irclog2html for #asterisk on 20051220

00:00.34Dr-Linuxwhat 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?
00:01.20twisted[asteria]well, if it's a fir wall, be sure to use the right drill bit, otherwise your firwall might split
00:01.33RyanWDr-Linux: tcpdump for it. SIP uses udp 5060 for starters if thats any help.
00:01.36Beirdohehehe
00:02.27riddlebox{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)
00:02.53Dr-LinuxRyanW: 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)
00:05.09Seldon1975hi guys
00:06.05Seldon1975please help! what does "pbx_extension_helper.c: cannot find extension context 'default'" mean
00:06.19Seldon1975the digium guy has screwed my config
00:06.25*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
00:06.32Dr-Linuxtwisted[asteria]: on the server i'm using >> -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
00:07.23Dr-Linuxits 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
00:07.38Dr-Linuxhow can i do, that what port is being blocked? :S
00:08.54RyanW<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)
00:12.38riddleboxSeldon1975:is your voicemail stuff looking for the context default?
00:13.20Seldon1975riddle: yes
00:14.13riddleboxSeldon1975:do you have a default context for extensions.conf?
00:14.20Seldon1975yes, at the top
00:15.02riddleboxcan you go to pastebin and paste your extensions.conf and voicemail.conf?
00:15.18SpaceBassI'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
00:15.20SpaceBasshttp://pastebin.ca/34282
00:15.35Seldon1975riddle: 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)
00:15.49YaroManHello
00:15.54Seldon1975riddle: does it really have to do with voicemail.conf?  Im ge4tting this issue when trying to dial out
00:15.55SpaceBassanyone ever heard of zap calls "jumping" context like that?
00:16.18YaroManDoes any one use here BroadVoice with Asterisk @ Home?
00:16.21riddleboxSeldon1975:just looking at all possiblities
00:16.41Seldon1975riddle: ok thx; can I get back to you I have the digium guy on the phone
00:16.49riddleboxyeah
00:16.53SpaceBassYaroMan yes - three accounts
00:17.23*** join/#asterisk ManxPower (n=ewieling@29.sub-70-219-77.myvzw.com)
00:17.32riddleboxSeldon1975:he is probably smarter than me lol
00:17.43SpaceBassYaroMan keep it here... I may get called away and others may have better advice/help then I
00:17.51SpaceBassbut yeah, glad to help YaroMan
00:18.23YaroManOk thanks!!! What I'm trying to do it is to learn how it is works before go in to production
00:18.44YaroManI download Asterisk @ Home 2.2 Iso and installed on my Old PC
00:18.50SpaceBassYaroMan gotcha
00:18.52YaroManP3 1 Ghz with 1 GB Ram
00:19.17YaroManI was following this instructions http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm
00:19.34twisted[asteria]yuck
00:20.08SpaceBassok...
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.
00:22.22SpaceBassYaroMan still there?
00:22.30YaroManyeah
00:22.40SpaceBassdid you have a question?
00:22.44YaroManhold on on a phone with a client
00:22.49SpaceBassoh
00:26.13YaroMani'm back
00:26.17SpaceBassk
00:26.23SpaceBassanyone have a sec to take a look at my zapata-auto ? http://pastebin.ca/34282
00:26.27SpaceBassonce someone calls zap/2 all calls take that context, even calls for zap/1
00:26.38SpaceBassYaroMan 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)
00:26.55YaroManso i setup box allredy
00:27.04SpaceBassok
00:28.35YaroMani install but i have a truble with a trunk
00:28.54SpaceBassso you got a broadvoice BYOD account?
00:30.31SpaceBassopen up an SSH session into that box so you can see the CLI
00:30.41SpaceBasstype sip show registery and see what it says
00:32.40YaroManok hold on
00:32.51YaroMan[root@asterisk1 ~]# sip
00:32.51YaroMan-bash: sip: command not found
00:33.08SpaceBassYaroMan you have to be in the asterisk CLI
00:33.10SpaceBasstype asterisk -r
00:33.12YaroMani got BYOD bussines unlimited
00:33.35YaroManConnected to Asterisk 1.2.1 currently running on asterisk1 (pid = 2631)
00:33.42YaroManasterisk1*CLI>
00:33.53YaroManI'm in ;)
00:33.53SpaceBassjust the output of the command: sip show registery
00:34.09Dr-Linuxmy Cisco phone doesn't work behind a firewall, but the softphone are working, what things need to be checked?
00:34.34SpaceBassDr-Linux using the same SIP account or 2 different ones?
00:34.57YaroManasterisk1*CLI> sip show registery
00:34.57YaroManNo such command 'sip show registery' (type 'help' for help)
00:35.04Dr-LinuxSpaceBass: same
00:35.16Alricits sip show registry, right?
00:35.17SpaceBassYaroMan sip show registry
00:35.21SpaceBasssorry I cannot spell tonight
00:35.30YaroMank
00:35.36SpaceBassAlric only if yous spellz good
00:35.40Dr-LinuxSpaceBass: talking to me?
00:35.59YaroManasterisk1*CLI> sip show registry
00:35.59YaroManHost                            Username       Refresh State
00:35.59YaroMansip.broadvoice.com:5060         8458671927@s       120 Request Sent
00:35.59YaroManasterisk1*CLI>
00:36.01SpaceBassDr-Linux not just then
00:36.08SpaceBassYaroMan behind a firewall?
00:36.12YaroManyes
00:36.15YaroManRouter
00:36.21Dr-LinuxSpaceBass: so what should i do?
00:36.30Dr-LinuxSpaceBass: should i change the SIP account?
00:36.40SpaceBassYaroMan did you forward ports 5060 UDP and 10,000-20,000 TCP  to your asterisk box?
00:36.50YaroMannope
00:36.55SpaceBassDr-Linux I'm thinking... not sure why 1 phone would work and 1 would not
00:37.00SpaceBassYaroMan thats your problem
00:37.00YaroMani'll do it now hold on please
00:37.47Dr-LinuxSpaceBass: one is softphone and otherone is Cisco hard phone :S
00:38.24SpaceBassDr-Linux so the phones are behind a firewall dialing out to an asterisk box ... is the asterisk box also behind a firewall?
00:38.38SpaceBassDr-Linux Try forwarding those same SIP ports to the IP of the cisco phone
00:38.46SpaceBassDr-Linux suspect that may be the problem
00:39.14*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
00:39.22Dr-LinuxSpaceBass: i have disabled the firewall
00:40.18Dr-LinuxSpaceBass: my ip phones work good on public IP
00:41.59SpaceBassDr-Linux so the cisco is getting a public IP?
00:42.51Dr-LinuxSpaceBass: 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)
00:44.06YaroMan_I'm back just restarted my router
00:44.17SpaceBassok
00:44.33SpaceBassDr-Linux exactly... thats the issue
00:44.50SpaceBassthe firewall is preventing communication to the SIP (asterisk) server
00:45.13SpaceBassDr-Linux if you have access to the Asterisk server you can try adding nat=yes to the configureation
00:45.22SpaceBassor trying forwarding the SIP ports to the phone
00:45.26YaroMan_ok should I restart now astrisk box?
00:45.29SpaceBassYaroMan any luck registering?
00:45.33SpaceBassYaroMan no need
00:45.39SpaceBassYaroMan in the asterisk CLI type: reload
00:45.47Dr-LinuxSpaceBass: i did nat=yes as well
00:46.01Dr-LinuxSpaceBass: on the PBX server firewall is disabled
00:46.03SpaceBassanyone using a Cisco phone with more than one Asterisk box  ?
00:46.11Dr-LinuxSpaceBass: do it still need to forward the ports?
00:46.19YaroMan_Done!
00:46.22SpaceBassDr-Linux when you say "disabled" what do you mean? is it still behind NAT?
00:46.53Dr-LinuxSpaceBass: no, PBX box is not behind the firewall
00:46.59SpaceBassDr-Linux if the phone and the asterisk box are getting private IPs (IE not public) then you have a nat firewall
00:47.08SpaceBassDr-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)
00:47.13SpaceBassYaroMan get it working?
00:47.15Dr-LinuxSpaceBass: yes
00:47.22YaroMan_i dont know how can I check it?
00:47.27Dr-LinuxSpaceBass: asterisk box has public IP
00:47.34YaroMan_let me try to login with my phone
00:47.34SpaceBassDr-Linux but the cisco phone has a private IP, correct?
00:47.39Dr-Linuxand the Phone has pvt IP address
00:47.42Dr-LinuxSpaceBass: yes
00:47.48nswintHey you guys.. I blocked my first telemarketer with the blacklist and zapteller feature!!!!
00:47.52*** join/#asterisk GXTi (i=realme@freenode/developer/GXTi)
00:48.05SpaceBassDr-Linux try forwarding the SIP ports (udp 5060 tcp 10000-20000) to the phone
00:48.17Dr-Linuxhhm..
00:48.20SpaceBass(you can narrow the 10,000 - 20,000 in /etc/asterisk/rtp.conf)
00:48.35Dr-LinuxSpaceBass: yeah these ports are there
00:48.48YaroMan_nope does not work ;(
00:48.50SpaceBassSIP is notorious for not traversing firewalls well
00:49.06SpaceBassDr-Linux they are there? are they forwarded to the phone or to your computer running the softphone?
00:49.42Dr-LinuxSpaceBass: how can i verify, they are just in rtp.conf , but iptables firewall is stopped
00:49.51SpaceBassYaroMan what does the sip show registry
00:50.06YaroMan_hold on
00:50.19YaroMan_asterisk1*CLI> sip show registry
00:50.19YaroMan_Host                            Username       Refresh State
00:50.19YaroMan_sip.broadvoice.com:5060         8458671927@s       120 Request Sent
00:50.19YaroMan_asterisk1*CLI>
00:51.10SpaceBassDr-Linux keep it in here... I may have to run soon
00:51.48YaroMan_also wich password should I use the one I register with or the one under devices on broadvoice?
00:52.30SpaceBassthe one from devices from BV
00:53.11*** part/#asterisk nswint (n=nswint@c-24-98-129-84.hsd1.ga.comcast.net)
00:53.24riddleboxYaroMan_:it is a different password that you have to ask BroadVoice for
00:53.25YaroMan_asterisk1*CLI> sip show registry
00:53.25YaroMan_Host                            Username       Refresh State
00:53.25YaroMan_sip.broadvoice.com:5060         8458671927@s       120 Request Sent
00:53.25YaroMan_asterisk1*CLI>
00:53.28SpaceBassDr-Linux keep it in here... I may have to run soon
00:53.52YaroMan_look like they support is not there ;(
00:54.05YaroMan_I send them an email this mornign no replay
00:54.13YaroMan_was trying to call same story
00:54.16YaroMan_on hold for ever
00:54.24ManxPower~docs
00:54.25jbotit 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
00:54.28ManxPower~mailinglist
00:54.29jbotmethinks 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
00:54.31SpaceBassbrb
00:54.35riddleboxYaroMan_:I have tried calling them too, but after 8 minutes it just hangs up on you
00:54.47riddleboxthey usually respond to my emails within a day
00:54.50YaroMan_no i was on a hold for good hr
00:54.56riddleboxouch
00:54.58YaroMan_I have at home Vonage ;)
00:55.14riddleboxI am using broadvoice
00:55.28YaroMan_i just signup for broad voice last night
00:55.33YaroMan_so i can try them
00:55.50SpaceBassYaroMan where were we?
00:55.50riddleboxI like them but they definetly need better customer service
00:56.02YaroMan_SpassBoss
00:56.03SpaceBassyou forwarded the ports right? and made sure 5060 was UDP ?
00:56.15YaroMan_i ded reload ans
00:56.15YaroMan_asterisk1*CLI> sip show registry
00:56.15YaroMan_Host                            Username       Refresh State
00:56.15YaroMan_sip.broadvoice.com:5060         8458671927@s       120 Request Sent
00:56.15YaroMan_asterisk1*CLI>
00:56.27YaroMan_yes i did it too
00:56.46Druken~pastebin
00:56.47jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/
00:57.00SpaceBassI 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)
00:57.02SpaceBassbut thats just me
00:57.20SpaceBassYaroMan_ for some reason its not getting the registeration back
00:57.37SpaceBasswhat does your register string look like? (remember to use * for the password)
00:58.01YaroMan_where can I find this?
00:58.06YaroMan_I was using AMP
00:58.21SpaceBassIts the last field in APM
00:58.23SpaceBassat the very bottom
00:58.51YaroMan_ok hold on
00:59.07stormfrhello, is there somebody here with experience with chan_bluetooth ?
00:59.17YaroMan_8458671927@sip.broadvoice.com:xxxxxxxxx :8458671927@sip.broadvoice.com
00:59.47SpaceBassmake sure you don't really have a space after the password :845....
01:00.10YaroMan_i dont have any space ;)
01:00.28SpaceBassok... b/c I saw that exact problem a week ago
01:01.45Dr-Linuxawww
01:01.50Dr-Linuxmy phone is working now ;)
01:02.39YaroMan_Dr-Linux what service do u use?
01:03.57chris-fngentlemen, can anyone help me with an Asterisk::Manager (Manager.pm) problem?
01:06.30YaroMan_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)
01:09.59riddleboxYaroMan_:did you sign up for the BYOD service with broadvoice?
01:10.17YaroMan_yes I did
01:10.26YaroMan_BYOD Unlimited Bussines
01:10.32riddleboxdid they send you a registration email?
01:10.35YaroMan_i got 2 phone numbers from thme
01:10.40YaroMan_let me check
01:11.02RyanWI 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?
01:11.06YaroMan_BroadVoice Services Activation Confirmation
01:11.53riddleboxYaroMan_:so you have the sip registration info they gave you right?
01:12.16YaroMan_no I find on a portal
01:13.01riddleboxYaroMan_:I was under the impression that they sent you the password and stuff that you need to connect asterisk to their network
01:14.16YaroMan_if you login to account there and click on Account then My Devices
01:14.24YaroMan_I find that inf there
01:16.15riddleboxYaroMan_: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
01:16.56YaroMan_can you help me to configurate my asterisk please?
01:19.20YaroMan_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)
01:27.34sbingnerwtf is UPS server?
01:28.12denonhe said VPS
01:31.44*** join/#asterisk infinity1 (n=brendon@solara.netcal.com)
01:32.01infinity1cna you do elseif with ael? i've tried it with no luck
01:33.51Qwellinfinity1: try else if?
01:36.04*** part/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
01:36.29infinity1Qwell: 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)
01:40.25lilo(whoops ;)
01:40.27*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:40.43Qwelllilo: What did you do this time? :p
01:40.48mog_homeQWELL
01:41.04QwellMOG!@
01:41.04*** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
01:41.27mog_homehow goes it
01:41.40Qwellgood, you?
01:41.41YaroMan_Is it posible to run Asterisk on VPS?
01:41.44mog_homefantastic
01:41.47mog_homevps?
01:41.58Qwellsounds like an IBM server or something
01:42.15mog_homevery pretty server?
01:42.24mog_homeit runs on my cobalt which is quite sexy
01:42.26Qwellmaybe
01:42.57*** join/#asterisk colle (n=colle@c-232be155.27-1-64736c10.cust.bredbandsbolaget.se)
01:42.59mog_homeits a tad slow
01:43.03YaroMan_no on Virtual Private Server?
01:43.05mog_homebut sense i added ram its a champ
01:43.10mog_homewhich is?
01:43.24YaroMan_I have a server wich I use for VPS hosting
01:43.28YaroMan_it is Dual Xeon
01:43.30Qwellahh, A VPS
01:43.31YaroMan_with 4 GB ram
01:43.33YaroMan_and SCSI
01:43.34Qwelllame
01:43.38QwellYou do NOT want to do that.
01:43.42mog_homeit will problably work just fine
01:43.51mog_homebut qwell is probably right
01:43.52Qwellmog_home: it's like segmented servers
01:43.56YaroMan_it has only 3 accounts there so far
01:43.56Qwelllame, lame, lame
01:44.01mog_homelike uml?
01:44.08mog_homeor jail?
01:44.08YaroMan_no i use H-SPhere VPS servers
01:44.15stormfrhello, is there somebody here with experience with chan_bluetooth ?
01:44.21YaroMan_they install CentOS
01:44.53mog_homeprobably not stormfr whats question thouhg
01:46.08stormfrmog_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)
01:52.30Weezeyrealtime's telling me to check the debug, but /var/log/asterisk/debug isn't there
01:52.42Weezeywhat do I need to enable to generate debug stuffs?
01:53.47stormfrgo to logging.conf and uncomment ;full to full
01:53.54Weezeythanks
01:53.55stormfrand type set debug 255 in CLI
01:54.04stormfr(as well set verbose 255)
01:55.32*** join/#asterisk ronaldl79 (n=ILuv2Tra@c-24-8-54-203.hsd1.co.comcast.net)
01:55.52ronaldl79Hello.
01:55.55Weezeyhi
01:56.04ronaldl79Hi, Weezey. How are you?
01:56.20ronaldl79(Your name made me think of the 'Jeffersons' TV show)
01:57.03Weezeyshe spelled it Weezie
01:57.12ronaldl79Anyone been following this Gizmo project? It's based on Jabber and Sip ... I think I'll sign up and connect it to *
01:57.52mog_homeno its not based on jabber i thought
01:58.02mog_homebut the jingle stuff is intresting as all hell
01:58.37ronaldl79I just read that it was, who knows.
01:58.52ronaldl79Google's Jingle, ahh, haven't digged into that yet, but I've read about it.
01:59.21ronaldl79I'm glad Google chose to build their IM network on an open protocol -- it's the future of IM, imho.
01:59.33mog_homeindeed
02:00.00SkramXHiya
02:00.02ronaldl79I was just thinking of someone implementing jingle into *.
02:00.06ronaldl79Hi, Skram.
02:00.28ronaldl79It would be nice to have the ability to speak with Google Talk users and clients built on Jingle.
02:00.35mog_homeyup
02:00.38ronaldl79Sweet, Mog.
02:00.41mog_homevery
02:00.42*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
02:00.53mog_homeit needs some tlc though
02:00.56Nuggetgoogle talk needs to hurry up and enable s2s
02:00.59FuriousGeorgehey all
02:01.13mog_homethey arent nugget
02:01.20mog_homeor at least i dont think they will
02:01.24mog_homebut they should
02:01.25Nuggetthey say they are.
02:01.30Nuggetbut they're vague about it
02:01.32mog_homeyeah thats what i hear
02:01.32ronaldl79Hey Furious
02:01.50*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
02:01.56SkramXanyone heard of calpop.com ? Are they reliable (if you know of them(
02:02.00ronaldl79Jabber is a neat protocol.
02:02.06SkramXMy company is looking for a good place for a seconday server
02:02.07mog_homeyup
02:02.34ronaldl79How's your jabber project coming along, mog?
02:02.40mog_homequite well actually
02:02.46ronaldl79Great.
02:02.48mog_homei hit amazing stuff just recently
02:03.01mog_homewish i had more time to work on it
02:03.07mog_homethankfully school is over for now
02:03.14ronaldl79If I had a lot of money, I'd contribute to a lot of open source projects.
02:03.34ronaldl79mog, have you checked out Gabcast?
02:03.39mog_homelol if i had more time id contribute more...
02:03.41mog_homeyeah
02:03.44mog_homevery creative
02:03.45ronaldl79What do you think?
02:03.55mog_homewill be much cooler when we hit wideband
02:04.01ronaldl79I think so too -- I only got excited about it because of * being used on the backend.
02:04.13ronaldl79Definitely -- I noticed the wideband branch.
02:04.18FuriousGeorgei 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
02:04.19mog_home8k mono doesnt sound amazing though
02:04.30mog_homebut wideband might be goodenough
02:04.33ronaldl79Wideband supports stereo, right?
02:04.37mog_homeno
02:04.37FuriousGeorgebut all the lights are on, on the back of the box
02:04.42mog_home16khz audio samples
02:04.45mog_homeas apposed to 8
02:04.58ronaldl79Yeah, I read that much about it...:P
02:05.06ronaldl79I 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)
02:05.15mog_home?
02:05.44ronaldl79Well, there seems to be a lot of companies using their technology for wideband voice.
02:05.44FuriousGeorgeum, is the first bay on the second wctdm not bay 5?  if not, what is it?
02:06.15mog_homeindeed
02:06.20mog_homei dont know
02:06.31mog_homei doubt it though
02:06.41ronaldl79I tell ya what, Asterisk is like a drug.
02:06.44ronaldl79I'm addicted.
02:06.47mog_homelol
02:06.50mog_homevery true
02:07.11ronaldl79It's exciting to use such an open and flexible platform.
02:07.30ronaldl79After almost two years, I'm still amazed...
02:09.11mog_homewow thats a long time
02:11.05ronaldl79hehehe
02:11.19ronaldl79Actually, maybe more like 1.5 years.
02:11.35ronaldl79I'm still learning....got really serious about it the last year
02:12.05mog_homeyup
02:12.49ronaldl79But 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.
02:13.07ronaldl79It's insane all you can do with * ... it's like a big toy to me that never goes out of style.
02:13.21YaroMan_wow spent almost all day but cant make my asterisk to vork with BroadVoice
02:13.35mog_homei work all day long on pbx stuff but my pbx is so lame
02:13.36ronaldl79YaroMan...I use BV on my * box.
02:13.57ronaldl79Are you an * consultant, mog?
02:14.07mog_homein support/dev
02:14.15ronaldl79Oh, cool.
02:14.15mog_homebut my pbx does very little at home
02:14.33mog_homeonly cool code is the jabber stuff im playing with
02:15.01ronaldl79I'm going to visit Digium next year ...
02:15.08mog_homeheh
02:15.12mog_homewe need to have tours
02:15.17mog_homeits a pretty cool place
02:15.29ronaldl79It would be a closer trip if I still lived in Nashville...
02:15.38ronaldl79I'm in Denver now ....
02:15.40mog_homewhere you live ronaldl79
02:15.42mog_homeahh
02:16.02ronaldl79Yeah, I lived in Nashville for 3 1/2 years until this past April
02:16.05mog_homethats a nice place
02:16.15ronaldl79I've accomplished more here for my business (Riverscape) than I did in Nashville
02:16.19mog_homeshould have stayed...
02:16.31mog_homewhy you comming to digium?
02:16.32ronaldl79I 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)
02:16.58ronaldl79To get a sense of the company's culture....
02:17.21Nuggetronaldl79 just likes visiting dollywood and twitty city.
02:17.21ronaldl79It's an acquisition interest
02:17.48ronaldl79lol
02:17.52ronaldl79Never been to dollywood
02:21.04h3x0ryes
02:21.07h3x0roops
02:21.27mog_homewow that was cold...
02:22.33*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
02:24.05*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
02:24.07PakiPenguinmorning
02:24.33mog_homemorning
02:24.39PakiPenguincan anyone tell me whether i should go for channel bank or for ip phones?
02:24.51[TK]D-FenderIP Phones if at all possible.
02:25.13[TK]D-FenderChannel bank will cost you more and provide less functionality.
02:25.13PakiPenguincool! and which ip phones do you suggest?
02:25.22[TK]D-FenderDepends ont he need.
02:25.31[TK]D-FenderWhat kind of setup?
02:25.43mog_homehow many phones you setting upo
02:25.47mog_homeare you already wired?
02:25.53PakiPenguin20
02:25.54PakiPenguinyeah
02:25.56*** part/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net)
02:26.10[TK]D-FenderYes,  Will you be able to run a 2nd wire for the phone seperate from the computer's?
02:26.12PakiPenguinsorry 30
02:26.27[TK]D-FenderWhat kind of call volume?
02:26.39mog_homei think most people are happier with ip
02:26.40PakiPenguinyes  we can do that , the place is pretty neatly wired up , 2 ports on every desk for ethernet
02:26.47mog_homechannel bank can be cheaper
02:26.57[TK]D-FenderOk, I'd suggest Polycom 100%
02:27.07PakiPenguincall volume is ~100 calls a day per agent , incoming , outgoing
02:27.14PakiPenguin[TK]D-Fender, can you link me please?
02:27.15h3x0ri saw some funny shit the other day
02:27.16[TK]D-FenderNeed speakerphone?
02:27.21PakiPenguinnot really
02:27.24*** join/#asterisk luisedo (n=luisedo@208.195.215.4)
02:27.30h3x0ra call center with a spa-1001 adapter and an analog phone on 30 desks.
02:27.37h3x0rlazy bastards
02:27.42[TK]D-FenderSo a basic call center with say hard-wired headsets?
