.33 | zoa | proxy |
09:26.44 | dioptre | is there a way to punch? |
09:26.57 | dioptre | or get info from asterisk to make a custom punch? |
09:26.59 | Assid | get 1 box to be publicly available |
09:27.05 | Assid | and you have a "punch" |
09:27.38 | dioptre | u mean one cant have nat/fw? |
09:27.41 | *** join/#asterisk reth (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb) |
09:27.52 | dioptre | (client) |
09:28.07 | Assid | the server needs dmz/fw if a client wants to access it remotely |
09:28.27 | *** part/#asterisk __a (n=__a@85.105.12.111) |
09:29.55 | reth | I get an error "405 Method Not Allowed" when my ata-box is registring. anyone know what to do? |
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09:38.32 | nfi|ermes | hi all |
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09:40.00 | dioptre_ | hey assid |
09:40.06 | dioptre_ | sorry i dropped out mate |
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09:44.53 | dioptre_ | well the idea is i need to assume a classic p2p scenario with 2 clients both behind fws/nats with access to a rendezvous server and stun and /or asterisk.... i was thinking of hole punching using no asterisk and a stun server - but was wondering whether i could get private addresses from asterisk - and if hole punching doesnt work then just let asterisk proxy |
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10:17.47 | Chotaire | good news.. new chan_capi for 1.2.1 should be finished within 2 days ;) |
10:18.17 | Chotaire | oej, you there? |
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10:24.28 | zoa | oej was here a second before |
10:24.49 | Chotaire | damn ;) |
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10:25.30 | zoa | he'll be back |
10:25.38 | Chotaire | that's ok. |
10:25.40 | Chotaire | I gotta discuss a tiny bug with oej about some rfc2833 madness. |
10:25.41 | kmilitzer | Hello ... |
10:25.44 | Chotaire | yup kai9 |
10:25.48 | zoa | Chotaire: can i bother you a little in private ? |
10:25.54 | Chotaire | zoa: sure, go ahead. |
10:29.04 | kmilitzer | I have some strange behavior, maybe someone can help me. I have my asterisk connected to a SS7 switch using chan_ss7 ... everything seems to work fine after some tweeking, but I am not able to send a ringing tone to callers |
10:29.52 | kmilitzer | If I use the m option in the dial comand the caller hears the music-on-hold as "early-media", but if i use r, nothing gets through |
10:30.15 | kmilitzer | Any ideas where to look for the problem? |
10:32.44 | mkl1525 | Hi, with "show queues" I can get some information abut the queues now I'd like to have this informations stored in a db and/or showed on a webpage - is there a way to get this values passed to an external script? |
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10:35.54 | Chotaire | I think yes. |
10:36.02 | Chotaire | (kmilitzer) |
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10:56.13 | Remosi | can I do a RoundRobin of SIP interfaces? Only Zap interfaces appear to support groups? |
10:56.33 | jalsot_ | hi |
10:57.17 | Remosi | hi |
10:58.09 | jalsot_ | does anybody know how much transcoding of g711-Speex/ilbc/GSM/g729 can be achieved with a single P4 3GHz box? |
10:58.35 | zoa | around 120 for speex ilbc and g729 |
10:58.41 | zoa | around some hundred with gsm |
10:58.46 | zoa | owz |
10:58.47 | jalsot_ | :o |
10:58.48 | zoa | half that |
10:58.52 | zoa | 1/2 |
10:58.54 | zoa | 60 |
10:58.57 | jalsot_ | really? |
10:58.59 | zoa | i thought dual xeon |
10:59.00 | zoa | yes |
10:59.12 | jalsot_ | so 60 concurrent calls with speex? |
10:59.19 | zoa | something like that yes |
10:59.23 | zoa | depending ont he settings |
10:59.33 | jalsot_ | I guess speex should be tuned for SSE |
11:01.23 | jalsot_ | that sounds great |
11:01.38 | jalsot_ | g729 uses less CPU, right? |
11:02.11 | RoyK | iirc that depends on the speex version |
11:02.21 | RoyK | the new cvs head stuff is supposed to be quite good |
11:02.34 | RoyK | but standard configured speex is sloow |
11:02.34 | jalsot_ | :) |
11:02.50 | RoyK | jalsot_: show translation recalc 60 |
11:02.55 | jalsot_ | how could it be tuned? |
11:03.04 | jalsot_ | I know that CLI |
11:03.14 | RoyK | ok |
11:03.16 | RoyK | see http://en.wikipedia.org/wiki/Speex |
11:03.24 | jalsot_ | however milliseconds don't tell mu how much calls can take |
11:03.32 | jalsot_ | or is there an equation? |
11:06.06 | jalsot_ | ok, my show translation recalc 60 gave: slin-speex 25 |
11:06.42 | jalsot_ | [but that's an old speex 1.0.4 Ubuntu] |
11:08.06 | tzafrir_laptop | we normally use speex for testing when we want to be sure * does some work. speex as a sure way to hig the CPU ;-) |
11:08.29 | jalsot | :) |
11:09.09 | jalsot | does anybody know if it is possible to calculate how many concurrent call be made from the translation table? |
11:09.38 | jalsot | somewhere I read that my 25ms would mean 40 calls [1s/25ms] |
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11:21.02 | cj-rm | does anyone here use jasterisk?? The Java<->Asterisk bindings?? |
11:24.41 | uchman | Java. :( |
11:32.11 | cj-rm | yeah, java... |
11:32.26 | cj-rm | RoyK: mmmm... nice |
11:33.22 | trixter | can you smoke some java in your wanpipe? |
11:42.41 | tzafrir_laptop | trixter, is that the reason for all of the jams you had in your connection? |
11:43.12 | trixter | I didnt have any jams in my connection |
11:43.21 | trixter | jam is made from fruit juice I like whole fruit like preserves |
11:43.24 | trixter | :P |
11:44.03 | trixter | actually that isnt quite true jam is half and half, jelly is fruit juice and preserves is more whole fruit.. but meh exlanations arent as funny |
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11:57.45 | jalsot | does anybody know how does the Intel IPP g729 codec perform in comparsion with Digium's g729? |
11:59.03 | zoa | i tried it |
11:59.08 | zoa | its +/- the same |
11:59.56 | jalsot | so Digium's version (did they write it or just distributing?) is not better optimized :( |
12:00.13 | zoa | no big difference between the two |
12:00.23 | jalsot | zoa: thanks! |
12:00.55 | zoa | but digiums is legal |
12:00.57 | zoa | and works on amd |
12:01.37 | RoyK | zoa: morning |
12:01.41 | jalsot | yes, I know |
12:04.03 | zoa | morning |
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12:05.51 | trixter | the ipp stuff can be legal if you pay the license fee, however its difficult to pay that given the new format for the licensing |
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12:06.33 | trixter | and then its questionable in certain countries (few) where algorithms, code, etc are not valid for patents - only physical devices of which this doesnt meet all the criteria |
12:07.25 | zoa | actually i think that if you pay to digium, you probably could get away with using ipp's |
12:07.47 | zoa | but that would require you not to use digiums then |
12:07.48 | trixter | you would have a valid license for the codec |
12:07.52 | zoa | but still questionalble |
12:07.59 | trixter | well not a total number greater than what you have paid for |
12:08.23 | trixter | because you arent afaik licensing specific code ... the problem with intels IPP code is that distributing anything off it requires an intel license as well |
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12:09.32 | trixter | one thing I do know for sure is that I have a chicken teriyaki rice bowl |
12:09.45 | zoa | true |
12:09.56 | jalsot | there is a licensing fee for the code/binary itself but you should pay royality fee as well, am I right? |
12:10.01 | zoa | yes |
12:10.53 | jalsot | I guess, getting per channel codec license is not cheap as well - probably it's not possible to buy just some |
12:11.02 | zoa | it is |
12:11.05 | zoa | with digium |
12:11.09 | RoyK | zoa: any news for the iax/sip jb? |
12:11.09 | zoa | 10$ / channel |
12:11.18 | jalsot | yep, I know that |
12:11.20 | zoa | i think its ready, let me yell to the other side |
12:11.26 | zoa | ah its not ready |
12:11.27 | RoyK | :) |
12:11.28 | RoyK | ah |
12:11.29 | RoyK | ok |
12:11.37 | RoyK | when? |
12:11.37 | zoa | seems like he has some coredumps with it |
12:11.43 | RoyK | hehe |
12:11.44 | trixter | well it depends on what you mean by licensing fee.. the binary can have one fee that goes to pay multiple things.. in theory it pays for digiums time to make a commercial product, it also goes to pay for the patent owner (the united nations - cause a percentage of the GDP of every country isnt enough) |
12:12.07 | trixter | I have no problem screwing the UN out of their license fee |
12:12.18 | trixter | they deserve it |
12:12.59 | jalsot | probably I will try to go with speex ;) |
12:13.11 | jalsot | or maybe ilbc |
12:13.21 | trixter | but that doesnt screw the UN out of their money |
12:13.26 | trixter | so what good is it :P |
12:14.17 | trixter | for those that dont know the UN owns the ITU that owns G.729 |
12:15.18 | trixter | and if you think that is bad if kofi (UN president) gets his way and takes control of the internet wait for the required use of patented protocols just to access the net or something else. The UN has not gotten involved in a project without charging additional fees because the billions of unaccounted dollars they receive annually just isnt enough |
12:16.45 | tzafrir_laptop | the ITU? there are a number of patent holders, right? |
12:16.45 | trixter | personally I would like to see a finance sheet of where my UN tax dollars go but that will never happen. |
12:16.56 | trixter | they are at least hte main one |
12:17.43 | trixter | my system is being squirley I cant look it up now for a list |
12:18.07 | trixter | but I know they are at least one and seem to be the front for it, aside from delegating license collection to some other company |
12:22.06 | jalsot | trixter: so you like g729 the most, right? :) |
12:22.17 | trixter | oh yeah |
12:22.19 | trixter | cant you tell? |
12:22.26 | trixter | I l;ove everything about it |
12:22.44 | jalsot | :D |
12:23.14 | jalsot | so it sounds better than anything else? |
12:24.36 | Ikarus | lol, configuring zaphfc is easy for BRI for me, don't hve to change a thing, although who ever defaulted nl might be a bit odd |
12:25.37 | trixter | it sounds better than default festival |
12:26.19 | jalsot | :) |
12:26.32 | jalsot | I've never tried festival... |
12:26.47 | jalsot | what in compare with speex or ilbc? |
12:26.51 | trixter | did you have speek and spell as a child? |
12:27.01 | trixter | er speak |
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12:27.32 | trixter | if you did, speak and spell sounds better than festival |
12:27.39 | trixter | especially to people with hearing loss |
12:28.19 | jalsot | hmmm, I should learn more English :) |
12:28.43 | trixter | speak and spell is a toy from the 1970s that would have kids spell words |
12:29.04 | jalsot | ahaaa |
12:29.09 | jalsot | :) |
12:29.45 | trixter | meaning that a fairly inexpensive toy from the 1970s sounds better, festival has some work to be done.. there are rumors that you can tweak it to sound a lot better, but that begs the question why doesnt festival just come with the enhancements? |
12:30.16 | coppice | everyone loves to attack festival. do you realise it is the basis for practically every commericial TTS |
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12:32.03 | iDunno | isn't it non-freeish? |
12:32.27 | tzafrir_laptop | well, it comes with Debian, so it must be free |
12:32.32 | trixter | I doubt that practically every commercial tts violates the gpl |
12:32.46 | coppice | festival is not GPL |
12:33.31 | trixter | what is it then? |
12:34.05 | coppice | sort of BSD like, if i remember correctly |
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12:35.40 | trixter | MIT/X Consortium License |
12:36.03 | coppice | pretty similar |
12:36.54 | trixter | when did festival start? |
12:37.45 | coppice | many years ago. it is one of the key research platforms for TTS, from one of the world's centre of excellence on the subject |
12:38.39 | mutilator | i could really go for a monterey chicken quesedilla right now |
12:38.41 | mutilator | anyone else? |
12:39.10 | trixter | well regardless it doesnt sound that good, I dont know anyone that would disagree with that, and if its the basis for other platforms those platforms prove that festival can be made better, which again and still and to repeat begs the question why doesnt it? |
12:39.13 | sivana | mutilator: ya |
12:39.56 | mutilator | no taco bell and only $6 in the pocket |
12:40.03 | mutilator | i think today is going to be mcdonalds dollar menu instead |
12:40.15 | sivana | heh |
12:40.17 | trixter | no del taco? |
12:40.20 | coppice | trixter: it has never tried to sound good. its a research platform. people just keep trying to use it for real systems. its only the voice itself which is poor. the rest is world class. |
12:40.21 | trixter | they are cheaper than taco bell |
12:40.28 | mutilator | theres only mcdonalds |
12:40.34 | mutilator | and a subway inside a shell gas station |
12:40.47 | Porks | hi :D |
12:40.54 | Porks | I have two X100P cards conected in PSTN (two lines). is it possible transfer for line two an incoming call in line one? |
12:41.21 | trixter | coppice: you are proving my point. if everything else is superior then why not upgrade the voice? my original comment was that you can hunt around to find upgrades and such to make it sound better, which still and always has and to repeat begs the question why hasnt it been so configured stock |
12:41.58 | coppice | can you find a better free voice to add to festival? |
12:41.59 | trixter | why make it seem inferior, becuase its for research is not a valid reason, that is an excuse -- just like saying 'we are all volunteer' when an open source project sucks horribly |
12:42.11 | trixter | that isnt the point |
12:42.17 | mutilator | because they don;t want to make THAT it easy for ppl to rip them off? |
12:42.43 | mutilator | anyone who has made a voice spent money to do it and won't give it away? |
12:42.43 | trixter | see now you have aparently run out of arguments and are just trying to say 'well its better htan nothing' which is like saying 'we are volunteers so our open source software sucks' which arent valid reasons |
12:42.56 | coppice | it is the point. the research people hav no motivation to give it a good voice. if you would like to make one and donate it, i expect they would be happen to accept |
12:42.58 | trixter | no the rumors are that you can find the info free here and there |
12:43.20 | trixter | nah its just an excuse |
12:43.25 | trixter | the same old open source crap |
12:43.53 | coppice | if you are too much of a lazy asshole to contribute, then you should expect nothing |
12:44.02 | trixter | look at handhelds.org they use the 'we are volunteers' a lot to cover things like lack of testing, improper coding, etc. you submit a bug about something they yell and scream saying 'how dare you find a bug' |
12:44.14 | trixter | and you are continuing that with your poor attitude |
12:44.21 | viperdude | hi guys |
12:44.24 | trixter | rather than have a real conversation about this you try to turn everything to your mental level |
12:44.26 | trixter | that of a child |
12:44.41 | mutilator | o_O |
12:44.47 | trixter | there is a difference between asking a real legitimate question and you calling names as a result and you coming up with a rather poor excuse |
12:44.49 | coppice | i don't have a poor attitude. you do. you are expecting others to do stuff that only benefits you, not them |
12:44.55 | trixter | calling names is being childish |
12:45.01 | mutilator | linux on the psp |
12:45.02 | trixter | which you proved that you do have |
12:45.05 | trixter | because you are a kid |
12:45.10 | trixter | which is quite funny |
12:45.22 | trixter | I have a poor attitude becuase you called me an asshole? |
12:45.26 | trixter | that makes sense |
12:45.37 | trixter | what I hear you saying is "I am rubber you are glue" |
12:45.44 | coppice | I said "if" you are |
12:45.44 | viperdude | what overhead does Asteirsk Manager access put on a server? I want multiple logins for CRM integration |
12:45.45 | trixter | again more childish nonsense from coppice |
12:45.59 | mutilator | coppice is eating right into this |
12:46.05 | mutilator | poor guy |
12:46.09 | trixter | becuase he is a kid |
12:46.11 | trixter | a little child |
12:46.14 | trixter | at least mentally |
12:46.24 | mutilator | erm no, you're just acting like a prick |
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12:46.43 | trixter | becuase he called me an asshole I am acting like a prick? |
12:46.46 | trixter | I can see how you would get that |
12:46.52 | mutilator | you wouldn't bash him with useless comments if you weren't sinking to 'his level' |
12:46.53 | trixter | he is a kid he doesnt know half of what he speaks |
12:47.01 | trixter | festival people have REFUSED any assistance in making the voice better |
12:47.09 | coppice | I want a licence like the GPL, but which allows an exception list for people like trixter. it really pains me to have to let people like that use the stuff I provide |
12:47.11 | trixter | yet he statses that if someone were to submit it they would want it |
12:47.18 | trixter | yet he states they dont want it becuase its for research |
12:47.28 | trixter | you provide nothing |
12:47.30 | trixter | nothing of value anyway |
12:47.36 | trixter | nothing anyone would actually want |
12:47.39 | trixter | of that I am sure |
12:47.44 | trixter | you provide names like asshole |
12:47.46 | trixter | who wants that? |
12:47.51 | trixter | you make excuses |
12:47.51 | mutilator | use sentences man |
12:47.52 | trixter | who wants that? |
12:48.08 | mutilator | show you actaully have some sort of IQ |
12:48.10 | trixter | you are obviously an idiot that is plain to see |
12:48.24 | trixter | the mere fact that I am typing proves that I have some sort of iq |
12:48.41 | trixter | however coppice has proven his worth tonight |
12:48.48 | trixter | woo hoo he reached new heights |
12:48.50 | mutilator | it's actually morning here.. |
12:48.58 | mutilator | been at work a good 2 hrs now |
12:49.05 | trixter | well good for you |
12:49.09 | trixter | that doesnt invalidate my statement |
12:49.16 | trixter | given frame of reference and all that |
12:49.38 | mutilator | yea |
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12:49.44 | mutilator | *shrug* |
12:49.58 | trixter | so coppice enlighten me what have you specifically used |
12:50.06 | trixter | er written |
12:50.11 | trixter | since you are the all mighty contributor |
12:50.28 | trixter | please fill us in we are dying to know |
12:50.39 | mutilator | i'm not particularly dieing to know.. |
12:50.48 | trixter | you were bragging about all that you have written that I am personally using so please tell me |
12:51.02 | trixter | yeah that is what I thought |
12:51.11 | mutilator | why do you insist on continuing with this trixter? |
12:51.18 | sivana | ya |
12:51.52 | mutilator | i see no reason why it matter whether he lied or not, he doesn't have to validate anything to you or anyone else |
12:52.09 | sivana | except me |
12:52.14 | mutilator | well ofcourse |
12:52.18 | mutilator | but anyway |
12:52.52 | mutilator | man we took our total control modem banks down last night to update em |
12:52.58 | mutilator | and it wrote some f'd up radius logs |
12:53.06 | mutilator | 389hour connections and stuff |
12:53.12 | mutilator | like 1200 records were messed up |
12:53.36 | mutilator | hope i don't need any of those for anything |
12:55.32 | *** join/#asterisk fugitivo (n=ajf@209.13.244.9) |
12:55.41 | fugitivo | morning |
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12:56.22 | mutilator | mornin |
12:56.55 | trixter | oh that reminds me I need to install my *free* cepstral liceses becuase at least that is better sounding than festival at least some good came of this |
12:57.16 | fugitivo | yeah, cepstral is better |
12:57.16 | trixter | and free |
12:57.18 | trixter | at least for me |
12:57.22 | trixter | :) |
12:57.27 | fugitivo | why free? |
12:57.29 | fugitivo | i paid for it |
12:57.31 | coppice | sure. its festival with different voices |
12:57.35 | mutilator | he's 1337 |
12:57.37 | trixter | I asked if they would givem e a free license and they did |
12:57.52 | fugitivo | oh :) |
12:58.01 | fugitivo | coppice: no, it's not festival |
12:58.18 | fugitivo | big difference |
12:58.31 | trixter | he was refering to the back end code |
12:58.32 | coppice | it is. slightly modified, but festival. look at who supplies it, and look at who wrote festival |
12:58.58 | fugitivo | well, it sounds better than festival |
12:59.01 | fugitivo | not only for the voices |
12:59.24 | mutilator | coppice: don't prove him wrong.. his world will fall apart.. |
12:59.31 | trixter | ibmTTS is free too if you use the program I wrote which if I recall is available on my page |
12:59.58 | trixter | and I think that one sounds better than cepstral |
13:00.44 | mutilator | man |
13:00.52 | mutilator | even cepstral can pronounce my name correct |
13:00.53 | trixter | course the code I wrote is really just an asterisk integrator, nothing more |
13:00.54 | mutilator | last name* |