02:27.49PakiPenguinyeah
02:27.54luisedohi every body
02:28.17[TK]D-Fenderh3x0r : 1 thing to keep in mind : REN <-
02:28.27luisedoi'm looking for some kind of guru who can help this soul
02:28.59[TK]D-FenderPakiPenguin : Then I'd suggest Polycom IP 301's for your agents (the majority I suspect), and IP 601's for managers.
02:29.16PakiPenguinHow much a IP301 cost? [TK]D-Fender
02:29.27luisedodoes any one of you have worked with festival?
02:29.34[TK]D-Fenderhttp://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm
02:29.37mog_homei have luisedo
02:29.42mog_homewhats 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
02:29.56luisedohi mog
02:30.03SkramX301 == okay?
02:30.12Dr-Linuxmy asterisk BOX is behind the firewall, canreinvite=no option in sip.conf should be set to no or yes?
02:30.16luisedoi'm trying to make an ivr wirh festival and asterisk
02:30.17[TK]D-FenderSkramX : for a minimal agent on headset, yes
02:30.20mog_homeyeah
02:30.24Dr-Linuxi can't hear ivr sounds etc
02:30.27PakiPenguinhmmms
02:30.32luisedobut when festival is loaded
02:30.33[TK]D-FenderAll Polycom's are very solid phones.
02:30.40mog_homeyeah
02:30.43luisedoloads the 100% of my processor
02:30.51mog_homeyou using punctuation?
02:31.00luisedoyep
02:31.03mog_homeare you using the 8khz voices?
02:31.10luisedofestival ('test to say')
02:31.22SkramXfestival sucks bro..
02:31.23mog_hometry taking out punctuations there is an old bug involved with punctuation
02:31.25SkramXseriously.
02:31.27mog_homebut its not the best
02:31.31luisedoi duno, where cain i check if i'm using 8k voices?
02:31.31mog_homeit works though
02:31.35mog_homeand is funny if its not your biz
02:31.37mog_homeif you dont knw
02:31.41mog_homeyou probably arent
02:31.46mog_homeand thats your problem
02:31.53mog_homevoip-info has a guide on it
02:31.59mog_homeim sorry i dont know off of top head
02:32.10luisedoyesp ive read voip-info already
02:32.37luisedoi ve other problem
02:32.40PakiPenguinhttp://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43410781952.htm <-- is this phone any good?
02:32.51PakiPenguinthe gxp - 2000
02:33.22[TK]D-FenderPakiPenguin : 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.....
02:33.35PakiPenguinoh okay
02:33.48myke420247i have a bunch of gxp2000's
02:33.49luisedoas you can c (buecause of my very poor english :D) the ivr that i want to make must be in spanish
02:33.49myke420247they're ok
02:33.52myke420247no sidetone tho
02:34.01luisedoso i want to set spanish as the default language on festival
02:34.58[TK]D-FenderI 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?
02:35.31franxhi, could you tell me where to find a guide for sip hardphones installation?
02:35.37myke420247polycom is $$$
02:35.46luisedoi'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-Fendermyke420247 : You do get what you pay for....  The feel, operate, and manage very well.
02:36.14Katty[TK]D-Fender: i'm making happy bread.
02:36.29[TK]D-FenderKatty : mew?
02:36.41Katty[TK]D-Fender: it's when you make a double batch.
02:36.50Katty[TK]D-Fender: and then, when it's done baking, take it to someone else
02:37.09[TK]D-FenderKatty : Strange to see you on at night too....
02:37.29Katty[TK]D-Fender: not really. i'm just usually talking in other channels on other irc servers.
02:39.36[TK]D-FenderAh....
02:39.43*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
02:39.52[TK]D-Fenderfine... stranger to see you HERE at night :)
02:42.10Katty(=
02:43.31*** join/#asterisk bjohnson (n=bjohnson@i216-58-43-89.cybersurf.com)
02:44.13PakiPenguinhmms
02:44.13myke420247kitty katty
02:44.16myke420247how many cats do you have?
02:44.21myke420247<- 6
02:44.24*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
02:46.12DrukenAHHHHHH!!!!!!!!!!!!!
02:46.42Katty...
02:46.49shido6ahh the good old budgetone
02:47.17luisedodoes anyone knows how to change the default language of festival?
02:47.19Drukenhow is Katty tonight?
02:47.53Kattymopey and depressed.
02:47.55myke420247gxp != budgetone
02:47.58Kattyhence the happy bread.
02:48.13Drukenwhy depressed?
02:48.23Kattydon't ask
02:48.32Kattyand i don't want fixed either, so don't bother trying
02:49.09shido6I was depressed
02:49.14shido6then I sold my cisco 7960s
02:49.25shido6and bought a trip to see the family
02:49.45DrukenKatty: aight..
02:50.05Kattykthx
02:50.22PakiPenguinmyke420247, do you have a gxp?
02:50.31myke420247paki, about a dozen
02:50.41Drukenshido6: why you get rid of the cisco's?
02:50.51Dandani have a gxp
02:50.54shido6because I dont need them anymore
02:50.57shido6Im selling 941's
02:51.00Dandanabout 70 :)
02:51.14myke420247damn
02:51.25[TK]D-Fendershido6 : and why are you selling 941's?
02:51.26Dandanyeah running on .13 beta :)
02:51.27shido6I only used 1 line on the cisco's really
02:51.30myke420247dandan, any way to get sidetone on them?  that's the biggest complaint so far
02:51.40shido6941's are cheaper
02:51.49Dandanmyke420247: no, same situation here
02:51.51shido6and u can configure up to 4 line appearances
02:51.52PakiPenguinDandan, are they nice phones? i mean quality / voice quality?
02:51.54Dandani am happy with blf
02:51.56shido61/2 the price of s 7960
02:52.04myke420247blf?
02:52.07[TK]D-Fendershido6 : when you say "selling", do you mean your personal ones, or as a business?
02:52.13DandanPakiPenguin: well, I do not have any complains from my users
02:52.16shido6business
02:52.17Dandanbusy light
02:52.27Dandanand auto answer
02:52.58[TK]D-Fendershido6 : the 941 is a great little phone, but with what you can get a Polycom IP 501 for these days, its a hard arguement.
02:53.07shido6my first sip phone here the budgetone works quite well with g729 on the NuFone net
02:53.11PakiPenguinthanks a lot [TK]D-Fender  :)
02:53.18shido6how much can you get polycom 501's for
02:53.19shido6?
02:53.22myke420247yeah i have a barbietone at home
02:53.32Dandani have evaluated a snom 190, poly 300 and those gxp's
02:53.34PakiPenguinbarbitone :p
02:53.40Dandanand i liked gxpes the most :)
02:53.58shido6yeah jerjer coined that one when he and I were screwing with the elcheapo phones
02:54.10[TK]D-Fender170$USD.  typically 941's =$150USD.  For 20$ you get a better speakerphone, Polycom quality and pixl display, PoE possible, a 2nd eth port.....
02:54.15shido6then more people started using the term and giggling even more
02:54.34shido6now we're waiting for Mattel or Grndstream to really make one
02:54.36Dandangood one shido6
02:54.42[TK]D-FenderI run 26 Polycom IP 600's and 1 x 601 at work.
02:54.42luisedoPLEASE 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...
02:54.59luisedodoes any one know how to change the default voice on festival?
02:55.01Dandanluisedo: we are all ears
02:55.12luisedothanks Dandan
02:55.21Nivex[TK]D-Fender: where?  The cheapest I've seen 501's new is $199
02:55.24Dandanlemmie see :)
02:55.29[TK]D-Fenderwww.atacomm.com
02:56.26luisedoi'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)
02:56.43Nivex[TK]D-Fender: thanks!
02:57.09*** join/#asterisk Zach^^ (n=Zachary@65.121.244.130)
02:57.15Zach^^how do i seup incoming calls to load holiday greeting then load mainmen options?
02:57.26Nivexit'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?
02:57.31Dandanedit main menu :)
02:57.54*** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca)
02:57.54Dandanluisedo: did you get it?
02:58.20Drukenrtp is udp right?
02:58.31shido6http://festvox.org/docs/manual-1.4.3/festival_7.html
02:58.32NivexDruken: yes
02:58.36Zach^^Dandan i setup holiday greeting.. to play a short msg then i want it to auto move to the next menu
02:58.39shido6--language LANG
02:58.39luisedoyes Dandan
02:58.41DrukenNivex: thought so :)
02:58.46luisedoi'm starting to read couse ma connection is too bad
02:58.51shido6currently LANG may be one of english spanish or welsh
02:58.59shido6depnding on what voices are actuallya vailable in your installation
02:59.19Drukenwelsh.. isn't that a wanabe english with a UK accent?
02:59.25luisedoyes i've tried --language spanish when i start festval server
02:59.40luisedobut it doesn't work when i use the command test2wave
02:59.45DandanZach^^: 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
02:59.45luisedotext2wave
03:00.23Zach^^Dandan... i have it working but there is a long delay
03:00.46Dandancan you paste it somewhere?
03:00.49Dandan~pastebin
03:00.51jbotsomebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.com/
03:01.04luisedooff course
03:01.08luisedogive me one sec
03:01.20Ateboyhi, 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...
03:01.30Dandanlol that was for zach and luisedo :)
03:01.45SkramXYes! I just got a small holiday bonus from the boss!
03:01.49AteboyI must admit I don't know much about asterisk but that is why I'm experimenting...
03:01.49DandanAteboy: i can't tell you anything, i used 0.7 :)
03:01.55[TK]D-FenderNivex : Its easy to get, just ask around.  I have about 1/2 dozen versions myself
03:01.59DandanSkramX: <= jealous!
03:02.23Dandan~docs
03:02.24jboti 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
03:02.24SkramXDandan: It wasnt much.. and its not even a real job, just a small support job for a small web hosting company
03:02.36DandanAteboy: http://www.oreilly.com/catalog/asterisk
03:02.45SkramXI have my own hosting company as well.. but it is targeted to asterisk and linux guru's/intermediates.
03:02.45Ateboyalready ordered
03:02.50DandanSkramX: <= still jealous
03:02.54SkramXDandan: :|
03:02.59Dandan:))))
03:03.00Ateboywith "switching to VoIP"
03:03.01Dandanj/k :)
03:03.06Dandangood luck :)
03:03.10SkramXIt was enough to buy a 941!!
03:03.13DandanAteboy: i have 3 books so far :)
03:03.29SkramXlol
03:03.44luisedoready
03:03.47luisedohttp://pastebin.com/471210
03:03.58luisedothat's my whole extensions.conf
03:04.00DandanAteboy: http://geekgazette.com/index.php?option=com_content&task=view&id=40&Itemid=2
03:04.04Zach^^Dandan what file is it?
03:04.11luisedoextensions.conf
03:04.17DandanZach^^: extensions.conf? i do not know
03:04.21Dandanmaybe you have some includes
03:04.33DandanAteboy: that book is free for download
03:04.39DandanZach^^: you should DL it too
03:04.59AteboyI know, but since it was reasonably priced, I bought it to encourage the project or the author
03:05.14Ateboywhy download?  for easy search?  I d/l it as well...
03:05.17Dandanluisedo: instead of background use playback
03:05.33Ateboywas waiting for a working asterisk setup to really go through it
03:05.39DandanAteboy: yeah, i strongly DISCOURAGE you from the YELLOW book
03:06.06AteboyI read a few how-tos, the wiki... so many stuff...
03:06.06DandanAteboy: switching to voip is 50% theory
03:06.16Ateboydandan: which yellow book?
03:06.17Dandanthen they get down to business
03:06.22DandanAteboy: hold on :)
03:06.27luisedoDanda, i use background buecause i can press a number while i'm hearing the file
03:06.29Ateboydandan: I should cancel my order?
03:06.31Zach^^Dandan http://pastebin.com/471216
03:06.35DandanAteboy: no
03:06.40Dandanthose books from oreilly are good
03:06.48Dandanluisedo: to do what?
03:06.53luisedoand i dont have to wait to finish te audio file to press a number
03:06.59DandanZach^^: one sec
03:07.14luisedoan ivr
03:07.27Ateboydandan: I don't follow you... I see you're busy though... I'll wait
03:08.39[TK]D-FenderI like my 941...
03:08.44*** join/#asterisk ManxPower (n=ewieling@29.sub-70-219-77.myvzw.com)
03:08.49SkramXI should get one..
03:09.29DandanAteboy: 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
03:09.41DandanZach^^: one sec, looking
03:10.13Ateboydandan: Ok, first time I see this one.  I'll stay far from it.
03:10.44DandanAteboy: i can recommend the oreilly ones i read them both
03:10.54AteboyI should get them soon
03:11.00DandanZach^^: boy, your extensions.conf is messy
03:11.11Zach^^Dandan i got it....
03:11.12Dandanwhy is background as priority 9?
03:11.14Zach^^next thing
03:11.15Dandan:)
03:11.33Zach^^when i press # it says no directory entires match your search
03:11.53Ateboywhat 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?
03:12.05Dandaninclude => app-directory
03:12.20DandanAteboy: i use sipps on two pcs
03:12.25Dandanand got a x100p clone from ebay
03:12.29Dandanfro 9.95
03:12.30Dandanafair
03:12.41DandanZach^^: look in app-directory
03:12.45Dandanwhat is going on there
03:12.55DandanZach^^: is that *@home?
03:13.00Zach^^yep
03:13.06DefrazIn 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.
03:13.08Ateboydandan: what is sipps ?
03:13.24DefrazCan I escape it with anything?
03:13.30Dandando you know what nero is?
03:13.44raeth`lol AMP must hate my guts.
03:13.48alephcomDefraz: You shouldn't need to escape it with anything.  I do something similar to that.
03:13.51DandanAteboy: http://ww2.nero.com/sippstar/enu/index.html
03:13.57Zach^^Dandan http://pastebin.com/471225
03:14.37Defrazhmmm yea amp hates it but I was curious if I could do it right in the extentions file instead.
03:14.41DandanZach^^: exten => #,2,AGI(directory,${DIR-CONTEXT},ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS}o)
03:14.47Dandancheck your contexts
03:14.55myke420247is sippstar based on asterisk?
03:14.56Dandan(sorry I am no an *@home expert)
03:14.57Zach^^?
03:15.04Dandanmyke420247: do not think so
03:15.10Dandanit is windows based
03:15.17DefrazI am going to prepend it with 6 digits then # then the number dialed (10 digits)
03:15.22myke420247what's good about it, for $400?
03:15.40Dandanmyke420247: hm, lets see :)
03:16.22*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
03:16.29Dandangotta find a place to DL it
03:16.29Dandan:)
03:16.32Zach^^Dandan if i call *411 it send me to the list and works
03:17.01DandanZach^^: what does your asterisk -rvvvvvv say?
03:17.31Dandanshmaltz: and...?
03:17.40Zach^^Dandan when i hit # from the main menu?
03:17.45shmaltzDndan, 3 work, 2 more to go
03:17.49DandanZach^^: yup
03:18.10*** join/#asterisk jahani2 (n=k@adsl-19-47-192-81.adsl.iam.net.ma)
03:18.10Zach^^<PROTECTED>
03:18.10Zach^^<PROTECTED>
03:18.10Zach^^<PROTECTED>
03:18.18Dandangood
03:18.28DandanZach^^: and...?
03:18.33Dandan(remember about pastebin)
03:18.38Zach^^yea
03:18.54shmaltzhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5838739062&rd=1&sspagename=STRK%3AMEWN%3AIT&rd=1
03:19.06Zach^^http://pastebin.com/471231
03:19.21shmaltzwell, the other 2 work as well, not a bad deal for $30.00
03:20.16DandanHistory:1 bid     (US $19.99 starting bid)
03:20.23Dandanlol the bid war was fierce!
03:20.36DandanZach^^: you gotta ask *@home gurus
03:20.47Dandani see that the agi directory had problems
03:20.49Dandanand exited
03:20.58mog_homeasterisk@home guru... lol
03:21.20Dandanshmaltz: Very Irresponsible , Not worth doing bussiness with !!! NOT A GOOD EBAYER !! :))) LOL
03:21.27Dandanmog_home: oxymoron :)
03:21.48Zach^^mog_home u can help ??? :D:D
03:22.00shmaltzDandan, you talking about that stupid fb that guy left me?
03:22.05mog_homewith?
03:22.23Dandanshmaltz: yah :)
03:23.03Zach^^when i press # for the company dir... it say no directory entires match your search
03:23.45shmaltzDandan, that idiot didn't answer my email for 5 days, and when it came to get the model number refused to do so.
03:24.15Dandanshmaltz: I know, eB is full of ppl like him :)
03:24.20mog_homewhats your dialplan look like
03:24.28Dandani got some fb from a$$h-les :)
03:24.43Zach^^mog_home have to explain.. i am neb to asterisk
03:24.50luisedothis is my dialplan http://pastebin.com/471210
03:25.36shmaltzDandan, 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.
03:25.38Zach^^neb=newb
03:25.46Dandanluisedo: exten=s,3,Festival('Para escuchar un poema marque 4') LOL :)
03:25.58shmaltzthe only time I ever wanted to sell something on eBay, they yanked it as an illegal auction
03:26.00Dandanshmaltz: don't worry pretty much all sellers disregard :)
03:26.12shmaltzDandan, I realized that
03:26.13Dandanshmaltz: were you selling your kidneys?
03:26.21luisedoDandan, what's wrong with that line?
03:26.24shmaltzDandan, not realy
03:26.38*** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au)
03:26.41franxi'i have a file called SIP000E08DAEA1E.cnf, should i upload it to the phone?
03:26.43Dandanluisedo: nothing, i am in a good mood really :)
03:26.53Dandannever encountered a press 4 for a poem
03:26.56luisedoooooooooooooooooo ic ;)
03:27.04shmaltzI was trying to sell a hand held keymaker/cutter, they told me it's illegal because you need a locksmith license for that
03:27.06luisedohahaha
03:27.19shmaltzin fact I'm still trying to get rid of it
03:27.41luisedoit's an example... what i must do is to give medical dates using voip
03:27.47Dandanshmaltz: I think if you are apprehended by pol1ce with thaose tools in your pocket you would get arrested :)
03:28.04shmaltzDnadan, not the key cutter
03:28.13luisedobut i'll use festival to read the hour of the date or the name of the doctor or the pacient...
03:28.25Dr-Linuxi faced this error "[root@i2c-pbx root]# Ouch ... error while writing audio data: : Broken pipe"
03:28.32Dr-Linuxwhat could be happend :S
03:28.38Dandani do not know how it works but i am pretty sure it is included in the paranoidal paradign of the police departments
03:28.40shmaltzDandan, 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
03:28.45Dr-Linuxi'm unable to start asterisk :S
03:28.48ManxPowerDr-Linux, thats usually a mpg123 error
03:29.02luisedoof course when the db is ready...
03:29.13Dandanluisedo: nice :)
03:29.24luisedoit's suposed that we well joint the whole proyect in one week
03:29.42Dandanwell good luck :)
03:29.55Dandani just bought an usb sound card to use with asterisk
03:30.01Dandananyone has any experience?
03:30.48ManxPowerdanalien, is it supported by Linux?
03:30.53ManxPowerif not, send itback
03:31.03Dandanactually anyone used alsa for an overhead announcer?
03:31.12mog_homewhy you need a sound card
03:31.13DandanManxPower: yah, i checked that beforehand
03:31.14Dandan:)
03:31.17Dr-Linuxi kill the mpg123 process, but still same happend:S
03:31.31Dandanmog_home: ^^
03:31.46mog_homeDandan:
03:31.47DandanDr-Linux: which version of mpg do you have?
03:32.17Dandanmog_home: ?
03:32.23mog_home???
03:32.31Dandanmog_home: actually anyone used alsa for an overhead announcer?
03:33.55mog_homeyeah
03:34.04mog_homeits pretty simple
03:34.11Dandando you have any scripts to share? :)
03:34.30mog_homeno need for scripts
03:35.07Dandani am just afraid that it doesn't work well with alsa
03:35.13Dandani read that * prefers oss
03:35.21mog_homenah
03:35.27mog_homeits the same
03:35.31Zach^^mog_home any idea about the company directory problem?
03:35.44Dandanmog_home: ty
03:37.42Zach^^mog u around?
03:37.54mog_homeno clue its got to be a dial plan issue
03:38.07Zach^^what file is the dialplan in?
03:38.15Qwell~wikis
03:38.16jbotrumour has it, wikis is http://www.voip-info.org
03:38.22QwellZach^^: You have some reading to do...
03:38.25mog_homeextensions.conf
03:38.40luisedoZach, extensions.conf
03:40.15Zach^^what do i need to look for?
03:40.59shmaltzZach^^, the wiki
03:41.32luisedoyou need to read a little...
03:41.48shmaltzFunny Quote of the Day - George Burns - "When I was a boy the Dead Sea was only sick."
03:42.07franxin 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)
03:42.13shmaltzFunny Quote of the Day - Jay London - "It all started when my dog began getting free roll over minutes."
03:42.20Dandanshmaltz: yeah, indeed :)
03:42.22shmaltzfranx, define there
03:42.57franxshmaltz: i mean to the phone
03:43.09shmaltzfranx, what phone?
03:43.14shmaltzwhat phone you using?
03:43.15franxthe ip phone
03:43.33Dandanfranx: which one?
03:43.36franxthe ip phone im trying to configure
03:43.37Dandanany pictures?
03:43.46franxspa 841
03:44.06franxsipura spa 841 (sorry for the dumb answers, im kinda slow)
03:44.10darwin35the 841 is easy
03:44.20*** join/#asterisk andu (n=andu@S0106000476ee2cfe.cg.shawcable.net)
03:44.23darwin35go to admin mode and advanced
03:44.29darwin35goto line 1
03:44.29SwKchan_zap.c:62:2: #error "You need newer libpri"
03:44.31SwKdamn it
03:44.40SwKi just checked this shit outta svn
03:44.40shmaltzfranx, 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
03:44.53mog_homedid you install?
03:44.58darwin35adn put in the proxy you connect to adn the usename  adn password
03:45.03shmaltzdarwin35, they never existed
03:45.03SwKyes
03:45.18SwKlibpri # svn up
03:45.18SwKAt revision 282.
03:45.30QwellSwK: that's hot
03:45.35mog_homeits probably nubbed somewhere swk
03:45.40SwKyeah
03:45.43mog_homeunless cresl1n is jokin on ya
03:45.45SwKits just pissing me off
03:46.24shmaltzhttp://www.wired.com/news/culture/0,1284,69861,00.html?tw=rss.TOP
03:46.34shmaltzI think we should send the 841 for contest
03:46.54Qwell841 shipped
03:47.25franxthanks a lot !!1
03:47.37Dandanshmaltz: Duke Nukem Forever :)
03:47.37shmaltzQwell, but it's discontinued,
03:47.44anduHello 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
03:47.46Qwelldoesn't mean it's not vaporware
03:47.50Qwellrather, doesn't mean it is
03:48.46mog_home?
03:48.51mog_home841?
03:48.58shmaltzI hope Zach^^ left because he was too busy reading :)
03:49.09shmaltzmog, yeah, whats wrong
03:49.41mog_homewhats 841 Qwell
03:49.56shmaltz~841
03:50.06shmaltzjbot is ignorant
03:50.11franxhehe
03:50.18franxi have a login problem :/
03:50.32darwin35jbot sex
03:50.34jbotupdatedb; 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/
03:50.37franxwhat is the default user/password
03:50.41mog_homewow
03:50.46mog_homegross
03:51.10darwin35for the 841 its blank
03:51.20*** join/#asterisk bmg505 (n=leon@dsl-146-15-60.telkomadsl.co.za)
03:51.31darwin35factory default
03:51.44mog_homewhats 841
03:52.05franxthanks
03:52.12Dandan~sippura
03:52.20rob0741xx is Tulsa, OK; 641xx is KC, MO ... 841 must be further west
03:52.21darwin35what ever the nmbr on the sipura hard phone is
03:52.25Dandan~grandstream
03:52.26jbot65USD /phone  email to voipsales@xvoip.com
03:52.42darwin35dont buy grandstream
03:52.47darwin35not worth it
03:52.47Dandanwhoa which one?
03:52.58Dandandarwin35: too late, already have 70 of them
03:53.06darwin35for the same money you can get a open src pa168 based phoone
03:53.10shmaltzmog, whats gross?
03:53.25mog_homejbot sex
03:53.26jbotupdatedb; 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/
03:53.28Dandandarwin35: which one?
03:53.37shmaltzmog, I think it's funny
03:53.43mog_homeit is
03:53.50mog_homethe first million times i read it
03:53.54darwin35I have the yughen ywh10
03:54.11anduis there anything wlse required in sip.conf besides subscribecontext = ... to make use of the hint extension?
03:54.16darwin35you can see them on iareaphone.net
03:54.23andudarwin35: I found those to be very slow
03:54.45darwin35no not with frimware upgrades
03:54.56shmaltzwell, I got this one once:
03:54.58shmaltzsex is like math, add the bed, subtract the cloths, devide the legs, leave your solution, and hope they don't multiply
03:55.09mog_homeuh hu
03:55.17andudarwin35: for a phone I mean there's a lot of lag refreshing the lcd etc, the ATA are far better IMO
03:55.51darwin35you have to turn off the verbosity in the software
03:55.58darwin35it boots  faster
03:56.17darwin35but you   have to know how . in the nnext ver I release it will be in fast boot mode
03:56.25darwin351/3 the time to boot
03:57.33andunext ver like in the redfox firmware or something else ?
03:58.07darwin35I use redfox src but modify it for my needs and turn off some things they leave on
03:58.51darwin35thus far I like how it has turned out
03:59.48darwin35but the only one I have is for the ywh10
03:59.58andudo you have a d/l link ? I'd like to give revive my ywh10
03:59.59darwin35but the only one I have is for the ywh10/12
04:00.10darwin35I have the passthrew port
04:02.04darwin35I will in a few  let me finish my last patch
04:02.09Ateboyquick question: for a beginner, should I go with centos 3 (kernel 2.4) or 4 (2.6)?
04:02.15anduthanks darwin35
04:02.23darwin35I am working on now to get the mwi to work right
04:02.45andulcd message ?
04:02.57anduor how did you go about it ?
04:03.04darwin35yeah the count is wrong
04:03.10darwin35and it should flash
04:06.43darwin35right now it comes up with message:255
04:06.49darwin35it should count
04:09.11*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
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04:21.53DandanAteboy: 3
04:21.53kimosabehi lol
04:31.46Ateboy3
04:31.50Ateboy3?
04:32.26Dandanyeah
04:32.26Dandan2.4
04:33.26Ateboydandan: rationale?
04:33.58Dandani still consider 2.6 beta
04:34.12*** join/#asterisk franx (n=Francisc@185-76-246-201.adsl.terra.cl)
04:34.22Dandando 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)
04:34.41Ateboywell, I will use the CentOS stock kernel
04:34.45Ateboy*would
04:34.49Dandanoh
04:35.00Dandani still have 2.4 on all my serious servers
04:35.00Dandan:)
04:35.13Dandanit just suits me better YMMV
04:35.17AteboyI have 2.6 on my prod server...
04:35.18Ateboyeh
04:35.27Dandan:)
04:35.37Dandani have 2.4 with grsec and I am happy :)
04:36.01AteboyI see
04:36.30AteboyI'm very insecure compared to you...
04:36.38darwin35www.digitalgunfire.com
04:41.00franxif 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?
04:41.26Dandanfranx: that depends on canreinvite=
04:41.30Dandanand your routing table
04:42.03franxdoes c connect directly to a or b?
04:42.17*** part/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca)
04:42.19Dandanfranx: it all depends
04:42.27Dandanif you have canreinvite=no
04:42.31Dandanthen your * box
04:42.51Dandanwill route calls through the asterisk
04:43.03Dandanif your canreinvite is set to yes then
04:43.17Dandanyour clients can establish independents call pathx
04:43.27Dandanthat is without any * middleman :)
04:43.49franxim having troubles doing that, i can send audio from a or b to c but from c to one of them not
04:44.37Dandanis C behind NAT (that is on the other side of NAT)?
04:44.49franxyes
04:45.09franxa and b are behind NAT
04:45.13franxc is outside
04:45.40Dandanhm
04:45.51Dandanthen read about: STUN, RTP UDP and problems with NAT
04:46.00Dandanvoip-info.org is your friend
04:46.08Dandan~stun
04:46.09jbotsomebody said stun was that feeling you get when you realise your SIP call actually got through!
04:46.18Dandan:>
04:46.25Dandan~nat
04:46.26jboti guess nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
04:46.39franxthanks a lot for your time
04:46.43DandanYEAH: See docs.
04:46.46Dandanno problem :)
04:48.17franxis there an official asterisk reference?
04:48.56Dandanofficial?
04:48.56Dandanwell
04:49.19franxwell something close
04:49.26DandanI find the books useful as a general introduction to asterisk, but voip-info.org is indespensible
04:49.26Dandan:)
04:49.38Dandan~books
04:49.42franxill take that into note
04:49.45Dandan~book
04:49.46jbotit has been said that book is on the table
04:49.48franxhee
04:49.57Dandanhttp://www.oreilly.com/catalog/asterisk/
04:50.03Dandanthat one and Switching to VOIP
04:50.17Dandanthis one you can DL for free
04:50.22franxi see
04:50.34Dandanhttp://geekgazette.com/index.php?option=com_content&task=view&id=40&Itemid=2
04:50.46Dandanbut I encourage you to buy it :)
04:51.14franxill do it
04:51.30franxoreily owns
04:51.37Dandanyes they do
04:52.29franxmust go, once again thanks a lot and ill rember buying those books :D
04:53.20mog_homeor dl em
04:55.24Dandanwell
04:55.26Dandanif you buy
04:55.34Dandanthat encourages writers to write
04:55.48Dandanthe more they write the better the material they produce (hopefully)
04:55.49Dandan:)
04:57.14*** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
04:57.51AyanoI have an asterisk server set up, and it wont dial from extension to extension.  I don't even know were to begin
05:00.06Dandanhm does your context include both extensions?
05:06.28Ayanoits an aah install.  So it should.  where do I check?
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05:09.40*** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net)
05:09.41Dandanno idea aah does not apply here
05:09.43Dandansorry
05:09.43Dandan:/
05:10.36Dandantime to go
05:10.37Dandan[d]
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05:25.00lehelhey
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05:46.31anduAny 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
05:47.43*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)
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05:54.32tecnicoHi. 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
05:55.02shido6contact your service provider
05:55.18tecnicoISP you mean ?
05:55.34shido6voip service provider
05:55.59tecnicolike I said, the problem is simulatenous with multiple VOIP providers
05:56.11shido6same versions?
05:56.25tecnicoTeliax, voxee, syxtel
05:56.32tecnicoall three at the same time
05:56.40shido6are they running asterisk?
05:56.56tecnicoMy guess is yes.
05:57.10tecnicoI'm using IAX for all of them
05:57.41shido6you're gonna end up at Shit Creek if you keep travelling up Assumption Rd.
05:58.34tecnicowell, 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
05:59.41tecnicoass soon as I dial.. the call gets accepted by the provider and not even a second later I get the error
06:00.01benjkshido6: how do I dial international on NuFone, 011 prefix doesn't seem to work
06:00.17Qwellbenjk: They need to enable it on your account
06:00.20Qwellspeaking of which
06:00.28Qwellshido6: Got a rates.csv which lists everything?
06:00.55benjkshido6: how do I get international dialing enabled?
06:03.41shido630 days
06:03.52benjkshido6: how come I am getting charged for calls on a DID that JerJer says is not yet activated
06:03.57shido6and a verification process
06:04.23benjkok
06:15.42shido6heh
06:15.49shido6are you making test calls to this number? :)
06:15.50*** join/#asterisk Igbothom_III (n=HiltonT@203-206-170-99.perm.iinet.net.au)
06:16.06shido6call it from the PSTN for testing
06:16.10shido6not through the network
06:16.19shido6we provision our end immediately
06:16.20benjkI make test calls yes, but from a different provider
06:16.37benjkin fact from several different providers
06:16.47shido6so whenever the legacy telco decides to get their act together we're set to go
06:17.02benjkI don't blame you
06:17.25benjkbut it says 10 days and its been 12 days now
06:17.56benjkor is that business days?
06:19.03benjkif saturdays and sundays don't count, then it will be one more day
06:19.05*** join/#asterisk kannan (n=kannan@dsl-Chn-static-223.45.101.203.touchtelindia.net)
06:19.29shido6sat sun, or holidays
06:19.47benjkthey don't count for the 10 days period?
06:20.23shido6no
06:22.41*** join/#asterisk Igbothom_III (n=HiltonT@203-206-170-99.perm.iinet.net.au)
06:23.26benjkok, I didnt recall what the website page said, only remember "10 days" and now I can't see that page anymore
06:23.40benjkbut thanks for the info
06:23.53benjkshould come online tomorrow then
06:24.17*** join/#asterisk michael123 (n=michael@202.22.163.104)
06:24.33michael123guys what is the best echo canceller
06:24.38michael123software
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06:32.23*** join/#asterisk AFK1 (n=itsme@203.81.209.175)
06:32.55AFK1hii guys, need help???????? can anyone help me with a recording solution in which digium can be used for just tapping the lines?
06:38.07mog_homewhat AFK1
06:38.17mog_homeyou can just do record
06:38.21mog_homebefore any dial
06:38.43*** join/#asterisk Primer (n=vi@sh.nu)
06:39.05AFK1mog_home: let me explain u the scenario,
06:39.21AFK1i have a customer with 24 analog lines running ,, theere is no PBX
06:39.31AFK1now they want to record these 24 analog lines
06:39.39Qwellouch
06:39.44Qwellcan't get a PRI?
06:39.46mog_homerecord
06:39.48AFK1one option is i put 2 x 24 cards, use one for in and one for out,
06:39.50Qwell24 analog lines must cost a ton
06:40.01AFK1but it increases the cost alot by putting 2 x 24 card
06:40.02mog_homeyeah
06:40.10PrimerAnyone 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?
06:40.15mog_homequite a pretty penny i imagine
06:40.27mog_homeman i need a jabber expert
06:40.34AFK1i want to know, if i can use 1 x 24 port card and just tap in the 24 lines?
06:40.56drrayafk1 - through a punchdownd block
06:41.01mog_homethis plain sasl stuff is pissin me off
06:41.03Qwell"tap in"?
06:41.08mog_homeget a break out box
06:41.13mog_homethey are like 50
06:41.20QwellI think he's saying like...
06:41.28drrayget a punchdown block, they are like $24
06:41.29Qwelluse the fxo and a splitter
06:41.42AFK1ahan, Qwell
06:41.49QwellAFK1: That would be horrible to do
06:42.36AFK1Qwell: sooooo :) any tap in solution that woul dbe able to work :)
06:42.46mog_homebreakout box
06:42.51QwellNo, I'm saying...don't do that.  heh
06:43.19QwellAFK1: You have 24 analog lines, and 24 analog phones?
06:43.34AFK1yeah thats true?
06:43.41QwellI don't know.  Is it?
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06:44.05AFK1yeah yeah 24 anolog lines connected to 24 analog phones:)
06:44.18Qwellthen you'll want 2 x 24 port cards
06:44.51Qwellor 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)
06:45.10QwellI can't even imagine what 24 analog lines would cost
06:45.13AFK1no :) i want to use 1 x 24 card to do this rather using 2 x 24 caerds :)
06:45.16h3x0rim gonna print some t-shirts that says "TDM SUCKS"
06:45.26QwellAFK1: You can do that, but you won't be able to hook up your phones
06:45.31QwellYou need one port per device.
06:45.34Qwelldevice being a line, or a phone
06:45.35drrayI heart my TDM
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06:45.48h3x0ron the front it will say "But ATM sucks more"
06:45.49QwellSo, in your case, you have 48 devices, which needs 48 ports
06:45.55Assidthis is really funny, i cant get moh working on 1 box.. but works perfect on my home box
06:45.57drrayswallows
06:46.06QwellAssid: got the right version of mpg123 on said box?
06:46.08AFK1okay Qwell thanks :)
06:46.17QwellAFK1: So, in other words...you need two cards
06:46.29Assidyeah.. but im tyring to use format_mp3
06:46.58Assidim actually gonna try using the same mp3 files on both.. to see why it hates me
06:47.10QwellAssid: format_mp3 hates some mp3s
06:47.19Qwellmake them 8khz mono, and they should work fine
06:47.34QwellAFK1: is there any reason you guys aren't using a PRI?
06:47.36Assidi did.. it is @ 8khz mono
06:47.44QwellAssid: no id3 tags?
06:47.50Assidnope.. all removed
06:47.52Qwellsupposedly that matters
06:48.48Assidwinamp shows the file at 8kbps/8khz/mono
06:49.36YaroMan_Hello everyone
06:49.52YaroMan_I need some help to setup my Asterisk @ Home with BroadVoice
06:50.37drrayI've never used asterisk @ home
06:51.39YaroMan_is it not the same as regular asterisk?
06:52.33mog_homenot quite
06:52.40mog_homeits a little lame
06:52.51mog_homelike it was shot in the leg
06:53.35YaroMan_mog_home what do you use>
06:53.46mog_homesvn trunk
06:53.53mog_homeactually not even that
06:53.54mog_homemy trunk
06:55.15YaroMan_nice
06:55.17mog_homethat and a debian box, and a gentoo box
06:55.20mog_homesome other junk
06:55.25YaroMan_so you have your own numbers?
06:55.41mog_homei have a did yeah
06:55.48mog_homebut i just run asterisk for my home
06:56.02mog_homeotherwise i work on asterisk code and help other people get it installed and going
06:56.42mog_homeqwell what are you still doing up so late
06:57.33AssidQwell: yeah looks like that particular set of mp3 files
06:57.36YaroMan_i went to the bar
06:57.41YaroMan_hook up with few girls
06:57.45YaroMan_and came back home
06:57.52Assidoh wait
06:59.26newlIf you came home alone, you didn't "hook" anything. :D
06:59.30Qwellmog_home: it's only 11
07:00.06mog_homeits 1 am out here
07:00.13mog_homeman you go to bed early
07:00.18Qwellheh
07:00.19drrayonly 11 here
07:00.21Qwellnot usually
07:00.29mog_homei didnt realize there was a 2 hour difference
07:00.40mog_homewell you always go to bed when its around 4 here
07:05.07AssidQwell: if i have to increase the volume do i add -v2.5 to the input options or to the output?
07:13.16AssidQwell: remade the mp3 files.. apparently they werent ready to run on my home box either
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07:18.29Assidbut 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!
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07:27.48uchmanIs there a way to redirect a call with my ZyXEL Prestige 2000W?
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08:16.05YaroMan_ok now i can recive a calls but I can't make calls ;(
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08:18.43eivindtrHi all. Does anyone know about a softphone that allows you to specify autoanswer for a specific set of CallerIDs, or other filtering mechanisms?
08:30.26shido6brb
08:30.31infinity1no
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08:47.35dioptrecan 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?
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08:49.48Chotairemorning.. anyone ever played with meetme talker detection? or even heard of it?
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09:04.31__aguys, any clues as to why a timeout parameter to Queue command wouldn't work?  A Queue just doesn't timeout and keeps ringing...
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09:05.36[Wiebel]hmm
09:05.45*** join/#asterisk oej_ (n=oej@apollo.webway.se)
09:06.24zoaheya oej
09:06.32dioptrecan 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?
09:06.34zoahows berlin ?
09:06.45oej_Berlin?
09:06.51[Wiebel]Hi
09:06.52zoadioptre, what do you mean with hole punching ?
09:07.00zoayou are not at berlin with kpf
09:07.00[Wiebel]anyone here ever tried playing with a cisco 7970 + video?
09:07.01zoa?
09:07.08dioptreusing iax thru firewalls?
09:07.12__azoa: any idea why wouldn't a queue timeout?
09:07.13zoaaaah
09:07.22zoa__a, i'd say a bug
09:07.46zoai don't trust the queue things, they always proved buggy to me in the past
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09:08.53dioptrecan 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?
09:10.25zoadioptre, i still dont get it
09:10.31zoaif you want to go through a firewall use iax
09:10.42zoaif you want to have meetme (rendezvous ?) then use asterisk
09:10.57dioptrebut does iax penetrate fws?
09:11.06zoayes
09:11.54dioptrecool, so how do they interchange info ? like with stun? to get each others ip addresses?
09:12.05dioptrethanks heaps for responding btw
09:13.05dioptredoes asterisk proxy all traffic for an iax conversation?
09:13.42zoaim sorry but i dont really get your questions
09:13.43zoaaaaah
09:13.46zoai guess i get it now
09:13.52zoaiax works in a different way
09:13.58zoait uses only 1 ports
09:14.05zoamaking it really nat friendly
09:14.10zoanat is probably what you mean
09:14.29zoaso asterisk will try to transfer the two ends to each other
09:14.36zoaand if that doesnt work, it will proxy the call
09:15.04dioptreomg great!
09:15.23dioptrethanks!
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09:15.30dioptreive read like a thousand pages to find that out :) ta
09:15.57Assidactually.. if im not mistaken atleast 1 box should be globally/directly accessible
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09:16.02zoayes true
09:16.04zoabut thats normal
09:16.08zoathere's no other way
09:16.21dioptreo so it does need hole punching?
09:16.34dioptrelike in sip?
09:16.53Assidsimilar..
09:16.56dioptreor does asterisk do that?
09:17.03dioptreor can it do that?
09:17.07Assidbut sip uses RTP.. can cause certain issues for some firewalls
09:17.30dioptrebut if i used stun would it work u reckon?
09:17.53Assidit should. yes. if the stun is public
09:18.15AssidBUT.. iax is a much safer bet if 1 box is globally available
09:18.26Assidbesides.. its iax.... effectively native to *
09:18.42dioptreok - but if its p2p
09:19.03dioptre...have to use stun eh?
09:19.14Assidthe main question is.. can any of those peers be available directly
09:19.26dioptreprobably not
09:19.32Assidstun wouldnt really do much then
09:19.38Assidatleast i dont think so
09:19.47zoastun wouldnt do anything for that
09:19.49dioptrebut thats what hole punching is for eh?
09:20.00zoadioptre, no
09:20.04Assidonly for the data connection
09:20.06zoathat wouldnt help you
09:20.07Assidnothing else
09:20.17zoaif your servers are both behind nat, they cannot find each other
09:20.22zoawhatever you try
09:20.26zoayou need something to proxy
09:20.36zoaand stun is not a proxy
09:20.43zoastun is just telling a client its own ip
09:20.48zoapublic ip
09:21.03zoayou don't need that with iax
09:21.12dioptreok - so what if i had clients behind nats/fw and a asterisk server public?
09:21.29dioptrewould that work like stun?
09:22.03Assidokay lets put it this way
09:22.22Assidstun is a tunnel.. but the car doesnt know where to go because it doesnt have a global destination
09:23.25Assiddioptre: why cant you just DMZ/port fw to the asterisk box?
09:24.47dioptrewell it may not be possible for p2p
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09:25.06Assidwell.. even p2p needs a server where peers connect to
09:25.12dioptretrue
09:25.12heroinehi
09:25.26dioptrebut i was hoping that could talk with asterisk
09:25.32dioptrewhich would be public
09:25.47dioptre(and i hoped i didnt have to use stun)
09:25.47Assidif any asterisk box is public.. yes, you can do it
09:25.54dioptreand both clients could be behind fw
09:25.56zoathey can without using stun
09:25.56Assidwithout using stun.
09:26.05Assidiax
09:26.24dioptrewould the asterisk proxy the conv?
09:26.26dioptreor tunnel?
09:26.27zoayes
09:26.33zoaproxy
09:26.44dioptreis there a way to punch?
09:26.57dioptreor get info from asterisk to make a custom punch?
09:26.59Assidget 1 box to be publicly available
09:27.05Assidand you have a "punch"
09:27.38dioptreu mean one cant have nat/fw?
09:27.41*** join/#asterisk reth (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb)
09:27.52dioptre(client)
09:28.07Assidthe server needs dmz/fw if a client wants to access it remotely
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09:29.55rethI get an error "405 Method Not Allowed" when my ata-box is registring. anyone know what to do?
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09:38.32nfi|ermeshi all
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09:40.00dioptre_hey assid
09:40.06dioptre_sorry i dropped out mate
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09:44.53dioptre_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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10:17.47Chotairegood news.. new chan_capi for 1.2.1 should be finished within 2 days ;)
10:18.17Chotaireoej, you there?
10:20.52*** join/#asterisk mkl1525 (n=daniel@82.100.204.245)
10:24.28zoaoej was here a second before
10:24.49Chotairedamn ;)
10:24.52*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
10:25.30zoahe'll be back
10:25.38Chotairethat's ok.
10:25.40ChotaireI gotta discuss a tiny bug with oej about some rfc2833 madness.
10:25.41kmilitzerHello ...
10:25.44Chotaireyup kai9
10:25.48zoaChotaire: can i bother you a little in private ?
10:25.54Chotairezoa: sure, go ahead.
10:29.04kmilitzerI 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
10:29.52kmilitzerIf 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
10:30.15kmilitzerAny ideas where to look for the problem?
10:32.44mkl1525Hi, 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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10:35.54ChotaireI think yes.
10:36.02Chotaire(kmilitzer)
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10:56.13Remosican I do a RoundRobin of SIP interfaces?  Only Zap interfaces appear to support groups?
10:56.33jalsot_hi
10:57.17Remosihi
10:58.09jalsot_does anybody know how much transcoding of g711-Speex/ilbc/GSM/g729 can be achieved with a single P4 3GHz box?
10:58.35zoaaround 120 for speex ilbc and g729
10:58.41zoaaround some hundred with gsm
10:58.46zoaowz
10:58.47jalsot_:o
10:58.48zoahalf that
10:58.52zoa1/2
10:58.54zoa60
10:58.57jalsot_really?
10:58.59zoai thought dual xeon
10:59.00zoayes
10:59.12jalsot_so 60 concurrent calls with speex?
10:59.19zoasomething like that yes
10:59.23zoadepending ont he settings
10:59.33jalsot_I guess speex should be tuned for SSE
11:01.23jalsot_that sounds great
11:01.38jalsot_g729 uses less CPU, right?
11:02.11RoyKiirc that depends on the speex version
11:02.21RoyKthe new cvs head stuff is supposed to be quite good
11:02.34RoyKbut standard configured speex is sloow
11:02.34jalsot_:)
11:02.50RoyKjalsot_: show translation recalc 60
11:02.55jalsot_how could it be tuned?
11:03.04jalsot_I know that CLI
11:03.14RoyKok
11:03.16RoyKsee http://en.wikipedia.org/wiki/Speex
11:03.24jalsot_however milliseconds don't tell mu how much calls can take
11:03.32jalsot_or is there an equation?
11:06.06jalsot_ok, my show translation recalc 60 gave: slin-speex 25
11:06.42jalsot_[but that's an old speex 1.0.4 Ubuntu]
11:08.06tzafrir_laptopwe normally use speex for testing when we want to be sure * does some work. speex as a sure way to hig the CPU ;-)
11:08.29jalsot:)
11:09.09jalsotdoes anybody know if it is possible to calculate how many concurrent call be made from the translation table?
11:09.38jalsotsomewhere I read that my 25ms would mean 40 calls [1s/25ms]
11:09.50*** join/#asterisk juice (n=juice@mo-67-77-176-14.dyn.sprint-hsd.net)
11:21.02cj-rmdoes anyone here use jasterisk??  The Java<->Asterisk bindings??
11:24.41uchmanJava. :(
11:32.11cj-rmyeah, java...
11:32.26cj-rmRoyK: mmmm... nice
11:33.22trixtercan you smoke some java in your wanpipe?
11:42.41tzafrir_laptoptrixter, is that the reason for all of the jams you had in your connection?
11:43.12trixterI didnt have any jams in my connection
11:43.21trixterjam is made from fruit juice I like whole fruit like preserves
11:43.24trixter:P
11:44.03trixteractually 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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11:57.45jalsotdoes anybody know how does the Intel IPP g729 codec perform in comparsion with Digium's g729?
11:59.03zoai tried it
11:59.08zoaits +/- the same
11:59.56jalsotso Digium's version (did they write it or just distributing?) is not better optimized :(
12:00.13zoano big difference between the two
12:00.23jalsotzoa: thanks!
12:00.55zoabut digiums is legal
12:00.57zoaand works on amd
12:01.37RoyKzoa: morning
12:01.41jalsotyes, I know
12:04.03zoamorning
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12:05.51trixterthe 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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12:06.33trixterand 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
12:07.25zoaactually i think that if you pay to digium, you probably could get away with using ipp's
12:07.47zoabut that would require you not to use digiums then
12:07.48trixteryou would have a valid license for the codec
12:07.52zoabut still questionalble
12:07.59trixterwell not a total number greater than what you have paid for
12:08.23trixterbecause 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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12:09.32trixterone thing I do know for sure is that I have a chicken teriyaki rice bowl
12:09.45zoatrue
12:09.56jalsotthere is a licensing fee for the code/binary itself but you should pay royality fee as well, am I right?