13:01.04 | mutilator | like.. not even humans can do that |
13:01.11 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
13:01.19 | trixter | mutilator: what nationality is your last name? |
13:01.24 | mutilator | german |
13:01.37 | fugitivo | hehe |
13:01.47 | trixter | ahh.. is it a vowel thing that causes the parsers to choke? |
13:01.53 | trixter | or something else? |
13:01.55 | mutilator | gusler |
13:02.09 | fugitivo | that's easy |
13:02.14 | mutilator | most people say guzzler |
13:02.15 | TheCops | how you can add licenses for a g729 codec ? (I already have one, but I dont know how to add some licenses) |
13:02.15 | trixter | most americans would say gus-ler |
13:02.27 | mutilator | or goosler |
13:02.47 | mutilator | it's rare someone says gus-ler |
13:02.49 | trixter | should it be goo-sler? |
13:02.55 | trixter | is gus-ler correct? |
13:03.00 | mutilator | ya |
13:03.14 | trixter | huh ... anyone I talk to on a regular basis would have said it that way |
13:03.28 | trixter | how else can you say it? its not a ts no z would seem inappropriate |
13:03.35 | mutilator | heh, not any techsupport/phone service of any kind |
13:03.55 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
13:03.56 | trixter | well in all honesty most of the call centers I have called arent english natives |
13:04.09 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:04.15 | trixter | so I can see why they would have a problem with german names |
13:04.17 | mutilator | well this is talking to like |
13:04.18 | mutilator | teh telco |
13:04.23 | trixter | people always toss extra letters into my name |
13:04.23 | mutilator | and electric co |
13:04.44 | trixter | both first and last and many cant say my last name correctly becuase they dont read what is spelled instead insist they know better than me how to spell the name |
13:04.56 | trixter | one person when I was a kid even told my mom she misspelled it |
13:05.02 | trixter | and insisted on correcting it for her |
13:05.40 | mutilator | heh |
13:06.28 | fugitivo | well, english natives have big problems with latin names |
13:06.42 | trixter | my name is scottish more or less |
13:06.54 | fugitivo | i'd like to hear how you say "fugitivo" ;) |
13:07.00 | mutilator | i guess back in the day mine used to be gossler |
13:07.06 | mutilator | and somehow turned into gusler |
13:07.17 | trixter | you would like to think that they could at least spell and say those names its not anything highly weird they can say mcdonalds well enough which is similar |
13:07.42 | trixter | ours changed when immigration in america couldnt spell |
13:08.10 | trixter | mutilator: went to school with a girl whose last name was 'kress' or something and that was german originally and got changed for the same reason |
13:08.26 | Porks | fugitivo... is it pt_BR? |
13:08.58 | fugitivo | Porks: pt_BR, it_IT, es_ES, es_MX, and a lot more |
13:09.13 | Porks | ah! ;) |
13:09.36 | coppice | fugitivo: well I guess its nothing like fugitive-o. |
13:09.39 | trixter | fugitivo: I live in mexifornia there are a lot of latin names particularly es_mx so people here generally can handle latin names :P |
13:10.08 | trixter | fugi-tivo the next generation pvr! |
13:10.19 | fugitivo | english natives like to say the last O like OU |
13:10.33 | fugitivo | so in spanish it'll sound something like fujitivouu |
13:10.51 | fugitivo | the g doesn't sound like the english g |
13:10.56 | trixter | most of that is where they put the accent |
13:11.13 | mutilator | first time he's worked graveyard 12-8 shift |
13:11.27 | trixter | the same problem exists with japanese |
13:11.36 | mutilator | and i'm going to be upgrading the sql server tomorrowso the site & radius will be down for 5-10 minutes |
13:11.46 | fugitivo | japanese and spanish have a similar phonetic |
13:11.47 | mutilator | so i'm not going to tell him i'm doin it and not answer my phone.. see what he does |
13:11.53 | trixter | although there are differences between english and japanese that gets a lot of english speaking people.. tori means bird torii is a gate (like the big red one at kyoto) |
13:11.54 | mutilator | muahaha |
13:12.04 | trixter | the double vowel means you say that sound slightly longer than a single vowel |
13:12.08 | trixter | a concept english doesnt have |
13:13.07 | trixter | there are also differences in how much you actually say a sound.. du desu ka - how are you the desu is pronounced more like 'des' with just barely saying the final 'u' |
13:13.14 | [TK]D-Fender | However english has a seperate future tense which japanese doesn't. See who's got the eye on the future? ;) |
13:13.17 | coppice | fugitivo: so, the f is not like f, and the u is not like u, and the g is no like g, and the i is not like i, and the t is not like t, and the i is not like i, and the is not like v, and the o is not like o, but other than that, your name is real easy for english speakers. right? :-) |
13:13.48 | fugitivo | coppice: :) |
13:13.51 | [TK]D-Fender | coppice : The "p" is silent.... like in swimming. :D |
13:14.28 | fugitivo | the nice thing about spanish, is that your say what you read, if you know how the letters sound, you can say any word |
13:14.50 | coppice | fugitivo: its the same with chinese |
13:15.06 | fugitivo | chinese is a pain in the ass :) |
13:15.53 | [TK]D-Fender | Much like Inuit... too many damn vowel / W sounds in a row :) |
13:15.53 | coppice | except if you speak cantonese, you'll say something completely different than if you speak mandarin :-) |
13:15.53 | trixter | english is a bastard language.. vowel substitutions to change tense is norse influenced.. run/ran where that is an exception becuase normally you add a suffix to make it past tense, runned for example, incorrect but otherwise follows the rules. a common mistake for a kid to make because they havent learned all the exceptions |
13:16.04 | *** join/#asterisk gevious (n=chatzill@196.31.11.194) |
13:16.18 | gevious | Hi Guys |
13:16.26 | fugitivo | coppice: doesn't chinese have different meanings for the same sound with different tone? |
13:16.35 | coppice | that's one of the three key reasons why english is so successful |
13:16.37 | gevious | Has anyone used gnudialer with Asterisk 1.2.1 |
13:17.04 | coppice | fugitivo: if the tone is different, its not really the same sound, is it? |
13:17.30 | gevious | Sorry I know this is an asterisk channel, but I am having some difficulty with Gnudialer and no one is ever on #gnudialer |
13:17.31 | fugitivo | coppice: not the sound, but it's the same word |
13:17.46 | fugitivo | gevious: now this is #languages |
13:18.10 | coppice | ah, what what does the same word mean? I can hear it sounds different |
13:18.19 | gevious | I see that |
13:18.29 | trixter | personally I think english is successful becuase to deal with america you need ot know english and its often easier for someone else to learn english than it is for americans to learn a foreign language |
13:18.41 | fugitivo | coppice: in spanish if you say the same word with different tones, it means the same thing |
13:19.03 | gevious | Anyone know where I can get Gnudialer help? |
13:19.09 | gevious | Besides gnudialer.org |
13:19.21 | gevious | Absolutely no doc on gnudialer anywhere |
13:19.45 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:19.53 | trixter | fugitivo: in english some words can have totally different meanings if you say them with different inflections.. take the word dude.. depending on how you say it it can mean anything from hello to you really messed up to are you in the closet with a knife |
13:20.07 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:20.08 | coppice | 3 reasons for english dominance: |
13:20.10 | coppice | The British spread it around their empire |
13:20.11 | coppice | America speaks it, and its dominant today |
13:20.13 | coppice | it flexible, and people munge it around to taste |
13:20.13 | trixter | gevious: what speciifcally are you having problems with? if you never say what you specific problem is no one will ever help you |
13:20.14 | coppice | I think if you take any one of those three away it wouldn't be nearly as widespread as it is today |
13:20.58 | gevious | I have got it compiled, but keep getting "terminate called after throwing an instance of 'xFileOpenError'" when I execute gnudialer |
13:21.12 | fugitivo | trixter: in chinese, you can say for example "a" in different tones, and it'll have different meanings |
13:21.22 | fugitivo | that's why chinese is so hard to learn |
13:21.48 | trixter | reading it isnt that hard though |
13:21.51 | mutilator | i say just force english on everyone |
13:21.59 | trixter | I have more problems reading arabic than chinese or japanese |
13:22.13 | coppice | fugitivo: that isn't actually a hard part of learning mandarin, but its a rather hard part of learning cantonese |
13:22.16 | gevious | I have traced it down to the addGlobalSettings function in queues.h but am stuck now |
13:22.17 | trixter | not sure if its the right to left part or if its the fact that the characters all look almost identical |
13:22.20 | fugitivo | trixter: but can you speak it correctly? |
13:22.33 | trixter | speak which? |
13:22.37 | fugitivo | chinese |
13:22.40 | trixter | which? |
13:22.46 | fugitivo | cantonese |
13:22.46 | mutilator | new |
13:22.55 | trixter | no I never learned to speak cantonese |
13:23.15 | trixter | I relaly only learned some chineese becuase of my desire to learn japanese |
13:23.22 | fugitivo | coppice: mandarin is different in that aspect? |
13:23.25 | trixter | some of the characters are the same, some arent |
13:23.32 | trixter | so it gave me a little boost |
13:24.18 | coppice | fugitivo: both use tones, but they don't cause a big headache with mandarin. they do with cantonese. learning to read chinese is actually easy. it just looks hard |
13:24.28 | trixter | gevious: that is beyond anything I know, not ever having used gnudialer.. initially I would have thought that it was a permission setting but if its queues.h that makes me wonder |
13:25.39 | fugitivo | coppice: i started some time ago to learn chinese, and the teacher told me "you'll not learn to speak it correctly, you'll only learn to write and read it" |
13:25.43 | mutilator | *NINJA VANISH* |
13:25.46 | trixter | are you sure that its writing to the corerct dir and that its running with enough permission to write properly to that dir? that would just be a guess I have never used gnudialer so I cant really say |
13:25.47 | mutilator | pewf |
13:25.50 | Remosi | I have several SIP FXO's, when I'm making an outbound call I want to choose a free fxo at random. At the moment I'm looking at using queues.conf, am I missing something? is this possible? |
13:26.10 | coppice | fugitivo: I have no idea what that means :-) |
13:26.23 | trixter | fugitivo: typically if you dont learn a language before age 6 you speak with an accent, you can overcome that but *most* people have a problem with that |
13:26.36 | trixter | they will always speak it with some accent if they learned a specific language after that age |
13:26.43 | trixter | after 13 makes it even harder |
13:27.03 | trixter | it could be that is what your teacher was refering to |
13:27.22 | fugitivo | i think she was refering to the tones |
13:27.47 | coppice | not really. many people learn languages in adult life, and are completely free of accent. it depends more on the person than the age. the speed of learning is massively faster below 6, though |
13:27.48 | fugitivo | in spanish, you say exactly what it's written, we don't have tones |
13:28.34 | mutilator | accents are cool |
13:28.45 | mutilator | move to the south for a year and then move back north, everyone says ya talk funny |
13:28.48 | coppice | we have tones in english. they provide punctuation. chinese can't do that, so they actually use words for punctuation |
13:28.49 | trixter | according to linguistic studies (something I have read up on personally for fun) there is an accent in the majority, I did allow for some (the minority of all that speak multiple languages) to learn it without accent, but ... |
13:28.51 | mutilator | and ya don't even realize it |
13:29.23 | trixter | I pick up local accents fairly quick, when I moved from texas to new jersey it was unknown that I lived in texas for 7 years |
13:29.43 | fugitivo | trixter: lol |
13:29.57 | [Wiebel] | trixter: well that's something you want to hide ;) |
13:30.19 | mutilator | the cowboy hat and boots and tight pants probly gave it away anyway tho |
13:30.21 | coppice | the speed with which a 4 or 5 year old can pick up a new language is amazing |
13:30.35 | trixter | until they made me take a dancing class in high school (*right* after moving up) and they wanted us to do square dancing and I said 'no one does that why do we have to learn it' and the teacher said 'you may go down south sometime' and I said 'I just came from there no one does it' at which point she started saying something about how open source developers are volunteers or something and they dont have to create with the same quality.. something lik |
13:30.35 | trixter | e that |
13:31.00 | trixter | mutilator: hey I livedi n texas in the 80s and parachute pants were popular there! |
13:31.21 | mutilator | still had the bowboy hat and boots tho i see |
13:31.22 | mutilator | :P |
13:31.23 | fugitivo | parachute pants, hehe |
13:31.25 | trixter | although mid 80s roper boots were all the rage |
13:31.25 | mutilator | cowboy* |
13:31.41 | trixter | oh god I haded those boots |
13:31.47 | mutilator | damnit my coffee is cold |
13:31.56 | trixter | and the hats I never really liked, late 80s I wore a flourecent green fedora |
13:31.56 | mutilator | now i have to walk outside in the freezing tundra to get more |
13:35.34 | docelm0 | w00t! Good MORNING #ASTERISK!!!! |
13:35.35 | mutilator | omfg hands are ice! |
13:36.15 | coppice | ice? but its 22C :-) |
13:36.18 | [Wiebel] | it's like 2:36 here ;) |
13:36.19 | [Wiebel] | PM |
13:36.32 | docelm0 | 50F here |
13:36.39 | [Wiebel] | although I wish i came out of bed just now :) |
13:36.44 | [Wiebel] | s/i/I/ |
13:37.10 | fugitivo | 18C Clear Skies, 10:37am |
13:37.19 | mutilator | pfft |
13:37.23 | mutilator | -15C here |
13:37.28 | fugitivo | -15C???? |
13:37.32 | fugitivo | where are you? |
13:37.37 | mutilator | northern michgan |
13:37.41 | coppice | 22C, can't see the sky as its 9:37PM |
13:37.43 | mutilator | wind blowin like crazy too |
13:37.48 | wasim | i went to school in houghton |
13:37.52 | mutilator | me too |
13:38.00 | mutilator | well not highschool |
13:38.03 | fugitivo | coppice: i don't see the sky neither, weather information in my kontact :) |
13:38.06 | mutilator | univ |
13:38.09 | wasim | college, but it still gives me the shivers |
13:38.14 | mutilator | yeh |
13:38.16 | wasim | bloody cold |
13:38.20 | mutilator | and windy as fuk |
13:38.29 | mutilator | specially by the eerc |
13:38.35 | wasim | yeah, if you stopped walking the wind would slide you back on the ice |
13:39.00 | mutilator | ah the memories |
13:39.14 | wasim | and then i left my car in the parking lot, and they piled snow all around it, so i didn't get it back till apr |
13:39.22 | mutilator | yea |
13:39.33 | mutilator | the snow drifts hanging off the edge of the car 2 feet or more |
13:39.52 | wasim | brrr .... |
13:39.55 | wasim | stop please ... |
13:39.59 | mutilator | break it off and what was showing of the car dissapears |
13:40.36 | coppice | wasim: so you had to walk 5 miles to and from college, uphill both ways, eat a crust of bread and drink ice cold water, and you were lucky |
13:41.01 | wasim | coppice: and to top it off, i had to take english as a foreign language exam ... |
13:41.04 | coppice | wasim: whats the lowest temp where you live? |
13:41.19 | wasim | coppice: it'll drop here to about 3C |
13:41.35 | coppice | wasim: that must have been a piece of cake for you :-\ |
13:41.50 | wasim | coppice: boring as hell ... nice spanish girl though |
13:42.00 | coppice | wasim: its gets to 4 or 5 here, but only for a week or two |
13:42.22 | coppice | wasim: how did you end up with english as a foreign language |
13:42.41 | mutilator | degree in anything wasim? |
13:42.46 | wasim | coppice: inspite of getting 673 in my toefl? no clue, some stupid university crap |
13:43.00 | wasim | mutilator: no, i got my butt out of there and to more temperate climate (iowa) |
13:43.08 | mutilator | how long ago were ya there? |
13:43.14 | wasim | mutilator: 1991 |
13:43.18 | mutilator | heh |
13:43.53 | coppice | in india I heard the US has a nasty trick now to extract more tuition fees. even if your english is great, they are marking on accents. suddenly indians who were not expecting any toefl issues have them |
13:44.29 | coppice | wasim: 1991 was the last time i spent winter in a cold place :-) |
13:44.51 | wasim | its the wind chill in houghton that killed you ... |
13:45.02 | wasim | -40 with wind chill ... horrid |
13:52.56 | *** join/#asterisk kimosabe (n=kimosabe@201.153.61.211) |
13:53.17 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
13:53.23 | kimosabe | can some one lead me to a good how to on fedora core and asterisk set up |
13:53.30 | kimosabe | thanks room |
13:55.52 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
13:55.56 | kimosabe | is asterisk running on any version of freebsd directlly from ports |
13:58.23 | [TK]D-Fender | FC3 apparently works well. FC4 has compiler issues. As for a how-to. Just install all the typical devel packages, download * from the SVN repositor and compile. |
13:58.35 | kimosabe | oki |
13:59.01 | viperdude | what overhead does Asteirsk Manager access put on a server? I want multiple logins for CRM integration |
13:59.16 | coppice | FC4 has a fussier compiler, but there shouldn't be issues with that any more |
13:59.35 | iCEBrkr | I dunno. FC4 has partition and boot manager issues as well. |
13:59.42 | iCEBrkr | ....depending on the hardward, obviously |
13:59.55 | *** join/#asterisk umay (n=chris@65-37-2-236.nrp2.roc.ny.frontiernet.net) |
14:01.10 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
14:01.13 | SpaceBass | morning |
14:01.20 | iCEBrkr | SpaceBass: hey |
14:01.23 | SpaceBass | anyone using a cisco phone with more than one sip server? |
14:01.43 | SpaceBass | (and by sip I mean asterisk) |
14:01.59 | SpaceBass | iCEBrkr, hey how are yoy |
14:02.02 | SpaceBass | s/yoy/you |
14:02.10 | iCEBrkr | x.x |
14:02.14 | iCEBrkr | 3hrs of sleep |
14:02.29 | iCEBrkr | and hoping these PRIs get turned-up today |
14:02.33 | SpaceBass | sounds like you are well rested... |
14:03.04 | iCEBrkr | haha |
14:03.07 | [TK]D-Fender | iCEBrkr : Got confirmation that they didn't finish thejob? |
14:03.16 | iCEBrkr | [TK]D-Fender: Actually, yes |
14:03.25 | iCEBrkr | I gotta make a few calls today to figure out WTF is going on |
14:03.55 | SpaceBass | Anyone using dring on a zaptel line? |
14:03.59 | [TK]D-Fender | heh, I *knew* there was no way the card was at fault :) |
14:04.45 | iCEBrkr | haha |
14:05.26 | [TK]D-Fender | Can't wait until their analog card comes out..... mmmm |
14:05.50 | [TK]D-Fender | Brilliant design |
14:07.37 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
14:07.42 | [TK]D-Fender | And apparently hardware EC on low density analog... yummm |
14:07.46 | *** join/#asterisk apardo (n=apardo@80.224.114.10) |
14:09.38 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
14:09.41 | SpaceBass | Anyone using dring on a zaptel line? |
14:09.44 | coppice | which analogue card? |
14:09.57 | iCEBrkr | ha |
14:09.59 | [TK]D-Fender | coppice : http://www.telephonyware.com/telephonyware/sangoma_aa.html |
14:10.14 | coppice | people tell me they are nice |
14:10.59 | [TK]D-Fender | Spec's are saying all theright things... |
14:12.02 | coppice | i think the backplane thing looks clunky. they apparently work well, though |
14:12.46 | [TK]D-Fender | It does have some potential drawbacks. If they are available in multiple lengths that'd help |
14:13.17 | [TK]D-Fender | Or if they modularize it. |
14:13.32 | coppice | I don't understand why they didn't make a card like digium's new one. 24 channels is a sweet spot, since an amphenol connector just fits nicely in a PCI slot |
14:13.57 | [TK]D-Fender | coppice : Could be in the works. Who knows... |
14:14.08 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
14:14.19 | [TK]D-Fender | Gotta start somewhere... |
14:15.12 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
14:15.48 | *** join/#asterisk fgravato (n=frank@office-nat.choopa.net) |
14:17.18 | [TK]D-Fender | Since they play nice with IRQ's and slot types they should still appeal to a mid-high range of installs below T1 requirements |
14:17.33 | *** join/#asterisk svenna_ (n=svenna@p548D30B8.dip0.t-ipconnect.de) |
14:17.41 | fgravato | is there any way of disabling transfer # in asterisk |
14:17.48 | fgravato | besides features.conf |
14:19.11 | [TK]D-Fender | Don't use tT in Dial commands? |
14:19.30 | [TK]D-Fender | I never do..... |
14:21.49 | fgravato | what about if you use call file? |
14:22.32 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
14:23.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:25.37 | [TK]D-Fender | no idea |
14:27.24 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
14:28.51 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
14:31.56 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
14:41.54 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
14:42.00 | asteriskmonkey | morning |
14:43.02 | oej | asteriskmonkey: Are you going to code a /proc interface or was it just something for the wishlist? |
14:43.12 | Ikarus | Bleah, zaptel blows up with "zaphfc: Unknown symbol" and then multiple entries with different symbols |
14:45.09 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
14:46.06 | heroine | i just setup an asterisk-1.2 but have some minor trouble with the voiceMailMain command in extension.conf as i prefix the dialed extension with s, but when the user dial into his mailbox, he's still asked for his mailbox number and password |
14:48.01 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
14:48.11 | tzafrir_laptop | sysfs is nicer to work with, although 2.6-specific |
14:48.12 | brad_mssw | heroine: you may have to supply your voicemail context like 112@default |
14:48.33 | heroine | hmm .. |
14:48.46 | Delvar | i used, exten => 100,1,voiceMailMain(${callerid(number)}@default) |
14:48.47 | heroine | i think i see where is the problem , thx for your tip :) |
14:49.00 | tzafrir_laptop | Ikarus, what version of kernel, zaptel and bristuff? |
14:49.02 | [TK]D-Fender | You may also want to be sure of exactly what you are passing to VoiceMailMain including the validity of the mailbox # itself |
14:49.28 | heroine | we use the mysql backend there. I think i should first look that point |
14:50.16 | Ikarus | tzafrir_laptop: I think I found it out myself |
14:50.46 | asteriskmonkey | oej |
14:51.09 | asteriskmonkey | oej: i was hoping someone would program that part and i make the external apps to run via cron :) |
14:51.50 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:51.50 | *** mode/#asterisk [+o anthm] by ChanServ |
14:52.36 | oej | asteriskmonkey: Ok, I'll close the bug report while waiting for code then. I think it has to be an external program that connects to the manager interface... |
14:53.14 | asteriskmonkey | oej: i thought it would be intenal code that outputs to a proc :P |
14:53.46 | RoyK | zoa: ping |
14:54.28 | Ikarus | tzafrir_laptop: ah well, it seems I'll have to upgrade to Debian unstable, or compile asterisk myself |
14:55.14 | Ikarus | And that second option is most likely, I heard about some issues with bristuff and Asterisk 1.2 |
14:55.18 | zoa | pong |
14:56.39 | coppice | zoa, man of jitters |
14:57.40 | tzafrir_laptop | Ikarus, I have debs of asterisk 1.2.0 . Asterisk Unstable is currently without bristuff |
14:58.13 | zoa | hey ho coppice, man of faxes! |
14:58.18 | tzafrir_laptop | In fact, the new debs I hope to soon upload will probably be without bristuff, unless I see a sign of life from kapejod |
14:58.39 | coppice | zoa: its all lies |
14:58.58 | Ikarus | tzafrir_laptop: bleah, I can't live without a decent ISDN layer (capi isn't decent in my tests) |
14:59.28 | coppice | the wonderful thing with ISDN layers is there are so many to choose from :-) |
14:59.42 | Ikarus | coppice: uhuh |
14:59.49 | tzafrir_laptop | Ikarus, in what ways is zapbri better than others? |
14:59.51 | Ikarus | coppice: well, vISDN is the only alternative |
15:00.09 | zoa | mISDN ? |
15:00.10 | Ikarus | tzafrir_laptop: it has almost no echo problems according to a friend of mine |
15:00.11 | brad_mssw | i ported the asterisk patch to 1.2.1 from bristuff ... but the libpri patch is a beotch, haven't had time |
15:00.49 | Ikarus | tzafrir_laptop: but vISDN comes with the same advantages and more, but it is still beta |
15:00.50 | coppice | Ikarus: look around. everyone seem to have their own ISDN layer, though many are based on the same original |
15:01.05 | Chotaire | brad: bristuff for 1.2.1 (incl. new chan_capi) to be released within 2-3 days. |
15:01.27 | brad_mssw | Chotaire: good to know |
15:01.39 | tzafrir_laptop | Chotaire, have you been in touch with the Kapejod himself? when? |
15:01.43 | Chotaire | today. |
15:01.57 | Chotaire | he promised me he'll make sure he gets finished before xmas. |
15:02.14 | Ikarus | coppice: well, I have seen mISDN, vISDN, bristuff and CAPI |
15:02.21 | coppice | did he say which christmas? |
15:02.25 | Chotaire | hehe |
15:02.41 | Chotaire | coppice: it's already working for 1.2.0, all he needs to do is make the changes for 1.2.1 and that's it. |
15:03.00 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
15:03.07 | Ikarus | coppice: and I just need one that supports overlap dialing and works well |
15:03.18 | Ikarus | even if that means running an older version of Asterisk |
15:03.41 | Chotaire | coppice: I will personally keep bothering him ;) I can't update to 1.2.* until he's finished ;) |
15:04.10 | asteriskmonkey | anyone know why asterisk would hang up on someone when it is supporst to go to vm? |
15:04.12 | Ikarus | Also DID is needed |
15:05.11 | Chotaire | and while at that... I seen changes to 1.2.1 regarding sip info dtmf debugging... except of more debugging output, has someone taken a closer look at my rfc2833 bug report for DID dtmf with pstn->sip->asterisk calls? it's broken since 1.0.4 as it seems. |
15:05.34 | SpaceBass | asteriskmonkey maybe VM is in a different context and its not being passed correctly |
15:06.41 | Chotaire | 1.2.* is giving me a major headache for sure. I wish I could update but I'll have to stick with 1.0.3+tenthousands-of-patches. |
15:07.36 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
15:07.40 | Chotaire | anyway, whoever is responsible for the decision to move meetme join/leave sounds into /var/lib/asterisk/sounds instead of using hex-encoded crap in some .h file: congratulations. |
15:08.44 | asteriskmonkey | SpaceBass: i have it like this exnt@context is that correct or has somthing changed since 1.0.9 on that :D |
15:09.18 | Chotaire | asteriskmonkey: what does the console say when its being disconnected? |
15:09.24 | SpaceBass | asteriskmonkey I'm not sure... never seen it coded like that |
15:09.30 | *** join/#asterisk MattB2 (n=Mattb@mail.tricycleinc.com) |
15:09.34 | MattB2 | hi all |
15:09.47 | MattB2 | possibly silly question - any idea why PRI NO DEBUG doesn't work in 1.2.1 |
15:09.57 | asteriskmonkey | SpaceBass: good idea, never thought to look at the console ! doh |
15:10.09 | Chotaire | i'm not spacebass, but that's ok. |
15:10.10 | MattB2 | Ahhhhhh |
15:10.15 | MattB2 | it was a silly question |
15:10.17 | MattB2 | ignore me ;) |
15:10.19 | MattB2 | oh you are! |
15:10.22 | asteriskmonkey | sorry chotaire: P |
15:10.28 | Chotaire | asteriskmonkey: asterisk -vvvvvvvvvvvvvvvvvvc and just paste what happens. |
15:10.29 | asteriskmonkey | my eyes are still not open |
15:11.37 | Seldon1975 | anyone know where I can get a GSM codec,converter or player for windows? |
15:11.45 | Chotaire | sox |
15:11.50 | Chotaire | seldon1975: there is sox for windows. |
15:11.51 | Seldon1975 | i cant play the Asterisk sounds on my XP machine |
15:11.55 | Chotaire | you can convert .gsm files to .wav |
15:11.56 | asteriskmonkey | this is what i got |
15:11.56 | Seldon1975 | aha |
15:11.56 | asteriskmonkey | Executing VoiceMail("Zap/13-1", "debitact201@debitact") in new stack |
15:11.56 | asteriskmonkey | <PROTECTED> |
15:12.04 | Seldon1975 | thanks Chotaire |
15:12.17 | Chotaire | seldon1975: check the sox website, there is a win32 binary which will make you happy. |
15:12.27 | Seldon1975 | nice one |
15:12.43 | *** join/#asterisk ManxPower (n=ewieling@200.sub-70-197-9.myvzw.com) |
15:13.18 | asteriskmonkey | Chotaire: did you see what i pasteds :) |
15:13.42 | Ikarus | Ah well, visdn doesn't even compile |
15:13.50 | [TK]D-Fender | asteriskmonkey : that mailbox doesn't look too legit |
15:14.08 | Chotaire | asteriskmonkey: yes... I seen it.. never seen a mailbox name "Zap/13.-1" ;) |
15:14.10 | [TK]D-Fender | unless you are doing wierd names... |
15:14.24 | asteriskmonkey | [TK]D-Fender: you cant use characters in voicemail address? worked before |
15:14.36 | Chotaire | oops... |
15:14.40 | [TK]D-Fender | You can, but does it exist as written? |
15:14.42 | Chotaire | well, you know what I mean ;) |
15:15.03 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
15:15.05 | asteriskmonkey | ah why the hell is it trying to dial the vm :P |
15:15.27 | *** join/#asterisk jcwunder (n=chris@a194.lrz.vpn.lrz-muenchen.de) |
15:16.18 | asteriskmonkey | this is the correct way to dial vm in 1.2 right? Voicemail(ext@context) |
15:17.31 | Chotaire | VoiceMail([s|u|b]extension[@context][&extension[@context]][...]) |
15:17.57 | Chotaire | what about this... == Auto fallthrough, channel 'Zap/13-1' status is 'UNKNOWN' |
15:18.07 | asteriskmonkey | im lost on that |
15:18.16 | asteriskmonkey | anyone know what thats about? |
15:20.14 | *** join/#asterisk jluk (n=njon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk) |
15:20.24 | ManxPower | No, Voicemail(vmbox@vmcontext) |
15:20.30 | Chotaire | well... |
15:20.34 | Chotaire | take a look above. |
15:20.51 | ManxPower | extensions are not the same as a mailbox. You can make them the same, but that's cosmetic. |
15:20.59 | Chotaire | exactly. |
15:21.12 | ManxPower | and the @vmcontext is the voicemail.conf context, not the extensions.conf context. |
15:21.20 | Chotaire | you must have mailbox number/name debitact201 in voicemail.conf |
15:21.38 | Chotaire | and @vmcontext you only need if you have multiple vmb systems. |
15:22.03 | Chotaire | e.g. multiple virtual vmb systems with same box numbers/names. |
15:22.11 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
15:22.48 | Chotaire | so... VoiceMail(201) will pretty much do the job. |
15:22.53 | Chotaire | i hope I got this right, manxpower? |
15:23.12 | Chotaire | all you need is a vmb "201" in voicemail.conf |
15:23.14 | asteriskmonkey | ManxPower: i know there is a diffent context in the voicemail.conf |
15:23.17 | asteriskmonkey | its bizzare |
15:23.35 | asteriskmonkey | the name of the box is the mailbox .. |
15:24.18 | asteriskmonkey | so why this dosnt work i dont understand.. further more i dont know why the error says its dialing it exten => 201,1,Playback(debitact-tryext) |
15:24.18 | asteriskmonkey | exten => 201,n,Voicemail(debitact201@debitact) |
15:24.38 | darwin_35 | in a static dial plan in a macro how would you make the macro read the protocal used ? |
15:24.45 | ManxPower | asteriskgeeks, show us the ACTUAL line from extensions.conf |
15:24.47 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
15:25.09 | SpaceBass | Anyone using dring on a zaptel line? |
15:25.22 | SpaceBass | anyone using a cisco phone with more than one sip server? |
15:25.38 | darwin_35 | Dial(protocal/${arg1}? |
15:26.10 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-243-94-102.dsl.tulsok.swbell.net) |
15:27.02 | Chotaire | darwin: maybe Dial(protocal/${arg1})? ;) |
15:28.44 | Seldon1975 | Hi, can someone suggest how to set up a dialplan to playback(all-ougoing-lines-unavailable) when al my trunks are busy? |
15:29.00 | darwin_35 | nope does not work |
15:29.00 | Seldon1975 | i mean we have t for timeout and i for invalid; but no b for busy? |
15:29.03 | darwin_35 | grrr |
15:29.34 | [TK]D-Fender | Seldon1975 : What does "busy" mean when you're expecting input? Busy is the result of DIALING. |
15:29.55 | Seldon1975 | yes, when all my Zap lines 18-23 are utilized already |
15:30.13 | Seldon1975 | but how can I determine this in my dialplan? |
15:30.28 | [TK]D-Fender | Seldon1975 : the dialresult would be "congestion" |
15:30.33 | Seldon1975 | ie: an internal user tries to dial out |
15:30.38 | Seldon1975 | oh |
15:30.39 | [TK]D-Fender | SEE ABOVE |
15:30.42 | *** join/#asterisk alexissoft (n=alexis@ws1.rtcn.be) |
15:30.42 | alexissoft | hi |
15:30.55 | darwin_35 | the rate things change inthe dial plan its hard to keep up |
15:32.18 | alexissoft | can i use a standard winmodem as FXO ? |
15:32.43 | darwin_35 | only the intel 579 I think it was |
15:32.58 | Seldon1975 | how do I respond to a specific DIALRESULT from a Dial() command? |
15:33.06 | alexissoft | and i can't use any standard modem |
15:33.19 | darwin_35 | nope |
15:33.30 | coppice | what is a standard winmodem? they don't follow any standards |
15:33.59 | alexissoft | hmm ... yes :) |
15:34.05 | [TK]D-Fender | Seldon1975 : Look at the STDEXTEN macro on the WIKI for some inspiration. |
15:34.10 | *** join/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net) |
15:34.13 | alexissoft | i think it's the same thing for a serial modem |
15:34.17 | Seldon1975 | D-Fender: thx |
15:34.35 | *** part/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net) |
15:35.07 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
15:36.18 | coppice | alexissoft: I'm not sure what you mean. the winmodems don't follow any standards. the modem software that comes with them makes them look like a standard modem. however, as a telephony interface there are no standards |
15:36.39 | alexissoft | ok, i see |
15:37.04 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
15:38.16 | asteriskmonkey | Chotaire: i do have a mailbox called that in my voicemail.conf under that context and it dont work for somereason |
15:38.23 | alexissoft | congratulations for asterisk, it's a very very very very very very good product !! |
15:38.30 | alexissoft | i love it :) |
15:39.02 | Katty | morning lads. |
15:40.12 | mog_work | thanks alexissoft |
15:40.17 | alexissoft | :) |
15:40.21 | mog_work | everyone appreciates it |
15:40.43 | Katty | well don't everyone say g'morning at one. |
15:40.46 | Katty | once, too. |
15:41.41 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
15:41.45 | [TK]D-Fender | GOOD MORNING SUNSHINE! |
15:42.42 | asteriskmonkey | morning |
15:43.11 | *** join/#asterisk Seldon19751 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
15:45.22 | Katty | asteriskmonkey: what do you want? |
15:45.25 | Katty | asteriskmonkey: get off my version. |
15:45.46 | asteriskmonkey | lol |
15:46.07 | asteriskmonkey | i use mirc was looking for names of other irc clients i could possibly use |
15:46.32 | Katty | mirc :<<<<<<<<<<<<<<<<<<<<<<<<< |
15:46.37 | Katty | mirc is /horror/ |
15:46.46 | Katty | and it's on WINDOWS |
15:46.49 | Katty | which means you can't screen it |
15:46.54 | Katty | horror horror horror |
15:46.56 | asteriskmonkey | lol |
15:47.26 | asteriskmonkey | well whats a good cli one for bsd and ill go compile it |
15:47.34 | alexissoft | irssi :) |
15:47.37 | alexissoft | or weechat |
15:47.45 | *** join/#asterisk backblue (n=moo@82.102.1.42) |
15:47.55 | asteriskmonkey | and that wont start building xwindows as a part of dependencies ? :) |
15:47.57 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
15:47.57 | alexissoft | asteriskgeeks, a graphic IRC client or a textual one ? |
15:48.07 | alexissoft | irssi & weechat are text-based |
15:48.12 | asteriskmonkey | sweet |
15:48.18 | Katty | yay for irssi |
15:48.22 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
15:48.26 | backblue | hi, does anyone knows (in cisco.com) where i can find the sip configuration manual for cisco 7970 & cisco 7940?? |
15:48.28 | shido6 | . |
15:48.58 | [Wiebel] | backblue: it's not included with the firmware? |
15:49.25 | backblue | [Wiebel]: i dont know, i dont have download new firmware. |
15:49.39 | [Wiebel] | ah |
15:50.06 | [Wiebel] | wel |
15:50.10 | [Wiebel] | it's not included :) |
15:50.14 | [Wiebel] | (I Just checked) |
15:50.39 | *** join/#asterisk Porks (n=nao@200.231.120.138) |
15:50.41 | [Wiebel] | I do have a default config for you |
15:50.54 | backblue | can you send me? |
15:51.19 | backblue | anyway, i need the installation manual anyway |
15:52.14 | [Wiebel] | http://www.wiebel.nl/asterisk/SEPexample.cnf |
15:54.32 | Katty | i could use support. |
15:54.40 | Katty | how do i kill an id after ps auxing? |
15:54.52 | Katty | kill $id? |
15:54.55 | mkrufky | yes |
15:55.01 | Katty | kthx |
15:55.02 | mkrufky | and if that doesnt work, |
15:55.05 | mkrufky | kill -9 $id |
15:56.56 | Katty | xmms borked. |
15:56.58 | Porks | hi? |
15:57.05 | Katty | asl. |
15:57.08 | Katty | </sarcasm> |
15:57.12 | Porks | i need some help :| |
15:57.16 | Katty | me too. |
15:57.23 | Katty | but i need a therapist. |
15:57.23 | Porks | hauehae |
15:57.27 | Porks | I have two lines of PSTN conected at two X100P cards... is possible transfer a call from line #1 to line #2? |
15:57.38 | Porks | I mean... people call me with the line #1's number... so I have to let the line #1 free.. so when somebody call me I wanna transfer the call to line #2 and leave line #1 free |
15:57.58 | Porks | is it magic? or is it possible? |
15:57.59 | Porks | :D |
15:58.43 | iDunno | I could do with some magic |
15:58.48 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
15:59.16 | Porks | me too... but not with asterisk |
16:00.02 | Porks | like.. the PBX... don't have many lines come in... and people call in only one number? |
16:02.26 | SpaceBass | anyone using a cisco phone with more than one * box? |
16:03.20 | iCEBrkr | -.- zzzZZZZZ |
16:03.24 | Katty | [TK]D-Fender: do you want a happy mew years card? |
16:03.38 | Katty | [TK]D-Fender: as part of my mass mailings. |
16:04.39 | tzafrir_laptop | Porks, this best be done at your provider |
16:04.59 | tzafrir_laptop | "leading number" or something similar. |
16:05.22 | tzafrir_laptop | Otherwise I can't see how this can be implemented |
16:05.25 | iCEBrkr | mew years. lol |
16:05.51 | iCEBrkr | [TK]D-Fender: yea, don't feel special.. It's a spammation of people she knows.. |
16:05.52 | darwin_35 | ICEBRKR same happened on the new box I built yesterdy |
16:05.59 | *** join/#asterisk Porks (n=nao@200.231.120.138) |
16:06.08 | iCEBrkr | ?? |
16:06.08 | Porks | argh... |
16:06.15 | darwin_35 | its not linking to the db |
16:06.18 | Katty | iCEBrkr: yes. |
16:06.21 | Katty | iCEBrkr: all my friends get cards. |
16:06.30 | tzafrir_laptop | Porks, this best be done at your provider |
16:06.35 | tzafrir_laptop | "leading number" or something similar. |
16:06.36 | iCEBrkr | darwin_35: Follow the freetds how-to |
16:06.36 | tzafrir_laptop | Otherwise I can't see how this can be implemented |
16:06.47 | Katty | file[desk]: which means you, pretty boy. |
16:06.55 | darwin_35 | whats the url |
16:07.07 | iCEBrkr | darwin_35: google |
16:07.30 | *** join/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com) |
16:08.38 | Porks | tzafrir_laptop uhmm... is the some for pbx? (the 'normal' way.. without asterisk or other stuff) |
16:08.57 | Porks | the telephon company provides this |
16:10.41 | iCEBrkr | 3hrs of sleep.. If that. I'm not sure how I'm going to make it through the day |
16:12.09 | darwin_35 | nothing for myodbc |
16:12.18 | darwin_35 | this sucks |
16:12.37 | darwin_35 | why does it work fine on 4 other boxes but not this one |
16:12.45 | iCEBrkr | darwin_35: If I got it working, you can too |
16:13.26 | darwin_35 | evrythign matches box for box |
16:13.29 | darwin_35 | 3 for 3 |
16:13.34 | darwin_35 | #for # |
16:14.09 | Dandan | hey all |
16:14.10 | Dandan | :) |
16:14.19 | trixter | since it would enhance the value of ebay (who now owns skype) I wonder if anyone there has thought of pbx integration for their business sellers. I am thinking that at etel I might hit up the paypal booth (dunno why its listed as paypal not ebay but meh) about this and see if any are interested.. even if it were closed source to issue a library that could be used to do skype would be handy. While I personally dont like skype I do like choices and d |
16:14.20 | trixter | o not feel that I should dictate to someone else how they should run their business or talk to their friends or ... |
16:14.25 | Dandan | got a problem, just started playing around with x100p: |
16:14.33 | Dandan | when I call out |
16:14.40 | Dandan | through zap/1 |
16:14.41 | trixter | or does anyone in here have a contact with ebay that would be handy in that regard? cause I think a general purpose library would be quite handy |
16:14.47 | Dandan | it is being automatically picked up |
16:14.59 | Dandan | and i end up talking to zap/1 which is myself?!? |
16:15.16 | smither | I'm using ast 1.2.1 and have a question. In receiving an incoming call over POTS using an ATA and the zaptel library, how do you control when the call is answered? There used to be a zap_waitcall function that in which you could specify the number of rings, but I cannot find that functionality now. Any suggestions? |
16:16.27 | iCEBrkr | smither: Where's your POTS line hooked up? To your astrisk box, right? |
16:16.29 | iCEBrkr | Dandan: ?? |
16:16.44 | Dandan | iCEBrkr: hm, hold on lemmie paste |
16:16.51 | iCEBrkr | NOooooooooo |
16:16.52 | smither | Yes - through an x100p. |
16:16.54 | iCEBrkr | Dandan: pastebin.ca |
16:17.04 | Dandan | *paste into pastebin :) |
16:17.07 | iCEBrkr | lol |
16:17.12 | Dandan | sorry that's what I meant |
16:17.45 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
16:17.52 | iCEBrkr | smither: Ok, so the x100p answers as soon as it rings and then starts running through your dialplan to figure out where to deliver the call. |
16:17.56 | iCEBrkr | smither: what's it not doing? |
16:18.18 | fiber0pti | Anyone have problems with asterisk dropping calls with a voip provider? People will be on the phone and the phone won't hang up but neither party can hear the other person. |
16:18.22 | *** join/#asterisk oelewapperke (i=4ce5f774@85.158.215.1) |