12:10.01zoayes
12:10.53jalsotI guess, getting per channel codec license is not cheap as well - probably it's not possible to buy just some
12:11.02zoait is
12:11.05zoawith digium
12:11.09RoyKzoa: any news for the iax/sip jb?
12:11.09zoa10$ / channel
12:11.18jalsotyep, I know that
12:11.20zoai think its ready, let me yell to the other side
12:11.26zoaah its not ready
12:11.27RoyK:)
12:11.28RoyKah
12:11.29RoyKok
12:11.37RoyKwhen?
12:11.37zoaseems like he has some coredumps with it
12:11.43RoyKhehe
12:11.44trixterwell 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)
12:12.07trixterI have no problem screwing the UN out of their license fee
12:12.18trixterthey deserve it
12:12.59jalsotprobably I will try to go with speex ;)
12:13.11jalsotor maybe ilbc
12:13.21trixterbut that doesnt screw the UN out of their money
12:13.26trixterso what good is it :P
12:14.17trixterfor those that dont know the UN owns the ITU that owns G.729
12:15.18trixterand 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
12:16.45tzafrir_laptopthe ITU? there are a number of patent holders, right?
12:16.45trixterpersonally I would like to see a finance sheet of where my UN tax dollars go but that will never happen.
12:16.56trixterthey are at least hte main one
12:17.43trixtermy system is being squirley I cant look it up now for a list
12:18.07trixterbut I know they are at least one and seem to be the front for it, aside from delegating license collection to some other company
12:22.06jalsottrixter: so you like g729 the most, right? :)
12:22.17trixteroh yeah
12:22.19trixtercant you tell?
12:22.26trixterI l;ove everything about it
12:22.44jalsot:D
12:23.14jalsotso it sounds better than anything else?
12:24.36Ikaruslol, 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
12:25.37trixterit sounds better than default festival
12:26.19jalsot:)
12:26.32jalsotI've never tried festival...
12:26.47jalsotwhat in compare with speex or ilbc?
12:26.51trixterdid you have speek and spell as a child?
12:27.01trixterer speak
12:27.05*** join/#asterisk coppice (n=chatzill@195.166.17.210.dyn.pacific.net.hk)
12:27.26*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
12:27.32trixterif you did, speak and spell sounds better than festival
12:27.39trixterespecially to people with hearing loss
12:28.19jalsothmmm, I should learn more English :)
12:28.43trixterspeak and spell is a toy from the 1970s that would have kids spell words
12:29.04jalsotahaaa
12:29.09jalsot:)
12:29.45trixtermeaning 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?
12:30.16coppiceeveryone loves to attack festival. do you realise it is the basis for practically every commericial TTS
12:31.51*** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org)
12:32.03iDunnoisn't it non-freeish?
12:32.27tzafrir_laptopwell, it comes with Debian, so it must be free
12:32.32trixterI doubt that practically every commercial tts violates the gpl
12:32.46coppicefestival is not GPL
12:33.31trixterwhat is it then?
12:34.05coppicesort of BSD like, if i remember correctly
12:34.17*** join/#asterisk Porks (n=nao@200.231.120.138)
12:35.40trixterMIT/X Consortium License
12:36.03coppicepretty similar
12:36.54trixterwhen did festival start?
12:37.45coppicemany years ago. it is one of the key research platforms for TTS, from one of the world's centre of excellence on the subject
12:38.39mutilatori could really go for a monterey chicken quesedilla right now
12:38.41mutilatoranyone else?
12:39.10trixterwell 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?
12:39.13sivanamutilator: ya
12:39.56mutilatorno taco bell and only $6 in the pocket
12:40.03mutilatori think today is going to be mcdonalds dollar menu instead
12:40.15sivanaheh
12:40.17trixterno del taco?
12:40.20coppicetrixter: 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.
12:40.21trixterthey are cheaper than taco bell
12:40.28mutilatortheres only mcdonalds
12:40.34mutilatorand a subway inside a shell gas station
12:40.47Porkshi :D
12:40.54PorksI have two X100P cards conected in PSTN (two lines). is it possible transfer for line two an incoming call in line one?
12:41.21trixtercoppice: 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
12:41.58coppicecan you find a better free voice to add to festival?
12:41.59trixterwhy 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
12:42.11trixterthat isnt the point
12:42.17mutilatorbecause they don;t want to make THAT it easy for ppl to rip them off?
12:42.43mutilatoranyone who has made a voice spent money to do it and won't give it away?
12:42.43trixtersee 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
12:42.56coppiceit 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
12:42.58trixterno the rumors are that you can find the info free here and there
12:43.20trixternah its just an excuse
12:43.25trixterthe same old open source crap
12:43.53coppiceif you are too much of a lazy asshole to contribute, then you should expect nothing
12:44.02trixterlook 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'
12:44.14trixterand you are continuing that with your poor attitude
12:44.21viperdudehi guys
12:44.24trixterrather than have a real conversation about this you try to turn everything to your mental level
12:44.26trixterthat of a child
12:44.41mutilatoro_O
12:44.47trixterthere 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
12:44.49coppicei don't have a poor attitude. you do. you are expecting others to do stuff that only benefits you, not them
12:44.55trixtercalling names is being childish
12:45.01mutilatorlinux on the psp
12:45.02trixterwhich you proved that you do have
12:45.05trixterbecause you are  a kid
12:45.10trixterwhich is quite funny
12:45.22trixterI have a poor attitude becuase you called me an asshole?
12:45.26trixterthat makes sense
12:45.37trixterwhat I hear you saying is "I am rubber you are glue"
12:45.44coppiceI said "if" you are
12:45.44viperdudewhat overhead does Asteirsk Manager access put on a server? I want multiple logins for CRM integration
12:45.45trixteragain more childish nonsense from coppice
12:45.59mutilatorcoppice is eating right into this
12:46.05mutilatorpoor guy
12:46.09trixterbecuase he is a kid
12:46.11trixtera little child
12:46.14trixterat least mentally
12:46.24mutilatorerm no, you're just acting like a prick
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12:46.30*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
12:46.43trixterbecuase he called me an asshole I am acting like a prick?
12:46.46trixterI can see how you would get that
12:46.52mutilatoryou wouldn't bash him with useless comments if you weren't sinking to 'his level'
12:46.53trixterhe is a kid he doesnt know half of what he speaks
12:47.01trixterfestival people have REFUSED any assistance in making the voice better
12:47.09coppiceI 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
12:47.11trixteryet he statses that if someone were to submit it they would want it
12:47.18trixteryet he states they dont want it becuase its for research
12:47.28trixteryou provide nothing
12:47.30trixternothing of value anyway
12:47.36trixternothing anyone would actually want
12:47.39trixterof that I am sure
12:47.44trixteryou provide names like asshole
12:47.46trixterwho wants that?
12:47.51trixteryou make excuses
12:47.51mutilatoruse sentences man
12:47.52trixterwho wants that?
12:48.08mutilatorshow you actaully have some sort of IQ
12:48.10trixteryou are obviously an idiot that is plain to see
12:48.24trixterthe mere fact that I am typing proves that I have some sort of iq
12:48.41trixterhowever coppice has proven his worth tonight
12:48.48trixterwoo hoo he reached new heights
12:48.50mutilatorit's actually morning here..
12:48.58mutilatorbeen at work a good 2 hrs now
12:49.05trixterwell good for you
12:49.09trixterthat doesnt invalidate my statement
12:49.16trixtergiven frame of reference and all that
12:49.38mutilatoryea
12:49.43*** join/#asterisk Cresl1n (n=matt@joltid-gw.joltid.org)
12:49.44mutilator*shrug*
12:49.58trixterso coppice enlighten me what have you specifically used
12:50.06trixterer written
12:50.11trixtersince you are the all mighty contributor
12:50.28trixterplease fill us in we are dying to know
12:50.39mutilatori'm not particularly dieing to know..
12:50.48trixteryou were bragging about all that you have written that I am personally using so please tell me
12:51.02trixteryeah that is what I thought
12:51.11mutilatorwhy do you insist on continuing with this trixter?
12:51.18sivanaya
12:51.52mutilatori see no reason why it matter whether he lied or not, he doesn't have to validate anything to you or anyone else
12:52.09sivanaexcept me
12:52.14mutilatorwell ofcourse
12:52.18mutilatorbut anyway
12:52.52mutilatorman we took our total control modem banks down last night to update em
12:52.58mutilatorand it wrote some f'd up radius logs
12:53.06mutilator389hour connections and stuff
12:53.12mutilatorlike 1200 records were messed up
12:53.36mutilatorhope i don't need any of those for anything
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12:55.41fugitivomorning
12:56.06*** join/#asterisk Porks (n=nao@200.231.120.138)
12:56.22mutilatormornin
12:56.55trixteroh 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
12:57.16fugitivoyeah, cepstral is better
12:57.16trixterand free
12:57.18trixterat least for me
12:57.22trixter:)
12:57.27fugitivowhy free?
12:57.29fugitivoi paid for it
12:57.31coppicesure. its festival with different voices
12:57.35mutilatorhe's 1337
12:57.37trixterI asked if they would givem e a free license and they did
12:57.52fugitivooh :)
12:58.01fugitivocoppice: no, it's not festival
12:58.18fugitivobig difference
12:58.31trixterhe was refering to the back end code
12:58.32coppiceit is. slightly modified, but festival. look at who supplies it, and look at who wrote festival
12:58.58fugitivowell, it sounds better than festival
12:59.01fugitivonot only for the voices
12:59.24mutilatorcoppice: don't prove him wrong.. his world will fall apart..
12:59.31trixteribmTTS is free too if you use the program I wrote which if I recall is available on my page
12:59.58trixterand I think that one sounds better than cepstral
13:00.44mutilatorman
13:00.52mutilatoreven cepstral can pronounce my name correct
13:00.53trixtercourse the code I wrote is really just an asterisk integrator, nothing more
13:00.54mutilatorlast name*
13:01.04mutilatorlike.. not even humans can do that
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13:01.19trixtermutilator: what nationality is your last name?
13:01.24mutilatorgerman
13:01.37fugitivohehe
13:01.47trixterahh..  is it a vowel thing that causes the parsers to choke?
13:01.53trixteror something else?
13:01.55mutilatorgusler
13:02.09fugitivothat's easy
13:02.14mutilatormost people say guzzler
13:02.15TheCopshow you can add licenses for a g729 codec ? (I already have one, but I dont know how to add some licenses)
13:02.15trixtermost americans would say gus-ler
13:02.27mutilatoror goosler
13:02.47mutilatorit's rare someone says gus-ler
13:02.49trixtershould it be goo-sler?
13:02.55trixteris gus-ler correct?
13:03.00mutilatorya
13:03.14trixterhuh ...  anyone I talk to on a regular basis would have said it that way
13:03.28trixterhow else can you say it?  its not a ts no z would seem inappropriate
13:03.35mutilatorheh, not any techsupport/phone service of any kind
13:03.55*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)
13:03.56trixterwell 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)
13:04.15trixterso I can see why they would have a problem with german names
13:04.17mutilatorwell this is talking to like
13:04.18mutilatorteh telco
13:04.23trixterpeople always toss extra letters into my name
13:04.23mutilatorand electric co
13:04.44trixterboth 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
13:04.56trixterone person when I was a kid even told my mom she misspelled it
13:05.02trixterand insisted on correcting it for her
13:05.40mutilatorheh
13:06.28fugitivowell, english natives have big problems with latin names
13:06.42trixtermy name is scottish more or less
13:06.54fugitivoi'd like to hear how you say "fugitivo" ;)
13:07.00mutilatori guess back in the day mine used to be gossler
13:07.06mutilatorand somehow turned into gusler
13:07.17trixteryou 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
13:07.42trixterours changed when immigration in america couldnt spell
13:08.10trixtermutilator: 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
13:08.26Porksfugitivo... is it pt_BR?
13:08.58fugitivoPorks: pt_BR, it_IT, es_ES, es_MX, and a lot more
13:09.13Porksah! ;)
13:09.36coppicefugitivo: well I guess its nothing like fugitive-o.
13:09.39trixterfugitivo: I live in mexifornia there are a lot of latin names particularly es_mx so people here generally can handle latin names :P
13:10.08trixterfugi-tivo  the next generation pvr!
13:10.19fugitivoenglish natives like to say the last O like OU
13:10.33fugitivoso in spanish it'll sound something like fujitivouu
13:10.51fugitivothe g doesn't sound like the english g
13:10.56trixtermost of that is where they put the accent
13:11.13mutilatorfirst time he's worked graveyard 12-8 shift
13:11.27trixterthe same problem exists with japanese
13:11.36mutilatorand i'm going to be upgrading the sql server tomorrowso the site & radius will be down for 5-10 minutes
13:11.46fugitivojapanese and spanish have a similar phonetic
13:11.47mutilatorso i'm not going to tell him i'm doin it and not answer my phone.. see what he does
13:11.53trixteralthough 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)
13:11.54mutilatormuahaha
13:12.04trixterthe double vowel means you say that sound slightly longer than a single vowel
13:12.08trixtera concept english doesnt have
13:13.07trixterthere 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-FenderHowever english has a seperate future tense which japanese doesn't.  See who's got the eye on the future? ;)
13:13.17coppicefugitivo: 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? :-)
13:13.48fugitivocoppice: :)
13:13.51[TK]D-Fendercoppice : The "p" is silent.... like in swimming. :D
13:14.28fugitivothe nice thing about spanish, is that your say what you read, if you know how the letters sound, you can say any word
13:14.50coppicefugitivo: its the same with chinese
13:15.06fugitivochinese is a pain in the ass :)
13:15.53[TK]D-FenderMuch like Inuit... too many damn vowel / W sounds in a row :)
13:15.53coppiceexcept if you speak cantonese, you'll say something completely different than if you speak mandarin :-)
13:15.53trixterenglish 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
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13:16.18geviousHi Guys
13:16.26fugitivocoppice: doesn't chinese have different meanings for the same sound with different tone?
13:16.35coppicethat's one of the three key reasons why english is so successful
13:16.37geviousHas anyone used gnudialer with Asterisk 1.2.1
13:17.04coppicefugitivo: if the tone is different, its not really the same sound, is it?
13:17.30geviousSorry I know this is an asterisk channel, but I am having some difficulty with Gnudialer and no one is ever on #gnudialer
13:17.31fugitivocoppice: not the sound, but it's the same word
13:17.46fugitivogevious: now this is #languages
13:18.10coppiceah, what what does the same word mean? I can hear it sounds different
13:18.19geviousI see that
13:18.29trixterpersonally 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
13:18.41fugitivocoppice: in spanish if you say the same word with different tones, it means the same thing
13:19.03geviousAnyone know where I can get Gnudialer help?
13:19.09geviousBesides gnudialer.org
13:19.21geviousAbsolutely no doc on gnudialer anywhere
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13:19.53trixterfugitivo: 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
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13:20.08coppice3 reasons for english dominance:
13:20.10coppiceThe British spread it around their empire
13:20.11coppiceAmerica speaks it, and its dominant today
13:20.13coppiceit flexible, and people munge it around to taste
13:20.13trixtergevious: what speciifcally are you having problems with?  if you never say what you specific problem is no one will ever help you
13:20.14coppiceI think if you take any one of those three away it wouldn't be nearly as widespread as it is today
13:20.58geviousI have got it compiled, but keep getting "terminate called after throwing an instance of 'xFileOpenError'" when I execute gnudialer
13:21.12fugitivotrixter: in chinese, you can say for example "a" in different tones, and it'll have different meanings
13:21.22fugitivothat's why chinese is so hard to learn
13:21.48trixterreading it isnt that hard though
13:21.51mutilatori say just force english on everyone
13:21.59trixterI have more problems reading arabic than chinese or japanese
13:22.13coppicefugitivo: that isn't actually a hard part of learning mandarin, but its a rather hard part of learning cantonese
13:22.16geviousI have traced it down to the addGlobalSettings function in queues.h but am stuck now
13:22.17trixternot sure if its the right to left part or if its the fact that the characters all look almost identical
13:22.20fugitivotrixter: but can you speak it correctly?
13:22.33trixterspeak which?
13:22.37fugitivochinese
13:22.40trixterwhich?
13:22.46fugitivocantonese
13:22.46mutilatornew
13:22.55trixterno I never learned to speak cantonese
13:23.15trixterI relaly only learned some chineese becuase of my desire to learn japanese
13:23.22fugitivocoppice: mandarin is different in that aspect?
13:23.25trixtersome of the characters are the same, some arent
13:23.32trixterso it gave me a little boost
13:24.18coppicefugitivo: 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
13:24.28trixtergevious: 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
13:25.39fugitivocoppice: 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"
13:25.43mutilator*NINJA VANISH*
13:25.46trixterare 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
13:25.47mutilatorpewf
13:25.50RemosiI 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?
13:26.10coppicefugitivo: I have no idea what that means :-)
13:26.23trixterfugitivo: 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
13:26.36trixterthey will always speak it with some accent if they learned a specific language after that age
13:26.43trixterafter 13 makes it even harder
13:27.03trixterit could be that is what your teacher was refering to
13:27.22fugitivoi think she was refering to the tones
13:27.47coppicenot 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
13:27.48fugitivoin spanish, you say exactly what it's written, we don't have tones
13:28.34mutilatoraccents are cool
13:28.45mutilatormove to the south for a year and then move back north, everyone says ya talk funny
13:28.48coppicewe have tones in english. they provide punctuation. chinese can't do that, so they actually use words for punctuation
13:28.49trixteraccording 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 ...
13:28.51mutilatorand ya don't even realize it
13:29.23trixterI 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
13:29.43fugitivotrixter: lol
13:29.57[Wiebel]trixter: well that's something you want to hide ;)
13:30.19mutilatorthe cowboy hat and boots and tight pants probly gave it away anyway tho
13:30.21coppicethe speed with which a 4 or 5 year old can pick up a new language is amazing
13:30.35trixteruntil 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
13:30.35trixtere that
13:31.00trixtermutilator: hey I livedi n texas in the 80s and parachute pants were popular there!
13:31.21mutilatorstill had the bowboy hat and boots tho i see
13:31.22mutilator:P
13:31.23fugitivoparachute pants, hehe
13:31.25trixteralthough mid 80s roper boots were all the rage
13:31.25mutilatorcowboy*
13:31.41trixteroh god I haded those boots
13:31.47mutilatordamnit my coffee is cold
13:31.56trixterand the hats I never really liked, late 80s I wore a flourecent green fedora
13:31.56mutilatornow i have to walk outside in the freezing tundra to get more
13:35.34docelm0w00t!   Good MORNING #ASTERISK!!!!
13:35.35mutilatoromfg hands are ice!
13:36.15coppiceice? but its 22C :-)
13:36.18[Wiebel]it's like 2:36 here ;)
13:36.19[Wiebel]PM
13:36.32docelm050F here
13:36.39[Wiebel]although I wish i came out of bed just now :)
13:36.44[Wiebel]s/i/I/
13:37.10fugitivo18C Clear Skies, 10:37am
13:37.19mutilatorpfft
13:37.23mutilator-15C here
13:37.28fugitivo-15C????
13:37.32fugitivowhere are you?
13:37.37mutilatornorthern michgan
13:37.41coppice22C, can't see the sky as its 9:37PM
13:37.43mutilatorwind blowin like crazy too
13:37.48wasimi went to school in houghton
13:37.52mutilatorme too
13:38.00mutilatorwell not highschool
13:38.03fugitivocoppice: i don't see the sky neither, weather information in my kontact :)
13:38.06mutilatoruniv
13:38.09wasimcollege, but it still gives me the shivers
13:38.14mutilatoryeh
13:38.16wasimbloody cold
13:38.20mutilatorand windy as fuk
13:38.29mutilatorspecially by the eerc
13:38.35wasimyeah, if you stopped walking the wind would slide you back on the ice
13:39.00mutilatorah the memories
13:39.14wasimand 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
13:39.22mutilatoryea
13:39.33mutilatorthe snow drifts hanging off the edge of the car 2 feet or more
13:39.52wasimbrrr ....
13:39.55wasimstop please ...
13:39.59mutilatorbreak it off and what was showing of the car dissapears
13:40.36coppicewasim: 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
13:41.01wasimcoppice: and to top it off, i had to take english as a foreign language exam ...
13:41.04coppicewasim: whats the lowest temp where you live?
13:41.19wasimcoppice: it'll drop here to about 3C
13:41.35coppicewasim: that must have been a piece of cake for you :-\
13:41.50wasimcoppice: boring as hell ... nice spanish girl though
13:42.00coppicewasim: its gets to 4 or 5 here, but only for a week or two
13:42.22coppicewasim: how did you end up with english as a foreign language
13:42.41mutilatordegree in anything wasim?
13:42.46wasimcoppice: inspite of getting 673 in my toefl? no clue, some stupid university crap
13:43.00wasimmutilator: no, i got my butt out of there and to more temperate climate (iowa)
13:43.08mutilatorhow long ago were ya there?
13:43.14wasimmutilator: 1991
13:43.18mutilatorheh
13:43.53coppicein 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
13:44.29coppicewasim: 1991 was the last time i spent winter in a cold place :-)
13:44.51wasimits the wind chill in houghton that killed you ...
13:45.02wasim-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)
13:53.23kimosabecan some one lead me to a good how to on fedora core and asterisk set up
13:53.30kimosabethanks room
13:55.52*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
13:55.56kimosabeis asterisk running on any version of freebsd directlly from ports
13:58.23[TK]D-FenderFC3 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.
13:58.35kimosabeoki
13:59.01viperdudewhat overhead does Asteirsk Manager access put on a server? I want multiple logins for CRM integration
13:59.16coppiceFC4 has a fussier compiler, but there shouldn't be issues with that any more
13:59.35iCEBrkrI dunno. FC4 has partition and boot manager issues as well.
13:59.42iCEBrkr....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)
14:01.13SpaceBassmorning
14:01.20iCEBrkrSpaceBass: hey
14:01.23SpaceBassanyone using a cisco phone with more than one sip server?
14:01.43SpaceBass(and by sip I mean asterisk)
14:01.59SpaceBassiCEBrkr, hey how are yoy
14:02.02SpaceBasss/yoy/you
14:02.10iCEBrkrx.x
14:02.14iCEBrkr3hrs of sleep
14:02.29iCEBrkrand hoping these PRIs get turned-up today
14:02.33SpaceBasssounds like you are well rested...
14:03.04iCEBrkrhaha
14:03.07[TK]D-FenderiCEBrkr : Got confirmation that they didn't finish thejob?
14:03.16iCEBrkr[TK]D-Fender: Actually, yes
14:03.25iCEBrkrI gotta make a few calls today to figure out WTF is going on
14:03.55SpaceBassAnyone using dring on a zaptel line?
14:03.59[TK]D-Fenderheh, I *knew* there was no way the card was at fault :)
14:04.45iCEBrkrhaha
14:05.26[TK]D-FenderCan't wait until their analog card comes out..... mmmm
14:05.50[TK]D-FenderBrilliant design
14:07.37*** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
14:07.42[TK]D-FenderAnd 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)
14:09.41SpaceBassAnyone using dring on a zaptel line?
14:09.44coppicewhich analogue card?
14:09.57iCEBrkrha
14:09.59[TK]D-Fendercoppice : http://www.telephonyware.com/telephonyware/sangoma_aa.html
14:10.14coppicepeople tell me they are nice
14:10.59[TK]D-FenderSpec's are saying all theright things...
14:12.02coppicei think the backplane thing looks clunky. they apparently work well, though
14:12.46[TK]D-FenderIt does have some potential drawbacks.  If they are available in multiple lengths that'd help
14:13.17[TK]D-FenderOr if they modularize it.
14:13.32coppiceI 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-Fendercoppice : 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-FenderGotta 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-FenderSince 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)
14:17.41fgravatois there any way of disabling transfer # in asterisk
14:17.48fgravatobesides features.conf
14:19.11[TK]D-FenderDon't use tT in Dial commands?
14:19.30[TK]D-FenderI never do.....
14:21.49fgravatowhat 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-Fenderno 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)
14:42.00asteriskmonkeymorning
14:43.02oejasteriskmonkey: Are you going to code a /proc interface or was it just something for the wishlist?