16:18.55 | smither | ecebrkr: exactly - but I would like it to not pick up until, say, 4 rings (like you can do with a dedicated answering machine). Is there anyway to do that? |
16:19.08 | iCEBrkr | Not that I know of. |
16:19.08 | smither | sorry - iCEBrkr. |
16:19.09 | SpaceBass | fiber0pti I have that problem with my wifi phone but not with a specific provider |
16:19.27 | SpaceBass | smither there is a way to delay pickup |
16:19.30 | iCEBrkr | smither: If you're gonna use asterisk use it.. Don't half-ass it. |
16:19.35 | iCEBrkr | :) |
16:19.37 | Dandan | http://pastebin.com/471784 |
16:19.46 | Dandan | that is weeeeird :/ |
16:20.34 | iCEBrkr | Dandan: umm, what's wrong with that? |
16:20.51 | Dandan | it seems like I am talking to myself |
16:20.54 | Dandan | :) |
16:20.56 | iCEBrkr | You're not. |
16:21.09 | Dandan | well it is not ringing, the phone call is established |
16:21.17 | iCEBrkr | Call progress |
16:21.56 | Dandan | yeah, but nothing happens after that last line... |
16:21.56 | iCEBrkr | and you shouldn't have to put any parameters on it.. drop the |60|r |
16:22.42 | smither | Don't really understand the reference to half-ass it, but I know there used to be this functionality in the old Zapata library - I think the call was zap_waitcall() or zap_waitanswer(), and one of the parameters was the number of rings to wait before going off hook. Was that call removed? |
16:23.08 | iCEBrkr | I'm trying to figure out why the heck you'd wanna wait 4rings for asterisk to answer |
16:23.50 | Dandan | iCEBrkr: it seems like it takes the phone line off hook but doesn't dial any digits |
16:24.00 | Dandan | I receive: the number you have dialed is incomplete |
16:24.13 | smither | iCEBrkr - to give me a chance to pick up a regular POTS phone before Ast takes over, again like you do with an answering machine. |
16:24.36 | iCEBrkr | smither: umm, make Asterisk answer the line, and ring your ATA |
16:24.51 | iCEBrkr | if you don't answer, have Asterisk transfer to VoiceMail() |
16:25.09 | iCEBrkr | ya know.. Like a real PBX.. Afterall, Asterisk is a PBX |
16:25.33 | iCEBrkr | Dandan: I'm assuming you're trying to 7digit dial in that areacode? |
16:25.40 | Dandan | yup |
16:26.01 | iCEBrkr | Dandan: Does it require a 1 first? |
16:26.08 | [TK]D-Fender | Dandan : Don't bother with the prefix in the Extens line. Just do a length check and add it if necessary |
16:26.25 | smither | That makes sense, and of course will work with a more complete system than I currently have. I understand the 'real PBX' reference, maybe I am 'half-ass'ing it :-). |
16:26.40 | iCEBrkr | smither: That's what I meant about doing it half-assed.. :D |
16:26.57 | iCEBrkr | smither: It's frightening at first, I know. I was the same way with Asterisk when I put it on my home land-line. |
16:27.37 | iCEBrkr | smither: I use Asterisk as a glorified answering machine myself. |
16:27.41 | Dandan | so what is your dial-out with x100p? |
16:28.09 | iCEBrkr | Dandan: I think it's more like you're not passing the correct number of digits out. |
16:28.27 | Dandan | if i use the pass through it works |
16:29.03 | iCEBrkr | I'm actually kinda confused. |
16:29.13 | iCEBrkr | Cuz ${EXTEN:3} is gonna strip off the areacode. |
16:29.14 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
16:29.26 | Dandan | gonna strip 724 |
16:29.28 | Dandan | as prefix |
16:29.29 | smither | OK, another question, perhaps related - in making an outgoing calls (like a phone tree with a recorded message) I would like to not start the recorded message until the called number (again, through the x110p to POTS lines) is answered. Is there anyway to detect when the POTS line is picked up at the other end? |
16:29.29 | iCEBrkr | right |
16:29.54 | iCEBrkr | smither: It'll do that. |
16:30.02 | iCEBrkr | smither: callprogress=yes |
16:30.21 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
16:30.23 | iCEBrkr | err something like that. It's what Dandan was confused about when Zap/1-1 Answered :P |
16:30.36 | iCEBrkr | Dandan: So is 724 a prefix or AC? |
16:30.43 | smither | iCEBrkr: That looks promising - which .conf is that put in? |
16:30.53 | Dandan | prefix to get pots |
16:31.11 | *** join/#asterisk razu_ (n=razu@ip220.cab17.mus.starman.ee) |
16:31.26 | |Vulture| | Anyone know how I might code inbound CID to remove an extra " from it? the telco is sending CID as ""Test Call" and it should be "Test Call" |
16:31.38 | iCEBrkr | smither: zapata.conf |
16:32.06 | smither | iCEBrkr: Thanks! I'm off to play with Asterisk. |
16:32.11 | iCEBrkr | Dandan: oh, like dialing 9.. ok |
16:32.16 | Dandan | yeah |
16:32.40 | Dandan | there is something wrong... |
16:32.42 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
16:32.45 | darwin_35 | <PROTECTED> |
16:32.46 | Dandan | * can't pick up the line properly |
16:33.01 | *** join/#asterisk Jick (n=Jick@209-83-240-53-static.dsl.oplink.net) |
16:33.05 | darwin_35 | the msql ver ar ediff |
16:33.08 | iCEBrkr | Dandan: and you're sure the number after you dial 724 is local? :P~~ |
16:33.11 | *** part/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com) |
16:33.16 | darwin_35 | .11 vvers .15 |
16:33.18 | Dandan | iCEBrkr: heh :) |
16:33.30 | *** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net) |
16:33.32 | t0ke | hi |
16:33.44 | Dandan | it is not even funny :) |
16:33.47 | iCEBrkr | haha |
16:34.16 | t0ke | anyone know if is possible that IAX calls use enblock and no overloap method for send digits? |
16:34.25 | iCEBrkr | darwin_35: This is why I don't run that bleeding-edge shit. |
16:34.41 | t0ke | it does slower calling via PRI |
16:35.58 | Dandan | Dec 20 11:33:56 WARNING[813]: chan_zap.c:6315 handle_init_event: Detected alarm on channel 1: Red Alarm |
16:36.02 | Dandan | now what? |
16:36.07 | *** join/#asterisk gniretar_work (n=mark@152.160.35.1) |
16:36.29 | iCEBrkr | Dandan: your phone line is unplugged. |
16:36.34 | iCEBrkr | Dandan: go pay your phone bill! Geesh |
16:36.36 | iCEBrkr | :D |
16:36.40 | gniretar_work | hi all |
16:37.06 | gniretar_work | hey, i'm on SuSe 10.0, what init script distro would u guys recommend i use? |
16:37.08 | Dandan | iCEBrkr: oh right, I am checking the cable :) |
16:37.51 | t0ke | anyone know how to activate enblock dialing mode in Asterisk? |
16:38.35 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
16:38.40 | TheCops | Hi |
16:38.40 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
16:39.17 | *** join/#asterisk santiago (n=santiago@208.195.215.4) |
16:39.51 | TheCops | I use Eyebeam with g729 codec to my asterisk server and trying to call another extension (A sip one) on the pbx who is using ulaw. The eyebeam client is working great. But on the other SIP Phone, we hear echo back when speaking. |
16:40.02 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
16:40.09 | TheCops | This is a translations issue ? |
16:42.25 | Seldon19751 | I downloaded and installed postgres; compiled asterisk and I started getting console errors when I made outgoing calls - has anyone else experienced this? |
16:42.46 | Seldon19751 | it seems that installing pgsql and recompiling * mucked it up |
16:43.20 | tzafrir_laptop | Seldon19751, doesn't your distro have pgsql? |
16:43.36 | Seldon19751 | it appears not, when I type 'pgsql' I get command not found |
16:43.43 | Seldon19751 | which is odd, since it's CentOS |
16:43.45 | tzafrir_laptop | what distro? |
16:43.54 | Seldon19751 | CentOS |
16:43.59 | Seldon19751 | 2.6.1 |
16:44.03 | tzafrir_laptop | considered using apt/yum to install it? |
16:44.20 | Seldon19751 | tzafrir: could you tell me how? |
16:44.32 | tzafrir_laptop | Most packages are not installed by default, because they're so easy to install with the package manager |
16:44.51 | Dandan | Dec 20 11:44:29 NOTICE[862]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
16:44.55 | tzafrir_laptop | I believe CentOS uses yum . try: yum install postgresql-devel |
16:44.56 | Dandan | iCEBrkr: what is that? |
16:45.12 | backblue | tzafrir_laptop: yes it uses |
16:45.18 | Seldon19751 | tzafrir: thanks |
16:45.25 | Seldon19751 | tzafrir: ill try it |
16:45.56 | Seldon19751 | tzafrir: if I get the same issue; how could I uninstall pgsql |
16:46.01 | tzafrir_laptop | Seldon19751, BTW: 2.6.1 is probably not the version of CentOS. CentOS is typically either 3.something or 4.something. |
16:46.28 | Seldon19751 | tzafrir: oh ok thx |
16:46.29 | *** join/#asterisk Igbothom_III (n=HiltonT@203-206-170-99.perm.iinet.net.au) [NETSPLIT VICTIM] |
16:46.47 | tzafrir_laptop | I'd try to get rid of the postgresql stuff in /usr/local... maybe there is a 'make uninstall' there |
16:46.52 | *** join/#asterisk Igbothom (n=HiltonT@203-206-170-99.perm.iinet.net.au) |
16:47.08 | backblue | j php |
16:47.18 | tzafrir_laptop | anyway, pastebin the exact error message |
16:47.20 | Seldon19751 | tzafrir: righto. Im doing this because I want to get call logging to the DB going |
16:47.58 | Seldon19751 | tzafrir: yum seems to have installed pgsql-devel ok - should I make install asterisk now? |
16:48.50 | Seldon19751 | tzafrir: can you tell me how to check that it's installed? |
16:48.52 | tzafrir_laptop | Seldon19751, AFAIK there are no decent CentOS asterisk packages. |
16:49.12 | tzafrir_laptop | rpm -q pgsql-devel |
16:49.23 | Seldon19751 | I realise now I shouldn't have; but I'm using an Asterisk@home distro |
16:49.29 | tzafrir_laptop | rpm -V pgsql-devel |
16:50.06 | tzafrir_laptop | The latter should give an empty output |
16:50.21 | Seldon19751 | it says: "pgsql-devel is not installed" do I need to do something else after yum downloads the packages? |
16:52.06 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
16:52.53 | *** join/#asterisk fugitivo (n=ajf@209.13.240.210) |
16:53.48 | darwin_35 | ICE I am stumped I have fallowed evrey thing |
16:54.03 | darwin_35 | its driving me insane |
16:54.56 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-61-91.38-151.net24.it) |
16:56.28 | [TK]D-Fender | darwin35 : Whats the problem? |
16:56.54 | *** join/#asterisk waddy (n=waddy@83.218.4.231) |
16:58.19 | waddy | I have moved from RH9 to ES4, when i do the install i cannot choose the indivdual packages like isdn utils etc? I look rh network and it dont exist for RHES4 ? Can anyone help please |
16:58.44 | waddy | also things like ncurses etc |
16:58.52 | Beirdo | waddy: hate to be rude, but why are you asking that here? |
16:58.55 | [TK]D-Fender | Perhaps you should try #RHEL |
16:59.13 | waddy | cause i want to get my isdn card working with asterisk |
16:59.23 | waddy | dead simple with rh9 |
16:59.56 | waddy | just thought there may be a simple thing im overlooking |
17:00.00 | Beirdo | likely is |
17:00.13 | Beirdo | but this isn't a RHEL channel, we don't likely know |
17:00.43 | waddy | might be easier to go back to RH9 |
17:00.55 | Beirdo | as [TK]D-Fender said, maybe try #RHEL |
17:01.02 | Beirdo | if it exists |
17:01.58 | waddy | yer ill give it a go |
17:02.00 | waddy | thanks |
17:02.07 | *** join/#asterisk slak- (i=slak@rewted.biz) |
17:02.08 | slak- | hi |
17:02.15 | slak- | can someone tell me what this is |
17:02.16 | slak- | Destroying call '9f298d4d-73bdc324@192.168.111.20' |
17:02.23 | slak- | i dont have a sip pgone at that IP |
17:02.26 | Beirdo | no prob, waddy |
17:02.26 | brad_mssw | does AEL support && or || in if() statments ?? |
17:02.27 | slak- | its a workstation |
17:02.33 | slak- | without any software phone |
17:03.38 | slak- | someone :) |
17:03.49 | slak- | clue me in why i see that log in the clit |
17:03.52 | slak- | er |
17:03.52 | slak- | cli |
17:05.27 | *** join/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com) |
17:06.00 | Enderson | hello |
17:07.34 | Enderson | I have only one FXS and it's plugged at my PABX at number 7, I've already called it, and made it call a SIP number and other stuff |
17:07.51 | Enderson | but now I wat to make it transfer the call, like I do pressing flash and then tha number |
17:07.53 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
17:08.09 | Equinox | <PROTECTED> |
17:08.09 | Equinox | <PROTECTED> |
17:08.16 | [TK]D-Fender | What kind of FXS? What is #7? |
17:08.26 | TheCops | hey [TK]D-Fender |
17:08.30 | Seldon19751 | caould someone tell me how to configure Asterisk to log files to a MySQL server? |
17:08.34 | [TK]D-Fender | hiya TheCops |
17:08.34 | Enderson | I've already tryied Flash() before a Dial() , that didn't work |
17:08.35 | Seldon19751 | can it be done? |
17:08.52 | Qwell | Seldon19751: I'm not thinking so, because where would it log if it can't write to sql server? |
17:08.52 | [TK]D-Fender | Seldon19751 : WIKI <- |
17:08.59 | Enderson | I tryed SendDTMF it called the other extension at 5, but it hangup |
17:09.00 | Qwell | bbl though |
17:09.09 | TheCops | [TK]D-Fender, g729 on dialup have a great quality! for 10$ it is a good thing to buy :) |
17:09.19 | Enderson | [TK]D-Fender: it's the number that * is plugged on PBX |
17:09.44 | Seldon19751 | [TAN] whats the best IRC client for Windows? |
17:09.46 | slak- | i have two sip extensions that are connectining via a sipura ATA, that lose registration and are unable to register unless I reset the ATA unit. Ive tried to swap ATA units and same thing happens, checked config and the two sip accounts look exactly as the rest, im stumped |
17:09.50 | slak- | can someone help me debug |
17:09.57 | [TK]D-Fender | Enderson : that made no sense to me, sorry... |
17:09.58 | slak- | the only thing that fixes it is if i reset the ATA |
17:10.09 | slak- | then it works for 24hrs or so |
17:10.19 | Enderson | [TK]D-Fender: that's my english, I don't know nome names =/ |
17:10.34 | Enderson | let me try again ... |
17:10.53 | slak- | [TK]D-Fender: :D |
17:10.55 | Equinox | Will my current 1.x config files work with 1.2.1? |
17:11.02 | Equinox | Or will I need to mess with them ? :) |
17:11.07 | slak- | :) |
17:11.12 | *** join/#asterisk Sixam (n=killa666@71.83.113.93) |
17:11.30 | slak- | Equinox: I had some issues with voicemail.conf |
17:11.35 | Enderson | asterisk is plugged via FXS port to my company PBX, I have to call 7 so asterisk to answer it on the channel 1 context |
17:11.46 | *** join/#asterisk vandien (i=sted@aditu.dahltronics.de) |
17:11.50 | Equinox | slak- Voicemail I can deal with.. Extensions.conf sip.conf iax2.conf would be bad things to break ;) |
17:12.09 | slak- | those were fine, i didnt test iax2 |
17:12.32 | [TK]D-Fender | Extensions.conf is what'll break. the rest it pretty much the same. |
17:12.33 | slak- | and MeetMe seems to be broken..sound distrotions |
17:12.38 | Enderson | I only have 1 channel, and the company PBX asterisk is plugged on, if youb answer a call, and need to trasnfer, it's just press "flash", and then the number |
17:12.52 | [TK]D-Fender | Oh and I've heard some Darth Vader MeetMe references :D |
17:12.55 | slak- | Equinox: also, in modules.conf you'll need to comment out chan_modem |
17:12.59 | Equinox | [TK]D-Fender, Hmm... Lots of stuff changed in extensions.conf? My setup is pretty simple. |
17:13.01 | slak- | or * wont start with the old config |
17:13.10 | slak- | and chan_modem is off by default in 1.2 |
17:13.23 | slak- | Equinox: my simple extensions.conf didnt change and worked |
17:13.30 | [TK]D-Fender | Equinox : Certain amount of things. Depreciated callerID, ASTDB, priority jumping, etc |
17:13.37 | ManxPower | I thought chan_modem was not even built in 1.2 |
17:13.40 | Equinox | Ahh I don't use astdb |
17:13.43 | slak- | its not ManxPower |
17:13.44 | Equinox | Only DB I use is for CDR records |
17:13.53 | Equinox | I use postgres for that tho, not mysql |
17:13.58 | slak- | thats why if he tries to use his old config, it wont start right |
17:14.04 | Equinox | What is priority jumping? |
17:14.07 | slak- | because the old config tries to load chan_modem |
17:14.24 | Equinox | I'll blow away chan_modem right now |
17:14.25 | [TK]D-Fender | Equinox : the old +101 thing for cmd's like DIAL. |
17:14.35 | SkramX | Hi All! |
17:14.40 | slak- | TK: i posted a real annoying problem a few screens up, your input would be appreciated |
17:14.54 | ManxPower | read UPGRADE.txt and SECURITY files. |
17:15.09 | [TK]D-Fender | slak- : no idea |
17:15.18 | slak- | well any clue how i can debug it? |
17:15.25 | Equinox | [TK]D-Fender, Oh... Hmm I'll have to see how that is handled now. |
17:15.29 | slak- | i cant get the ATA to register without resetting its power |
17:15.31 | Equinox | Gotchya |
17:15.31 | slak- | really annoying |
17:15.35 | slak- | and its not the ATA |
17:15.47 | [TK]D-Fender | slak- : Really not sure what to say... |
17:16.02 | slak- | OK, seperate issue: |
17:16.03 | slak- | Destroying call '9f298d4d-73bdc324@192.168.111.20' |
17:16.08 | slak- | that is a workstation |
17:16.12 | slak- | without any software phone |
17:16.15 | morale | what are hints for? can i use it to check to make sure my SIP phone is plugged in and only ring it if it is? or if someone is on the line don't ring it and ring my other extension? |
17:16.18 | slak- | why would one see that line |
17:17.46 | slak- | is there any way to debug sipura ata's to see why they cant register |
17:17.46 | [TK]D-Fender | morale : Hint is for presences detection so that you can see if other phones are busy (off-hook, ringing, etc) |
17:17.46 | [TK]D-Fender | slak- : Try SIP debug |
17:17.46 | [TK]D-Fender | slak- : in CLI |
17:17.48 | slak- | its on |
17:17.51 | slak- | nothing useful |
17:17.57 | file[desk] | woot MSN is b0rken |
17:19.51 | slak- | file why would i see this line if theres no sip phone on this workstation: Destroying call '9f298d4d-73bdc324@192.168.111.20' |
17:19.55 | slak- | i cant figure it out |
17:20.15 | file[desk] | there has to be. |
17:20.57 | slak- | what about this Destroying call '2c821ad0686f8f9c43f19b7f212dd78e@192.168.111.1' |
17:21.01 | slak- | thats my * box |
17:21.10 | file[desk] | why do you care anyway? |
17:21.11 | slak- | and theres even an entry for localhost like that |
17:21.20 | slak- | well because it makes no sense |
17:21.21 | file[desk] | those are only visible under debug |
17:22.03 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj1i.dialup.mindspring.com) |
17:22.07 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
17:22.19 | kink0 | hello |
17:22.40 | slak- | [TK]D-Fender: i found another way to solve that ATA problem without resetting its power, and thats "Remove Last Reg: YES" |
17:22.43 | slak- | under SIP tab |
17:22.47 | *** join/#asterisk fanatic (n=jparrott@pcp01488192pcs.limstn01.de.comcast.net) |
17:22.54 | slak- | does that ring any bells? |
17:23.05 | slak- | it registered immediately |
17:23.26 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
17:24.06 | darwin_35 | anyone here have a call back busy macro they are willing to share |
17:25.23 | slak- | look in the sample extensions.ael |
17:25.28 | kink0 | anyone can help about PRI(E1)-> GSM(movile) implementation ? ( I just have a quad PRI digium card ) |
17:25.41 | KriS83 | Just out of curiosity, what do you mean by "callback busy macro"? |
17:26.12 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
17:26.46 | bjohnson_ | awesome: |
17:26.47 | bjohnson_ | http://www.bell.ca/shop/wireline.portal?_nfpb=true&_pageLabel=PrsShpPnsPro_WithoutLeftNav&content=/portlets/personal/phoneservices/promotion/uniden/index.jsp&metaKey=PrsShpPnsPro_Uniden&language=en®ion=ON&myurl=CSQ&mobility_upgrade=false |
17:27.01 | bjohnson_ | good combination of bt and cell |
17:27.18 | bjohnson_ | stick it on a SIP ATA and you're off to the races |
17:27.41 | slak- | but its not gsm capable |
17:27.47 | bjohnson_ | even a baby monitor .. hahaha |
17:27.51 | bjohnson_ | no note gsm |
17:27.55 | bjohnson_ | not gsm |
17:28.01 | bjohnson_ | Bell isn't gsm |
17:28.57 | kink0 | I see, there not GSM mobile, just wireless |
17:29.23 | bjohnson_ | cdma cellular |
17:29.26 | bjohnson_ | not wireless |
17:29.40 | bjohnson_ | oops .. maybe not |
17:29.52 | bjohnson_ | I thought I was in their cellular phone sales section |
17:29.59 | bjohnson_ | damn |
17:30.20 | bjohnson_ | not so cool then |
17:30.31 | slak- | i have a cdma from verizon |
17:30.45 | bjohnson_ | I thought it was CDMA cellular and BT |
17:30.48 | slak- | i think i prefer the sim cards better |
17:31.06 | darwin_35 | IceBrkr ... more input |
17:31.12 | bjohnson_ | for me, it all comes down to the service packages |
17:31.28 | bjohnson_ | I don't think service is much different in my area |
17:31.54 | slak- | verizon is the only service provider with a good signal everywhere |
17:32.10 | darwin_35 | this odbc is is not making sense |
17:32.10 | slak- | i snapped my cingular phone in half it pissed me off so much ;/ |
17:32.11 | darwin_35 | its pissing me off |
17:32.35 | slak- | everywhere in my state i should say |
17:32.54 | *** part/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com) |
17:33.21 | RoyK | darwin35: why odbc? |
17:33.42 | darwin_35 | iodbc/myodbc |
17:33.56 | darwin_35 | used for connection |
17:33.59 | RoyK | why use odbc to connect to mysql? |
17:34.07 | RoyK | why not use a native driver? |
17:34.15 | darwin_35 | only for the fax interface |
17:34.22 | RoyK | fax? |
17:34.38 | RoyK | fax over mysql? :) |
17:34.43 | RoyK | nah |
17:34.45 | RoyK | fax over odbc |
17:35.26 | RoyK | ~seen zoa |