14:43.12IkarusBleah, 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)
14:46.06heroinei 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)
14:48.11tzafrir_laptopsysfs is nicer to work with, although 2.6-specific
14:48.12brad_msswheroine: you may have to supply your voicemail context   like   112@default
14:48.33heroinehmm ..
14:48.46Delvari used, exten => 100,1,voiceMailMain(${callerid(number)}@default)
14:48.47heroinei think i see where is the problem , thx for your tip :)
14:49.00tzafrir_laptopIkarus, what version of kernel, zaptel and bristuff?
14:49.02[TK]D-FenderYou may also want to be sure of exactly what you are passing to VoiceMailMain including the validity of the mailbox # itself
14:49.28heroinewe use the mysql backend there. I think i should first look that point
14:50.16Ikarustzafrir_laptop: I think I found it out myself
14:50.46asteriskmonkeyoej
14:51.09asteriskmonkeyoej: 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
14:52.36oejasteriskmonkey: 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...
14:53.14asteriskmonkeyoej: i thought it would be intenal code that outputs to a proc :P
14:53.46RoyKzoa: ping
14:54.28Ikarustzafrir_laptop: ah well, it seems I'll have to upgrade to Debian unstable, or compile asterisk myself
14:55.14IkarusAnd that second option is most likely, I heard about some issues with bristuff and Asterisk 1.2
14:55.18zoapong
14:56.39coppicezoa, man of jitters
14:57.40tzafrir_laptopIkarus, I have debs of asterisk 1.2.0 . Asterisk Unstable is currently without bristuff
14:58.13zoahey ho coppice, man of faxes!
14:58.18tzafrir_laptopIn fact, the new debs I hope to soon upload will probably be without bristuff, unless I see a sign of life from kapejod
14:58.39coppicezoa: its all lies
14:58.58Ikarustzafrir_laptop: bleah, I can't live without a decent ISDN layer (capi isn't decent in my tests)
14:59.28coppicethe wonderful thing with ISDN layers is there are so many to choose from :-)
14:59.42Ikaruscoppice: uhuh
14:59.49tzafrir_laptopIkarus, in what ways is zapbri better than others?
14:59.51Ikaruscoppice: well, vISDN is the only alternative
15:00.09zoamISDN ?
15:00.10Ikarustzafrir_laptop: it has almost no echo problems according to a friend of mine
15:00.11brad_msswi ported the asterisk patch to 1.2.1 from bristuff ... but the libpri patch is a beotch, haven't had time
15:00.49Ikarustzafrir_laptop: but vISDN comes with the same advantages and more, but it is still beta
15:00.50coppiceIkarus: look around. everyone seem to have their own ISDN layer, though many are based on the same original
15:01.05Chotairebrad: bristuff for 1.2.1 (incl. new chan_capi) to be released within 2-3 days.
15:01.27brad_msswChotaire: good to know
15:01.39tzafrir_laptopChotaire, have you been in touch with the Kapejod himself? when?
15:01.43Chotairetoday.
15:01.57Chotairehe promised me he'll make sure he gets finished before xmas.
15:02.14Ikaruscoppice: well, I have seen mISDN, vISDN, bristuff and CAPI
15:02.21coppicedid he say which christmas?
15:02.25Chotairehehe
15:02.41Chotairecoppice: 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)
15:03.07Ikaruscoppice: and I just need one that supports overlap dialing and works well
15:03.18Ikaruseven if that means running an older version of Asterisk
15:03.41Chotairecoppice: I will personally keep bothering him ;) I can't update to 1.2.* until he's finished ;)
15:04.10asteriskmonkeyanyone know why asterisk would hang up on someone when it is supporst to go to vm?
15:04.12IkarusAlso DID is needed
15:05.11Chotaireand 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.
15:05.34SpaceBassasteriskmonkey maybe VM is in a different context and its not being passed correctly
15:06.41Chotaire1.2.* 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)
15:07.40Chotaireanyway, 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.
15:08.44asteriskmonkeySpaceBass: i have it like this exnt@context is that correct or has somthing changed since 1.0.9 on that :D
15:09.18Chotaireasteriskmonkey: what does the console say when its being disconnected?
15:09.24SpaceBassasteriskmonkey I'm not sure... never seen it coded like that
15:09.30*** join/#asterisk MattB2 (n=Mattb@mail.tricycleinc.com)
15:09.34MattB2hi all
15:09.47MattB2possibly silly question - any idea why PRI NO DEBUG doesn't work in 1.2.1
15:09.57asteriskmonkeySpaceBass: good idea, never thought to look at the console ! doh
15:10.09Chotairei'm not spacebass, but that's ok.
15:10.10MattB2Ahhhhhh
15:10.15MattB2it was a silly question
15:10.17MattB2ignore me ;)
15:10.19MattB2oh you are!
15:10.22asteriskmonkeysorry chotaire: P
15:10.28Chotaireasteriskmonkey: asterisk -vvvvvvvvvvvvvvvvvvc and just paste what happens.
15:10.29asteriskmonkeymy eyes are still not open
15:11.37Seldon1975anyone know where I can get a GSM codec,converter or player for windows?
15:11.45Chotairesox
15:11.50Chotaireseldon1975: there is sox for windows.
15:11.51Seldon1975i cant play the Asterisk sounds on my XP machine
15:11.55Chotaireyou can convert .gsm files to .wav
15:11.56asteriskmonkeythis is what i got
15:11.56Seldon1975aha
15:11.56asteriskmonkeyExecuting VoiceMail("Zap/13-1", "debitact201@debitact") in new stack
15:11.56asteriskmonkey<PROTECTED>
15:12.04Seldon1975thanks Chotaire
15:12.17Chotaireseldon1975: check the sox website, there is a win32 binary which will make you happy.
15:12.27Seldon1975nice one
15:12.43*** join/#asterisk ManxPower (n=ewieling@200.sub-70-197-9.myvzw.com)
15:13.18asteriskmonkeyChotaire: did you see what i pasteds :)
15:13.42IkarusAh well, visdn doesn't even compile
15:13.50[TK]D-Fenderasteriskmonkey : that mailbox doesn't look too legit
15:14.08Chotaireasteriskmonkey: yes... I seen it.. never seen a mailbox name "Zap/13.-1" ;)
15:14.10[TK]D-Fenderunless you are doing wierd names...
15:14.24asteriskmonkey[TK]D-Fender: you cant use characters in voicemail address? worked before
15:14.36Chotaireoops...
15:14.40[TK]D-FenderYou can, but does it exist as written?
15:14.42Chotairewell, you know what I mean ;)
15:15.03*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)
15:15.05asteriskmonkeyah 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)
15:16.18asteriskmonkeythis is the correct way to dial vm in 1.2 right? Voicemail(ext@context)
15:17.31ChotaireVoiceMail([s|u|b]extension[@context][&extension[@context]][...])
15:17.57Chotairewhat about this... == Auto fallthrough, channel 'Zap/13-1' status is 'UNKNOWN'
15:18.07asteriskmonkeyim lost on that
15:18.16asteriskmonkeyanyone know what thats about?
15:20.14*** join/#asterisk jluk (n=njon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk)
15:20.24ManxPowerNo, Voicemail(vmbox@vmcontext)
15:20.30Chotairewell...
15:20.34Chotairetake a look above.
15:20.51ManxPowerextensions are not the same as a mailbox.  You can make them the same, but that's cosmetic.
15:20.59Chotaireexactly.
15:21.12ManxPowerand the @vmcontext is the voicemail.conf context, not the extensions.conf context.
15:21.20Chotaireyou must have mailbox number/name debitact201 in voicemail.conf
15:21.38Chotaireand @vmcontext you only need if you have multiple vmb systems.
15:22.03Chotairee.g. multiple virtual vmb systems with same box numbers/names.
15:22.11*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
15:22.48Chotaireso... VoiceMail(201) will pretty much do the job.
15:22.53Chotairei hope I got this right, manxpower?
15:23.12Chotaireall you need is a vmb "201" in voicemail.conf
15:23.14asteriskmonkeyManxPower: i know there is a diffent context in the voicemail.conf
15:23.17asteriskmonkeyits bizzare
15:23.35asteriskmonkeythe name of the box is the mailbox ..
15:24.18asteriskmonkeyso 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)
15:24.18asteriskmonkeyexten => 201,n,Voicemail(debitact201@debitact)
15:24.38darwin_35in a static dial plan in a macro how would you make the macro read the protocal used ?
15:24.45ManxPowerasteriskgeeks, show us the ACTUAL line from extensions.conf
15:24.47*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
15:25.09SpaceBassAnyone using dring on a zaptel line?
15:25.22SpaceBassanyone using a cisco phone with more than one sip server?
15:25.38darwin_35Dial(protocal/${arg1}?
15:26.10*** join/#asterisk bkw_ (n=bkw_@ppp-70-243-94-102.dsl.tulsok.swbell.net)
15:27.02Chotairedarwin: maybe Dial(protocal/${arg1})? ;)
15:28.44Seldon1975Hi, can someone suggest how to set up a dialplan to playback(all-ougoing-lines-unavailable) when al my trunks are busy?
15:29.00darwin_35nope does not work
15:29.00Seldon1975i mean we have t for timeout and i for invalid; but no b for busy?
15:29.03darwin_35grrr
15:29.34[TK]D-FenderSeldon1975 : What does "busy" mean when you're expecting input?  Busy is the result of DIALING.
15:29.55Seldon1975yes, when all my Zap lines 18-23 are utilized already
15:30.13Seldon1975but how can I determine this in my dialplan?
15:30.28[TK]D-FenderSeldon1975 : the dialresult would be "congestion"
15:30.33Seldon1975ie: an internal user tries to dial out
15:30.38Seldon1975oh
15:30.39[TK]D-FenderSEE ABOVE
15:30.42*** join/#asterisk alexissoft (n=alexis@ws1.rtcn.be)
15:30.42alexissofthi
15:30.55darwin_35the rate things change inthe dial plan its hard to keep up
15:32.18alexissoftcan i use a standard winmodem as FXO ?
15:32.43darwin_35only the  intel 579 I think it was
15:32.58Seldon1975how do I respond to a specific DIALRESULT from a Dial() command?
15:33.06alexissoftand i can't use any standard modem
15:33.19darwin_35nope
15:33.30coppicewhat is a standard winmodem? they don't follow any standards
15:33.59alexissofthmm ... yes :)
15:34.05[TK]D-FenderSeldon1975 : 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)
15:34.13alexissofti think it's the same thing for a serial modem
15:34.17Seldon1975D-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)
15:36.18coppicealexissoft: 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
15:36.39alexissoftok, i see
15:37.04*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
15:38.16asteriskmonkeyChotaire: i do have a mailbox called that in my voicemail.conf under that context and it dont work for somereason
15:38.23alexissoftcongratulations for asterisk, it's a very very very very very very good product !!
15:38.30alexissofti love it :)
15:39.02Kattymorning lads.
15:40.12mog_workthanks alexissoft
15:40.17alexissoft:)
15:40.21mog_workeveryone appreciates it
15:40.43Kattywell don't everyone say g'morning at one.
15:40.46Kattyonce, too.
15:41.41*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
15:41.45[TK]D-FenderGOOD MORNING SUNSHINE!
15:42.42asteriskmonkeymorning
15:43.11*** join/#asterisk Seldon19751 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com)
15:45.22Kattyasteriskmonkey: what do you want?
15:45.25Kattyasteriskmonkey: get off my version.
15:45.46asteriskmonkeylol
15:46.07asteriskmonkeyi use mirc was looking for names of other irc clients i could possibly use
15:46.32Kattymirc :<<<<<<<<<<<<<<<<<<<<<<<<<
15:46.37Kattymirc is /horror/
15:46.46Kattyand it's on WINDOWS
15:46.49Kattywhich means you can't screen it
15:46.54Kattyhorror horror horror
15:46.56asteriskmonkeylol
15:47.26asteriskmonkeywell whats a good cli one for bsd and ill go compile it
15:47.34alexissoftirssi :)
15:47.37alexissoftor weechat
15:47.45*** join/#asterisk backblue (n=moo@82.102.1.42)
15:47.55asteriskmonkeyand that wont start building xwindows as a part of dependencies ? :)
15:47.57*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:47.57alexissoftasteriskgeeks, a graphic IRC client or a textual one ?
15:48.07alexissoftirssi & weechat are text-based
15:48.12asteriskmonkeysweet
15:48.18Kattyyay for irssi
15:48.22*** join/#asterisk oej_ (n=oej@apollo.webway.se)
15:48.26backbluehi, does anyone knows (in cisco.com) where i can find the sip configuration manual for cisco 7970 & cisco 7940??
15:48.28shido6.
15:48.58[Wiebel]backblue: it's not included with the firmware?
15:49.25backblue[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
15:50.54backbluecan you send me?
15:51.19backblueanyway, i need the installation manual anyway
15:52.14[Wiebel]http://www.wiebel.nl/asterisk/SEPexample.cnf
15:54.32Kattyi could use support.
15:54.40Kattyhow do i kill an id after ps auxing?
15:54.52Kattykill $id?
15:54.55mkrufkyyes
15:55.01Kattykthx
15:55.02mkrufkyand if that doesnt work,
15:55.05mkrufkykill -9 $id
15:56.56Kattyxmms borked.
15:56.58Porkshi?
15:57.05Kattyasl.
15:57.08Katty</sarcasm>
15:57.12Porksi need some help :|
15:57.16Kattyme too.
15:57.23Kattybut i need a therapist.
15:57.23Porkshauehae
15:57.27PorksI have two lines of PSTN conected at two X100P cards... is possible transfer a call from line #1 to line #2?
15:57.38PorksI 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
15:57.58Porksis it magic? or is it possible?
15:57.59Porks:D
15:58.43iDunnoI could do with some magic
15:58.48*** join/#asterisk kpettit (n=keith@69.15.174.114)
15:59.16Porksme too... but not with asterisk
16:00.02Porkslike.. the PBX... don't have many lines come in... and people call in only one number?
16:02.26SpaceBassanyone using a cisco phone with more than one * box?
16:03.20iCEBrkr-.- zzzZZZZZ
16:03.24Katty[TK]D-Fender: do you want a happy mew years card?
16:03.38Katty[TK]D-Fender: as part of my mass mailings.
16:04.39tzafrir_laptopPorks, this best be done at your provider
16:04.59tzafrir_laptop"leading number" or something similar.
16:05.22tzafrir_laptopOtherwise I can't see how this can be implemented
16:05.25iCEBrkrmew years. lol
16:05.51iCEBrkr[TK]D-Fender: yea, don't feel special.. It's a spammation of people she knows..
16:05.52darwin_35ICEBRKR same happened on the new box I built yesterdy
16:05.59*** join/#asterisk Porks (n=nao@200.231.120.138)
16:06.08iCEBrkr??
16:06.08Porksargh...
16:06.15darwin_35its not linking to the db
16:06.18KattyiCEBrkr: yes.
16:06.21KattyiCEBrkr: all my friends get cards.
16:06.30tzafrir_laptopPorks, this best be done at your provider
16:06.35tzafrir_laptop"leading number" or something similar.
16:06.36iCEBrkrdarwin_35: Follow the freetds how-to
16:06.36tzafrir_laptopOtherwise I can't see how this can be implemented
16:06.47Kattyfile[desk]: which means you, pretty boy.
16:06.55darwin_35whats the url
16:07.07iCEBrkrdarwin_35: google
16:07.30*** join/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com)
16:08.38Porkstzafrir_laptop uhmm... is the some for pbx? (the 'normal' way.. without asterisk or other stuff)
16:08.57Porksthe telephon company provides this
16:10.41iCEBrkr3hrs of sleep.. If that.  I'm not sure how I'm going to make it through the day
16:12.09darwin_35nothing for myodbc
16:12.18darwin_35this sucks
16:12.37darwin_35why does it work fine on 4 other boxes but not this one
16:12.45iCEBrkrdarwin_35: If I got it working, you can too
16:13.26darwin_35evrythign matches box for box
16:13.29darwin_353 for 3
16:13.34darwin_35#for #
16:14.09Dandanhey all
16:14.10Dandan:)
16:14.19trixtersince 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
16:14.20trixtero not feel that I should dictate to someone else how they should run their business or talk to their friends or ...
16:14.25Dandangot a problem, just started playing around with x100p:
16:14.33Dandanwhen I call out
16:14.40Dandanthrough zap/1
16:14.41trixteror 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
16:14.47Dandanit is being automatically picked up
16:14.59Dandanand i end up talking to zap/1 which is myself?!?
16:15.16smitherI'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?
16:16.27iCEBrkrsmither: Where's your POTS line hooked up? To your astrisk box, right?
16:16.29iCEBrkrDandan: ??
16:16.44DandaniCEBrkr: hm, hold on lemmie paste
16:16.51iCEBrkrNOooooooooo
16:16.52smitherYes - through an x100p.
16:16.54iCEBrkrDandan: pastebin.ca
16:17.04Dandan*paste into pastebin :)
16:17.07iCEBrkrlol
16:17.12Dandansorry that's what I meant
16:17.45*** join/#asterisk fiber0pti (n=John@invinine.com)
16:17.52iCEBrkrsmither: 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.
16:17.56iCEBrkrsmither: what's it not doing?
16:18.18fiber0ptiAnyone 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)
16:18.55smitherecebrkr: 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?
16:19.08iCEBrkrNot that I know of.
16:19.08smithersorry - iCEBrkr.
16:19.09SpaceBassfiber0pti I have that problem with my wifi phone but not with a specific provider
16:19.27SpaceBasssmither there is a way to delay pickup
16:19.30iCEBrkrsmither: If you're gonna use asterisk use it.. Don't half-ass it.
16:19.35iCEBrkr:)
16:19.37Dandanhttp://pastebin.com/471784
16:19.46Dandanthat is weeeeird :/
16:20.34iCEBrkrDandan: umm, what's wrong with that?
16:20.51Dandanit seems like I am talking to myself
16:20.54Dandan:)
16:20.56iCEBrkrYou're not.
16:21.09Dandanwell it is not ringing, the phone call is established
16:21.17iCEBrkrCall progress
16:21.56Dandanyeah, but nothing happens after that last line...
16:21.56iCEBrkrand you shouldn't have to put any parameters on it.. drop the |60|r
16:22.42smitherDon'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?
16:23.08iCEBrkrI'm trying to figure out why the heck you'd wanna wait 4rings for asterisk to answer
16:23.50DandaniCEBrkr: it seems like it takes the phone line off hook but doesn't dial any digits
16:24.00DandanI receive: the number you have dialed is incomplete
16:24.13smitheriCEBrkr - to give me a chance to pick up a regular POTS phone before Ast takes over, again like you do with an answering machine.
16:24.36iCEBrkrsmither: umm, make Asterisk answer the line, and ring your ATA
16:24.51iCEBrkrif you don't answer, have Asterisk transfer to VoiceMail()
16:25.09iCEBrkrya know.. Like a real PBX.. Afterall, Asterisk is a PBX
16:25.33iCEBrkrDandan: I'm assuming you're trying to 7digit dial in that areacode?
16:25.40Dandanyup
16:26.01iCEBrkrDandan: Does it require a 1 first?
16:26.08[TK]D-FenderDandan : Don't bother with the prefix in the Extens line.  Just do a length check and add it if necessary
16:26.25smitherThat 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 :-).
16:26.40iCEBrkrsmither: That's what I meant about doing it half-assed.. :D
16:26.57iCEBrkrsmither: It's frightening at first, I know.  I was the same way with Asterisk when I put it on my home land-line.
16:27.37iCEBrkrsmither: I use Asterisk as a glorified answering machine myself.
16:27.41Dandanso what is your dial-out with x100p?
16:28.09iCEBrkrDandan: I think it's more like you're not passing the correct number of digits out.
16:28.27Dandanif i use the pass through it works
16:29.03iCEBrkrI'm actually kinda confused.
16:29.13iCEBrkrCuz ${EXTEN:3} is gonna strip off the areacode.
16:29.14*** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net)
16:29.26Dandangonna strip 724
16:29.28Dandanas prefix
16:29.29smitherOK, 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?
16:29.29iCEBrkrright
16:29.54iCEBrkrsmither: It'll do that.
16:30.02iCEBrkrsmither: callprogress=yes
16:30.21*** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net)
16:30.23iCEBrkrerr something like that.  It's what Dandan was confused about when Zap/1-1 Answered :P
16:30.36iCEBrkrDandan: So is 724 a prefix or AC?
16:30.43smitheriCEBrkr: That looks promising - which .conf is that put in?
16:30.53Dandanprefix 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"
16:31.38iCEBrkrsmither: zapata.conf
16:32.06smitheriCEBrkr: Thanks!  I'm off to play with Asterisk.
16:32.11iCEBrkrDandan: oh, like dialing 9.. ok
16:32.16Dandanyeah
16:32.40Dandanthere is something wrong...
16:32.42*** join/#asterisk SwK[Work] (n=SwK@64.89.118.139)
16:32.45darwin_35<PROTECTED>
16:32.46Dandan* can't pick up the line properly
16:33.01*** join/#asterisk Jick (n=Jick@209-83-240-53-static.dsl.oplink.net)
16:33.05darwin_35the msql ver ar ediff
16:33.08iCEBrkrDandan: 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)
16:33.16darwin_35.11 vvers .15
16:33.18DandaniCEBrkr: heh :)
16:33.30*** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net)
16:33.32t0kehi
16:33.44Dandanit is not even funny :)
16:33.47iCEBrkrhaha
16:34.16t0keanyone know if is possible that IAX calls use enblock and no overloap method for send digits?
16:34.25iCEBrkrdarwin_35: This is why I don't run that bleeding-edge shit.
16:34.41t0keit does slower calling via PRI
16:35.58DandanDec 20 11:33:56 WARNING[813]: chan_zap.c:6315 handle_init_event: Detected alarm on channel 1: Red Alarm
16:36.02Dandannow what?
16:36.07*** join/#asterisk gniretar_work (n=mark@152.160.35.1)
16:36.29iCEBrkrDandan: your phone line is unplugged.
16:36.34iCEBrkrDandan: go pay your phone bill! Geesh
16:36.36iCEBrkr:D
16:36.40gniretar_workhi all
16:37.06gniretar_workhey, i'm on SuSe 10.0, what init script distro would u guys recommend i use?
16:37.08DandaniCEBrkr: oh right, I am checking the cable :)
16:37.51t0keanyone know how to activate enblock dialing mode in Asterisk?
16:38.35*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
16:38.40TheCopsHi
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)
16:39.51TheCopsI 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)
16:40.09TheCopsThis is a translations issue ?
16:42.25Seldon19751I downloaded and installed postgres; compiled asterisk and I started getting console errors when I made outgoing calls - has anyone else experienced this?
16:42.46Seldon19751it seems that installing pgsql and recompiling * mucked it up
16:43.20tzafrir_laptopSeldon19751, doesn't your distro have pgsql?
16:43.36Seldon19751it appears not, when I type 'pgsql' I get command not found
16:43.43Seldon19751which is odd, since it's CentOS
16:43.45tzafrir_laptopwhat distro?
16:43.54Seldon19751CentOS
16:43.59Seldon197512.6.1
16:44.03tzafrir_laptopconsidered using apt/yum to install it?
16:44.20Seldon19751tzafrir: could you tell me how?
16:44.32tzafrir_laptopMost packages are not installed by default, because they're so easy to install with the package manager
16:44.51DandanDec 20 11:44:29 NOTICE[862]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
16:44.55tzafrir_laptopI believe CentOS uses yum . try: yum install postgresql-devel
16:44.56DandaniCEBrkr: what is that?
16:45.12backbluetzafrir_laptop: yes it uses
16:45.18Seldon19751tzafrir: thanks
16:45.25Seldon19751tzafrir: ill try it
16:45.56Seldon19751tzafrir: if I get the same issue; how could I uninstall pgsql
16:46.01tzafrir_laptopSeldon19751, BTW: 2.6.1 is probably not the version of CentOS. CentOS is typically either 3.something or 4.something.
16:46.28Seldon19751tzafrir: oh ok thx
16:46.29*** join/#asterisk Igbothom_III (n=HiltonT@203-206-170-99.perm.iinet.net.au) [NETSPLIT VICTIM]
16:46.47tzafrir_laptopI'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)
16:47.08backbluej php
16:47.18tzafrir_laptopanyway, pastebin the exact error message
16:47.20Seldon19751tzafrir: righto.  Im doing this because I want to get call logging to the DB going
16:47.58Seldon19751tzafrir: yum seems to have installed pgsql-devel ok - should I make install asterisk now?