17:35.29 | jbot | zoa is currently on #asterisk (1d 7h 46m 53s). Has said a total of 119 messages. Is idling for 2h 35m 20s |
17:35.29 | darwin_35 | it was thiw way before I got here |
17:35.29 | iDunno | fax over carrier pigeon. |
17:35.44 | iDunno | fax over tcp over smoke signals is more fun, though. |
17:35.49 | RoyK | darwin35: then change it :) |
17:36.02 | RoyK | darwin35: if you need a mysql connection, use nativer drivers |
17:36.09 | RoyK | far better than odbc imho |
17:36.27 | RoyK | native, even |
17:36.28 | darwin_35 | [macro-faxreceive] |
17:36.28 | darwin_35 | exten => s,1,SetVar(FAXFILE=/usr/local/asterisk-fax/${CALLEDFAX}/${UNIQUEID}.tif |
17:36.28 | darwin_35 | exten => s,2,ODBCget(EXTEMAIL=${MACRO_EXTEN}/xEmail) |
17:36.28 | darwin_35 | exten => s,3,NoOP() |
17:36.28 | darwin_35 | exten => s,4,ODBCget(EXTNAME=${MACRO_EXTEN}/xName) |
17:36.29 | darwin_35 | exten => s,5,NoOP() |
17:37.31 | darwin_35 | I am just trying to get working what we have |
17:37.57 | darwin_35 | and it works on 4 other boxes but not this new box |
17:38.01 | RoyK | why did it stop working? |
17:38.07 | RoyK | ah |
17:38.08 | RoyK | ok |
17:38.20 | RoyK | prolly priorities 3 and 5 blocking it |
17:38.31 | darwin_35 | ? |
17:38.35 | darwin_35 | explain |
17:38.57 | RoyK | obviously, noop makes the previous line never to operate! |
17:38.57 | darwin_35 | I am not a database person |
17:39.03 | RoyK | :D |
17:39.09 | Dandan | argh |
17:39.14 | RoyK | what sort of error do you get? |
17:39.24 | darwin_35 | the issues is the odbc is not connecting to the main dbserver |
17:39.28 | Dandan | it seems like i have a problem with x100p not receiving digits fast enough |
17:39.38 | Dandan | anyone had that problem? |
17:40.01 | RoyK | darwin_35: ah. have you checked the acl on the server? |
17:40.06 | RoyK | mysql grants |
17:40.06 | RoyK | etc |
17:40.51 | darwin_35 | Dec 20 10:40:13 NOTICE[7215]: res_odbc.c:294 load_odbc_config: registered database handle 'DSNvoicemail' dsn->[DSNvoicemail] |
17:40.51 | darwin_35 | Dec 20 10:40:13 NOTICE[7215]: res_odbc.c:552 odbc_obj_connect: Connecting DSNvoicemail |
17:40.51 | darwin_35 | Dec 20 10:40:13 WARNING[7215]: res_odbc.c:563 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=1045 [MySQL][ODBC 3.51 Driver]Access denied for user 'teliax'@'localhost' (using password: YES) |
17:41.04 | RoyK | http://dev.mysql.com/doc/refman/5.0/en/show-grants.html |
17:41.05 | darwin_35 | same for all parts of the db |
17:41.15 | RoyK | localhost? |
17:41.28 | darwin_35 | where its getting local host I cant find |
17:41.29 | RoyK | do all servers run with local mysql installations? |
17:41.37 | RoyK | res_odbc.conf perhaps |
17:42.30 | darwin_35 | nope |
17:42.53 | RoyK | then the odbc config |
17:43.00 | *** join/#asterisk apardo (n=apardo@62-15-237-162.inversas.jazztel.es) |
17:43.00 | RoyK | find /etc -iname "*odbc*" |
17:43.01 | darwin_35 | roy can I paste pvt for a min |
17:43.06 | RoyK | pastebin |
17:43.09 | RoyK | ~pb |
17:43.11 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ |
17:43.26 | RoyK | hm. pastebin.ca is better imho |
17:43.35 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
17:43.45 | Dandan | RoyK: it is just that they use broken ECN |
17:44.09 | RoyK | Dandan: odbc uses broken ECN? |
17:44.14 | Dandan | no |
17:44.15 | Dandan | pb.ca |
17:44.16 | darwin_35 | http://pastebin.com/471892 |
17:44.24 | Blackthorn | Any suggestions on why a sipura spa-2000 that is behind nat would drop 30 seconds into the call about 50% of the time, but does not always drop? |
17:44.41 | darwin_35 | we are on mysql 4.1.15 |
17:44.56 | RoyK | danalien: wtf is ecn? |
17:45.31 | *** join/#asterisk chris-fn (n=chris@netblock-66-51-202-171.dslextreme.com) |
17:45.59 | chris-fn | i'm having a tad bit of trouble with Asterisk::Manager, anyone available for a sec? |
17:46.06 | darwin_35 | we have the server defined |
17:46.12 | RoyK | darwin35: what about udbc.ini? |
17:46.26 | darwin_35 | udbc.ini ? |
17:46.31 | RoyK | asdf.ini |
17:46.34 | RoyK | odbc.ini :) |
17:46.46 | Dandan | RoyK: explicit congestion notification |
17:47.02 | RoyK | ahki |
17:47.03 | RoyK | ic |
17:47.09 | Dandan | :) |
17:47.33 | RoyK | darwin_35: see http://www.minisoft.com/pages/middleware/ODBC_UNIX/odbc_for_unix_Tracing.htm |
17:47.39 | RoyK | darwin_35: under examples |
17:48.17 | RoyK | darwin_35: http://pastebin.com/471898 |
17:49.52 | *** join/#asterisk arcraig (n=arcraig@c-67-187-184-184.hsd1.ca.comcast.net) |
17:50.06 | darwin_35 | http://pastebin.com/471900 |
17:51.11 | Dandan | Dec 20 12:50:58 WARNING[1091]: channel.c:2530 ast_request: No channel type registered for 'Console' |
17:51.11 | Dandan | Dec 20 12:50:58 NOTICE[1091]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Console' (cause 66 - Channel not implemented) |
17:51.14 | Dandan | what the hell is that? |
17:51.18 | Dandan | doesn't like my oss? |
17:51.35 | RoyK | darwin_35: sorry. dunno. i don't use odbc... |
17:51.53 | darwin_35 | ok |
17:52.01 | asteriskmonkey | can anyone tell me why zap id dialling the voicemail .. here is a small dump VoiceMail("Zap/6-1", "201@debitact") |
17:53.02 | *** join/#asterisk ManxPower (n=ewieling@200.sub-70-197-9.myvzw.com) |
17:53.10 | *** join/#asterisk freezer (i=leetiden@ACB20633.ipt.aol.com) |
17:53.12 | freezer | hi |
17:54.58 | asteriskmonkey | hey.. why is this happening my extensions i have exten => 201,2,VoiceMail(su201@debitact) and i get in the concolse VoiceMail("Zap/6-1", "201@debitact").. and it dosnt go to vm.. anyone |
17:55.21 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
17:55.53 | RoyK | zoa: ping |
17:57.26 | docelm0 | PONG! |
17:57.32 | *** join/#asterisk bmg505 (n=leon@dsl-146-5-195.telkomadsl.co.za) |
17:57.40 | AndyCap | bjohnson_: was it you who mentioned a sip phone supporting encryption a while back? |
17:58.09 | kink0 | what is the best way to gateway to GSM mobile from asterisk ? |
17:59.30 | AndyCap | kink0: there is something called chan_bluetooth, but I don't know if it's any good. |
18:03.13 | fugitivo | kink0: www.2n.cz |
18:05.05 | kink0 | fugitivo, yes, the no easy is to find something less than 6000 dls |
18:05.31 | fugitivo | cost of that is 800 euros |
18:05.38 | fugitivo | it's like 1200 dls |
18:05.51 | fugitivo | i think it's the only sip/gsm gateway |
18:06.19 | fugitivo | you can get analog gsm gateways, much cheaper |
18:06.49 | msw | 7 |
18:06.51 | msw | doh |
18:08.28 | kink0 | fugitivo, I was thinking about the Stargate, 30 GSM channels gateway |
18:09.49 | kink0 | AndyCap, reding about that channel... appears interesting. |
18:10.11 | *** join/#asterisk loick (n=loick@APuteaux-151-1-65-183.w81-249.abo.wanadoo.fr) |
18:10.48 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
18:10.48 | *** mode/#asterisk [+o drumkilla] by ChanServ |
18:11.16 | *** join/#asterisk loick (n=loick@APuteaux-151-1-65-183.w81-249.abo.wanadoo.fr) |
18:15.41 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
18:16.13 | kink0 | chan_bluetouch is able to manage multiple phones ? are AT ( data channel ) on the same bluetouch communication than voice ? |
18:16.33 | Blackthorn | Question: if your sip.conf has nat=yes and qualify=yes and your spa-2000 unit is behind nat. Do you need to set nat-keep-alive=yes on the spa as well? |
18:18.15 | kink0 | I have version 1.2.0-rc1 but appears there nothing about chan_bluetouch |
18:20.15 | Dandan | bluetouch? or bluetooth? |
18:20.36 | puzzled | kink0: chan_bluetooth is not part of the asterisk source |
18:20.45 | [TK]D-Fender | chan_bluetooth is NOT part of Asterisk main. Its a 3rd party add-on. |
18:22.32 | fugitivo | chan_deathtouch |
18:22.51 | SkramX | chan_wrathofdeath |
18:23.27 | mog_work | anyone know libcurl? |
18:24.04 | fugitivo | mog_work: no, but i disable it at compilation time |
18:24.27 | mog_work | heh thanks |
18:25.36 | Dandan | anyone wanna share his overhead paging extensions syntax? |
18:28.09 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
18:28.10 | *** join/#asterisk saftsack (n=oliver@p54A7FDDB.dip.t-dialin.net) |
18:28.32 | mog_work | just dial(console/dsp) |
18:28.38 | mog_work | and set it to auto answer |
18:28.56 | mog_work | or thats what i thought it was |
18:28.59 | saftsack | can i plug a normal fax telephone on my asterisk server without using soandsdp? |
18:29.12 | fugitivo | yes |
18:29.21 | saftsack | and it works? :) |
18:29.30 | fugitivo | maybe |
18:29.53 | Blackthorn | Question: if your sip.conf has nat=yes and qualify=yes and your spa-2000 unit is behind nat. Do you need to set nat-keep-alive=yes on the spa as well? |
18:30.04 | saftsack | fugitivo, ? ^ |
18:30.16 | fugitivo | Blackthorn: try it |
18:30.56 | fugitivo | saftsack: maybe it'll work 100%, maybe 50%, what hardware are you using? |
18:31.07 | saftsack | i have a tdm wildcard |
18:31.35 | fugitivo | tdm400? |
18:31.40 | Dandan | mog_work: i do not hear anything |
18:31.44 | Dandan | through speakers :/ |
18:31.55 | Dandan | neither it plays any GSM files |
18:31.57 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
18:32.30 | saftsack | fugitivo, yes |
18:32.33 | SkramX | Hi, ctooley... We spoke a couple weeks ago.. |
18:32.49 | fugitivo | saftsack: well, probably it'll work, but it depends on your line and the line who's calling |
18:32.53 | *** join/#asterisk osirus (i=osirus@dhcp-100.fresno-dc2.brandxnet.com) |
18:33.01 | fugitivo | saftsack: i have a customer with that config, and it works 98% |
18:33.29 | saftsack | ok |
18:33.36 | saftsack | what do you mean with 98%? ^^ |
18:34.15 | fugitivo | from 100 transmissions, 98 faxes work ok, 2 fail |
18:34.16 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
18:34.19 | saftsack | ok |
18:34.26 | saftsack | are they work on the 2. try? |
18:35.05 | osirus | we use it, and the faxes do fail from time to time |
18:35.17 | ctooley | SkramX, yeah I remember |
18:35.23 | ctooley | SkramX, how are you? |
18:35.41 | *** join/#asterisk kimosabe (n=kimosabe@dsl-201-129-75-8.prod-infinitum.com.mx) |
18:35.41 | SkramX | I am alright |
18:35.47 | SkramX | Sorry, I had to get the door. |
18:35.48 | osirus | with the same percentile, just about |
18:35.50 | saftsack | osirus, hmm are the faxes directly connected with the wildcard or do you use spandsp? |
18:35.59 | SkramX | We just got a huge box of grapefruits.. must be a holiday present |
18:36.10 | osirus | directly connected |
18:36.24 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
18:36.32 | saftsack | with spandsp it would be more crappy or? |
18:36.54 | SkramX | ctooley: I will be right back.. you can PM me if you arent busy.. is there a reason I was never emailed back :| |
18:37.03 | osirus | I couldnt tell you |
18:37.08 | fugitivo | spandsp works perfectly for me |
18:37.35 | kimosabe | is the zircom instant asterisk disc good can ii also add gui to thius |
18:37.35 | fugitivo | but i use the x100p for that |
18:37.56 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
18:38.38 | saftsack | has someone of you a isdn deviceacces? |
18:38.44 | osirus | has anyone had any experience with dropped calls being sent over pstn from covad? |
18:39.01 | bjohnson_ | AndyCap: no. I haven't seen an encrypted sip phone yet |
18:39.30 | bjohnson_ | AndyCap: any encryption I've heard of in an implemented system is with a VPN |
18:39.35 | SkramX | -back0 |
18:39.39 | SkramX | *-back- |
18:40.36 | *** join/#asterisk jake1932 (n=jake1932@pool-68-236-10-151.phil.east.verizon.net) |
18:41.23 | [TK]D-Fender | AndyCap : Sipura SPA-941 supports SRTP (Secure RealTime Protocol) |
18:41.23 | osirus | i have a customer that transfers calls to me from a covad line, and we're getting lots of complaints about dropped calls |
18:42.33 | saftsack | i have three telephones here on three places. and if someone is in a room asterisk should know this. can asterisk "see" if an isdn telephone is connected? |
18:42.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
18:43.47 | rob0 | grapefruit can be good weapons! Especially if overripe. |
18:44.05 | *** part/#asterisk Sixam (n=killa666@71.83.113.93) |
18:44.06 | osirus | or frozen |
18:45.25 | saftsack | can i "ping" an isdn telephone? |
18:45.40 | jake1932 | is there any way I can turn off one T100P card (have 2 in total) without affecting the other on a live system? |
18:46.06 | darwin_35 | has the mysql leak with res_mysql been fixed |
18:46.40 | darwin_35 | the only reason my boss says we are usiing odbc |
18:46.52 | darwin_35 | was the mem leak |
18:47.16 | AndyCap | [TK]D-Fender: ok. thanks. got to google I guess. it was a not a well known brand. |
18:48.01 | [TK]D-Fender | Not well known? Thats "interesting". Sipura is quite highly regarded.... |
18:48.54 | fugitivo | hehe |
18:49.09 | darwin_35 | I had issues withthe sipura spa-841 |
18:49.22 | darwin_35 | <PROTECTED> |
18:49.23 | [TK]D-Fender | What kind of issues? |
18:49.27 | [TK]D-Fender | hmm |
18:49.44 | darwin_35 | and some times pick up a call and it would reboot |
18:49.46 | [TK]D-Fender | WEll the 941 has proven to be a pretty solid business phone so far... |
18:49.50 | AndyCap | [TK]D-Fender: yeah, I know sipura . but I was looking for a phone someone mentioned to me a few months ago and that one was not a well known brand |
18:49.57 | fugitivo | AndyCap: also, sipura is now cisco |
18:50.20 | AndyCap | fugitivo: argh, is nothing sacred? |
18:50.20 | AndyCap | :) |
18:50.22 | [TK]D-Fender | Or rather Sipura is now LInksys who is now Cisco ;) |
18:50.25 | darwin_35 | I have moved to pa-168 based chip phones now |
18:50.35 | [TK]D-Fender | PA168? EW |
18:50.38 | fugitivo | yeah, what d-fender said :) |
18:50.50 | AndyCap | just so they could get their hands on the next model ATA? :-P |
18:50.51 | [TK]D-Fender | UBER cheap junk... |
18:50.54 | fugitivo | darwin_35: that sucks |
18:51.29 | [TK]D-Fender | Only phones I recommend at this point is the SPA-941 and anything Polycom. |
18:53.47 | jake1932 | let me revise that last q... i have 2 TE110P cards in a live system. is there a way I can disable one card without affecting the calls on the other? |
18:54.31 | AndyCap | [TK]D-Fender: and snom? |
18:54.36 | fugitivo | jake1932: disable it in which way? |
18:54.47 | fugitivo | jake1932: unplug the cable? |
18:54.55 | jake1932 | i'm not at the location |
18:55.05 | jake1932 | but yes, essentially unplug the cable |
18:55.17 | jake1932 | (without actually unplugging it) |
18:55.21 | [TK]D-Fender | AndyCap : What about Snom? |
18:55.56 | AndyCap | [TK]D-Fender: just wondering if they'd make your not recommended list or unkown |
18:56.11 | fugitivo | jake1932: you could try to deactivate it with setpci, but it's dangerous on a production system :) |
18:56.58 | jake1932 | the one card is in and out of yellow alarm. i just don't want it to take any more calls |
18:56.58 | fugitivo | jake1932: you could remove the config from /etc/zaptel.conf and reload the modules, but you don't want to do that, right? |
18:57.01 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
18:57.19 | jake1932 | will that affect the calls on the working card? |
18:58.00 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
18:58.18 | [TK]D-Fender | AndyCap : Sno screen looks terrible, awkward little buttons.The functionality seems ok though. Still... I'd rather suggest a Polycom over it. |
18:58.48 | *** join/#asterisk kop_ (n=kopcicle@71-37-20-222.tukw.qwest.net) |
18:59.20 | fugitivo | jake1932: unloading the modules, yes |
19:00.05 | jake1932 | and ztcfg - prob will also cause the calls to drop.. |
19:01.18 | fugitivo | jake1932: maybe with zap destroy from the cli |
19:01.26 | fugitivo | jake1932: you'll need to destroy each channel |
19:01.41 | jake1932 | ok |
19:01.45 | slak- | drink a beer |
19:01.49 | slak- | then think about it later |
19:07.41 | osirus | ! |
19:07.50 | jake1932 | i destoyed them |
19:08.07 | jake1932 | now how do i get them back when they fix the yellow alarm issue? |
19:08.27 | saftsack | some germans here? |
19:08.37 | osirus | hah |
19:08.39 | osirus | wha |
19:08.40 | osirus | t |
19:08.52 | saftsack | ? |
19:09.04 | osirus | germans |
19:09.04 | osirus | where! |
19:11.03 | saftsack | osirus, r u german? |
19:11.12 | saftsack | or better, do you speak german? |
19:11.38 | trixter | ich keine spreche deutch und nicht seir gut! |
19:11.45 | osirus | no |
19:11.47 | saftsack | hrhr |
19:11.55 | osirus | im hella whiteboy |
19:12.41 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
19:15.04 | *** part/#asterisk osirus (i=osirus@dhcp-100.fresno-dc2.brandxnet.com) |
19:15.10 | *** join/#asterisk RoyK (n=roy@host-n39-140.homerun.telia.com) |
19:15.29 | *** join/#asterisk SDGL (n=sdgl@64.5.206.131) |
19:15.40 | RoyK | ~seen zoa |
19:15.51 | jbot | zoa is currently on #asterisk (1d 9h 27m 15s). Has said a total of 119 messages. Is idling for 4h 15m 42s |
19:17.15 | *** join/#asterisk Insanity5 (n=feaw@ip68-111-5-23.sv.om.cox.net) |
19:17.49 | Insanity5 | Am I the only one who has noticed that finding a reliable origination provdider is just so damn difficult? Finding someone staple, where you phone number won't disappear in a few months, is damn near impossible. |
19:18.11 | fugitivo | vonage |
19:18.23 | Insanity5 | fugitivo $$$ -- rather just call the local telco |
19:18.26 | Insanity5 | And it won't work with * |
19:18.31 | *** join/#asterisk techie (i=gus@antibala.com) |
19:18.43 | fugitivo | well, then it's not impossible :) |
19:18.54 | fugitivo | you have to play their rules |
19:19.31 | darwin_35 | icebrkr you alive |
19:19.40 | Insanity5 | lol, running it through the ata back into a card? no thanks |
19:20.40 | iCEBrkr | -.- zzZZZ |
19:21.12 | *** join/#asterisk tomtom_ (n=tom@bender.linugen.com) |
19:21.16 | tomtom_ | hi |
19:21.57 | Insanity5 | hi |
19:22.10 | fugitivo | hello |
19:22.33 | saftsack | Dec 20 20:18:00 NOTICE[2176]: app_dial.c:764 dial_exec: Unable to create channel of type 'misdn' |
19:22.40 | saftsack | but theres no reason given :( |
19:23.08 | Primer | Anyone here with a cisco 7920? I've been trying to get it to work with my WPA enabled AP to no avail. Anyone have a 7920 working with WPA? |
19:23.35 | saftsack | ok i know what was wrong |
19:24.48 | saftsack | i defined a group in my misdn.conf and in the extensions.conf i call the group ,Dial(misdn/g1/${EXTEN}) |
19:24.53 | saftsack | but it doesnt work |
19:25.00 | saftsack | is the dialstring false? |
19:25.18 | tomtom_ | anyone knows what's causing these: Dec 20 20:18:07 WARNING[3955]: RTP Read error: Bad file descriptor ? |
19:25.38 | tomtom_ | I get them sometimes when transferring a call to a meetme room |
19:25.38 | *** join/#asterisk gene846 (n=gene@dslA-2.millry.net) |
19:30.42 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
19:30.43 | SpaceBass | hey folks |
19:31.09 | SpaceBass | I'm migrating my AAH setup to a new box but don't want to move my zaptel hardware yet... can I keep the * box with the zaptel stuff and just route it to my new box? |
19:32.07 | SpaceBass | I'm thinking just create a trunk and some routing... |
19:32.25 | fugitivo | yes |
19:32.37 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
19:33.56 | iCEBrkr | trunking is kinda neat |
19:35.40 | *** join/#asterisk saft (n=saft@ip-202-37-230-210.internet.co.nz) |
19:36.41 | *** join/#asterisk mungojam (n=admin@87.194.16.84) |
19:36.52 | iCEBrkr | Could it be any warmer in here?! |
19:37.00 | iCEBrkr | Geesh, you'd think it's 65 degrees outside. |
19:37.07 | mutilator | bleh |
19:37.20 | iCEBrkr | ...and where the hell are my PRIs?! |
19:37.24 | mutilator | 200 lines of code just to apply payments and thats not even doing any of the financing options |
19:37.26 | iCEBrkr | Device name | Protocol | Station | Status | |
19:37.26 | iCEBrkr | wanpipe1 | AFT HDLC | N/A | Disconnected | |
19:37.26 | iCEBrkr | wanpipe2 | AFT HDLC | N/A | Disconnected | |
19:37.27 | iCEBrkr | No love. |
19:37.36 | mutilator | i r teh hate accounting |
19:37.50 | mutilator | and english o_O |
19:37.58 | mutilator | :P~ |
19:38.00 | iCEBrkr | engrish? |
19:38.15 | *** join/#asterisk apardo (n=apardo@54.Red-83-50-238.dynamicIP.rima-tde.net) |
19:39.58 | [TK]D-Fender | iCEBrkr : Whats the telco have to say? |
19:40.01 | saft | wheeeeeee, i have an interesting issue here, i only get audio one way going Dlink DG104s -MGCP- * -iax- * -chan_sccp- Cisco VIP 30 |
19:40.24 | iCEBrkr | [TK]D-Fender: I put in a second call, I'm waiting for someone to call me back.. AGAIN |
19:40.49 | [TK]D-Fender | DUMB%#$^s |
19:40.57 | SpaceBass | so which box has the trunk and which has the account? |
19:41.01 | iCEBrkr | My rep never calls me back |
19:41.11 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
19:41.18 | iCEBrkr | SpaceBass: Think client-server :) |
19:41.21 | SpaceBass | If box A has the zaptel and will answer those calls and forward to box B... then B has the trunk |
19:41.38 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
19:41.39 | iCEBrkr | I'd make B register with A. |
19:41.40 | *** join/#asterisk Laibsch (n=Laibsch@p54B9B627.dip0.t-ipconnect.de) |
19:41.43 | SpaceBass | iCEBrkr thats what Im trying to do... mostly thinking out loud here |
19:41.43 | shanky | hi, good evening |
19:42.03 | shanky | anyone uses areski-stats ? |
19:42.25 | *** join/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net) |
19:42.41 | SpaceBass | B registers with A (zaptel box) ... thinking... that works for incoming and outgoing |