16:48.50Seldon19751tzafrir: can you tell me how to check that it's installed?
16:48.52tzafrir_laptopSeldon19751, AFAIK there are no decent CentOS asterisk packages.
16:49.12tzafrir_laptoprpm -q pgsql-devel
16:49.23Seldon19751I realise now I shouldn't have; but I'm using an Asterisk@home distro
16:49.29tzafrir_laptoprpm -V pgsql-devel
16:50.06tzafrir_laptopThe latter should give an empty output
16:50.21Seldon19751it 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)
16:53.48darwin_35ICE I am stumped I have fallowed evrey thing
16:54.03darwin_35its driving me insane
16:54.56*** join/#asterisk BladeRunner05 (n=feelme@adsl-61-91.38-151.net24.it)
16:56.28[TK]D-Fenderdarwin35 : Whats the problem?
16:56.54*** join/#asterisk waddy (n=waddy@83.218.4.231)
16:58.19waddyI 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
16:58.44waddyalso things like ncurses etc
16:58.52Beirdowaddy: hate to be rude, but why are you asking that here?
16:58.55[TK]D-FenderPerhaps you should try #RHEL
16:59.13waddycause i want to get my isdn card working with asterisk
16:59.23waddydead simple with rh9
16:59.56waddyjust thought there may be a simple thing im overlooking
17:00.00Beirdolikely is
17:00.13Beirdobut this isn't a RHEL channel, we don't likely know
17:00.43waddymight be easier to go back to RH9
17:00.55Beirdoas [TK]D-Fender said, maybe try #RHEL
17:01.02Beirdoif it exists
17:01.58waddyyer ill give it a go
17:02.00waddythanks
17:02.07*** join/#asterisk slak- (i=slak@rewted.biz)
17:02.08slak-hi
17:02.15slak-can someone tell me what this is
17:02.16slak-Destroying call '9f298d4d-73bdc324@192.168.111.20'
17:02.23slak-i dont have a sip pgone at that IP
17:02.26Beirdono prob, waddy
17:02.26brad_msswdoes AEL support && or || in if() statments ??
17:02.27slak-its a workstation
17:02.33slak-without any software phone
17:03.38slak-someone :)
17:03.49slak-clue me in why i see that log in the clit
17:03.52slak-er
17:03.52slak-cli
17:05.27*** join/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com)
17:06.00Endersonhello
17:07.34EndersonI 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
17:07.51Endersonbut 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)
17:08.09Equinox<PROTECTED>
17:08.09Equinox<PROTECTED>
17:08.16[TK]D-FenderWhat kind of FXS?  What is #7?
17:08.26TheCopshey [TK]D-Fender
17:08.30Seldon19751caould someone tell me how to configure Asterisk to log files to a MySQL server?
17:08.34[TK]D-Fenderhiya TheCops
17:08.34EndersonI've already tryied Flash() before a Dial() , that didn't work
17:08.35Seldon19751can it be done?
17:08.52QwellSeldon19751: I'm not thinking so, because where would it log if it can't write to sql server?
17:08.52[TK]D-FenderSeldon19751 : WIKI <-
17:08.59EndersonI tryed SendDTMF it called the other extension at 5, but it hangup
17:09.00Qwellbbl though
17:09.09TheCops[TK]D-Fender, g729 on dialup have a great quality! for 10$ it is a good thing to buy :)
17:09.19Enderson[TK]D-Fender: it's the number that * is plugged on PBX
17:09.44Seldon19751[TAN] whats the best IRC client for Windows?
17:09.46slak-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
17:09.50slak-can someone help me debug
17:09.57[TK]D-FenderEnderson : that made no sense to me, sorry...
17:09.58slak-the only thing that fixes it is if i reset the ATA
17:10.09slak-then it works for 24hrs or so
17:10.19Enderson[TK]D-Fender: that's my english, I don't know nome names =/
17:10.34Endersonlet me try again ...
17:10.53slak-[TK]D-Fender:  :D
17:10.55EquinoxWill my current 1.x config files work with 1.2.1?
17:11.02EquinoxOr will I need to mess with them ? :)
17:11.07slak-:)
17:11.12*** join/#asterisk Sixam (n=killa666@71.83.113.93)
17:11.30slak-Equinox: I had some issues with voicemail.conf
17:11.35Endersonasterisk 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)
17:11.50Equinoxslak- Voicemail I can deal with.. Extensions.conf sip.conf iax2.conf would be bad things to break ;)
17:12.09slak-those were fine, i didnt test iax2
17:12.32[TK]D-FenderExtensions.conf is what'll break.  the rest it pretty much the same.
17:12.33slak-and MeetMe seems to be broken..sound distrotions
17:12.38EndersonI 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-FenderOh and I've heard some Darth Vader MeetMe references :D
17:12.55slak-Equinox: also, in modules.conf you'll need to comment out chan_modem
17:12.59Equinox[TK]D-Fender, Hmm... Lots of stuff changed in extensions.conf?  My setup is pretty simple.
17:13.01slak-or * wont start with the old config
17:13.10slak-and chan_modem is off by default in 1.2
17:13.23slak-Equinox: my simple extensions.conf didnt change and worked
17:13.30[TK]D-FenderEquinox : Certain amount of things.  Depreciated callerID, ASTDB, priority jumping, etc
17:13.37ManxPowerI thought chan_modem was not even built in 1.2
17:13.40EquinoxAhh I don't use astdb
17:13.43slak-its not ManxPower
17:13.44EquinoxOnly DB I use is for CDR records
17:13.53EquinoxI use postgres for that tho, not mysql
17:13.58slak-thats why if he tries to use his old config, it wont start right
17:14.04EquinoxWhat is priority jumping?
17:14.07slak-because the old config tries to load chan_modem
17:14.24EquinoxI'll blow away chan_modem right now
17:14.25[TK]D-FenderEquinox : the old +101 thing for cmd's like DIAL.
17:14.35SkramXHi All!
17:14.40slak-TK: i posted a real annoying problem a few screens up, your input would be appreciated
17:14.54ManxPowerread UPGRADE.txt and SECURITY files.
17:15.09[TK]D-Fenderslak- : no idea
17:15.18slak-well any clue how i can debug it?
17:15.25Equinox[TK]D-Fender, Oh... Hmm I'll have to see how that is handled now.
17:15.29slak-i cant get the ATA to register without resetting its power
17:15.31EquinoxGotchya
17:15.31slak-really annoying
17:15.35slak-and its not the ATA
17:15.47[TK]D-Fenderslak- : Really not sure what to say...
17:16.02slak-OK, seperate issue:
17:16.03slak-Destroying call '9f298d4d-73bdc324@192.168.111.20'
17:16.08slak-that is a workstation
17:16.12slak-without any software phone
17:16.15moralewhat 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?
17:16.18slak-why would one see that line
17:17.46slak-is there any way to debug sipura ata's to see why they cant register
17:17.46[TK]D-Fendermorale : Hint is for presences detection so that you can see if other phones are busy (off-hook, ringing, etc)
17:17.46[TK]D-Fenderslak- : Try SIP debug
17:17.46[TK]D-Fenderslak- : in CLI
17:17.48slak-its on
17:17.51slak-nothing useful
17:17.57file[desk]woot MSN is b0rken
17:19.51slak-file why would i see this line if theres no sip phone on this workstation: Destroying call '9f298d4d-73bdc324@192.168.111.20'
17:19.55slak-i cant figure it out
17:20.15file[desk]there has to be.
17:20.57slak-what about this Destroying call '2c821ad0686f8f9c43f19b7f212dd78e@192.168.111.1'
17:21.01slak-thats my * box
17:21.10file[desk]why do you care anyway?
17:21.11slak-and theres even an entry for localhost like that
17:21.20slak-well because it makes no sense
17:21.21file[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)
17:22.19kink0hello
17:22.40slak-[TK]D-Fender: i found another way to solve that ATA problem without resetting its power, and thats "Remove Last Reg: YES"
17:22.43slak-under SIP tab
17:22.47*** join/#asterisk fanatic (n=jparrott@pcp01488192pcs.limstn01.de.comcast.net)
17:22.54slak-does that ring any bells?
17:23.05slak-it registered immediately
17:23.26*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
17:24.06darwin_35anyone here have a call back busy macro they are willing to share
17:25.23slak-look in the sample extensions.ael
17:25.28kink0anyone can help about PRI(E1)-> GSM(movile) implementation ? ( I just have a quad PRI digium card )
17:25.41KriS83Just out of curiosity, what do you mean by "callback busy macro"?
17:26.12*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
17:26.46bjohnson_awesome:
17:26.47bjohnson_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&region=ON&myurl=CSQ&mobility_upgrade=false
17:27.01bjohnson_good combination of bt and cell
17:27.18bjohnson_stick it on a SIP ATA and you're off to the races
17:27.41slak-but its not gsm capable
17:27.47bjohnson_even a baby monitor .. hahaha
17:27.51bjohnson_no note gsm
17:27.55bjohnson_not gsm
17:28.01bjohnson_Bell isn't gsm
17:28.57kink0I see, there not GSM mobile, just wireless
17:29.23bjohnson_cdma cellular
17:29.26bjohnson_not wireless
17:29.40bjohnson_oops .. maybe not
17:29.52bjohnson_I thought I was in their cellular phone sales section
17:29.59bjohnson_damn
17:30.20bjohnson_not so cool then
17:30.31slak-i  have a cdma from verizon
17:30.45bjohnson_I thought it was CDMA cellular and BT
17:30.48slak-i think i prefer the sim cards better
17:31.06darwin_35IceBrkr ... more input
17:31.12bjohnson_for me, it all comes down to the service packages
17:31.28bjohnson_I don't think service is much different in my area
17:31.54slak-verizon is the only service provider with a good signal everywhere
17:32.10darwin_35this odbc is is not making sense
17:32.10slak-i snapped my cingular phone in half it pissed me off so much ;/
17:32.11darwin_35its pissing me off
17:32.35slak-everywhere in my state i should say
17:32.54*** part/#asterisk smither (n=smither@cpe-68-203-132-96.houston.res.rr.com)
17:33.21RoyKdarwin35: why odbc?
17:33.42darwin_35iodbc/myodbc
17:33.56darwin_35used for connection
17:33.59RoyKwhy use odbc to connect to mysql?
17:34.07RoyKwhy not use a native driver?
17:34.15darwin_35only for the fax interface
17:34.22RoyKfax?
17:34.38RoyKfax over mysql? :)
17:34.43RoyKnah
17:34.45RoyKfax over odbc
17:35.26RoyK~seen zoa
17:35.29jbotzoa is currently on #asterisk (1d 7h 46m 53s).  Has said a total of 119 messages.  Is idling for 2h 35m 20s
17:35.29darwin_35it was thiw way before I got here
17:35.29iDunnofax over carrier pigeon.
17:35.44iDunnofax over tcp over smoke signals is more fun, though.
17:35.49RoyKdarwin35: then change it :)
17:36.02RoyKdarwin35: if you need a mysql connection, use nativer drivers
17:36.09RoyKfar better than odbc imho
17:36.27RoyKnative, even
17:36.28darwin_35[macro-faxreceive]
17:36.28darwin_35exten => s,1,SetVar(FAXFILE=/usr/local/asterisk-fax/${CALLEDFAX}/${UNIQUEID}.tif
17:36.28darwin_35exten => s,2,ODBCget(EXTEMAIL=${MACRO_EXTEN}/xEmail)
17:36.28darwin_35exten => s,3,NoOP()
17:36.28darwin_35exten => s,4,ODBCget(EXTNAME=${MACRO_EXTEN}/xName)
17:36.29darwin_35exten => s,5,NoOP()
17:37.31darwin_35I am just trying to get working what we have
17:37.57darwin_35and it works on 4 other boxes but not this new box
17:38.01RoyKwhy did it stop working?
17:38.07RoyKah
17:38.08RoyKok
17:38.20RoyKprolly priorities 3 and 5 blocking it
17:38.31darwin_35?
17:38.35darwin_35explain
17:38.57RoyKobviously, noop makes the previous line never to operate!
17:38.57darwin_35I am not a database person
17:39.03RoyK:D
17:39.09Dandanargh
17:39.14RoyKwhat sort of error do you get?
17:39.24darwin_35the issues is the odbc is not connecting to the main dbserver
17:39.28Dandanit seems like i have a problem with x100p not receiving digits fast enough
17:39.38Dandananyone had that problem?
17:40.01RoyKdarwin_35: ah. have you checked the acl on the server?
17:40.06RoyKmysql grants
17:40.06RoyKetc
17:40.51darwin_35Dec 20 10:40:13 NOTICE[7215]: res_odbc.c:294 load_odbc_config: registered database handle 'DSNvoicemail' dsn->[DSNvoicemail]
17:40.51darwin_35Dec 20 10:40:13 NOTICE[7215]: res_odbc.c:552 odbc_obj_connect: Connecting DSNvoicemail
17:40.51darwin_35Dec 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)
17:41.04RoyKhttp://dev.mysql.com/doc/refman/5.0/en/show-grants.html
17:41.05darwin_35same for all parts of the db
17:41.15RoyKlocalhost?
17:41.28darwin_35where its getting local host I cant find
17:41.29RoyKdo all servers run with local mysql installations?
17:41.37RoyKres_odbc.conf perhaps
17:42.30darwin_35nope
17:42.53RoyKthen the odbc config
17:43.00*** join/#asterisk apardo (n=apardo@62-15-237-162.inversas.jazztel.es)
17:43.00RoyKfind /etc -iname "*odbc*"
17:43.01darwin_35roy can I paste pvt for a min
17:43.06RoyKpastebin
17:43.09RoyK~pb
17:43.11jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/
17:43.26RoyKhm. pastebin.ca is better imho
17:43.35*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)
17:43.45DandanRoyK: it is just that they use broken ECN
17:44.09RoyKDandan: odbc uses broken ECN?
17:44.14Dandanno
17:44.15Dandanpb.ca
17:44.16darwin_35http://pastebin.com/471892
17:44.24BlackthornAny 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?
17:44.41darwin_35we are on mysql 4.1.15
17:44.56RoyKdanalien: wtf is ecn?
17:45.31*** join/#asterisk chris-fn (n=chris@netblock-66-51-202-171.dslextreme.com)
17:45.59chris-fni'm having a tad bit of trouble with Asterisk::Manager, anyone available for a sec?
17:46.06darwin_35we have the server defined
17:46.12RoyKdarwin35: what about udbc.ini?
17:46.26darwin_35udbc.ini ?
17:46.31RoyKasdf.ini
17:46.34RoyKodbc.ini :)
17:46.46DandanRoyK: explicit congestion notification
17:47.02RoyKahki
17:47.03RoyKic
17:47.09Dandan:)
17:47.33RoyKdarwin_35: see http://www.minisoft.com/pages/middleware/ODBC_UNIX/odbc_for_unix_Tracing.htm
17:47.39RoyKdarwin_35: under examples
17:48.17RoyKdarwin_35: http://pastebin.com/471898
17:49.52*** join/#asterisk arcraig (n=arcraig@c-67-187-184-184.hsd1.ca.comcast.net)
17:50.06darwin_35http://pastebin.com/471900
17:51.11DandanDec 20 12:50:58 WARNING[1091]: channel.c:2530 ast_request: No channel type registered for 'Console'
17:51.11DandanDec 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)
17:51.14Dandanwhat the hell is that?
17:51.18Dandandoesn't like my oss?
17:51.35RoyKdarwin_35: sorry. dunno. i don't use odbc...
17:51.53darwin_35ok
17:52.01asteriskmonkeycan 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)
17:53.12freezerhi
17:54.58asteriskmonkeyhey.. 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)
17:55.53RoyKzoa: ping
17:57.26docelm0PONG!
17:57.32*** join/#asterisk bmg505 (n=leon@dsl-146-5-195.telkomadsl.co.za)
17:57.40AndyCapbjohnson_: was it you who mentioned a sip phone supporting encryption a while back?
17:58.09kink0what is the best way to gateway to GSM mobile from asterisk ?
17:59.30AndyCapkink0: there is something called chan_bluetooth, but I don't know if it's any good.
18:03.13fugitivokink0: www.2n.cz
18:05.05kink0fugitivo, yes, the no easy is to find something less than 6000 dls
18:05.31fugitivocost of that is 800 euros
18:05.38fugitivoit's like 1200 dls
18:05.51fugitivoi think it's the only sip/gsm gateway
18:06.19fugitivoyou can get analog gsm gateways, much cheaper
18:06.49msw7
18:06.51mswdoh
18:08.28kink0fugitivo, I was thinking about the Stargate, 30 GSM channels gateway
18:09.49kink0AndyCap, 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)
18:16.13kink0chan_bluetouch is able to manage multiple phones ? are AT ( data channel ) on the same bluetouch communication than voice ?
18:16.33BlackthornQuestion: 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?
18:18.15kink0I have version 1.2.0-rc1 but appears there nothing about chan_bluetouch
18:20.15Dandanbluetouch? or bluetooth?
18:20.36puzzledkink0: chan_bluetooth is not part of the asterisk source
18:20.45[TK]D-Fenderchan_bluetooth is NOT part of Asterisk main.  Its a 3rd party add-on.
18:22.32fugitivochan_deathtouch
18:22.51SkramXchan_wrathofdeath
18:23.27mog_workanyone know libcurl?
18:24.04fugitivomog_work: no, but i disable it at compilation time
18:24.27mog_workheh thanks
18:25.36Dandananyone 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)
18:28.32mog_workjust dial(console/dsp)
18:28.38mog_workand set it to auto answer
18:28.56mog_workor thats what i thought it was
18:28.59saftsackcan i plug a normal fax telephone on my asterisk server without using soandsdp?
18:29.12fugitivoyes
18:29.21saftsackand it works? :)
18:29.30fugitivomaybe
18:29.53BlackthornQuestion: 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?
18:30.04saftsackfugitivo, ? ^
18:30.16fugitivoBlackthorn: try it
18:30.56fugitivosaftsack: maybe it'll work 100%, maybe 50%, what hardware are you using?
18:31.07saftsacki have a tdm wildcard
18:31.35fugitivotdm400?
18:31.40Dandanmog_work: i do not hear anything
18:31.44Dandanthrough speakers :/
18:31.55Dandanneither it plays any GSM files
18:31.57*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
18:32.30saftsackfugitivo, yes
18:32.33SkramXHi, ctooley... We spoke a couple weeks ago..
18:32.49fugitivosaftsack: 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)
18:33.01fugitivosaftsack: i have a customer with that config, and it works 98%
18:33.29saftsackok
18:33.36saftsackwhat do you mean with 98%? ^^
18:34.15fugitivofrom 100 transmissions, 98 faxes work ok, 2 fail
18:34.16*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
18:34.19saftsackok
18:34.26saftsackare they work on the 2. try?
18:35.05osiruswe use it, and the faxes do fail from time to time
18:35.17ctooleySkramX, yeah I remember
18:35.23ctooleySkramX, how are you?
18:35.41*** join/#asterisk kimosabe (n=kimosabe@dsl-201-129-75-8.prod-infinitum.com.mx)
18:35.41SkramXI am alright
18:35.47SkramXSorry, I had to get the door.
18:35.48osiruswith the same percentile, just about
18:35.50saftsackosirus, hmm are the faxes directly connected with the wildcard or do you use spandsp?
18:35.59SkramXWe just got a huge box of grapefruits.. must be a holiday present
18:36.10osirusdirectly connected
18:36.24*** join/#asterisk amir_ (n=amir@gentoo/developer/amir)
18:36.32saftsackwith spandsp it would be more crappy or?
18:36.54SkramXctooley: I will be right back.. you can PM me if you arent busy.. is there a reason I was never emailed back :|
18:37.03osirusI couldnt tell you
18:37.08fugitivospandsp works perfectly for me
18:37.35kimosabeis the zircom instant asterisk disc good can ii also add gui to thius
18:37.35fugitivobut i use the x100p for that
18:37.56*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
18:38.38saftsackhas someone of you a isdn deviceacces?
18:38.44osirushas anyone had any experience with dropped calls being sent over pstn from covad?
18:39.01bjohnson_AndyCap: no.  I haven't seen an encrypted sip phone yet
18:39.30bjohnson_AndyCap: any encryption I've heard of in an implemented system is with a VPN
18:39.35SkramX-back0
18:39.39SkramX*-back-
18:40.36*** join/#asterisk jake1932 (n=jake1932@pool-68-236-10-151.phil.east.verizon.net)
18:41.23[TK]D-FenderAndyCap : Sipura SPA-941 supports SRTP (Secure RealTime Protocol)
18:41.23osirusi have a customer that transfers calls to me from a covad line, and we're getting lots of complaints about dropped calls
18:42.33saftsacki 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)
18:43.47rob0grapefruit can be good weapons! Especially if overripe.
18:44.05*** part/#asterisk Sixam (n=killa666@71.83.113.93)
18:44.06osirusor frozen
18:45.25saftsackcan i "ping" an isdn telephone?
18:45.40jake1932is there any way I can turn off one T100P card (have 2 in total) without affecting the other on a live system?
18:46.06darwin_35has the mysql leak with res_mysql been fixed
18:46.40darwin_35the only reason my boss says we are usiing odbc
18:46.52darwin_35was the mem leak
18:47.16AndyCap[TK]D-Fender: ok. thanks. got to google I guess. it was a not a well known brand.
18:48.01[TK]D-FenderNot well known?  Thats "interesting".  Sipura is quite highly regarded....
18:48.54fugitivohehe
18:49.09darwin_35I had issues withthe sipura spa-841
18:49.22darwin_35<PROTECTED>
18:49.23[TK]D-FenderWhat kind of issues?
18:49.27[TK]D-Fenderhmm
18:49.44darwin_35and some times pick up a call and it would reboot
18:49.46[TK]D-FenderWEll the 941 has proven to be a pretty solid business phone so far...
18:49.50AndyCap[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
18:49.57fugitivoAndyCap: also, sipura is now cisco
18:50.20AndyCapfugitivo: argh, is nothing sacred?
18:50.20AndyCap:)
18:50.22[TK]D-FenderOr rather Sipura is now LInksys who is now Cisco ;)
18:50.25darwin_35I have moved to pa-168 based chip phones now
18:50.35[TK]D-FenderPA168?  EW
18:50.38fugitivoyeah, what d-fender said :)
18:50.50AndyCapjust so they could get their hands on the next model ATA? :-P
18:50.51[TK]D-FenderUBER cheap junk...
18:50.54fugitivodarwin_35: that sucks
18:51.29[TK]D-FenderOnly phones I recommend at this point is the SPA-941 and anything Polycom.
18:53.47jake1932let 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?
18:54.31AndyCap[TK]D-Fender: and snom?
18:54.36fugitivojake1932: disable it in which way?
18:54.47fugitivojake1932: unplug the cable?
18:54.55jake1932i'm not at the location
18:55.05jake1932but yes, essentially unplug the cable
18:55.17jake1932(without actually unplugging it)
18:55.21[TK]D-FenderAndyCap : What about Snom?
18:55.56AndyCap[TK]D-Fender: just wondering if they'd make your not recommended list or unkown
18:56.11fugitivojake1932: you could try to deactivate it with setpci, but it's dangerous on a production system :)
18:56.58jake1932the one card is in and out of yellow alarm.  i just don't want it to take any more calls
18:56.58fugitivojake1932: 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)
18:57.19jake1932will that affect the calls on the working card?
18:58.00*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi)
18:58.18[TK]D-FenderAndyCap : 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)
18:59.20fugitivojake1932: unloading the modules, yes
19:00.05jake1932and ztcfg - prob will also cause the calls to drop..
19:01.18fugitivojake1932: maybe with zap destroy from the cli
19:01.26fugitivojake1932: you'll need to destroy each channel
19:01.41jake1932ok
19:01.45slak-drink a beer
19:01.49slak-then think about it later
19:07.41osirus!
19:07.50jake1932i destoyed them
19:08.07jake1932now how do i get them back when they fix the yellow alarm issue?
19:08.27saftsacksome germans here?
19:08.37osirushah
19:08.39osiruswha
19:08.40osirust
19:08.52saftsack?
19:09.04osirusgermans
19:09.04osiruswhere!
19:11.03saftsackosirus, r u german?
19:11.12saftsackor better, do you speak german?
19:11.38trixterich keine spreche deutch und nicht seir gut!