19:43.08 | shanky | I have a problem with the cdr-report, it use a sql sentences over cdr table which contains 'userfield' field and I don't have that field in the cdr table, must I alter the cdr table? |
19:43.59 | Laibsch | Hi, I am trying to compile the ztdummy module for a 2.6 kernel. I have the appropriate kernel sources and a symlink /usr/src/linux-2.6 to it as well as suggested by the README.Linux26. Still, when I do make linux26 I get a "You do not appear to have the kernel sources for your current kernel installed." Why is that? |
19:44.03 | mungojam | http://pastebin.ca/34375 |
19:44.24 | Dandan | anyone has any experience with OSS? |
19:44.28 | Dandan | i am still stuck :/ |
19:44.33 | fugitivo | oss? |
19:44.34 | *** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no) |
19:44.42 | Dandan | as in console/dsp |
19:44.42 | fugitivo | open sound system? |
19:44.53 | fugitivo | that's obsolete, use alsa |
19:45.12 | RoyK | ~seen zoa |
19:45.16 | jbot | zoa is currently on #asterisk (1d 9h 56m 40s). Has said a total of 119 messages. Is idling for 4h 45m 7s |
19:45.20 | RoyK | argh |
19:45.20 | Dandan | alsa... hm lets try :) |
19:46.04 | iCEBrkr | RAAWWWWWRR! |
19:46.08 | saftsack | <PROTECTED> |
19:46.15 | Dandan | iCEBrkr: ? |
19:46.23 | saftsack | this is my dialstring but if i dial a number it just calls the first two digits |
19:46.27 | saftsack | because it doesnt wait |
19:46.30 | iCEBrkr | [TK]D-Fender: Apparently, the remote/colo PRI is turned-up.. Yet, I can't dial |
19:47.07 | iCEBrkr | [TK]D-Fender: It would appear I have the Digium card configured correctly down there since zttools reports 'OK' |
19:47.15 | jake1932 | saftsack: did you do a show dialplan? |
19:47.27 | iCEBrkr | Dandan: I'm going round and round with our provider to get PRIs installed and turned-up. |
19:47.39 | saftsack | jake1932, show dialplan? |
19:47.44 | jake1932 | saftsack: if not, can you pastebin your "show dialplan" |
19:47.51 | jake1932 | in the CLI |
19:48.14 | saftsack | ok |
19:48.16 | saftsack | moment |
19:48.26 | *** part/#asterisk Primer (n=vi@sh.nu) |
19:48.36 | [TK]D-Fender | iCEBrkr : PM |
19:48.43 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
19:48.45 | saftsack | http://pastebin.com/472098 |
19:48.59 | jake1932 | what do you dial? |
19:49.12 | saftsack | i dial in context raus |
19:49.17 | saftsack | 0mynumber |
19:49.28 | saftsack | and asterisk should do that too |
19:49.34 | saftsack | but it just takes 05 |
19:49.35 | jake1932 | does it wait? |
19:49.37 | Dandan | ALSA usbaudio.c:870: timeout: still 8 active urbs.. |
19:49.42 | Dandan | alsa :/ |
19:49.52 | jake1932 | saftsack: soe your phone have a dialplan? |
19:49.54 | jake1932 | does |
19:50.04 | saftsack | what for a phone? |
19:50.12 | saftsack | yes i think so |
19:50.22 | jake1932 | saftsack: what phone are you using? |
19:50.24 | saftsack | <PROTECTED> |
19:50.33 | saftsack | an isdn telephone |
19:50.58 | jake1932 | that you're calling from, to, or both? |
19:51.47 | mungojam | Hi, having searched the wiki and isdn4l list I am unclear on just how stable or easy to get working a BRI card is within linux and asterisk, and they are rather expensive to find out for myself. Any experiences? |
19:52.09 | jake1932 | mungojam: just in the US - for BRI |
19:52.13 | shido6 | anyone in michigan? |
19:52.50 | jake1932 | mungojam: wasn't hard - just time consuming pinning down all the info. what card are you using? |
19:52.58 | mungojam | jake: I don't quite understand, I thought BRI wasn't available in the US, or is it that it works differently? |
19:53.21 | jake1932 | mungojam: definately available -PIA to get (and get rid of) :) |
19:53.50 | mungojam | Hi, well I don't have a card, and I don't have the money to buy one unless i can be 95% sure that I will be able to get it working without reading every mailing list in the world |
19:54.02 | jake1932 | mungojam: can't guarantee either |
19:54.42 | jake1932 | mungojam: i have an AVG card, though that I got pretty much working, however, I soon after realized I couldn't use it in the US and went to a DIVA card |
19:54.55 | mungojam | :) yes ofcourse, I guess I am asking: is all the isdn support in linux and asterisk still at beta quality? |
19:55.12 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
19:55.17 | mungojam | I am in the UK |
19:56.50 | jake1932 | mungojam: for non-us, http://www.junghanns.net/en/chan_capi.htmlt |
19:57.14 | fugitivo | mungojam: try it yourself, it's opensource, no guarantee |
19:58.05 | jake1932 | <PROTECTED> |
19:58.15 | fugitivo | avg is the antivirus :) |
19:58.19 | jake1932 | right |
19:59.00 | jake1932 | AVM Fritz is the one I got (that I have to resell) |
19:59.33 | RoyK | chan_capi works with the fritz |
19:59.45 | jake1932 | just not in the US |
19:59.52 | RoyK | maybe :) |
20:00.10 | jake1932 | i know from testing |
20:00.28 | RoyK | oh well |
20:00.44 | jake1932 | <PROTECTED> |
20:01.31 | mungojam | fugitivo: I am normally quite willing to try stuff for myself, but the cost of a 4 port bri card is just way too risky if they are generally iffy |
20:01.37 | SpaceBass | anyone using NuFone and AMP? |
20:01.48 | Ariel_ | SpaceBass, yes |
20:01.59 | jake1932 | saftsack: you still there? |
20:02.07 | saftsack | yes |
20:02.12 | SpaceBass | Ariel_ did you have to manually add the dial string to extensions_custom.conf or did you set up the trunk in AMP? |
20:02.17 | mungojam | somewhere around £900 |
20:02.22 | tomtom_ | Nobody any info on this error Dec 20 20:18:07 WARNING[3955]: RTP Read error: Bad file descriptor ? |
20:02.31 | jake1932 | <PROTECTED> |
20:02.34 | mungojam | thanks for the help, bye |
20:02.35 | *** part/#asterisk mungojam (n=admin@87.194.16.84) |
20:02.44 | saftsack | jake1932, yes |
20:03.43 | *** join/#asterisk backblue (n=moo@87-196-10-196.net.novis.pt) |
20:04.08 | RoyK | can someone please take a look at this patch to help me find wtf is wrong with it? http://karlsbakk.net/asterisk/scripts/asterisk-mrtg |
20:04.20 | RoyK | er |
20:04.22 | RoyK | sorry |
20:04.29 | RoyK | http://karlsbakk.net/asterisk/patches/1.2.x/zoa-jb-2005-12-20.patch |
20:04.30 | RoyK | that one |
20:05.27 | SpaceBass | anyone using a cisco phone and registering to more than one asterisk box? |
20:06.26 | fugitivo | RoyK: is that a patch? it looks like a windows service pack ;) |
20:06.32 | backblue | SpaceBass: why do you want to register in more then one asterisk? |
20:07.00 | SpaceBass | backblue b/c I have 2 servers running currently and it would facalitate testing trunking b/t them |
20:07.02 | RoyK | fugitivo: heh. the latter is sip jitterbuffer patch |
20:08.26 | *** join/#asterisk Dutts (n=dutts@81.168.70.41) |
20:08.36 | *** part/#asterisk Dutts (n=dutts@81.168.70.41) |
20:08.45 | *** join/#asterisk netvulture (n=vulture@adsl-63-197-17-60.dsl.snfc21.pacbell.net) |
20:08.54 | backblue | SpaceBass: b/c? |
20:09.19 | backblue | SpaceBass: you can forward calls bettwen them, dont need to register on both. |
20:09.27 | SpaceBass | b/c I want to :) |
20:09.37 | saftsack | jake1932, any ideas? |
20:09.39 | backblue | when does your cisco should know when to use *1 or *2? |
20:09.43 | marcus2 | anyone have polycom sip 1.6.3 firmware handy? |
20:09.53 | [TK]D-Fender | I do. |
20:09.59 | SpaceBass | b/c I have a 7060 that can (in theroy) register with more than one sip server and I need to test calls b/t the two and that's how I'd prefer to do it |
20:10.09 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
20:10.19 | SpaceBass | Also thinking about using it for extension based routing |
20:11.00 | SpaceBass | backblue I'm using an online cgi based tool to generate the cisco config file and it supports a server/port for each extension, but I'm having problems getting it to register |
20:11.04 | netvulture | whats up guys - I have an issue with calls from Teliax going to Voicemail - Dec 20 11:27:30 WARNING[28102] app.c: No audio available on IAX2/teliax - Any ideas?? Is it teliax? Is it me? Is it my connection? Is it app_voicemail? Is it asterisk in general? |
20:11.25 | darwin_35 | codecs |
20:11.41 | netvulture | g711u |
20:11.45 | backblue | SpaceBass: i dont see why do you need yor 7960 to register in both * |
20:11.56 | backblue | jus forward calls bettwen them |
20:12.02 | fugitivo | netvulture: check codecs |
20:12.07 | netvulture | should I try to force the codec before answering? |
20:12.13 | SpaceBass | well, frankly, that is what I'm trying to do and curious if anyone has expirence with it |
20:13.07 | backblue | SpaceBass: but you need 2 clients, not 1 client, and 2 servers. |
20:13.08 | harryvv | vonage has been given a extra 250 mill in cash. You would think if its been around this long it would have enough profits to not ask for more. |
20:13.22 | backblue | SpaceBass: and by the way, that its so very simple |
20:13.22 | netvulture | ok - thanks - i'll check those out |
20:13.32 | backblue | you do that, and have examples |
20:13.36 | backblue | in voip-info |
20:13.53 | backblue | like, connecting to any voip free provider, its exacly the same. |
20:14.08 | backblue | just insted, use only your two * box |
20:14.18 | SpaceBass | I'm really not a nube... |
20:15.36 | jake1932 | <PROTECTED> |
20:16.01 | saftsack | jake1932, ok thanks |
20:16.10 | saftsack | hmm i read that this is a normal misdn issue |
20:16.21 | saftsack | but i can remember, that it works some days ago |
20:17.17 | *** join/#asterisk loick (n=loick@APuteaux-151-1-65-183.w81-249.abo.wanadoo.fr) |
20:18.42 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
20:18.45 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
20:19.20 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
20:22.52 | *** join/#asterisk justinu (n=j2@72.18.13.34) |
20:23.35 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
20:23.38 | SpaceBass | will IAX2 pass DID (trunking from one box to aother) or do I need to a /<exten> to my register string |
20:24.37 | saftsack | my asterisk doesnt hangup if the timeout was reached :( |
20:24.43 | saftsack | <PROTECTED> |
20:24.43 | saftsack | <PROTECTED> |
20:24.50 | saftsack | after 10 seconds it should hanguo |
20:27.22 | shido6 | anyone want a heart attack? |
20:27.33 | shido6 | http://media.spikedhumor.com/8944/Jingle_Bells_Reversed.swf |
20:28.14 | justinu | http://video.google.com/videoplay?docid=4845715794200371561 |
20:28.17 | justinu | fast forward to the end |
20:28.35 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) |
20:28.45 | Madkiss | hi all; what's the current chan_capi version and where do I get it? |
20:29.17 | fugitivo | shido6: GOD |
20:29.37 | arcraig | 0 /exit |
20:29.38 | arcraig | cd |
20:29.41 | arcraig | sorry :x |
20:32.44 | *** join/#asterisk santiago (n=santiago@208.195.215.247) |
20:35.11 | *** join/#asterisk bmg505 (n=leon@dsl-146-5-195.telkomadsl.co.za) |
20:37.37 | *** join/#asterisk JohnJacob (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
20:38.21 | saftsack | howto play the congested tone to the caller? |
20:39.22 | harryvv | google it should come up |
20:39.33 | harryvv | congestion asterisk |
20:39.35 | saftsack | ive read playtone but i dont thing that its correct or? |
20:39.42 | justinu | Congestion |
20:39.54 | saftsack | i have congestion already |
20:40.00 | saftsack | but it doesnt work :( |
20:40.10 | saftsack | maybe because theres a tk before my asterisk? |
20:40.52 | saftsack | <PROTECTED> |
20:40.52 | saftsack | <PROTECTED> |
20:40.53 | saftsack | so? |
20:41.21 | justinu | misdn... can't help you there |
20:41.29 | saftsack | ok thanks |
20:41.40 | saftsack | i have the feeling, that misdn is very buggy :( |
20:41.55 | justinu | could be... is the inbound call BRI also? |
20:42.24 | SpaceBass | I'm trunking b/t two * boxes, the client registeres successfully but inbound routing is not working... does IAX2 pass DID info by default? |
20:42.28 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
20:43.52 | *** part/#asterisk mog_work (n=mogorman@gateway.digium.com) |
20:44.59 | lesouvage | FWD suddenly stopped working on my Asterisk box. 612 (the time) isn't answered while it is while using sjphone. Is there a problem with FWD? |
20:45.19 | fugitivo | there're always problems with FWD |
20:45.27 | xachen | yeah |
20:45.34 | xachen | I never can get FWD to work with my asterisk box |
20:45.46 | xachen | IAXtel is a bugger too. I just use e164.org :) |
20:46.05 | lesouvage | xachen: I had it working, didn't change it and now it stopped working. |
20:46.40 | darwin_35 | we get it to work but only 1/3rd the time |
20:46.41 | darwin_35 | fwd and iaxtel |
20:47.05 | darwin_35 | I have yet to figure ot enum |
20:47.12 | darwin_35 | not had time |
20:47.17 | darwin_35 | or dundi |
20:47.53 | *** join/#asterisk Psykick (n=anon@smtp.phoenixone.co.nz) |
20:49.23 | lesouvage | So there is a good change that there is an external reason for the failure of my fwd account? |
20:50.25 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
20:50.27 | asteriskmonkey | hey |
20:50.38 | darwin_35 | what now monkey |
20:50.38 | *** join/#asterisk DJ-Pyro (n=DJ-Pyro@lan-gw.brevient.net) |
20:50.54 | darwin_35 | get off my back will you I am getting sore |
20:50.59 | asteriskmonkey | can anyone help me with this.. i have alot of people using iaxys and after the upgrade to 1.2.1 alot of there calls result in the caller not being able to hear them talk |
20:51.06 | asteriskmonkey | lol |
20:51.31 | iCEBrkr | There's a when and why to upgrade.. |
20:51.42 | iCEBrkr | Sounds like you disobeyed the when. |
20:51.44 | DJ-Pyro | anyone ever experience bad echo when the callers are coming in via DS1's? we have a DS3 worth of DS1s coming in and we have occasional complaints about really bad echo for the user, we have echocancel=yes in zapata but it's not helping |
20:51.58 | darwin_35 | back step to 1.2 and test if the issue is not there then its a bug |
20:52.11 | asteriskmonkey | DJ-Pyro |
20:52.19 | asteriskmonkey | DJ-Pyro: change the echo can |
20:52.21 | darwin_35 | IceBRkr |
20:52.29 | saftsack | if i want to dial ALL my telephones on my NT ports and if the telephones have the msn 10 and 20 i have to do so or? |
20:52.30 | asteriskmonkey | DJ-Pyro: and tweak the rx |
20:52.34 | saftsack | ISDNalle=misdn/g:NTPORTS/20&misdn/g:NTPORTS/10 |
20:52.35 | DJ-Pyro | something other than the default 128? |
20:52.38 | saftsack | or can i do this in one string? |
20:52.42 | iCEBrkr | ? |
20:52.51 | darwin_35 | I have to shoot oyu |
20:53.01 | darwin_35 | we have tried everythign and still no go |
20:53.11 | DJ-Pyro | asteriskmonkey: I thought rx was typically non TDM circuits |
20:53.16 | darwin_35 | its killing me |
20:53.26 | asteriskmonkey | no , pri's have gain :D |
20:53.27 | iCEBrkr | shoot me? you're the ones who opt'd to use ODBC :) |
20:53.29 | darwin_35 | I fallowed the odbc page youpointe dout |
20:53.32 | *** join/#asterisk Hmmhesays (i=Blorp@66.173.103.100) |
20:53.44 | asteriskmonkey | can i rollback to 1.0.9? |
20:53.45 | darwin_35 | no that was the person befor me |
20:53.50 | asteriskmonkey | do i have to rm everything first? |
20:54.00 | Hmmhesays | anyone in the uk I can call to see if my callerid is getting across BT? |
20:54.13 | darwin_35 | and now i am cleaning up a mess |
20:54.39 | *** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no) |
20:54.57 | darwin_35 | can you point out any ideas more that might help you said you got it working |
20:55.18 | iCEBrkr | darwin_35: screw real-time? |
20:55.28 | darwin_35 | I plan to move to postgress but I have to have this working like tonight |
20:55.34 | darwin_35 | cant |
20:55.40 | darwin_35 | or I would |
20:55.54 | asteriskmonkey | i can roll back to 1.0.9 right?? |
20:56.08 | [TK]D-Fender | DJ-Pyro : HOw many T1's? |
20:56.38 | Hmmhesays | UK anyone? |
20:56.39 | DJ-Pyro | [TK]D-Fender: 28 |
20:56.46 | DJ-Pyro | 7 asterisk servers with 4 t1's in each |
20:57.06 | darwin_35 | I beg of you o iced one make me your padawond and show me the ways to get it working |
20:57.08 | iCEBrkr | darwin_35: I dunno, sounds like more of a pain in the ass to me. |
20:57.15 | darwin_35 | yeah it is |
20:57.58 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
20:59.05 | *** join/#asterisk RoyK (n=roy@host-n39-140.homerun.telia.com) |
20:59.13 | RoyK | ~seen zoa |
20:59.24 | jbot | zoa is currently on #asterisk (1d 11h 10m 48s). Has said a total of 119 messages. Is idling for 5h 59m 15s |
20:59.29 | asteriskmonkey | is there anyone that has gone from 1.2.1 back to 1.0.9? |
20:59.48 | RoyK | a few times, yes |
20:59.53 | iCEBrkr | asteriskmonkey: whyd' you go to 1.2.1 anyhow? |
21:00.04 | RoyK | after trying to upgrade and than found 1.2.1 too unstable...... |
21:00.08 | asteriskmonkey | I wanted to have the more extended iax2 info stuff |
21:00.20 | iCEBrkr | look where that got'cha |
21:00.27 | asteriskmonkey | what do i have to do to roll back to 1.0.9 |
21:00.28 | Hmmhesays | UK anyone? |
21:00.29 | RoyK | 1.2 isn't really stable yet |
21:00.30 | denon | anyone have some 7960s they want to sell? |
21:00.35 | SpaceBass | I'm having problems with NuFone ... just using peer settings but I'm getting all circuits busy whenI try and dial out |
21:00.43 | *** join/#asterisk tidify (n=tidify@24-182-200-159.dhcp.ftwo.tx.charter.com) |
21:00.50 | asteriskmonkey | denon: are you in canada? |
21:00.53 | denon | nope |
21:00.58 | iCEBrkr | asteriskmonkey: copy your 1.0.9 bin's back into place. |
21:00.58 | *** join/#asterisk darby_t (i=darby_t@dlc120.neoplus.adsl.tpnet.pl) |
21:01.06 | denon | asteriskmonkey: close though, Minnesota. :) |
21:01.09 | iCEBrkr | asteriskmonkey: undo your 1.2.1 extensions.conf changes. |
21:01.11 | RoyK | asteriskgeeks: just install remove everything under /usr/lib/asterisk/modules/* and libpri, zaptel and asteterisk, possibly -addons and restart |
21:01.14 | Hmmhesays | haha you too huh denon? |
21:01.28 | Psykick | anyone used or had any success with iaxclient library? |
21:01.29 | RoyK | asteriskmonkey: that one was for you..... |
21:01.33 | denon | 'course, all the cool people live in MN |
21:01.36 | asteriskmonkey | what about my pri/zaptel stuff do i have to remake and remodprobe? |
21:01.40 | Hmmhesays | where about? |
21:01.43 | iCEBrkr | denon: freezing cool people.. |
21:01.54 | RoyK | asteriskmonkey: running zaptel stuff? |
21:01.59 | iCEBrkr | asteriskmonkey: yup. make install zaptall/libpri |
21:02.07 | denon | Hmmhesays: well, I'm in mpls at the moment, but I get around |
21:02.18 | denon | you? |
21:02.25 | Hmmhesays | i'm sitting in moorhead right now |
21:02.37 | denon | ah .. college kid? |
21:02.42 | Hmmhesays | naw |
21:02.45 | Hmmhesays | grew up here |
21:02.48 | asteriskmonkey | ah ... i hope the iaxy.bins that got upped arnt the cause of the issue either :P |
21:02.49 | denon | ic |
21:03.00 | denon | asteriskmonkey: you have some phones you want to sell over the border |
21:03.00 | denon | ? |
21:03.10 | Hmmhesays | lookin for a freaking Uk test number |
21:03.12 | Hmmhesays | so I can go home |
21:03.14 | harryvv | is it possible to have a asterisk or some script to test the integredy of a voip call though a service before the call is actually made? I want it to test for duplex latency ect. |
21:03.26 | iCEBrkr | LOL |
21:03.29 | denon | harryvv: you mean like sip qualification? |
21:03.44 | justinu | i think what he wants is like a MOS score of a test call |
21:03.49 | denon | yeah .. |
21:03.54 | denon | but thats not gonna happen :) |
21:03.56 | iCEBrkr | harryvv: yea, too bad your link won't be stable enough to test and even then it could degrade in the middle of a call. |
21:04.02 | justinu | not without doing some serious work :) |
21:04.07 | harryvv | denon, I suspect. can you break down the meaning of qualification? Twice I have made calls in the last few days and other side did not hear me. |
21:04.21 | denon | harryvv: ah well, that you could test .. |
21:04.21 | justinu | i've got the RTCP stuff working |
21:04.26 | Hmmhesays | gotta love the public internet |
21:04.35 | Hmmhesays | people expect it to be perfect |
21:04.43 | justinu | but i found both polycom and eyebeam send incorrect RTCP info in their sender reports. |
21:04.47 | justinu | Hmmhesays: why shouldn't it be? |
21:04.55 | justinu | if you just accept crap, you'll get crap |
21:05.02 | Hmmhesays | heh, why should it be |
21:05.05 | harryvv | hmm, so basicly there is no perfect voip model...or close to be perfect if there is no voip service terminating calls? |
21:05.21 | SpaceBass | anyone else having problems with NuFone? |
21:05.28 | Hmmhesays | anyone in here from the UK? |
21:05.31 | iCEBrkr | justinu: Cuz yea, you know. I'm the end user of the end user of the end user of Global-X.. and when my VoIP drops out, I'm gonna make a tech call to complain.. |
21:05.52 | justinu | global-x rocks |
21:05.59 | iCEBrkr | harryvv: leased-lines between your offices.. That'd be about it |
21:06.11 | iCEBrkr | justinu: that's why i used them as an example :) |
21:06.11 | harryvv | I guess the most reliable voip call and its probebly been dicussed in here before is having asterisk pbx's on both sites and terminating though them. |
21:06.29 | justinu | you could use some kind of teir 1 VPN solution |
21:06.37 | justinu | that'd probably get the QoS up to a decent level |
21:06.49 | harryvv | iCEBrkr yea, and how much do those lines cost? I mean if you want near 100% reliability of the physical link thats probebly it. |
21:07.05 | iCEBrkr | justinu: I actually had an issue with them.. One of their routers was dropping bits on packets. My FTP would eventually die, or if I got the file it was corrupt. Took 12hrs to debug. |
21:07.21 | asteriskmonkey | voip is really good if you can maintain a 50ms or lower route to the voip server |
21:07.25 | justinu | wow |
21:07.31 | justinu | that sounds like a serious pain in the ass |
21:07.44 | justinu | i've got 5ms latency to my PSTN gateway :) |
21:07.46 | asteriskmonkey | some people say high as 250, but to be hoest 50> you will occasional echo etc |
21:07.49 | iCEBrkr | haha |
21:08.00 | iCEBrkr | I don't get echo |
21:08.03 | harryvv | so 50 ms then |
21:08.04 | asteriskmonkey | justinu: i have somepeople that are 600k away with 15ms times :D |
21:08.05 | iCEBrkr | You people are nuts :D |
21:08.13 | Hmmhesays | or if you're from africa anything under 750ms is acceptable |
21:08.15 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
21:08.17 | justinu | lol |
21:08.20 | asteriskmonkey | lol |
21:08.23 | iCEBrkr | I had echo on my line once. |
21:08.23 | Hmmhesays | and you think i'm joking |
21:08.28 | justinu | i believe it |
21:08.43 | iCEBrkr | My VoIP just cracks and drops out. But never any echo |
21:08.52 | iCEBrkr | NEVER! |
21:08.53 | harryvv | I had one network reseller outright reject voip because of dropped packets. |
21:08.54 | asteriskmonkey | cracks are bad too :( |
21:08.55 | Hmmhesays | echo doesn't come from the voip side |
21:09.13 | asteriskmonkey | we are trying to get better quailty aswell as price over the pstn right? not come up with something hack |
21:09.15 | Hmmhesays | unless you've called an echo app |
21:09.20 | harryvv | its the electronics in the cards. |
21:09.21 | iCEBrkr | Hmmhesays: my X100P doesn't echo either :) |
21:09.25 | asteriskmonkey | crap msn chat has better sounding voice in some cases |
21:09.48 | justinu | don't use G711 if your network isn't perfect |
21:09.49 | harryvv | iCEBrkr my card used to echo..i haaardly hear my self on the other end but need to pay attention. |
21:10.13 | Hmmhesays | i could really use a UK number to send a call to, i vant to go home |
21:10.41 | harryvv | Hmmhesays where are you now |
21:10.45 | asteriskmonkey | my te110p used to slam echo like a dirty whore until i changed the echo can |
21:10.47 | Hmmhesays | Minnesota |
21:11.10 | harryvv | i see |
21:11.12 | Hmmhesays | I need to test this gateway to see if BT is accepting callerid |
21:11.19 | harryvv | what you doing in the frozen land of the north? |
21:11.24 | justinu | +44 208 237 3000 |
21:11.32 | justinu | (coca cola great britain) |
21:11.43 | Hmmhesays | accept i need to know what number comes in |
21:11.51 | harryvv | Hmmhesays what u doing there? |
21:11.51 | justinu | oh |
21:11.57 | Hmmhesays | in MN? i live here |
21:12.27 | Hmmhesays | lets see, i work, play in a band, drink and hit on chicks... most of the time hot depending on how much i've had to drink |
21:12.35 | justinu | lol |
21:12.45 | justinu | 6 pack and a light switch |
21:12.48 | justinu | universal solution |
21:12.53 | [TK]D-Fender | :O |
21:12.59 | Hmmhesays | LOL |
21:13.14 | harryvv | and your from england? |
21:13.19 | Hmmhesays | hell no |
21:13.26 | Hmmhesays | born in minneapolis |
21:13.31 | Hmmhesays | got a gateway in england |
21:13.39 | Hmmhesays | dass freaking 2 ugh |
21:13.53 | kink0 | Hmmhesays, BT ? |
21:14.01 | Hmmhesays | kink0 aye |
21:14.04 | kink0 | I am very interesting about bluetooch |
21:14.20 | kink0 | I did fews experiments ussing ATA and sound cards to connect GSM |
21:14.23 | justinu | dass-2, lol |
21:14.31 | justinu | gotta love BT |
21:14.38 | Hmmhesays | I think we might have our dass2 cid problem fixed, however I can't test it until i get a number in the uk to call |
21:14.38 | kink0 | but just today, start reading about chan_bluetooth |
21:14.59 | kink0 | ( someday I will learn how is written "blueto**" ) :) |
21:16.14 | fugitivo | bluetooth |
21:16.14 | fugitivo | repeat with me |
21:16.14 | kink0 | fugitivo, xD |
21:16.14 | fugitivo | bluetooth |
21:16.16 | Hmmhesays | bluetooch sounds like it could be part of the female anatomy |
21:16.16 | kink0 | Hmmhesays, have you success with bt ? just one or several channels ? |
21:16.18 | harryvv | gateway in england? |
21:16.18 | Hmmhesays | just one bastard channel |
21:16.18 | fugitivo | bluepuss |
21:16.18 | Hmmhesays | or span i should say |
21:16.27 | [TK]D-Fender | Hmmhesays : What kind of functionality do you get from it? |
21:16.27 | harryvv | Hmm did you ever see that movie that just came out called north country? |
21:16.35 | Hmmhesays | and i'll find out how successful as soon as a brit stumbles in here |
21:16.36 | justinu | it's like a PRI |
21:16.47 | Hmmhesays | a sick PRI |
21:16.53 | kink0 | like a PRI ? |
21:17.00 | Hmmhesays | 30 channels |
21:17.15 | kink0 | ahhh , about the number of channels. |
21:17.23 | asteriskmonkey | should be 32 |
21:17.24 | Hmmhesays | dass2 E1 |
21:17.29 | asteriskmonkey | 1 signalling right |
21:18.03 | kink0 | Hmmhesays, what hard did you use to setup your bt ? |
21:18.17 | Hmmhesays | the only dass2 gateway I could find a quintum dx2030 |
21:19.38 | kink0 | ok, but you runs asterisk in a linux box , right ? and this linux box has a bt->USB , right ? |
21:19.51 | Hmmhesays | um no |
21:20.09 | Hmmhesays | asterisk-->lan-->gateway-->bT |
21:20.34 | justinu | lol, some of you are talking about bluetooth (bt), and someone else is talking about british telecom (bt)? |
21:20.57 | Hmmhesays | oh HAHA |
21:20.59 | kink0 | justinu, I means bluetooth |
21:21.06 | Hmmhesays | i'm talking british telecom |
21:21.09 | justinu | lol |
21:21.12 | justinu | very confusing :) |
21:21.13 | Hmmhesays | britishtooch? |
21:21.14 | kink0 | lol !!! |
21:21.17 | saft | im talking brick tamland! |
21:21.20 | Hmmhesays | LOL |
21:21.25 | saft | I love lamp! |
21:21.41 | SpaceBass | im trying to trunk b/t two boxes... the client box registers but when i try and call into or out of the trunk it fails |
21:21.43 | SpaceBass | same lan |
21:22.02 | asteriskmonkey | spacebass: is it 1.2? |
21:22.07 | SpaceBass | asteriskmonkey yeah |
21:22.10 | asteriskmonkey | there is a bug reprot on that |
21:22.22 | asteriskmonkey | check the bugs.digium.com under iax i think |
21:22.34 | SpaceBass | sip trunking working? |
21:23.12 | kink0 | anyone has experiences with Valiant GSM gateways ? |
21:23.20 | Hmmhesays | ugh yes |
21:23.40 | kink0 | bad ? |
21:23.49 | harryvv | space...is this to test it before deloying the box at the remote site? |
21:23.59 | Hmmhesays | i want to jam a pen through the eye of the man that thought up a voip/gsm gateway |
21:24.22 | kink0 | Hmmhesays, why ? what happenes ? |
21:24.38 | SpaceBass | harryvv sorta... migrating to new box for my home setup but don't want to move zaptel hardware yet |
21:24.41 | Hmmhesays | so many failure points |
21:25.00 | kink0 | with Valiant or in general with any GSM gateway ? |
21:25.03 | SpaceBass | harryvv so I wanted to keep zaptel in the current box and move everything else to the new one and just trunk the zaptel calls over |
21:25.09 | Hmmhesays | GSM in general |
21:25.15 | Hmmhesays | especially in the middle east |
21:25.46 | kink0 | Hmmhesays, where you are from ? |
21:25.49 | saftsack | someone of you who is running asterisk not as rootß |
21:26.03 | Hmmhesays | geebus kink0 pay attention |
21:26.08 | Hmmhesays | i'm from Minnesota |
21:26.27 | kink0 | ahh, here Spain. |
21:26.45 | Hmmhesays | how about you give me a phone number and you tell me what callerid shows up |
21:26.54 | kink0 | I have read you need even to rotate IMEI on USA, or you are blocked. |
21:27.30 | kink0 | Hmmhesays, do you need my phone number ? I can see you caller id at my mobile, if you want try. |
21:27.35 | trixter | on gsm you need to rotate imei in the US? why would they block you if you dont? I didnt catch that part |
21:27.37 | Hmmhesays | sure you don't have to answer |
21:27.46 | kink0 | ( I will not pick up, so will cost nothing ) |
21:27.50 | kink0 | ok... ok. |
21:28.00 | Laibsch | Where do I get a ztdummy module for a 2.6 kernel? |
21:28.13 | Laibsch | Debian 2.6 kernel, stock version. |
21:28.28 | trixter | apt-get install asterisk |
21:28.31 | trixter | that includes ztdummy |
21:28.39 | dippo | can anyone recommend a good IAX trunking service for high-minute usage? |
21:28.43 | brimstone | anyone called international with nufone? |
21:28.58 | kink0 | I will not answer, but don't send over 5 ring |
21:28.59 | dippo | we are currently with teliax, but i am not sure how accommodating their price will be compared to a T1 or something for the amount of minutes we need (around 8000/mo) |
21:29.11 | trixter | nufone doesnt relaly like to let people call outside US/CA since they got taken for $450k in international calls |
21:29.23 | trixter | if you are a grandfathered customer they might let you |
21:29.35 | saftsack | someone of you who is able to show me howto give a user permissions to use zaptel? :) |
21:29.56 | trixter | or a customer that has a long standing relationship, they basically are trying to make it harder to call outside countrycode 1 (and even to 15 countries within 1) |
21:30.13 | Laibsch | trixter: Hm, OK but then I won't have amportal which I would like to have for the beginning. |
21:30.20 | Laibsch | I will try on a second machine. |
21:30.25 | brimstone | trixter: i had heard about that |
21:30.34 | brimstone | has anyone tried with iax.cc? |
21:31.36 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
21:31.38 | trixter | yes |
21:31.42 | trixter | with sixtel/iax.cc |
21:31.52 | brimstone | trixter: i can't seem to get this call to germany to go through |
21:32.00 | trixter | some of their routes dont work that they advertise though.. +448xx for example, afganistan doesnt work afaik either |
21:32.05 | trixter | but most do for the advertised rate |
21:32.23 | trixter | basically their upstream providers (all of em) wont let them call anything where there is a variable rate for the call |
21:32.36 | trixter | they can turn it on for select customers though |
21:32.45 | trixter | talk to chris sixtel9@aim iirc |
21:32.47 | justinu | i called north korea with level3 |
21:33.05 | trixter | brimstone what exactly are you trying to dial and with which provider? |
21:33.13 | trixter | msg me if you dont want the number open |
21:33.31 | trixter | or hit up sixtel9@aim that is their tech support IM acct :) |
21:33.50 | brimstone | trixter: just trying to call a girlfriend on holidday |
21:33.54 | xachen | sixtel |
21:33.56 | xachen | is evil |
21:34.07 | Hmmhesays | i had a girlfriend one |
21:34.09 | Hmmhesays | *once |
21:34.13 | xachen | I wouldn't recommend it *period* |
21:34.26 | justinu | Hmmhesays: were you drunk? |
21:34.30 | Hmmhesays | i'm a confirmed bachelor now |
21:34.37 | Hmmhesays | other than that engaged chick i'm seeing |
21:34.44 | trixter | what specifically is evil about them? |
21:34.47 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
21:34.49 | trixter | I havent had any problems with them |
21:34.54 | xachen | dude |
21:34.59 | xachen | They never answered their phones for months |
21:35.05 | justinu | everyone here complains about them |
21:35.06 | xachen | I filed the first BBB complaint in April |
21:35.08 | xachen | now there is 4 more |
21:35.09 | trixter | oh I have no problems getting hold of people there |
21:35.14 | xachen | they didn't anwer ANY of them |
21:35.23 | xachen | and their toll free numbers I had stopped working for me |
21:35.24 | xachen | with a busy tone |
21:35.29 | Hmmhesays | uk anyone? |
21:35.30 | xachen | I wans't even getting a connection to my server |
21:35.41 | xachen | I opened many "emergency" tickets |
21:35.46 | xachen | Finally just ran chargeback |
21:35.55 | trixter | I havent had to open any ticket I just ask for it to be done and it is |
21:36.10 | xachen | they must favour you then |
21:36.23 | xachen | I first found about them @ pastebin.ca :P |
21:36.27 | trixter | my friend has had the same experience |
21:36.35 | xachen | the owner of pastebin.ca took their ad down cause they didn't pay for their ads and had outstanding fees |
21:37.04 | trixter | I dunno about that, sounds a little sketchy that the owner would tell you that |
21:37.19 | netvulture | voicemail codec problem - when voicemail answers my teliax calls it has a write format of 2 (GSM) but when it starts recording is switches to write format 4 (ulaw) - The read and native are allways in 4 (Ulaw) - The write format should never be in 2 (gsm) - any ideas? My iax.conf has disallow=all, allow=ulaw in both the general and teliax contexts. |
21:37.19 | xachen | I know the owner :P |
21:37.23 | harryvv | might be a good site to advertise voip equipment for sale. |
21:37.43 | harryvv | how many of you here go though xo for termination? |
21:37.53 | trixter | ahh well my friend is friends with a higher up at sixtel :P |
21:37.54 | netvulture | The caller from teliax can hear the vm prompt just fine. |
21:38.12 | xachen | netvulture: Make sure you set your codecs int he account settings rea |
21:38.14 | trixter | why I have been able to get prefered rates from them as well as some other perks :) I get $0.0044/min from them with no problems in service |
21:38.17 | trixter | :D |
21:38.18 | harryvv | trixter, i have sixtel...not been the most reliabile service |
21:38.27 | kink0 | harryvv, what site ? I am seeking for PRI(E1) -> GSM gateway |
21:38.38 | netvulture | I have set teliax to use onlu ulaw |
21:38.55 | xachen | trixter: Send a message of word off. Tell them to answer their phones to actual customers who have problems :P |
21:38.57 | netvulture | both on my asterisk side and via the web portal at teliax |
21:39.00 | trixter | course the fact that I am pushing major traffic through them doesnt hurt either |
21:39.02 | harryvv | xo is a voip carrier that is a large backbone..or am I not correct on that? |
21:39.15 | SpaceBass | arrruuggg stupid trunking |
21:39.26 | trixter | xo is a clec and has a fiber backbone |
21:39.26 | xachen | netvulture: What is your voicemail.conf settings? |
21:39.43 | trixter | they do voip as well, and if you do enough traffic xo will give you $0.0035/min to most of the US |
21:39.55 | trixter | well population wise most anyway |
21:40.00 | xachen | format=wav49|gsm|wav |
21:40.01 | netvulture | format=wav49|wav |
21:40.05 | xachen | hmm |
21:40.23 | netvulture | i think wav49 is gsm isn't it |
21:40.29 | harryvv | trixster really...how much traffic are we talking about? |
21:41.01 | trixter | hmm.. someone is sending me 2 packages.. a 10 pound and a 5 pound and I have no idea who they are.. it may be the free ATAs and gxp2000s that I am getting as prizes for the sacramento asterisk users group |
21:41.03 | xachen | netvulture: Just try wav |
21:41.09 | trixter | hopefully cuase that means that I will have em really soon :) |
21:41.32 | netvulture | trying now - thanx |
21:41.57 | trixter | harryvv: I have one customer that does 75k minutes and another that is doing a 600kmin/mo test ... the test is going to be over in janurary when its expected they will bump up to 10M minutes a month when I will prolly leave sixtel for better rates |
21:42.13 | xachen | they never honoured their unused portion refund policy |
21:42.16 | xachen | so they have no trust with me |
21:42.20 | trixter | since they dont own the transmission lines they cant compare to someone who does own those lines |
21:42.52 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
21:43.27 | xachen | anyways bbiab |
21:43.44 | SpaceBass | anyone know if nufone allows setcallerid name? |
21:43.57 | Hmmhesays | uk uk uk uk |
21:44.15 | netvulture | no dice - while ringing all formats are 4 - vm answers write=2 - vm starts record - write=4 |
21:45.28 | jake1932 | using a TE100p - can anyone tell me why i would receive double digits intermittantly (00 instead of 0)? |
21:45.49 | netvulture | vm is definatly changing codec to 2 - i have a answer - wait(1) in place and codec stays at 4 until vm is called |
21:49.42 | SpaceBass | so, is IAX2 trunking broken in 1.2? mine is not obeying my incoming routing and my outgoing rings all circuits busy |
21:54.39 | trixter | woo thevoipconnection.com confirmed that they shipped all my goodies today. 5 ATAs and 3 gxp2000s ... woo hoo sacaug.org has free prizes!! and we have a tdm410 (1 fxs module) from digium |
21:54.52 | marcus2 | eww gxp2000s |
21:55.27 | trixter | free prizes and they are really the lowest end phone I would use |
21:55.55 | trixter | at least thevoipconnection.com supports the community |
21:55.56 | trixter | :) |
21:55.59 | justinu | they're ok for testing and such |
21:56.02 | justinu | and basic users |
21:56.18 | trixter | for $85 they arent bad |
21:56.23 | justinu | yeah |
21:57.13 | trixter | is the $35 savings for a bugetone 10x really worth it? |
21:57.29 | justinu | the budgetone is a real piece of crap |
21:57.31 | justinu | very flimsey |
21:57.45 | justinu | might be something you'd see on the desk of a 13 year old girl |
21:58.00 | justinu | my first telephone |
21:58.18 | trixter | first? at 13? |
21:58.21 | trixter | ha! |
21:58.33 | trixter | at 13 they are spending no less than 4 hours a day on the phone |
21:58.44 | justinu | hahah, ok, 7 year old then |
22:01.39 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
22:01.57 | Seldon19751 | I'm gonna get my self connected |
22:02.18 | Seldon19751 | .. eventually :} |
22:02.22 | *** join/#asterisk Kokey (n=Kokey@201.155.164.201) |
22:04.42 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
22:05.09 | marcus2 | the question is, is the gxp2000 worth the $30 savings over a polycom 301 ;) |
22:06.42 | justinu | does the 301 have a built in ethernet switch? |
22:07.04 | [hC] | Man this spa-941 would be absolutely -perfect- if it did PoE and had an ethernet switch in it |
22:07.09 | trixter | I think its a good phone for its price.. if you are looking at 15 (fairly small setup) that would be a $450 savings and on a smaller system like that I can see how money would be a concern |
22:07.12 | justinu | the 941 doesn't do PoE? |
22:07.17 | [hC] | Nope |
22:07.23 | justinu | wow, very lame |
22:07.25 | [hC] | Just tried |
22:07.30 | [hC] | its basically a sipura in a phone |
22:07.33 | LostFrog | Did it smoke? |
22:07.36 | justinu | yeah, i have the 841 |
22:07.47 | [hC] | nope, no smoke |
22:07.57 | [hC] | poe doesnt just 'send power' |
22:07.57 | [hC] | well |
22:07.58 | *** join/#asterisk xtrvd (n=test@S010600131035338a.cc.shawcable.net) |
22:07.58 | [hC] | good poe doesnt. |
22:07.59 | LostFrog | Damn.. I was looking for a little excitement. :) |
22:08.00 | justinu | 802.3af PoE is smart enough not to send power to devices that don't need it |
22:08.10 | [hC] | same with cisco CDP PoE |
22:08.23 | *** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net) |
22:08.33 | justinu | speaking of cisco CDP PoE, anyone know if polycom 501 will work with CDP PoE without a special cable? |
22:08.58 | [hC] | no idea. I use a netgear switch that works with everything, figures it out on its own |
22:09.19 | justinu | i have a dlink des-1526, won't power the polycom501 with a straight patch cable |
22:09.34 | justinu | nfw am I paying 40 bucks for a "PoE cable" |
22:09.43 | [hC] | make it yourself |
22:09.47 | [hC] | pinout instructions on voip-info |
22:09.55 | justinu | i didn't come across that in my search |
22:09.57 | darwin_35 | its punk magic |
22:10.01 | justinu | (i did for cisco 7960) |
22:10.22 | [hC] | search for poe cable maybe |
22:11.24 | trixter | just make sure its not a pos cable |
22:11.27 | trixter | those arent quite the same |
22:11.56 | justinu | any cable I make turns out to be a PoS cable ;) |
22:11.57 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
22:12.08 | trixter | heh |
22:15.00 | *** join/#asterisk loick (n=loick@APuteaux-151-1-65-183.w81-249.abo.wanadoo.fr) |
22:15.21 | darwin_35 | how do you set dialing so you dont have to hit # to make the call dial with out he wait time |
22:16.12 | *** join/#asterisk saftsack (n=saftsack@p54A7F9A9.dip.t-dialin.net) |
22:16.46 | saftsack | hi |
22:17.33 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
22:17.52 | trixter | darwin: that is often a dialplan setting in the sip device. some allow you to set patterns so it knows that you are done right away |
22:17.59 | trixter | others dont if you cant do that then you are stuck |
22:18.48 | mog_work | anyone in germany |
22:18.52 | mog_work | i need to terminate a call |
22:18.58 | mog_work | just one ^_^ |
22:20.01 | LostFrog | One 900 Hour call? |
22:20.30 | trixter | mog: I have a german number if that will help ya |
22:20.44 | trixter | I kinda wanna test that it works too :P |
22:21.36 | mog_work | no i need someone with a box in germany |
22:21.43 | mog_work | i dont want to terminate the long distance call |
22:22.39 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
22:24.18 | trixter | you need to place a call to a specific number in germany? or what? |
22:24.25 | trixter | I am not quite understanding what it is that you want |
22:25.33 | mog_work | i was trying to help a friend call his girlfriend who is in germany at the moment |
22:25.39 | trixter | ahh |
22:25.40 | mog_work | but jerjer hooked me up |