19:11.45osirusno
19:11.47saftsackhrhr
19:11.55osirusim 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)
19:15.40RoyK~seen zoa
19:15.51jbotzoa 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)
19:17.49Insanity5Am 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.
19:18.11fugitivovonage
19:18.23Insanity5fugitivo $$$ -- rather just call the local telco
19:18.26Insanity5And it won't work with *
19:18.31*** join/#asterisk techie (i=gus@antibala.com)
19:18.43fugitivowell, then it's not impossible :)
19:18.54fugitivoyou have to play their rules
19:19.31darwin_35icebrkr you alive
19:19.40Insanity5lol, running it through the ata back into a card?  no thanks
19:20.40iCEBrkr-.- zzZZZ
19:21.12*** join/#asterisk tomtom_ (n=tom@bender.linugen.com)
19:21.16tomtom_hi
19:21.57Insanity5hi
19:22.10fugitivohello
19:22.33saftsackDec 20 20:18:00 NOTICE[2176]: app_dial.c:764 dial_exec: Unable to create channel of type 'misdn'
19:22.40saftsackbut theres no reason given :(
19:23.08PrimerAnyone 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?
19:23.35saftsackok i know what was wrong
19:24.48saftsacki defined a group in my misdn.conf and in the extensions.conf i call the group ,Dial(misdn/g1/${EXTEN})
19:24.53saftsackbut it doesnt work
19:25.00saftsackis the dialstring false?
19:25.18tomtom_anyone knows what's causing these: Dec 20 20:18:07 WARNING[3955]: RTP Read error: Bad file descriptor ?
19:25.38tomtom_I get them sometimes when transferring a call to a meetme room
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19:30.43SpaceBasshey folks
19:31.09SpaceBassI'm migrating my AAH setup to a new box but don't want to move my zaptel hardware yet... can I keep the * box with the zaptel stuff and just route it to my new box?
19:32.07SpaceBassI'm thinking just create a trunk and some routing...
19:32.25fugitivoyes
19:32.37*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
19:33.56iCEBrkrtrunking is kinda neat
19:35.40*** join/#asterisk saft (n=saft@ip-202-37-230-210.internet.co.nz)
19:36.41*** join/#asterisk mungojam (n=admin@87.194.16.84)
19:36.52iCEBrkrCould it be any warmer in here?!
19:37.00iCEBrkrGeesh, you'd think it's 65 degrees outside.
19:37.07mutilatorbleh
19:37.20iCEBrkr...and where the hell are my PRIs?!
19:37.24mutilator200 lines of code just to apply payments and thats not even doing any of the financing options
19:37.26iCEBrkrDevice name | Protocol | Station | Status        |
19:37.26iCEBrkrwanpipe1    | AFT HDLC | N/A     | Disconnected  |
19:37.26iCEBrkrwanpipe2    | AFT HDLC | N/A     | Disconnected  |
19:37.27iCEBrkrNo love.
19:37.36mutilatori r teh hate accounting
19:37.50mutilatorand english o_O
19:37.58mutilator:P~
19:38.00iCEBrkrengrish?
19:38.15*** join/#asterisk apardo (n=apardo@54.Red-83-50-238.dynamicIP.rima-tde.net)
19:39.58[TK]D-FenderiCEBrkr : Whats the telco have to say?
19:40.01saftwheeeeeee, i have an interesting issue here, i only get audio one way going Dlink DG104s -MGCP- * -iax- * -chan_sccp- Cisco VIP 30
19:40.24iCEBrkr[TK]D-Fender: I put in a second call, I'm waiting for someone to call me back.. AGAIN
19:40.49[TK]D-FenderDUMB%#$^s
19:40.57SpaceBassso which box has the trunk and which has the account?
19:41.01iCEBrkrMy rep never calls me back
19:41.11*** join/#asterisk SkramX (n=skramy@vistech.org)
19:41.18iCEBrkrSpaceBass: Think client-server :)
19:41.21SpaceBassIf 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)
19:41.39iCEBrkrI'd make B register with A.
19:41.40*** join/#asterisk Laibsch (n=Laibsch@p54B9B627.dip0.t-ipconnect.de)
19:41.43SpaceBassiCEBrkr thats what Im trying to do... mostly thinking out loud here
19:41.43shankyhi, good evening
19:42.03shankyanyone uses areski-stats ?
19:42.25*** join/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net)
19:42.41SpaceBassB registers with A (zaptel box) ... thinking... that works for incoming and outgoing
19:43.08shankyI 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?
19:43.59LaibschHi, 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?
19:44.03mungojamhttp://pastebin.ca/34375
19:44.24Dandananyone has any experience with OSS?
19:44.28Dandani am still stuck :/
19:44.33fugitivooss?
19:44.34*** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no)
19:44.42Dandanas in console/dsp
19:44.42fugitivoopen sound system?
19:44.53fugitivothat's obsolete, use alsa
19:45.12RoyK~seen zoa
19:45.16jbotzoa is currently on #asterisk (1d 9h 56m 40s).  Has said a total of 119 messages.  Is idling for 4h 45m 7s
19:45.20RoyKargh
19:45.20Dandanalsa... hm lets try :)
19:46.04iCEBrkrRAAWWWWWRR!
19:46.08saftsack<PROTECTED>
19:46.15DandaniCEBrkr: ?
19:46.23saftsackthis is my dialstring but if i dial a number it just calls the first two digits
19:46.27saftsackbecause it doesnt wait
19:46.30iCEBrkr[TK]D-Fender: Apparently, the remote/colo PRI is turned-up.. Yet, I can't dial
19:47.07iCEBrkr[TK]D-Fender: It would appear I have the Digium card configured correctly down there since zttools reports 'OK'
19:47.15jake1932saftsack: did you do a show dialplan?
19:47.27iCEBrkrDandan: I'm going round and round with our provider to get PRIs installed and turned-up.
19:47.39saftsackjake1932, show dialplan?
19:47.44jake1932saftsack: if not, can you pastebin your "show dialplan"
19:47.51jake1932in the CLI
19:48.14saftsackok
19:48.16saftsackmoment
19:48.26*** part/#asterisk Primer (n=vi@sh.nu)
19:48.36[TK]D-FenderiCEBrkr : PM
19:48.43*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:48.45saftsackhttp://pastebin.com/472098
19:48.59jake1932what do you dial?
19:49.12saftsacki dial in context raus
19:49.17saftsack0mynumber
19:49.28saftsackand asterisk should do that too
19:49.34saftsackbut it just takes 05
19:49.35jake1932does it wait?
19:49.37DandanALSA usbaudio.c:870: timeout: still 8 active urbs..
19:49.42Dandanalsa :/
19:49.52jake1932saftsack: soe your phone have a dialplan?
19:49.54jake1932does
19:50.04saftsackwhat for a phone?
19:50.12saftsackyes i think so
19:50.22jake1932saftsack: what phone are you using?
19:50.24saftsack<PROTECTED>
19:50.33saftsackan isdn telephone
19:50.58jake1932that you're calling from, to, or both?
19:51.47mungojamHi, 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?
19:52.09jake1932mungojam: just in the US - for BRI
19:52.13shido6anyone in michigan?
19:52.50jake1932mungojam: wasn't hard - just time consuming pinning down all the info.  what card are you using?
19:52.58mungojamjake: I don't quite understand, I thought BRI wasn't available in the US, or is it that it works differently?
19:53.21jake1932mungojam: definately available -PIA to get (and get rid of) :)
19:53.50mungojamHi, 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
19:54.02jake1932mungojam: can't guarantee either
19:54.42jake1932mungojam: 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
19:54.55mungojam:) 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)
19:55.17mungojamI am in the UK
19:56.50jake1932mungojam: for non-us, http://www.junghanns.net/en/chan_capi.htmlt
19:57.14fugitivomungojam: try it yourself, it's opensource, no guarantee
19:58.05jake1932<PROTECTED>
19:58.15fugitivoavg is the antivirus :)
19:58.19jake1932right
19:59.00jake1932AVM Fritz is the one I got (that I have to resell)
19:59.33RoyKchan_capi works with the fritz
19:59.45jake1932just not in the US
19:59.52RoyKmaybe :)
20:00.10jake1932i know from testing
20:00.28RoyKoh well
20:00.44jake1932<PROTECTED>
20:01.31mungojamfugitivo: 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
20:01.37SpaceBassanyone using NuFone and AMP?
20:01.48Ariel_SpaceBass, yes
20:01.59jake1932saftsack: you still there?
20:02.07saftsackyes
20:02.12SpaceBassAriel_ did you have to manually add the dial string to extensions_custom.conf or did you set up the trunk in AMP?
20:02.17mungojamsomewhere around £900
20:02.22tomtom_Nobody any info on this error Dec 20 20:18:07 WARNING[3955]: RTP Read error: Bad file descriptor ?
20:02.31jake1932<PROTECTED>
20:02.34mungojamthanks for the help, bye
20:02.35*** part/#asterisk mungojam (n=admin@87.194.16.84)
20:02.44saftsackjake1932, yes
20:03.43*** join/#asterisk backblue (n=moo@87-196-10-196.net.novis.pt)
20:04.08RoyKcan someone please take a look at this patch to help me find wtf is wrong with it? http://karlsbakk.net/asterisk/scripts/asterisk-mrtg
20:04.20RoyKer
20:04.22RoyKsorry
20:04.29RoyKhttp://karlsbakk.net/asterisk/patches/1.2.x/zoa-jb-2005-12-20.patch
20:04.30RoyKthat one
20:05.27SpaceBassanyone using a cisco phone and registering to more than one asterisk box?
20:06.26fugitivoRoyK: is that a patch? it looks like a windows service pack ;)
20:06.32backblueSpaceBass: why do you want to register in more then one asterisk?
20:07.00SpaceBassbackblue b/c I have 2 servers running currently and it would facalitate testing trunking b/t them
20:07.02RoyKfugitivo: 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)
20:08.54backblueSpaceBass: b/c?
20:09.19backblueSpaceBass: you can forward calls bettwen them, dont need to register on both.
20:09.27SpaceBassb/c I want to :)
20:09.37saftsackjake1932, any ideas?
20:09.39backbluewhen does your cisco should know when to use *1 or *2?
20:09.43marcus2anyone have polycom sip 1.6.3 firmware handy?
20:09.53[TK]D-FenderI do.
20:09.59SpaceBassb/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)
20:10.19SpaceBassAlso thinking about using it for extension based routing
20:11.00SpaceBassbackblue 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
20:11.04netvulturewhats 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?
20:11.25darwin_35codecs
20:11.41netvultureg711u
20:11.45backblueSpaceBass: i dont see why do you need yor 7960 to register in both *
20:11.56backbluejus forward calls bettwen them
20:12.02fugitivonetvulture: check codecs
20:12.07netvultureshould I try to force the codec before answering?
20:12.13SpaceBasswell, frankly, that is what I'm trying to do and curious if anyone has expirence with it
20:13.07backblueSpaceBass: but you need 2 clients, not 1 client, and 2 servers.
20:13.08harryvvvonage 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.
20:13.22backblueSpaceBass: and by the way, that its so very simple
20:13.22netvultureok - thanks - i'll check those out
20:13.32backblueyou do that, and have examples
20:13.36backbluein voip-info
20:13.53backbluelike, connecting to any voip free provider, its exacly the same.
20:14.08backbluejust insted, use only your two * box
20:14.18SpaceBassI'm really not a nube...
20:15.36jake1932<PROTECTED>
20:16.01saftsackjake1932, ok thanks
20:16.10saftsackhmm i read that this is a normal misdn issue
20:16.21saftsackbut 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)
20:23.38SpaceBasswill IAX2 pass DID (trunking from one box to aother) or do I need to a /<exten> to my register string
20:24.37saftsackmy asterisk doesnt hangup if the timeout was reached :(
20:24.43saftsack<PROTECTED>
20:24.43saftsack<PROTECTED>
20:24.50saftsackafter 10 seconds it should hanguo
20:27.22shido6anyone want a heart attack?
20:27.33shido6http://media.spikedhumor.com/8944/Jingle_Bells_Reversed.swf
20:28.14justinuhttp://video.google.com/videoplay?docid=4845715794200371561
20:28.17justinufast forward to the end
20:28.35*** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss)
20:28.45Madkisshi all; what's the current chan_capi version and where do I get it?
20:29.17fugitivoshido6: GOD
20:29.37arcraig0  /exit
20:29.38arcraigcd
20:29.41arcraigsorry :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)
20:38.21saftsackhowto play the congested tone to the caller?
20:39.22harryvvgoogle it should come up
20:39.33harryvvcongestion asterisk
20:39.35saftsackive read playtone but i dont thing that its correct or?
20:39.42justinuCongestion
20:39.54saftsacki have congestion already
20:40.00saftsackbut it doesnt work :(
20:40.10saftsackmaybe because theres a tk before my asterisk?
20:40.52saftsack<PROTECTED>
20:40.52saftsack<PROTECTED>
20:40.53saftsackso?
20:41.21justinumisdn... can't help you there
20:41.29saftsackok thanks
20:41.40saftsacki have the feeling, that misdn is very buggy :(
20:41.55justinucould be... is the inbound call BRI also?
20:42.24SpaceBassI'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)
20:44.59lesouvageFWD 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?
20:45.19fugitivothere're always problems with FWD
20:45.27xachenyeah
20:45.34xachenI never can get FWD to work with my asterisk box
20:45.46xachenIAXtel is a bugger too. I just use e164.org :)
20:46.05lesouvagexachen: I had it working, didn't change it and now it stopped working.
20:46.40darwin_35we get it to work but only 1/3rd the time
20:46.41darwin_35fwd and iaxtel
20:47.05darwin_35I have yet to figure ot enum
20:47.12darwin_35not had  time
20:47.17darwin_35or dundi
20:47.53*** join/#asterisk Psykick (n=anon@smtp.phoenixone.co.nz)
20:49.23lesouvageSo 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)
20:50.27asteriskmonkeyhey
20:50.38darwin_35what now monkey
20:50.38*** join/#asterisk DJ-Pyro (n=DJ-Pyro@lan-gw.brevient.net)
20:50.54darwin_35get off my back will you I am getting sore
20:50.59asteriskmonkeycan 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
20:51.06asteriskmonkeylol
20:51.31iCEBrkrThere's a when and why to upgrade..
20:51.42iCEBrkrSounds like you disobeyed the when.
20:51.44DJ-Pyroanyone 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
20:51.58darwin_35back step to 1.2 and test if the issue is not there then its a bug
20:52.11asteriskmonkeyDJ-Pyro
20:52.19asteriskmonkeyDJ-Pyro: change the echo can
20:52.21darwin_35IceBRkr
20:52.29saftsackif 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?
20:52.30asteriskmonkeyDJ-Pyro: and tweak the rx
20:52.34saftsackISDNalle=misdn/g:NTPORTS/20&misdn/g:NTPORTS/10
20:52.35DJ-Pyrosomething other than the default 128?
20:52.38saftsackor can i do this in one string?
20:52.42iCEBrkr?
20:52.51darwin_35I have to shoot oyu
20:53.01darwin_35we have tried everythign and still no go
20:53.11DJ-Pyroasteriskmonkey: I thought rx was typically non TDM circuits
20:53.16darwin_35its killing me
20:53.26asteriskmonkeyno , pri's have gain :D
20:53.27iCEBrkrshoot me? you're the ones who opt'd to use ODBC :)
20:53.29darwin_35I fallowed the odbc page youpointe dout
20:53.32*** join/#asterisk Hmmhesays (i=Blorp@66.173.103.100)
20:53.44asteriskmonkeycan i rollback to 1.0.9?
20:53.45darwin_35no that was the person befor  me
20:53.50asteriskmonkeydo i have to rm everything first?
20:54.00Hmmhesaysanyone in the uk I can call to see if my callerid is getting across BT?
20:54.13darwin_35and now i am cleaning up a mess
20:54.39*** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no)
20:54.57darwin_35can you point out any ideas more that might help you said you got it working
20:55.18iCEBrkrdarwin_35: screw real-time?
20:55.28darwin_35I plan to move to postgress but I have to have this working like tonight
20:55.34darwin_35cant
20:55.40darwin_35or I would
20:55.54asteriskmonkeyi can roll back to 1.0.9 right??
20:56.08[TK]D-FenderDJ-Pyro : HOw many T1's?
20:56.38HmmhesaysUK anyone?
20:56.39DJ-Pyro[TK]D-Fender: 28
20:56.46DJ-Pyro7 asterisk servers with 4 t1's in each
20:57.06darwin_35I beg of you o iced one make me your padawond and show me the ways to get it working
20:57.08iCEBrkrdarwin_35: I dunno, sounds like more of a pain in the ass to me.
20:57.15darwin_35yeah 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)
20:59.13RoyK~seen zoa
20:59.24jbotzoa is currently on #asterisk (1d 11h 10m 48s).  Has said a total of 119 messages.  Is idling for 5h 59m 15s
20:59.29asteriskmonkeyis there anyone that has gone from 1.2.1 back to 1.0.9?
20:59.48RoyKa few times, yes
20:59.53iCEBrkrasteriskmonkey: whyd' you go to 1.2.1 anyhow?
21:00.04RoyKafter trying to upgrade and than found 1.2.1 too unstable......
21:00.08asteriskmonkeyI wanted to have the more extended iax2 info stuff
21:00.20iCEBrkrlook where that got'cha
21:00.27asteriskmonkeywhat do i have to do to roll back to 1.0.9
21:00.28HmmhesaysUK anyone?
21:00.29RoyK1.2 isn't really stable yet
21:00.30denonanyone have some 7960s they want to sell?
21:00.35SpaceBassI'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)
21:00.50asteriskmonkeydenon: are you in canada?
21:00.53denonnope
21:00.58iCEBrkrasteriskmonkey: 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)
21:01.06denonasteriskmonkey: close though, Minnesota. :)
21:01.09iCEBrkrasteriskmonkey: undo your 1.2.1 extensions.conf changes.
21:01.11RoyKasteriskgeeks: just install remove everything under /usr/lib/asterisk/modules/* and libpri, zaptel and asteterisk, possibly -addons and restart
21:01.14Hmmhesayshaha you too huh denon?
21:01.28Psykickanyone used or had any success with iaxclient library?
21:01.29RoyKasteriskmonkey: that one was for you.....
21:01.33denon'course, all the cool people live in MN
21:01.36asteriskmonkeywhat about my pri/zaptel stuff do i have to remake and remodprobe?
21:01.40Hmmhesayswhere about?
21:01.43iCEBrkrdenon: freezing cool people..
21:01.54RoyKasteriskmonkey: running zaptel stuff?
21:01.59iCEBrkrasteriskmonkey: yup. make install zaptall/libpri
21:02.07denonHmmhesays: well, I'm in mpls at the moment, but I get around
21:02.18denonyou?
21:02.25Hmmhesaysi'm sitting in moorhead right now
21:02.37denonah .. college kid?
21:02.42Hmmhesaysnaw
21:02.45Hmmhesaysgrew up here
21:02.48asteriskmonkeyah ... i hope the iaxy.bins that got upped arnt the cause of the issue either :P
21:02.49denonic
21:03.00denonasteriskmonkey: you have some phones you want to sell over the border
21:03.00denon?
21:03.10Hmmhesayslookin for a freaking Uk test number
21:03.12Hmmhesaysso I can go home
21:03.14harryvvis 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.
21:03.26iCEBrkrLOL
21:03.29denonharryvv: you mean like sip qualification?
21:03.44justinui think what he wants is like a MOS score of a test call
21:03.49denonyeah ..
21:03.54denonbut thats not gonna happen :)
21:03.56iCEBrkrharryvv: yea, too bad your link won't be stable enough to test and even then it could degrade in the middle of a call.
21:04.02justinunot without doing some serious work :)
21:04.07harryvvdenon, 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.
21:04.21denonharryvv: ah well, that you could test ..
21:04.21justinui've got the RTCP stuff working
21:04.26Hmmhesaysgotta love the public internet
21:04.35Hmmhesayspeople expect it to be perfect
21:04.43justinubut i found both polycom and eyebeam send incorrect RTCP info in their sender reports.
21:04.47justinuHmmhesays: why shouldn't it be?
21:04.55justinuif you just accept crap, you'll get crap
21:05.02Hmmhesaysheh, why should it be
21:05.05harryvvhmm, so basicly there is no perfect voip model...or close to be perfect if there is no voip service terminating calls?
21:05.21SpaceBassanyone else having problems with NuFone?
21:05.28Hmmhesaysanyone in here from the UK?
21:05.31iCEBrkrjustinu: 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..
21:05.52justinuglobal-x rocks
21:05.59iCEBrkrharryvv: leased-lines between your offices.. That'd be about it
21:06.11iCEBrkrjustinu: that's why i used them as an example :)
21:06.11harryvvI 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.
21:06.29justinuyou could use some kind of teir 1 VPN solution
21:06.37justinuthat'd probably get the QoS up to a decent level
21:06.49harryvviCEBrkr yea, and how much do those lines cost? I mean if you want near 100% reliability of the physical link thats probebly it.
21:07.05iCEBrkrjustinu: 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.
21:07.21asteriskmonkeyvoip is really good if you can maintain a 50ms or lower route to the voip server
21:07.25justinuwow
21:07.31justinuthat sounds like a serious pain in the ass
21:07.44justinui've got 5ms latency to my PSTN gateway :)
21:07.46asteriskmonkeysome people say high as 250, but to be hoest 50> you will occasional echo etc
21:07.49iCEBrkrhaha
21:08.00iCEBrkrI don't get echo
21:08.03harryvvso 50 ms then
21:08.04asteriskmonkeyjustinu: i have somepeople that are 600k away with 15ms times :D
21:08.05iCEBrkrYou people are nuts :D
21:08.13Hmmhesaysor if you're from africa anything under 750ms is acceptable
21:08.15*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
21:08.17justinulol
21:08.20asteriskmonkeylol
21:08.23iCEBrkrI had echo on my line once.
21:08.23Hmmhesaysand you think i'm joking
21:08.28justinui believe it
21:08.43iCEBrkrMy VoIP just cracks and drops out.  But never any echo
21:08.52iCEBrkrNEVER!
21:08.53harryvvI had one network reseller outright reject voip because of dropped packets.
21:08.54asteriskmonkeycracks are bad too :(
21:08.55Hmmhesaysecho doesn't come from the voip side
21:09.13asteriskmonkeywe are trying to get better quailty aswell as price over the pstn right? not come up with something hack
21:09.15Hmmhesaysunless you've called an echo app
21:09.20harryvvits the electronics in the cards.
21:09.21iCEBrkrHmmhesays: my X100P doesn't echo either :)
21:09.25asteriskmonkeycrap msn chat has better sounding voice in some cases
21:09.48justinudon't use G711 if your network isn't perfect
21:09.49harryvviCEBrkr my card used to echo..i haaardly hear my self on the other end but need to pay attention.
21:10.13Hmmhesaysi could really use a UK number to send a call to, i vant to go home
21:10.41harryvvHmmhesays where are you now
21:10.45asteriskmonkeymy te110p used to slam echo like a dirty whore until i changed the echo can
21:10.47HmmhesaysMinnesota
21:11.10harryvvi see
21:11.12HmmhesaysI need to test this gateway to see if BT is accepting callerid
21:11.19harryvvwhat you doing in the frozen land of the north?
21:11.24justinu+44 208 237 3000
21:11.32justinu(coca cola great britain)
21:11.43Hmmhesaysaccept i need to know what number comes in
21:11.51harryvvHmmhesays what u doing there?
21:11.51justinuoh
21:11.57Hmmhesaysin MN? i live here
21:12.27Hmmhesayslets 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
21:12.35justinulol
21:12.45justinu6 pack and a light switch
21:12.48justinuuniversal solution
21:12.53[TK]D-Fender:O
21:12.59HmmhesaysLOL
21:13.14harryvvand your from england?
21:13.19Hmmhesayshell no
21:13.26Hmmhesaysborn in minneapolis
21:13.31Hmmhesaysgot a gateway in england
21:13.39Hmmhesaysdass freaking 2 ugh
21:13.53kink0Hmmhesays, BT ?
21:14.01Hmmhesayskink0 aye
21:14.04kink0I am very interesting about bluetooch
21:14.20kink0I did fews experiments ussing ATA and sound cards to connect GSM
21:14.23justinudass-2, lol
21:14.31justinugotta love BT
21:14.38HmmhesaysI 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
21:14.38kink0but just today, start reading about chan_bluetooth
21:14.59kink0( someday I will learn how is written "blueto**" ) :)
21:16.14fugitivobluetooth
21:16.14fugitivorepeat with me
21:16.14kink0fugitivo, xD
21:16.14fugitivobluetooth
21:16.16Hmmhesaysbluetooch sounds like it could be part of the female anatomy
21:16.16kink0Hmmhesays, have you success with bt ? just one or several channels ?
21:16.18harryvvgateway in england?
21:16.18Hmmhesaysjust one bastard channel
21:16.18fugitivobluepuss
21:16.18Hmmhesaysor span i should say
21:16.27[TK]D-FenderHmmhesays : What kind of functionality do you get from it?
21:16.27harryvvHmm did you ever see that movie that just came out called north country?
21:16.35Hmmhesaysand i'll find out how successful as soon as a brit stumbles in here
21:16.36justinuit's like a PRI
21:16.47Hmmhesaysa sick PRI
21:16.53kink0like a PRI ?
21:17.00Hmmhesays30 channels
21:17.15kink0ahhh , about the number of channels.
21:17.23asteriskmonkeyshould be 32
21:17.24Hmmhesaysdass2 E1
21:17.29asteriskmonkey1 signalling right
21:18.03kink0Hmmhesays, what hard did you use to setup your bt ?
21:18.17Hmmhesaysthe only dass2 gateway I could find a quintum dx2030
21:19.38kink0ok, but you runs asterisk in a linux box , right ? and this linux box has a bt->USB , right ?
21:19.51Hmmhesaysum no
21:20.09Hmmhesaysasterisk-->lan-->gateway-->bT
21:20.34justinulol, some of you are talking about bluetooth (bt), and someone else is talking about british telecom (bt)?
21:20.57Hmmhesaysoh HAHA
21:20.59kink0justinu, I means bluetooth
21:21.06Hmmhesaysi'm talking british telecom
21:21.09justinulol
21:21.12justinuvery confusing :)
21:21.13Hmmhesaysbritishtooch?
21:21.14kink0lol !!!
21:21.17saftim talking brick tamland!
21:21.20HmmhesaysLOL
21:21.25saftI love lamp!
21:21.41SpaceBassim 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
21:21.43SpaceBasssame lan
21:22.02asteriskmonkeyspacebass: is it 1.2?
21:22.07SpaceBassasteriskmonkey yeah
21:22.10asteriskmonkeythere is a bug reprot on that
21:22.22asteriskmonkeycheck the bugs.digium.com under iax i think
21:22.34SpaceBasssip trunking working?
21:23.12kink0anyone has experiences with Valiant GSM gateways ?
21:23.20Hmmhesaysugh yes
21:23.40kink0bad ?
21:23.49harryvvspace...is this to test it before deloying the box at the remote site?
21:23.59Hmmhesaysi want to jam a pen through the eye of the man that thought up a  voip/gsm gateway
21:24.22kink0Hmmhesays, why ? what happenes ?
21:24.38SpaceBassharryvv sorta... migrating to new box for my home setup but don't want to move zaptel hardware yet
21:24.41Hmmhesaysso many failure points
21:25.00kink0with Valiant or in general with any GSM gateway ?
21:25.03SpaceBassharryvv 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
21:25.09HmmhesaysGSM in general
21:25.15Hmmhesaysespecially in the middle east
21:25.46kink0Hmmhesays, where you are from ?
21:25.49saftsacksomeone of you who is running asterisk not as rootß
21:26.03Hmmhesaysgeebus kink0 pay attention
21:26.08Hmmhesaysi'm from Minnesota
21:26.27kink0ahh, here Spain.
21:26.45Hmmhesayshow about you give me a phone number and you tell me what callerid shows up
21:26.54kink0I have read you need even to rotate IMEI on USA, or you are blocked.
21:27.30kink0Hmmhesays, do you need my phone number ? I can see you caller id at my mobile, if you want try.
21:27.35trixteron gsm you need to rotate imei in the US?  why would they block you if you dont?  I didnt catch that part
21:27.37Hmmhesayssure you don't have to answer
21:27.46kink0( I will not pick up, so will cost nothing )
21:27.50kink0ok... ok.
21:28.00LaibschWhere do I get a ztdummy module for a 2.6 kernel?
21:28.13LaibschDebian 2.6 kernel, stock version.
21:28.28trixterapt-get install asterisk
21:28.31trixterthat includes ztdummy
21:28.39dippocan anyone recommend a good IAX trunking service for high-minute usage?
21:28.43brimstoneanyone called international with nufone?
21:28.58kink0I will not answer, but don't send over 5 ring
21:28.59dippowe 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)
21:29.11trixternufone doesnt relaly like to let people call outside US/CA since they got taken for $450k in international calls
21:29.23trixterif you are a grandfathered customer they might let you
21:29.35saftsacksomeone of you who is able to show me howto give a user permissions to use zaptel? :)
21:29.56trixteror 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)
21:30.13Laibschtrixter: Hm, OK but then I won't have amportal which I would like to have for the beginning.
21:30.20LaibschI will try on a second machine.
21:30.25brimstonetrixter: i had heard about that
21:30.34brimstonehas anyone tried with iax.cc?
21:31.36*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
21:31.38trixteryes
21:31.42trixterwith sixtel/iax.cc
21:31.52brimstonetrixter: i can't seem to get this call to germany to go through
21:32.00trixtersome of their routes dont work that they advertise though..  +448xx for example, afganistan doesnt work afaik either
21:32.05trixterbut most do for the advertised rate
21:32.23trixterbasically their upstream providers (all of em) wont let them call anything where there is a variable rate for the call
21:32.36trixterthey can turn it on for select customers though
21:32.45trixtertalk to chris sixtel9@aim iirc
21:32.47justinui called north korea with level3
21:33.05trixterbrimstone what exactly are you trying to dial and with which provider?
21:33.13trixtermsg me if you dont want the number open
21:33.31trixteror hit up sixtel9@aim that is their tech support IM acct :)
21:33.50brimstonetrixter: just trying to call a girlfriend on holidday
21:33.54xachensixtel
21:33.56xachenis evil
21:34.07Hmmhesaysi had a girlfriend one
21:34.09Hmmhesays*once
21:34.13xachenI wouldn't recommend it *period*
21:34.26justinuHmmhesays: were you drunk?
21:34.30Hmmhesaysi'm a confirmed bachelor now
21:34.37Hmmhesaysother than that engaged chick i'm seeing
21:34.44trixterwhat specifically is evil about them?
21:34.47*** join/#asterisk kpettit (n=keith@69.15.174.114)
21:34.49trixterI havent had any problems with them
21:34.54xachendude
21:34.59xachenThey never answered their phones for months
21:35.05justinueveryone here complains about them
21:35.06xachenI filed the first BBB complaint in April
21:35.08xachennow there is 4 more
21:35.09trixteroh I have no problems getting hold of people there
21:35.14xachenthey didn't anwer ANY of them
21:35.23xachenand their toll free numbers I had stopped working for me
21:35.24xachenwith a busy tone
21:35.29Hmmhesaysuk anyone?
21:35.30xachenI wans't even getting a connection to my server
21:35.41xachenI opened many "emergency" tickets
21:35.46xachenFinally just ran chargeback
21:35.55trixterI havent had to open any ticket I just ask for it to be done and it is
21:36.10xachenthey must favour you then
21:36.23xachenI first found about them @ pastebin.ca :P
21:36.27trixtermy friend has had the same experience
21:36.35xachenthe owner of pastebin.ca took their ad down cause they didn't pay for their ads and had outstanding fees
21:37.04trixterI dunno about that, sounds a little sketchy that the owner would tell you that
21:37.19netvulturevoicemail 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.
21:37.19xachenI know the owner :P
21:37.23harryvvmight be a good site to advertise voip equipment for sale.
21:37.43harryvvhow many of you here go though xo for termination?
21:37.53trixterahh well my friend is friends with a higher up at sixtel :P
21:37.54netvultureThe caller from teliax can hear the vm prompt just fine.
21:38.12xachennetvulture: Make sure you set your codecs int he account settings rea
21:38.14trixterwhy 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
21:38.17trixter:D
21:38.18harryvvtrixter, i have sixtel...not been the most reliabile service
21:38.27kink0harryvv, what site ? I am seeking for PRI(E1) -> GSM gateway
21:38.38netvultureI have set teliax to use onlu ulaw
21:38.55xachentrixter: Send a message of word off. Tell them to answer their phones to actual customers who have problems :P
21:38.57netvultureboth on my asterisk side and via the web portal at teliax
21:39.00trixtercourse the fact  that I am pushing major traffic through them doesnt hurt either
21:39.02harryvvxo is a voip carrier that is a large backbone..or am I not correct on that?
21:39.15SpaceBassarrruuggg stupid trunking
21:39.26trixterxo is a clec and has a fiber backbone
21:39.26xachennetvulture: What is your voicemail.conf settings?
21:39.43trixterthey do voip as well, and if you do enough traffic xo will give you $0.0035/min to most of the US
21:39.55trixterwell population wise most anyway
21:40.00xachenformat=wav49|gsm|wav
21:40.01netvultureformat=wav49|wav
21:40.05xachenhmm
21:40.23netvulturei think wav49 is gsm isn't it
21:40.29harryvvtrixster really...how much traffic are we talking about?
21:41.01trixterhmm..  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
21:41.03xachennetvulture: Just try wav
21:41.09trixterhopefully cuase that means that I will have em really soon :)
21:41.32netvulturetrying now - thanx
21:41.57trixterharryvv: 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
21:42.13xachenthey never honoured their unused portion refund policy
21:42.16xachenso they have no trust with me
21:42.20trixtersince 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)
21:43.27xachenanyways bbiab
21:43.44SpaceBassanyone know if nufone allows setcallerid name?
21:43.57Hmmhesaysuk uk uk uk
21:44.15netvultureno dice - while ringing all formats are 4 - vm answers write=2 - vm starts record - write=4
21:45.28jake1932using a TE100p - can anyone tell me why i would receive double digits intermittantly (00 instead of 0)?
21:45.49netvulturevm is definatly changing codec to 2 - i have a answer - wait(1) in place and codec stays at 4 until vm is called
21:49.42SpaceBassso, is IAX2 trunking broken in 1.2? mine is not obeying my incoming routing and my outgoing rings all circuits busy
21:54.39trixterwoo 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
21:54.52marcus2eww gxp2000s
21:55.27trixterfree prizes and they are really the lowest end phone I would use
21:55.55trixterat least thevoipconnection.com supports the community
21:55.56trixter:)
21:55.59justinuthey're ok for testing and such
21:56.02justinuand basic users
21:56.18trixterfor $85 they arent bad
21:56.23justinuyeah
21:57.13trixteris the $35 savings for a bugetone 10x really worth it?
21:57.29justinuthe budgetone is a real piece of crap
21:57.31justinuvery flimsey
21:57.45justinumight be something you'd see on the desk of a 13 year old girl
21:58.00justinumy first telephone
21:58.18trixterfirst?  at 13?
21:58.21trixterha!
21:58.33trixterat 13 they are spending no less than 4 hours a day on the phone
21:58.44justinuhahah, ok, 7 year old then
22:01.39*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
22:01.57Seldon19751I'm gonna get my self connected
22:02.18Seldon19751.. 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)
22:05.09marcus2the question is, is the gxp2000 worth the $30 savings over a polycom 301 ;)
22:06.42justinudoes 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
22:07.09trixterI 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
22:07.12justinuthe 941 doesn't do PoE?
22:07.17[hC]Nope
22:07.23justinuwow, very lame
22:07.25[hC]Just tried
22:07.30[hC]its basically a sipura in a phone
22:07.33LostFrogDid it smoke?
22:07.36justinuyeah, 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.
22:07.59LostFrogDamn.. I was looking for a little excitement. :)
22:08.00justinu802.3af 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)
22:08.33justinuspeaking 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
22:09.19justinui have a dlink des-1526, won't power the polycom501 with a straight patch cable
22:09.34justinunfw 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
22:09.55justinui didn't come across that in my search
22:09.57darwin_35its punk magic
22:10.01justinu(i did for cisco 7960)
22:10.22[hC]search for poe cable maybe
22:11.24trixterjust make sure its not a pos cable
22:11.27trixterthose arent quite the same
22:11.56justinuany cable I make turns out to be a PoS cable ;)
22:11.57*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)
22:12.08trixterheh
22:15.00*** join/#asterisk loick (n=loick@APuteaux-151-1-65-183.w81-249.abo.wanadoo.fr)
22:15.21darwin_35how 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)
22:16.46saftsackhi
22:17.33*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
22:17.52trixterdarwin: 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
22:17.59trixterothers dont if you cant do that then you are stuck
22:18.48mog_workanyone in germany
22:18.52mog_worki need to  terminate a call
22:18.58mog_workjust one ^_^
22:20.01LostFrogOne 900 Hour call?
22:20.30trixtermog: I have a german number if that will help ya
22:20.44trixterI kinda wanna test that it works too :P
22:21.36mog_workno i need someone with a box in germany
22:21.43mog_worki 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)
22:24.18trixteryou need to place a call to a specific number in germany?  or what?
22:24.25trixterI am not quite understanding what it is that you want
22:25.33mog_worki was trying to help a friend call his girlfriend who is in germany at the moment
22:25.39trixterahh
22:25.40mog_workbut jerjer hooked me up
22:25.46trixtervoipbuster.com I believe gives free to germany
22:26.12*** join/#asterisk marv[work] (n=timr@64.89.118.139)
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22:28.34*** part/#asterisk chapeaurouge (n=chap@85.201.81.201)
22:29.06maarkenhas 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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22:34.12dippoanyone know of a good cordless SIP phone?
22:35.32wasimdippo: yeah, any good cordless phone with a good ata
22:35.36SkramXcordless 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)
22:39.44l1nuxhi :)
22:41.23l1nuxasterisk- Jingle not ready ? :P
22:41.31jake1932is 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)
22:41.56l1nuxhttp://code.google.com/apis/talk/index.html
22:42.01*** join/#asterisk RoyK (n=roy@host-n39-140.homerun.telia.com)
22:42.01harryvvdippy hitachi
22:42.08RoyK~seen zoa
22:42.19jbotzoa is currently on #asterisk (1d 12h 53m 43s).  Has said a total of 119 messages.  Is idling for 7h 42m 10s
22:42.21harryvvhitachi wireless voip wifi phone
22:45.15dippowasim: well, i was sorta hoping i wouldn't have to go that route
22:45.55dippoi was hoping to find a phone with a voip base with 10baseT that then 5.8GHz analog to a handset or something
22:45.59dippomaybe there's just not a market for that
22:46.16Remosil1nux, there are people working on it
22:46.36l1nuxgood news :D
22:46.42*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
22:46.49l1nuxthanks Remosi :)
22:47.33Remosinp
22:47.42maarkenor just get a ATA and a regular 5.8Ghz. :)
22:47.52dippoyeah
22:48.17dippoi guess those aren't as pricey as I thought
22:48.27dippo$60 for a single FXS sound about right?
22:48.41[TK]D-Fender$70 for dual
22:48.51l1nuxRemosi, where get recent news about asterisk-Jingle ?
22:49.07Remosino idea
22:49.13l1nuxok :)
22:49.34*** join/#asterisk deb_user (n=frank@71-36-59-120.albq.qwest.net)
22:49.52deb_userhello all
22:49.59deb_useranybody up for talking a little about echo?
22:50.25deb_useri 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-Fenderwhich?
22:51.03|Vulture|anyone here work with XO Flex package yet?
22:51.05deb_userright 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-FenderWhat ver of * and what settings in zapata?
22:51.28|Vulture|deb_user: adust rx/tx
22:51.45deb_uservulture, I've tried that
22:51.54deb_userbut I'm not quite sure what I'm doing
22:52.00dogtaniananyone 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
22:52.16deb_uservulture, how do I get ztmonitor working?
22:52.46dippo<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
22:54.18deb_useri see noise, vulture
22:54.28deb_userand 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
22:54.36dogtanian:/
22:54.41|Vulture|deb_user: you shouldn't see noise with * running either
22:54.45dogtanianis it easy enough to get hold of the firmware?
22:55.09|Vulture|dogtanian: if you have a license agreement with cisco... otherwise no
22:55.28dogtanianah
22:55.29deb_uservulture: 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
22:55.41deb_userstatic?
22:55.53dogtanianany 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
22:56.08dogtaniani was thinking of buying from ebay(uk)
22:56.22deb_userhmmm...
22:56.47deb_userso 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)
22:57.16deb_useri suppose I could try changing the cord from the TDM to the wall jack
22:57.17|Vulture|deb_user: pastebin your zapata.conf
22:57.24deb_userok
22:57.28|Vulture|deb_user: what card do you have in there?
22:57.29dogtanianooh. are the features as good as the 7960? perhaps i'll buy one of those instead :)
22:57.44deb_userits a clone X100P
22:57.51harryvvare there any routers that offer duplex wifi capability?
22:57.51deb_useri 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
22:58.09dogtaniancheers!
22:58.24deb_uservulture: 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-Fenderdippo : recommendations for... ?
22:58.55|Vulture|the newest zaptel drivers 1.2.1 are actually really nice on the TDMs
22:59.09FuriousGeorgei 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
22:59.11deb_uservulture: those are the drivers I'm using
22:59.14FuriousGeorgeis that true?
22:59.27deb_uservulture: 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?
23:00.07dippo[TK]D-Fender: a good dual FXS adapter. but I think I found one .. the linksys Sipura SPA-2002 for $70
23:00.09deb_uservulture: don't understand the question
23:00.10dippowhich seems to be pretty decent
23:00.24|Vulture|deb_user: is this for a home or office?
23:00.28deb_uservulture: office
23:00.39|Vulture|how many phones and how many phone lines?
23:00.41deb_uservulture: the one I've got now is at home...but its just a starter to learn on
23:00.59deb_uservulture: that's a tough question to answer....
23:01.11|Vulture|expansion is the real question here
23:01.15deb_uservulture: we've got a legacy pbx system that I need it to interface with
23:01.28deb_uservulture: so do all those phones count too?
23:01.53[TK]D-Fenderdippo : 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
23:02.18deb_uservulture: but the analog phone lines plug straight into the pbx
23:02.28deb_userthe 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
23:02.55deb_useri 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
23:03.23deb_uservulture: I believe you
23:03.42deb_userI 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
23:04.25deb_userWell, that would be a tough sell in my office
23:04.26|Vulture|now we have 8 offices all interconnected
23:04.31FuriousGeorgei guess ill find out how vmware does
23:04.36deb_userbecause we sank at least $5,000 into the pbx only two years ago
23:04.56deb_userso 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?
23:05.06deb_uservulture: not even
23:05.13deb_uservulture: its all digital, vodavi
23:05.21|Vulture|hmm not familliar with them
23:05.33deb_uservulture: 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?
23:06.01deb_userinterconnection of international offices
23:06.07deb_userfree voice mail
23:06.22deb_userwell, there's tons of reasons to go *, I don't have to tell you
23:06.28deb_userits 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.
23:07.11deb_userI don't think seamless integration will be possible
23:07.17|Vulture|agreed
23:07.27deb_userat least, I'm not willing to invest the time and effort
23:07.43|Vulture|I wouldn't either
23:08.05deb_userbut, 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
23:08.34deb_userwhy?
23:08.36trixterchange your business?  they said that about the internet and it didnt change anything :P
23:08.44deb_usera 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
23:09.04deb_userFXO in...FXS out
23:09.30deb_userwell, over four we could just hit another TDM, que no?
23:09.39deb_userthen up to 8 lines
23:09.45|Vulture|you need 1 FXO and 1 FXS per inbound line
23:09.52deb_userright
23:09.57|Vulture|which means for 4 inbound lines you need 2 TDM cards
23:10.11deb_userreally?
23:10.13|Vulture|and from everywhere I have seen they say not to use more than 2 TDM cards
23:10.14mog_workor tdm2400p...
23:10.27|Vulture|mog_work: oh is that the new breakout one?
23:10.29deb_useroh, sure
23:10.30[TK]D-FenderThat makes no sense....
23:10.31deb_userthat makes sense
23:10.34mog_workyes
23:10.39|Vulture|[TK]D-Fender: why not?
23:10.52deb_userThe 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-FenderWhy on earth would you need 1 FXS per FXO for incoming lines?
23:11.12deb_userFender: I think vulture is right
23:11.13dogtanian|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-FenderOH... 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
23:11.34deb_userFender: haha...i've gotta do the best with what I've got!
23:11.53|Vulture|dogtanian: not for polycom
23:11.59deb_uservulture: so tell me about channel banks
23:12.01|Vulture|dogtanian: for cisco yes
23:12.08deb_userthus far I've only looked at TDMs
23:12.43[TK]D-FenderChannel 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
23:12.53jake1932mog_was I talking to you about the T100P issue a few minutes ago?
23:13.07dogtanian|Vulture|: bah :/ any clues as to what I else I might need the license for?
23:13.18deb_uservulture: 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
23:13.44deb_userbut i don't understand the whole T1 concept
23:13.56deb_userwhat gets plugged into the T1 card?
23:13.57jake1932i 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
23:14.11deb_userohhhh
23:14.22deb_userso I plug all my analog lines into the bank?
23:14.31deb_userand 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
23:14.46deb_userand that gets plugged into the t1 card?
23:14.50|Vulture|that plugs into a T1 card
23:14.59deb_userbet the sound quality is fantastic
23:15.09jake1932does 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
23:15.32deb_userso I have all my incoming analogs plugged into the FXO mods
23:15.53deb_userand then the outgoing lines go to the legacy PBX via FXS
23:15.53dogtanian|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
23:16.24deb_userand then the entire channel bank plugs into the t1 card
23:16.34deb_userwell, 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
23:17.57deb_userok
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
23:19.17harryvvvulture like what modeles
23:19.34|Vulture|harryvv: the FXO modules for the TSU-600
23:19.36deb_useryeah...I see it just comes with FXS mods
23:19.38|Vulture|the used ones
23:19.45deb_userno mention of FXO
23:19.54|Vulture|deb_user: if you call them they will find some for you
23:20.26deb_uservulture: 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
23:20.48deb_userlet 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 *
23:21.07netvultureIts nice to see another Vulture in town
23:21.15|Vulture|lol sup netvulture
23:21.31deb_userand then, if its really paying off on that one line
23:21.41deb_userwe'll add a channel bank
23:21.51deb_userand go pure voip
23:22.01|Vulture|deb_user: you can fix those issues with the TDM
23:22.05|Vulture|1s
23:22.09deb_uservulture: you have a voip service provider for outbounds
23:22.11deb_user?
23:23.17deb_userwhat 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
23:26.39deb_useri've tried that
23:26.49|Vulture|really? hmmm
23:26.55|Vulture|like -1?
23:26.58deb_userbut 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
23:27.45deb_userone sec
23:27.56deb_useroh yeah
23:28.01deb_userRX is almost halfway!
23:28.19|Vulture|if it were TX I would say deff the card
23:28.42deb_userTX is quiet
23:28.42deb_usernothing on TX
23:28.47deb_userbut RX is really active
23:29.11|Vulture|adjust rxgain=-1
23:30.02deb_userok
23:30.08deb_userleave txgain at what?
23:30.13deb_userdefault?
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
23:32.57deb_userone 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)
23:34.40deb_usermodprobe 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
23:36.00deb_useraright
23:37.02*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
23:37.48CunningPikeDoes 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)
23:39.06feliskHello everyone !
23:39.34feliskI bought an hardware SIP phone ! BeWAN VoIP Phone S2
23:39.38feliskI have strange question
23:39.55feliskwhen I call my phone from cellular, everything is ok
23:40.09feliskbut when I call from SIP phone the cellular
23:40.17feliskI have quite big latency
23:40.26feliskany ideas where to search ?
23:40.58jake1932if 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
23:42.08deb_userno 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
23:42.37feliskwhy can I have latency when calling from inside and not when called ??????
23:42.40deb_usertake care
23:42.42deb_userthanks for your help
23:42.44|Vulture|np
23:45.26marcus2Dec 15 16:20:11 ryle zaptel Disabled echo canceller because of tone (tx) on channel 55
23:45.31marcus2what 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)
23:55.43harryvvSayDigit works for 0-9 what about *#?
23:57.06*** join/#asterisk Primer (n=vi@sh.nu)
23:57.43PrimerAnyone 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
23:57.57PrimerI have the debug server and upgrade server running
23:58.25*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
23:58.26Primerthe debug server shows things, so it's definitely communicating
23:58.58Kattyhi.
23:59.25*** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)

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