22:25.46 | trixter | voipbuster.com I believe gives free to germany |
22:26.12 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
22:27.30 | *** join/#asterisk maarken (n=panties@moist235.drizzle.com) |
22:28.34 | *** part/#asterisk chapeaurouge (n=chap@85.201.81.201) |
22:29.06 | maarken | has anyone ever gotten g729 working with openbsd? |
22:29.52 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
22:30.17 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
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22:33.26 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
22:34.12 | dippo | anyone know of a good cordless SIP phone? |
22:35.32 | wasim | dippo: yeah, any good cordless phone with a good ata |
22:35.36 | SkramX | cordless as in wireless as in 802.11X or what |
22:36.54 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-89.cybersurf.com) |
22:39.37 | *** join/#asterisk l1nux (i=moi@54.138.103-84.rev.gaoland.net) |
22:39.44 | l1nux | hi :) |
22:41.23 | l1nux | asterisk- Jingle not ready ? :P |
22:41.31 | jake1932 | is their a utility to monitor noise on a line connected to a TE100P? |
22:41.37 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
22:41.37 | *** mode/#asterisk [+o anthm] by ChanServ |
22:41.49 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
22:41.56 | l1nux | http://code.google.com/apis/talk/index.html |
22:42.01 | *** join/#asterisk RoyK (n=roy@host-n39-140.homerun.telia.com) |
22:42.01 | harryvv | dippy hitachi |
22:42.08 | RoyK | ~seen zoa |
22:42.19 | jbot | zoa is currently on #asterisk (1d 12h 53m 43s). Has said a total of 119 messages. Is idling for 7h 42m 10s |
22:42.21 | harryvv | hitachi wireless voip wifi phone |
22:45.15 | dippo | wasim: well, i was sorta hoping i wouldn't have to go that route |
22:45.55 | dippo | i was hoping to find a phone with a voip base with 10baseT that then 5.8GHz analog to a handset or something |
22:45.59 | dippo | maybe there's just not a market for that |
22:46.16 | Remosi | l1nux, there are people working on it |
22:46.36 | l1nux | good news :D |
22:46.42 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
22:46.49 | l1nux | thanks Remosi :) |
22:47.33 | Remosi | np |
22:47.42 | maarken | or just get a ATA and a regular 5.8Ghz. :) |
22:47.52 | dippo | yeah |
22:48.17 | dippo | i guess those aren't as pricey as I thought |
22:48.27 | dippo | $60 for a single FXS sound about right? |
22:48.41 | [TK]D-Fender | $70 for dual |
22:48.51 | l1nux | Remosi, where get recent news about asterisk-Jingle ? |
22:49.07 | Remosi | no idea |
22:49.13 | l1nux | ok :) |
22:49.34 | *** join/#asterisk deb_user (n=frank@71-36-59-120.albq.qwest.net) |
22:49.52 | deb_user | hello all |
22:49.59 | deb_user | anybody up for talking a little about echo? |
22:50.25 | deb_user | i could use some insight on its relationship to hardware, namely the hardware I'm using to connect to POTS |
22:50.49 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
22:50.57 | [TK]D-Fender | which? |
22:51.03 | |Vulture| | anyone here work with XO Flex package yet? |
22:51.05 | deb_user | right now I've got an X100P clone...and when I use it to connect to the PSTN I get an annoying echo on my side |
22:51.24 | [TK]D-Fender | What ver of * and what settings in zapata? |
22:51.28 | |Vulture| | deb_user: adust rx/tx |
22:51.45 | deb_user | vulture, I've tried that |
22:51.54 | deb_user | but I'm not quite sure what I'm doing |
22:52.00 | dogtanian | anyone know how easy it is to update the firmware on a CISCO 7960 so that it supports SIP? I don't think the ones I'm acquiring are going to have a license - so is this likely to be a problem for me? |
22:52.07 | |Vulture| | use ztmonitor |
22:52.16 | deb_user | vulture, how do I get ztmonitor working? |
22:52.46 | dippo | <PROTECTED> |
22:53.04 | |Vulture| | deb_user: ./ztmonitor (chan num) -v |
22:53.13 | |Vulture| | in the /usr/src/zaptel dir |
22:53.33 | |Vulture| | you want normal conversations in the mid range |
22:53.41 | |Vulture| | and you don't want any noise when the line is hung up |
22:53.52 | |Vulture| | that can be a sign of bad wiring or bad hardware |
22:54.18 | deb_user | i see noise, vulture |
22:54.28 | deb_user | and I don't even have asterisk running right now! |
22:54.30 | |Vulture| | dogtanian: easy if you have the SIP firmware, usually you have to start with one of the oldest SIP firmwares though then slowly upgrade |
22:54.36 | dogtanian | :/ |
22:54.41 | |Vulture| | deb_user: you shouldn't see noise with * running either |
22:54.45 | dogtanian | is it easy enough to get hold of the firmware? |
22:55.09 | |Vulture| | dogtanian: if you have a license agreement with cisco... otherwise no |
22:55.28 | dogtanian | ah |
22:55.29 | deb_user | vulture: yeah, the noise is easily up to an eighth of the entire bar |
22:55.34 | |Vulture| | and its copyrighted material so its not freeware |
22:55.41 | deb_user | static? |
22:55.53 | dogtanian | any idea how much licences are... off the top of ur head? |
22:55.58 | |Vulture| | deb_user: yea like bad wiring in the area your in, or possibly your TDM card |
22:56.08 | dogtanian | i was thinking of buying from ebay(uk) |
22:56.22 | deb_user | hmmm... |
22:56.47 | deb_user | so you think that's what's causing the echo? |
22:56.48 | |Vulture| | dogtanian: search the wiki I think they have some info on it, I switched to polycom as I prefer them and get them for the same price as used 7960s |
22:56.52 | *** part/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
22:57.16 | deb_user | i suppose I could try changing the cord from the TDM to the wall jack |
22:57.17 | |Vulture| | deb_user: pastebin your zapata.conf |
22:57.24 | deb_user | ok |
22:57.28 | |Vulture| | deb_user: what card do you have in there? |
22:57.29 | dogtanian | ooh. are the features as good as the 7960? perhaps i'll buy one of those instead :) |
22:57.44 | deb_user | its a clone X100P |
22:57.51 | harryvv | are there any routers that offer duplex wifi capability? |
22:57.51 | deb_user | i just got something cheap to learn on |
22:57.52 | |Vulture| | dogtanian: yes get a IP-501 or 601 if you REALLY need it but I don't think you will |
22:58.09 | dogtanian | cheers! |
22:58.24 | deb_user | vulture: I paid 10 bucks for it, but I suppose I've got to upgrade for production |
22:58.26 | |Vulture| | deb_user: that could be causing it... |
22:58.42 | |Vulture| | deb_user: yea I deff. would not recommend that for production |
22:58.49 | [TK]D-Fender | dippo : recommendations for... ? |
22:58.55 | |Vulture| | the newest zaptel drivers 1.2.1 are actually really nice on the TDMs |
22:59.09 | FuriousGeorge | i was just trying to install asterisk on colinux and there are some issues with netowkring only b/t host and guest OS, so its not really a question. i was thinking of using vmware, but i heard (i think) that vmware is (even) lesss suited for asterisk than colinux |
22:59.11 | deb_user | vulture: those are the drivers I'm using |
22:59.14 | FuriousGeorge | is that true? |
22:59.27 | deb_user | vulture: you recommend a TDM card? |
22:59.31 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
22:59.45 | |Vulture| | deb_user: where are you putting this box? |
23:00.07 | dippo | [TK]D-Fender: a good dual FXS adapter. but I think I found one .. the linksys Sipura SPA-2002 for $70 |
23:00.09 | deb_user | vulture: don't understand the question |
23:00.10 | dippo | which seems to be pretty decent |
23:00.24 | |Vulture| | deb_user: is this for a home or office? |
23:00.28 | deb_user | vulture: office |
23:00.39 | |Vulture| | how many phones and how many phone lines? |
23:00.41 | deb_user | vulture: the one I've got now is at home...but its just a starter to learn on |
23:00.59 | deb_user | vulture: that's a tough question to answer.... |
23:01.11 | |Vulture| | expansion is the real question here |
23:01.15 | deb_user | vulture: we've got a legacy pbx system that I need it to interface with |
23:01.28 | deb_user | vulture: so do all those phones count too? |
23:01.53 | [TK]D-Fender | dippo : yup thats the one to get |
23:01.53 | |Vulture| | deb_user: no but if you are interfacing I would recommend a T1 card with a Channel Bank |
23:02.14 | |Vulture| | deb_user: if you ever plan on having more than 4 incoming phone lines |
23:02.18 | deb_user | vulture: but the analog phone lines plug straight into the pbx |
23:02.28 | deb_user | the legacy pbx, that is |
23:02.39 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
23:02.53 | |Vulture| | deb_user: you would plug the PSTN into the * box then have FXS lines from the * box to the legacy PBX |
23:02.55 | deb_user | i was thinking to just run them first through an FXO module, and then run them back out of an FXS |
23:03.05 | |Vulture| | deb_user: trust me in the long run its so much nicer to go full VoIP |
23:03.23 | deb_user | vulture: I believe you |
23:03.42 | deb_user | I mean, seriously, I wouldn't have invested so much in a legacy pbx 2 years ago if I knew then what I know now |
23:03.54 | |Vulture| | we trashed all our Norstars |
23:03.58 | |Vulture| | well ebayed them |
23:04.05 | |Vulture| | and they sold for as much as the * system |
23:04.14 | |Vulture| | and it has crazy expansion and customization |
23:04.25 | deb_user | Well, that would be a tough sell in my office |
23:04.26 | |Vulture| | now we have 8 offices all interconnected |
23:04.31 | FuriousGeorge | i guess ill find out how vmware does |
23:04.36 | deb_user | because we sank at least $5,000 into the pbx only two years ago |
23:04.56 | deb_user | so I have to hold on to all of that stuff |
23:04.57 | |Vulture| | deb_user: owch... one of those half Analog half VoIP ones? |
23:05.06 | deb_user | vulture: not even |
23:05.13 | deb_user | vulture: its all digital, vodavi |
23:05.21 | |Vulture| | hmm not familliar with them |
23:05.33 | deb_user | vulture: but its all proprietary |
23:05.40 | |Vulture| | oh... I see |
23:05.45 | |Vulture| | so why the * interface? |
23:05.55 | |Vulture| | do a lot of internation/LD? |
23:06.01 | deb_user | interconnection of international offices |
23:06.07 | deb_user | free voice mail |
23:06.22 | deb_user | well, there's tons of reasons to go *, I don't have to tell you |
23:06.28 | deb_user | its the future |
23:06.47 | |Vulture| | thats going to be hell to interface... due to the fact that its not a common system and getting the phones to interface with the * voicemail notification system might be tricky |
23:07.04 | |Vulture| | you will loose some of the VM features of your PBX like a blinking light etc. |
23:07.11 | deb_user | I don't think seamless integration will be possible |
23:07.17 | |Vulture| | agreed |
23:07.27 | deb_user | at least, I'm not willing to invest the time and effort |
23:07.43 | |Vulture| | I wouldn't either |
23:08.05 | deb_user | but, what I'm thinking is if I run all incoming first through asterisk, and then straight to the PBX via FXS to get the operator |
23:08.12 | |Vulture| | full * with polycom would change your business if you had international branches |
23:08.22 | |Vulture| | yea thats possible |
23:08.28 | |Vulture| | but * would need a channel bank |
23:08.34 | deb_user | why? |
23:08.36 | trixter | change your business? they said that about the internet and it didnt change anything :P |
23:08.44 | deb_user | a TDM wouldn't do the trick? |
23:08.44 | |Vulture| | :O |
23:09.02 | |Vulture| | deb_user: yea it would... but only if you have less than 4 incoming lines |
23:09.04 | deb_user | FXO in...FXS out |
23:09.30 | deb_user | well, over four we could just hit another TDM, que no? |
23:09.39 | deb_user | then up to 8 lines |
23:09.45 | |Vulture| | you need 1 FXO and 1 FXS per inbound line |
23:09.52 | deb_user | right |
23:09.57 | |Vulture| | which means for 4 inbound lines you need 2 TDM cards |
23:10.11 | deb_user | really? |
23:10.13 | |Vulture| | and from everywhere I have seen they say not to use more than 2 TDM cards |
23:10.14 | mog_work | or tdm2400p... |
23:10.27 | |Vulture| | mog_work: oh is that the new breakout one? |
23:10.29 | deb_user | oh, sure |
23:10.30 | [TK]D-Fender | That makes no sense.... |
23:10.31 | deb_user | that makes sense |
23:10.34 | mog_work | yes |
23:10.39 | |Vulture| | [TK]D-Fender: why not? |
23:10.52 | deb_user | The analog line coming in gets plugged in |
23:10.55 | |Vulture| | he is trying to interface * with an existing PBX |
23:10.56 | [TK]D-Fender | Why on earth would you need 1 FXS per FXO for incoming lines? |
23:11.12 | deb_user | Fender: I think vulture is right |
23:11.13 | dogtanian | |Vulture|: this is probably a really dumb question, but am I likely to need a license for any other reason apart from to get hold of firmware? |
23:11.13 | [TK]D-Fender | OH... Now why on earth would you WANT to do that ? |
23:11.15 | |Vulture| | so * would just be a bridge |
23:11.21 | [TK]D-Fender | :) |
23:11.26 | |Vulture| | [TK]D-Fender: hahaha we just went over that |
23:11.34 | deb_user | Fender: haha...i've gotta do the best with what I've got! |
23:11.53 | |Vulture| | dogtanian: not for polycom |
23:11.59 | deb_user | vulture: so tell me about channel banks |
23:12.01 | |Vulture| | dogtanian: for cisco yes |
23:12.08 | deb_user | thus far I've only looked at TDMs |
23:12.43 | [TK]D-Fender | Channel bank comes out pricy... |
23:12.46 | |Vulture| | deb_user: TSU-600 is what I use... you plug it into a T1 card (they run ~$500) the TSU is about $300 with a bunch of FXO/FXS modules off ebay |
23:12.53 | jake1932 | mog_was I talking to you about the T100P issue a few minutes ago? |
23:13.07 | dogtanian | |Vulture|: bah :/ any clues as to what I else I might need the license for? |
23:13.18 | deb_user | vulture: pricey |
23:13.21 | |Vulture| | but allows you to have 12 incoming lines for your system |
23:13.38 | |Vulture| | dogtanian: not for * unless you plan on using g729 codec |
23:13.44 | deb_user | but i don't understand the whole T1 concept |
23:13.56 | deb_user | what gets plugged into the T1 card? |
23:13.57 | jake1932 | i turned on the Monitor cmd and I can clearly hear the guy say "aw f**k", but touchtones are distorted |
23:14.01 | |Vulture| | deb_user: the channel bank mimics a T1 |
23:14.07 | |Vulture| | deb_user: the channelbank |
23:14.11 | deb_user | ohhhh |
23:14.22 | deb_user | so I plug all my analog lines into the bank? |
23:14.31 | deb_user | and then have a single output which is a t1? |
23:14.33 | |Vulture| | deb_user: yea you get modules for the channel bank |
23:14.45 | |Vulture| | deb_user: correct. the channel bank is like a huge TDM card |
23:14.46 | deb_user | and that gets plugged into the t1 card? |
23:14.50 | |Vulture| | that plugs into a T1 card |
23:14.59 | deb_user | bet the sound quality is fantastic |
23:15.09 | jake1932 | does asterisk try to mute the touch tones on the monitor file? |
23:15.31 | |Vulture| | deb_user: about the same a TDM... PRI is the only place sound quality gets really nice |
23:15.32 | deb_user | so I have all my incoming analogs plugged into the FXO mods |
23:15.53 | deb_user | and then the outgoing lines go to the legacy PBX via FXS |
23:15.53 | dogtanian | |Vulture|: thanks for your help... you've just made my life a lot easier :) |
23:15.57 | |Vulture| | deb_user: all inbound to FXO modules on the channel bank... and FXS into those on the channel bank |
23:16.00 | |Vulture| | dogtanian: np |
23:16.23 | |Vulture| | deb_user: you got it... but if your staying under 4 inbound lines then go TDM |
23:16.24 | deb_user | and then the entire channel bank plugs into the t1 card |
23:16.34 | deb_user | well, its a lot cheaper |
23:16.47 | |Vulture| | deb_user: yes |
23:17.10 | |Vulture| | deb_user: but it needs to be a L2 Adtran TSU-600 |
23:17.14 | |Vulture| | deb_user: Ill link you |
23:17.57 | deb_user | ok |
23:18.05 | |Vulture| | http://cgi.ebay.com/AdTran-TSU-600-1200-076L2-24-FXS-ports-tested-warranty_W0QQitemZ5844438329QQcategoryZ51271QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
23:18.07 | |Vulture| | something like that |
23:18.12 | |Vulture| | but thats 24 FXS ports |
23:18.24 | |Vulture| | if you call that company they will give you a price quote I order from them a lot |
23:18.55 | |Vulture| | or wait for a good ebay auction cause you will pay a little more... but it is hard to find FXO banks they are in high demand |
23:19.17 | harryvv | vulture like what modeles |
23:19.34 | |Vulture| | harryvv: the FXO modules for the TSU-600 |
23:19.36 | deb_user | yeah...I see it just comes with FXS mods |
23:19.38 | |Vulture| | the used ones |
23:19.45 | deb_user | no mention of FXO |
23:19.54 | |Vulture| | deb_user: if you call them they will find some for you |
23:20.26 | deb_user | vulture: after this conversation I'm starting to think a TDM for a single line |
23:20.33 | |Vulture| | deb_user: I just got 2x L2 TSU-600 units with 2xFXO and 1xFXS modules in them ~$350 |
23:20.48 | deb_user | let the company see the value of it for a little while, start small |
23:20.53 | |Vulture| | deb_user: yea its just gunna get more complex especially for someone not already using * |
23:21.07 | netvulture | Its nice to see another Vulture in town |
23:21.15 | |Vulture| | lol sup netvulture |
23:21.31 | deb_user | and then, if its really paying off on that one line |
23:21.41 | deb_user | we'll add a channel bank |
23:21.51 | deb_user | and go pure voip |
23:22.01 | |Vulture| | deb_user: you can fix those issues with the TDM |
23:22.05 | |Vulture| | 1s |
23:22.09 | deb_user | vulture: you have a voip service provider for outbounds |
23:22.11 | deb_user | ? |
23:23.17 | deb_user | what issues? the echo I've got on my x100p clone? |
23:23.21 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
23:24.56 | |Vulture| | sorry I have been working on an issue |
23:25.05 | |Vulture| | deb_user: yea 1second |
23:25.43 | |Vulture| | deb_user: in zapata.conf try adjusting rxgain and txgain |
23:26.39 | deb_user | i've tried that |
23:26.49 | |Vulture| | really? hmmm |
23:26.55 | |Vulture| | like -1? |
23:26.58 | deb_user | but didn't really know what I was doing when I did it |
23:27.20 | |Vulture| | when you do ztmonitor is your RX or TX really high? |
23:27.24 | |Vulture| | for RX? |
23:27.27 | |Vulture| | your RX |
23:27.45 | deb_user | one sec |
23:27.56 | deb_user | oh yeah |
23:28.01 | deb_user | RX is almost halfway! |
23:28.19 | |Vulture| | if it were TX I would say deff the card |
23:28.42 | deb_user | TX is quiet |
23:28.42 | deb_user | nothing on TX |
23:28.47 | deb_user | but RX is really active |
23:29.11 | |Vulture| | adjust rxgain=-1 |
23:30.02 | deb_user | ok |
23:30.08 | deb_user | leave txgain at what? |
23:30.13 | deb_user | default? |
23:30.49 | *** part/#asterisk l1nux (i=moi@54.138.103-84.rev.gaoland.net) |
23:31.53 | |Vulture| | yea |
23:31.56 | |Vulture| | don't change |
23:32.57 | deb_user | one sec |
23:33.30 | |Vulture| | deb_user: then make sure you reload zaptel |
23:33.36 | |Vulture| | and execute a ztcfg -vv |
23:34.03 | *** join/#asterisk |cleric| (n=dacleric@p5482B901.dip0.t-ipconnect.de) |
23:34.40 | deb_user | modprobe zaptel? |
23:35.41 | |Vulture| | rmmod wcfxs;rmmod zaptel |
23:35.44 | |Vulture| | then modprobe |
23:35.52 | |Vulture| | lsmod to make sure zaptel is unloaded |
23:36.00 | deb_user | aright |
23:37.02 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
23:37.48 | CunningPike | Does anyone know what lots of these errors means in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal |
23:38.58 | *** join/#asterisk felisk (i=Remek@pro75-3-82-234-175-208.fbx.proxad.net) |
23:39.06 | felisk | Hello everyone ! |
23:39.34 | felisk | I bought an hardware SIP phone ! BeWAN VoIP Phone S2 |
23:39.38 | felisk | I have strange question |
23:39.55 | felisk | when I call my phone from cellular, everything is ok |
23:40.09 | felisk | but when I call from SIP phone the cellular |
23:40.17 | felisk | I have quite big latency |
23:40.26 | felisk | any ideas where to search ? |
23:40.58 | jake1932 | if I'm listening on recordings from Monitor and I hear noise on the out file (directly from asterisk), is this IRQ issues? |
23:41.06 | |Vulture| | bbl |
23:42.08 | deb_user | no change vulture |
23:42.22 | |Vulture| | deb_user: seems like a hardware issue/conflict |
23:42.25 | |Vulture| | I got to head out |
23:42.25 | |Vulture| | ttyl |
23:42.37 | felisk | why can I have latency when calling from inside and not when called ?????? |
23:42.40 | deb_user | take care |
23:42.42 | deb_user | thanks for your help |
23:42.44 | |Vulture| | np |
23:45.26 | marcus2 | Dec 15 16:20:11 ryle zaptel Disabled echo canceller because of tone (tx) on channel 55 |
23:45.31 | marcus2 | what does that mean? |
23:45.55 | *** part/#asterisk deb_user (n=frank@71-36-59-120.albq.qwest.net) |
23:53.38 | *** join/#asterisk jcwunder (n=chris@ppp-82-135-2-181.mnet-online.de) |
23:55.43 | harryvv | SayDigit works for 0-9 what about *#? |
23:57.06 | *** join/#asterisk Primer (n=vi@sh.nu) |
23:57.43 | Primer | Anyone here upgrade a cisco ata 186 recently? The instructions, which read something like "press the ATA button, then dial _this_", and pressing said ATA button doesn't do anything |
23:57.57 | Primer | I have the debug server and upgrade server running |
23:58.25 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
23:58.26 | Primer | the debug server shows things, so it's definitely communicating |
23:58.58 | Katty | hi. |
23:59.25 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |