| 00:06.12 | *** join/#asterisk Understudy (n=bhorne@adsl-065-012-216-193.sip.bct.bellsouth.net) |
| 00:06.20 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 00:07.15 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:07.22 | Understudy | Hello |
| 00:07.55 | Understudy | Is Asterisk HIPAA complaint? |
| 00:08.06 | _DAW | are you kidding? |
| 00:08.24 | Understudy | No |
| 00:08.31 | _DAW | chan_compliance |
| 00:08.52 | Understudy | I am trying to find documentation that shows it is HIPAA compliant |
| 00:10.04 | Exstatica | ok i'm having odd errors |
| 00:10.32 | Exstatica | http://pastebin.ca/456761 |
| 00:10.49 | Exstatica | getting Returned from dialparties with no extensions to call |
| 00:12.32 | tessier_ | Security vulnerability in asterisk's T38 handling. Wow. Asterisk does T38 now? I've been out of the asterisk loop for a while. Last I heard asterisk would likely never do T38. |
| 00:12.45 | _VoiceMeUp_COM | prolyl 1.4 |
| 00:12.48 | _VoiceMeUp_COM | or not |
| 00:12.56 | tessier_ | Understudy: What would it need to be HIPAA compliant? |
| 00:13.26 | Understudy | Because it is in a medical faclity |
| 00:13.47 | tessier_ | Understudy: Does everything in a medical facility need HIPAA compliance? |
| 00:14.00 | BSD_Tech | what is all the stop retransmission crap |
| 00:14.09 | Understudy | Any data storage system must be HIPAA compliant |
| 00:14.15 | tessier_ | Understudy: Paper? |
| 00:14.19 | _DAW | How do the phones fall into HIPPA? and what are you storing on the PBX? |
| 00:14.20 | tessier_ | Understudy: Does asterisk store data? |
| 00:14.29 | Understudy | voicemail |
| 00:14.41 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 00:14.43 | tessier_ | Understudy: What are the requirements of HIPAA with respect to voicemail? |
| 00:14.58 | tessier_ | Understudy: it may be HIPAA compliant. You'll have to understand what is required by HIPAA and check for yourself. |
| 00:15.24 | _DAW | Compliance mainly revolves around your ability to report on what you are supposed to be doing. |
| 00:15.53 | Understudy | The idea is that the system is listed as HIPAA complaint just like it is SIP compliant |
| 00:16.02 | _DAW | Then no.. |
| 00:16.15 | tessier_ | Understudy: Listed by who? |
| 00:16.51 | tessier_ | Understudy: Nobody has listed it as SIP compliant except the asterisk coders. Are you saying that if they just put "HIPAA compliant" on the website it would solve your problem? |
| 00:16.52 | Understudy | http://www.hipaadvisory.com/tech/voip.htm |
| 00:17.02 | tessier_ | ah, a url. |
| 00:17.16 | russellb | *nobody* is SIP compliant by the standards |
| 00:17.20 | russellb | there are too damn many of them ... |
| 00:17.31 | russellb | and many are vague, and many conflict .. |
| 00:17.38 | tessier_ | Understudy: Hmm...that doesn't say what is required for compliance. |
| 00:18.18 | Understudy | I am trying to find the exact listing. The customer is askig me if asterisk is HIPAA compliant. I don't know if it is or not |
| 00:18.36 | tessier_ | Understudy: First you have to know what it means to be HIPAA compliant. |
| 00:19.16 | *** part/#asterisk magikxx (n=tawandax@209.88.93.62) |
| 00:19.37 | Understudy | I was hoping the Asterisk already was. because I am not sure myself |
| 00:20.03 | Understudy | http://www.vonage-forum.com/ftopic8194.html |
| 00:22.55 | _VoiceMeUp_COM | lol |
| 00:23.01 | _VoiceMeUp_COM | you can use SRTP yes |
| 00:23.07 | _VoiceMeUp_COM | but man |
| 00:23.16 | _VoiceMeUp_COM | that if you have no regular phone in house |
| 00:23.25 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 00:23.29 | _VoiceMeUp_COM | coz they can lift your hook when they want |
| 00:23.54 | Understudy | http://articles.techrepublic.com.com/5100-1035_11-6134488.html Here it shows what HIPAA is |
| 00:23.56 | _VoiceMeUp_COM | i have the nsa cpecs somewhere , and they do require 7u in a rack space for them in case they need to carnivore you |
| 00:25.29 | _DAW | Everything in that last doc is related to best practices. You can take any system and make it vulnerable through a poor implementation. |
| 00:25.39 | flewid | [Apr 24 19:42:01] WARNING[32539]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 |
| 00:25.44 | flewid | are those warnings safe to ignore? |
| 00:25.56 | Sweeper | man, the asterisk wikipedia page needs more pictures |
| 00:26.13 | _DAW | Compliance is more than a stamp of certification. It is a whole methodology. |
| 00:27.04 | flewid | h3h |
| 00:27.10 | flewid | okay so apparently something else is the issue |
| 00:27.18 | flewid | i just installed 1.4 remote and it won't register there either |
| 00:27.25 | flewid | so again i'm back to the nat issue i guess |
| 00:27.27 | flewid | damn cisco! |
| 00:28.11 | BSD_Tech | cico drools |
| 00:28.12 | Understudy | Yeah , but the problem is the doctors office is requiring a HIPAA listing for the sytem. |
| 00:28.24 | BSD_Tech | firewall |
| 00:28.30 | BSD_Tech | routing |
| 00:28.32 | flewid | there's no fw on the remote box |
| 00:28.46 | flewid | local i'm behind a soekris + pfsense, connected to a cisco 2950 with a vlan for the phones |
| 00:28.54 | flewid | and as i said before, softphone registers dine |
| 00:28.55 | flewid | fine |
| 00:30.02 | _DAW | Well, there is no HIPAA seal of approval for asterisk. |
| 00:30.24 | BSD_Tech | hipaa? |
| 00:30.47 | _DAW | Health care info regulations.. |
| 00:30.55 | aydiosmio | health privacy regulations |
| 00:31.06 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:31.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:31.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 00:32.51 | BSD_Tech | I would not use asterisk in a hospital or medical instatution as the main pbx |
| 00:33.02 | *** join/#asterisk cv_guest (n=chatzill@71.36.229.227) |
| 00:33.20 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:33.20 | BSD_Tech | not with out 4 months of ongoing testing and making sure you had proper fail over |
| 00:33.33 | cv_guest | Is anyone in here famliar with writing a program to test for udp latency |
| 00:34.03 | Understudy | So basically Asterisk is not HIPAA compliant than. Bummer I won't get this contract. |
| 00:34.33 | _VoiceMeUp_COM | yeah |
| 00:34.42 | _VoiceMeUp_COM | imagine a deadlock while taking a ambulance clal |
| 00:34.43 | _DAW | It could be if you implemented in a compliant way. But there is no seal of approval. |
| 00:34.43 | _VoiceMeUp_COM | lol |
| 00:34.54 | _VoiceMeUp_COM | i would use ciscos |
| 00:35.19 | BSD_Tech | Nortel |
| 00:35.34 | _VoiceMeUp_COM | yeah |
| 00:35.36 | _VoiceMeUp_COM | anything |
| 00:35.40 | _VoiceMeUp_COM | that is not software only |
| 00:35.48 | BSD_Tech | agree |
| 00:35.55 | _DAW | asterisk can be VERY reliable if set up correctly |
| 00:36.01 | flewid | grr |
| 00:36.14 | flewid | * see' the phone, sip show users/sip show peers sees it, it just won't register |
| 00:36.17 | flewid | ehwtf |
| 00:36.21 | Understudy | I don't doubt it's reliablity. I worry about auditors |
| 00:36.23 | BSD_Tech | yes it can but in that instance where it fails it could cost the medical facility millions |
| 00:36.34 | _VoiceMeUp_COM | imagine a nurse calling a code blue on intercom.. deadlock or crash... patient dies.. IT company sued.. ... IT guys gets heart attack. goes to hospital.. nurse calls a code blue... back to square one |
| 00:36.52 | _DAW | Reporting would be a big issue. How would you prove you are doing it correctly even if you were? |
| 00:36.52 | _VoiceMeUp_COM | still we saw last month on 1 packet could bring it down |
| 00:37.27 | _VoiceMeUp_COM | but for support staff i guess its ok.. |
| 00:37.31 | _VoiceMeUp_COM | accoutning, etc |
| 00:37.37 | _VoiceMeUp_COM | just not in the ER |
| 00:37.40 | BSD_Tech | welcome to the world and the world gets smaller |
| 00:37.44 | _VoiceMeUp_COM | ;) |
| 00:37.45 | _DAW | Protect your SIP.. |
| 00:39.42 | _VoiceMeUp_COM | btw |
| 00:40.00 | _VoiceMeUp_COM | therse now way to use group() |
| 00:40.04 | _VoiceMeUp_COM | its makes deadlocks.. |
| 00:41.26 | _VoiceMeUp_COM | not all the time .. but if manager i guess ( cacti script) makes a show channels) then if its tiemd with a detruct of channel it dies |
| 00:42.01 | _VoiceMeUp_COM | comwave commercials are soooooooooooo funny |
| 00:42.16 | _VoiceMeUp_COM | 4 digit marketing budget |
| 00:42.21 | _VoiceMeUp_COM | 19.99 |
| 01:08.36 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:18.45 | fbffff | not to mention the 3 security vulns that were just reported and the large much of crash deadlocks vulns in the bug tracker |
| 01:18.53 | fbffff | s/much/amount |
| 01:38.40 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 01:47.14 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 01:48.46 | flewid | hi again - still unable to get cisco remote registration working - sip 401 unauth - my phone is behind a nat router, trying to register to a remote server that is behind no nat |
| 01:48.48 | flewid | using 1.4.2 |
| 01:49.01 | flewid | phone registers fine locally behind my own firewall just not to remote boxes it appears |
| 01:49.45 | BSD_Tech | ok |
| 01:49.50 | BSD_Tech | in the sip.conf |
| 01:50.10 | flewid | sec i'll paste to pasebin |
| 01:50.14 | BSD_Tech | onthe exten you want to register put nat=yes and qualify=yes |
| 01:50.42 | BSD_Tech | just to help it along |
| 01:51.13 | flewid | yup, i've tried nat=yes, nat=no, nat=never, and qualify=yes/no |
| 01:51.24 | flewid | ast does see the phone when i do sip show peers and sip show users |
| 01:51.28 | flewid | though it just wont' register |
| 01:51.52 | BSD_Tech | does it say 1000/1000 or jusr 1000 |
| 01:51.56 | flewid | on my firewall here, i have port 5060 tcp/udp portforwarded to my ast box, and 10k-20k udp for RTP also port forwarded |
| 01:51.57 | flewid | sec |
| 01:52.11 | BSD_Tech | if just 1000/ then it does not see the phone |
| 01:52.21 | BSD_Tech | it just seees you have built a exten |
| 01:52.33 | flewid | yup, it sees 125/125 |
| 01:52.53 | flewid | let me turn nat=yes and qualify=yes |
| 01:52.54 | BSD_Tech | sip show peers shows it |
| 01:53.17 | *** join/#asterisk hfb (n=hfb@adsl-75-39-161-90.dsl.irvnca.sbcglobal.net) |
| 01:53.30 | flewid | sip show peers and sip show users i see the ext |
| 01:53.36 | flewid | sip show peers is ext/ext |
| 01:53.40 | flewid | sho show users is just ext |
| 01:54.39 | BSD_Tech | ok then its registerd |
| 01:54.49 | BSD_Tech | if it shows 125/125 |
| 01:54.59 | BSD_Tech | sip show registry wont show a thing |
| 01:55.00 | flewid | can't make any calls, sip debug shows me "401 unauthorized" if i turn debugging on |
| 01:56.23 | flewid | SIP/2.0 401 Unauthorized |
| 01:56.25 | flewid | :/ |
| 02:02.16 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:05.27 | kusr | How do I debug failed re-INVITEs? Sound disappears in one direction. Who is to blame? |
| 02:05.51 | tzafrir | NAT |
| 02:05.58 | kusr | Really? |
| 02:06.49 | kusr | But the call works before it is reconnected (across NAT) then it fails after the INVITE (no NAT). |
| 02:06.55 | tzafrir | is there any NAT in the picture? |
| 02:06.58 | kusr | Yes. |
| 02:07.05 | kusr | My server is NAT:ed. |
| 02:07.29 | kusr | But it transmits its proper public IP in SIP messages. |
| 02:07.30 | tzafrir | SIP (the control connection) works file. The problem is with the secondary RTP connections |
| 02:07.39 | kusr | Yes. Precisely. |
| 02:08.11 | kusr | But is not the IP to use for RTP in the SIP messages. |
| 02:08.14 | kusr | ? |
| 02:08.39 | tzafrir | But the server on the other side needs to open a new connection back to that phone/server/whatever |
| 02:09.02 | kusr | AFAIK there should not even be a way to see that I am behind a NAT? |
| 02:10.00 | tzafrir | no. But if I try to connect to a port on the NAT server it won't magically connect to a port on your computer |
| 02:10.10 | kusr | We have server 1 (public Internet), server 2 (public Internet) and my server (NAT - but it sends its public IP in all SIP messages). |
| 02:10.27 | kusr | Ports are forwarded. |
| 02:10.34 | tzafrir | can yoou set nat=yes in the public server? |
| 02:10.44 | kusr | 5060 and the RTP ports. |
| 02:11.02 | kusr | I only control my server, behind the NAT. |
| 02:11.33 | kusr | I was thinking maybe one of the other servers simply do not support redirection? |
| 02:11.57 | kusr | My firts thoughts were entirely about NAT, but now I'm not so sure. |
| 02:13.36 | kusr | Then again, perhaps I should look at the SIP debug output once more... |
| 02:15.14 | kusr | tzafrir: Thanks. I'll look at it again tomorrow... |
| 02:15.38 | tzafrir | kusr, also look at the nat/sip page on pkg-voip |
| 02:16.20 | kusr | Where is that? |
| 02:16.35 | *** join/#asterisk rmayorga (n=rmayorga@168.243.73.11) |
| 02:19.01 | kusr | tzafrir: The debian team? That's what Google finds for pkg-voip... |
| 02:19.21 | tzafrir | sorry, I mean the wiki: voip-info |
| 02:19.36 | kusr | OK. Thanks. |
| 02:19.43 | kusr | Bye for now! |
| 02:20.34 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:22.18 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:22.21 | tzafrir | http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:22.27 | tzafrir | too late |
| 02:22.32 | tzafrir | ~sipnat |
| 02:22.34 | jbot | it has been said that sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:22.40 | tzafrir | duh |
| 02:26.32 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:26.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:29.02 | BSD_Tech | its like this |
| 02:29.25 | BSD_Tech | I am he who was is and will be |
| 02:32.03 | *** join/#asterisk flewid (n=flewid@216.145.22.66) |
| 02:36.34 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:39.30 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 02:40.30 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:40.52 | shmaltz | hi, is a PCI-X 64-bit compatible with Digium cards? |
| 02:43.16 | shmaltz | test, do you hera me? |
| 02:43.22 | shmaltz | *hear |
| 02:44.20 | fakhir | yes we do |
| 02:45.26 | shmaltz | fakhir, you have any clue if a PCI-X 64-bit slot will work with digium cards? |
| 02:46.04 | *** join/#asterisk voltagex (n=voltagex@124-254-99-2-dsl.ispone.net.au) |
| 02:46.08 | fakhir | nope thats why i did not answer ;) |
| 02:46.18 | voltagex | are there any sip/iax providers in Australia that do not suck? |
| 02:46.21 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:48.50 | dfgas | is there a service to where i can just call like 800 numbers? that is free |
| 02:49.27 | voltagex | dfgas: FreeWorldDialup |
| 02:49.43 | voltagex | SIPBroker or Voxalot |
| 02:49.45 | voltagex | there's plenty |
| 02:49.46 | dfgas | 800 numbers will work on there? |
| 02:49.46 | voltagex | Skype |
| 02:50.14 | dfgas | like for example calling att? |
| 02:50.41 | voltagex | 1800callatt? |
| 02:50.46 | voltagex | yep, sipbroker or voxalot |
| 02:50.57 | voltagex | or a regular phone :P |
| 02:51.35 | JerJer[mobile] | IAXTel used to have a toll-free gateway |
| 02:51.43 | voltagex | IAXTel still do |
| 02:51.44 | voltagex | them too |
| 02:51.46 | dfgas | calling to a reg phone, FWD says toll free numbers need to proceed by an asterisks then it hangs up |
| 02:52.00 | voltagex | yes |
| 02:52.00 | dfgas | hmmm |
| 02:52.02 | JerJer[mobile] | yes |
| 02:52.03 | voltagex | so *1800CALLATT |
| 02:52.07 | dfgas | ahh |
| 02:52.18 | JerJer[mobile] | call western union :) |
| 02:52.31 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 02:52.36 | dfgas | lol |
| 02:52.38 | *** join/#asterisk vosque (n=vosque@ip68-3-20-129.ph.ph.cox.net) |
| 02:53.18 | dfgas | cool |
| 02:53.19 | dfgas | ty |
| 02:53.55 | voltagex | dfgas: they mean an actual asterisk, not Asterisk the PBX |
| 02:54.05 | dfgas | i know |
| 02:54.12 | dfgas | i got it :D |
| 02:55.29 | *** join/#asterisk troy_psx (n=tex@c95332b5.virtua.com.br) |
| 02:55.31 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 02:57.57 | zuez | hi all. just learning by trying out FWD .. can't properly register, receiving the following: [Apr 24 22:56:34] NOTICE[9466]: chan_iax2.c:7368 socket_process: Registration of '841968' rejected: 'Registration Refused' from: '192.246.69.186' |
| 02:58.09 | zuez | password is 100% correct, anyone familiar with this? |
| 02:58.39 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 02:59.21 | drfreeze | Can someone point me to some instructions on getting a TDM card going. I don't have my Asterisk book with me and don't remember how to start up the service |
| 03:00.23 | BSD_Tech | clear |
| 03:01.41 | JerJer[mobile] | drfreeze: download the book |
| 03:01.54 | drfreeze | JerJer[mobile]: have a URL? |
| 03:01.54 | Sweeper | are there any good papers on asterisk in the corporate environment? |
| 03:04.04 | blitzrage | www.asteriskdocs.org |
| 03:04.09 | blitzrage | drfreeze: ^^^ |
| 03:04.42 | drfreeze | blitzrage: cool, thx |
| 03:14.08 | Frosh | How would I make phone calls using Asterisk? |
| 03:14.22 | BSD_Tech | write dialplan to allow you to |
| 03:14.36 | BSD_Tech | get a sip provider |
| 03:14.36 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 03:15.15 | asteriskguy | [TK]Fender are you there? |
| 03:15.52 | BSD_Tech | nope he is awol |
| 03:15.53 | *** join/#asterisk NormanAthol (n=Norman@CPE-2032087183.bal.ncable.net.au) |
| 03:15.55 | dfgas | k, in asterisk, when you dial a 800 number is there a way to have asterisk add a * in front of it? |
| 03:16.10 | asteriskguy | yeah |
| 03:16.28 | BSD_Tech | *+ in your trunking rules |
| 03:16.52 | *** join/#asterisk Mavvie (n=edwin@ppp246-56.lns3.syd6.internode.on.net) |
| 03:16.52 | aptura | _*XXX or what ever. |
| 03:16.58 | asteriskguy | exten => _800NXXXXXX,1,Set(EXTEN=*+${EXTEN}) |
| 03:17.20 | aptura | _*800NXXXXXX,1,Set(EXTEN=*+${EXTEN}) |
| 03:17.40 | asteriskguy | exten => _800,n,Apps() |
| 03:17.42 | aptura | why would you and unless you strip that off it wont go though. |
| 03:18.08 | asteriskguy | well, he's asking if there's a way to add |
| 03:18.29 | NormanAthol | i ahve 2 pbx's set up and wanted to know if i connect them through iax will calls from *a to *b be streated as incomming calls on the destination server or will they be treated as ionternal calls |
| 03:19.34 | drfreeze | blitzrage: can't seem to find what I'm looking for. |
| 03:19.44 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 03:20.02 | drfreeze | blitzrage: Do you know how to get a TDM up and running - or can point me to some docs? |
| 03:20.15 | asteriskguy | anyone have setup HylaFAX + IAXMODEM + ASTERISK? |
| 03:20.43 | dfgas | k, last question, i have port 5060 open in my router, can someone use for say X-lite to connect to it so they are on my asterisk server? |
| 03:20.58 | asteriskguy | not quite dfgas |
| 03:21.15 | dfgas | no? |
| 03:21.18 | asteriskguy | SIP uses 5060 for signalling, however, RTP needs a random port from 10000 - 20000 |
| 03:21.22 | asteriskguy | UDP |
| 03:21.48 | asteriskguy | you can however set a limit and lower the range in one of the conf files |
| 03:21.53 | dfgas | so open 10000 to 20000 to it also? lol |
| 03:21.58 | dfgas | dmz? |
| 03:22.46 | asteriskguy | well, you can limit it to a smaller range |
| 03:23.49 | NormanAthol | dfgas, you can change the rtp port range in rtp.conf |
| 03:24.12 | asteriskguy | especially if you don't have a lot of people using the service |
| 03:25.06 | NormanAthol | just a question on thsi topic asteriskguy do you know how many ports a single call will use is both incomming and outgoing on the one rtp ort or is it in seprate |
| 03:25.08 | dfgas | hmmm |
| 03:25.29 | *** join/#asterisk bkw__ (i=brian@adsl-70-142-43-193.dsl.tul2ok.sbcglobal.net) |
| 03:25.58 | dfgas | well i am using freepbx cause i am not a guru on the configs yet. i am going to have to do a voip class down the road |
| 03:26.11 | asteriskguy | two end points will more then likely use different rtp ports |
| 03:26.37 | asteriskguy | if you load up ethereal and capture the packets going to both end points you can see it |
| 03:26.43 | NormanAthol | so when selecting a port range i should allow 2 ports per call and that will give me my max number of calls |
| 03:27.21 | NormanAthol | i mgith have to do that |
| 03:29.56 | Sweeper | dfgas: honestly, you're better off starting with vanilla asterisk, if you actually want to get good at it. freepbx isn't going to help you learn :/ |
| 03:30.58 | asteriskguy | Agree with Sweeper |
| 03:30.58 | bkw__ | pfft |
| 03:31.01 | blitzrage | drfreeze: how about "Read the book online" right in the menu on the left? |
| 03:31.13 | dfgas | Sweeper: try |
| 03:31.16 | dfgas | tue |
| 03:31.18 | blitzrage | sometimes people don't care about learning |
| 03:31.21 | dfgas | arghh |
| 03:31.23 | dfgas | true |
| 03:31.23 | bkw__ | blitzrage, so true |
| 03:31.25 | blitzrage | they just want something up and running |
| 03:31.38 | asteriskguy | In my case I needed both |
| 03:31.39 | JT | unless you've got a good reason, leave the rtp port range as 10000 and 20000 |
| 03:31.48 | blitzrage | asteriskguy: freepbx won't give you both -- it gives you the latter |
| 03:31.52 | dfgas | k, if i change rtp to 10000 - 20000 to 10000 - 10100 that would be good? |
| 03:31.53 | JT | people who say it's "more secure" to reduce that portrange are retarded |
| 03:31.59 | asteriskguy | I want to learn but the scope of the project didn't give me enough time |
| 03:32.00 | blitzrage | you won't learn a bloody thing about "asterisk" with freepbx, even though it uses asterisk underneath |
| 03:32.02 | bkw__ | JT, not really.. |
| 03:32.13 | bkw__ | its about securing your network |
| 03:32.22 | bkw__ | and they are right less ports open is more secure |
| 03:32.34 | asteriskguy | So I pulled 12-18 hrs days for a week, got it up and running |
| 03:32.37 | blitzrage | too many people want asterisk to solve their administrative ability/duty |
| 03:32.39 | JT | then make sure it's more secure, by keeping software up to date |
| 03:32.42 | asteriskguy | yo JT, what's up? |
| 03:32.44 | JT | instead of wanking with portblocking |
| 03:32.44 | *** join/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 03:32.47 | flenders | jt: go to the pub! enjoy your anzac day mate~ |
| 03:32.52 | dfgas | ? |
| 03:32.59 | bkw__ | JT, what? |
| 03:33.07 | bkw__ | portblocking things is good practice |
| 03:33.15 | bkw__ | default deny is your friend |
| 03:33.22 | JT | bkw__: the main advantage of inbound port blocking is to protect insecure hosts |
| 03:33.25 | JT | uhuh |
| 03:33.35 | bkw__ | no its not |
| 03:33.43 | bancus | I have an asterisk PBX I'm trying to set up with hardlines. I have an FXO connected to a hardline from Verizon, and an FXS connected to a cordless phone here. I can call out from the phone, but if the other side hangs up before I do, rather than my phone hanging up, it gets a busy signal. Hints? |
| 03:33.51 | bkw__ | JT, so you don't think firewalls are a good thing? |
| 03:33.59 | asteriskguy | found a bug in Asterisk BE last week |
| 03:34.06 | JT | bkw__: they're a necessary evil with so many insecure hosts about |
| 03:34.20 | bkw__ | JT, but they exist to protect you. |
| 03:34.25 | asteriskguy | JT, any experience with HylaFAX? |
| 03:34.27 | bkw__ | no matter if the service is secure |
| 03:34.35 | bkw__ | asteriskguy, I have extensive exp. with Hylafax |
| 03:34.50 | asteriskguy | yes, maybe you can help me? |
| 03:34.59 | bkw__ | your better off running hylafax+ hylafax.sf.net |
| 03:35.08 | JT | bkw__: i prefer services actually be secure |
| 03:35.11 | bkw__ | lee howard's improved version |
| 03:35.13 | asteriskguy | oh really? |
| 03:35.15 | JT | instead of relying on a firewall |
| 03:35.16 | bkw__ | JT, then do not run asterisk |
| 03:35.23 | asteriskguy | I was looking at that this afternoon |
| 03:35.25 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:35.38 | bkw__ | JT, the only secure host is one that is off and unplugged |
| 03:35.46 | JT | bkw__: i know that |
| 03:35.51 | bkw__ | no matter how secure the services are |
| 03:35.54 | asteriskguy | does HylaFAX+ work with *? |
| 03:35.56 | JT | yes |
| 03:36.03 | JT | that's why security is all relative |
| 03:36.06 | bancus | aha, think I figured it out |
| 03:36.18 | bkw__ | asteriskguy, the guy that does Hylafax+ wrote IAXModem |
| 03:36.21 | bkw__ | and that will work with Asterisk |
| 03:36.26 | blitzrage | it looks like you're arguing, but it seems like you're agreeing |
| 03:36.28 | Sweeper | blitzrage: that's why I qualified my statement with "if you actually want to get good at it" |
| 03:36.31 | JT | but a lot of noobs hide behind their firewalls instead of employing good security practice |
| 03:36.38 | bkw__ | JT, so true |
| 03:36.44 | Sweeper | firewalls ARE good security practice... |
| 03:36.50 | bkw__ | NEXT!!! |
| 03:36.55 | blitzrage | firewalls are for pussies |
| 03:36.55 | JT | Sweeper: at the host level |
| 03:37.04 | MooingLemur | meow |
| 03:37.08 | Sweeper | at the network level! |
| 03:37.13 | JT | Sweeper: poorly implemented firewalls cause more trouble than good |
| 03:37.16 | bkw__ | yep |
| 03:37.21 | asteriskguy | great. Just what I'm looking for |
| 03:37.23 | Sweeper | so do poorly implemented light switches |
| 03:37.27 | bkw__ | blitzrage, go nag someone to fix the rtp debug in asterisk |
| 03:37.34 | bkw__ | printing negative timestamps is dumb |
| 03:37.38 | asteriskguy | HylaFAX+ can act as an fax -> email gateway right? |
| 03:37.55 | bkw__ | blitzrage, it can because the ts is an unsigned int |
| 03:37.58 | blitzrage | I only use rtp debug to make sure i have traffic going where I want, then turn it off |
| 03:38.01 | bkw__ | and the log line has a %d in it |
| 03:38.14 | blitzrage | sure seems like you could just make a quick patch and submit it to the bug tracker |
| 03:38.19 | Frosh | with asterisk, do you need to set up with only softphones? |
| 03:38.24 | bkw__ | I don't submit patches to asterisk |
| 03:38.26 | bkw__ | I stopped that. |
| 03:38.31 | Frosh | does real ip-phones work? |
| 03:38.42 | blitzrage | then don't complain |
| 03:38.45 | bkw__ | Frosh, no their is a multi million dollar market selling phones that don't work |
| 03:39.03 | bkw__ | blitzrage, its not a complaint but if thats the attitude people have then so be it |
| 03:39.12 | asteriskguy | JT how's it going man? |
| 03:39.12 | bkw__ | I'll bug Qwell about it later |
| 03:39.18 | JT | asteriskguy: not bad |
| 03:39.21 | blitzrage | that's *MY* attitude, not other peoples attitude |
| 03:39.32 | *** join/#asterisk saravia (n=jovannot@69.79.144.206) |
| 03:39.41 | kavit | ah anzac day! |
| 03:39.44 | dfgas | hmmm, i got the mp3 up for music on hold, but it doesn't play |
| 03:39.48 | kavit | finally got to sleep in |
| 03:39.50 | asteriskguy | bkw: who's selling phones that don't work? |
| 03:39.50 | kavit | !!!!!!!!!!!!! |
| 03:40.07 | bkw__ | asteriskguy, it was sarcasim |
| 03:40.13 | kavit | and what a sleep in it was \m/ |
| 03:40.24 | asteriskguy | oh |
| 03:40.26 | bkw__ | l8tr |
| 03:40.41 | asteriskguy | how's your project going JT? |
| 03:40.56 | JT | which project? |
| 03:41.12 | asteriskguy | ITSP |
| 03:41.24 | asteriskguy | or was it the fax one? |
| 03:41.32 | JT | that wasn't a prority |
| 03:42.05 | asteriskguy | man, I've been trying to get Fax to Email to work with * |
| 03:42.12 | asteriskguy | two days with no results |
| 03:42.34 | kavit | i have decided to write a clustering solution for asterisk.... well just using heartbeat and drbd for a hot standby initially...i am going to write an admin interface for it! |
| 03:43.06 | dfgas | is there an easy command to restart asterisk? |
| 03:43.10 | asteriskguy | kavit, I would definitely be interested in that |
| 03:43.18 | kavit | it works pretty well... i have been testing it for the last month |
| 03:43.18 | asteriskguy | dfgas: RESTART NOW |
| 03:43.39 | asteriskguy | but with most * configs you only to reload the appropriate resource |
| 03:43.55 | kavit | asteriskguy: I am going to replace our office system with the test setup without telling anyone |
| 03:43.59 | dfgas | heh |
| 03:44.03 | dfgas | just found it |
| 03:44.08 | dfgas | didn't see that before sorry |
| 03:44.25 | asteriskguy | lol, that's a heck of a risk |
| 03:44.27 | kavit | asteriskguy: and I am going to fail it over during live calls... |
| 03:44.36 | kavit | asteriskguy: yeah well... I am the boss... they cant fire me |
| 03:44.37 | asteriskguy | I have two asterisk systems at home I can play with |
| 03:45.11 | Sweeper | kavit: no, but you can make everyone else hate you :P |
| 03:45.19 | asteriskguy | kavit, what's drbd? |
| 03:45.27 | kavit | Sweeper: too late ... haha |
| 03:45.55 | asteriskguy | dfgas, type help in the CLI |
| 03:45.59 | kavit | asteriskguy: http://linux-ha.org/DataRedundancyByDrbd |
| 03:46.12 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 03:46.15 | kavit | asteriskguy: it lets you do RAID 1 over tcpip |
| 03:46.22 | kavit | asteriskguy: works like a charm! |
| 03:46.51 | kavit | i have only broke it once and that was because I did something it wasnt designed to do.. |
| 03:47.06 | asteriskguy | that's cool |
| 03:47.12 | *** join/#asterisk flashnet (n=flashnet@85.89.27.239) |
| 03:47.24 | flenders | kavit, you run a crossover between boxes? |
| 03:47.50 | asteriskguy | how difficult is it to implement ha? |
| 03:47.58 | kavit | flenders: nope... normal network cable... one of them is vmware and the other is an old shitty box |
| 03:48.19 | kavit | flenders: but if i was doing it in production... yeah I would run a cross over cable |
| 03:48.35 | flenders | you run data and voice on the same network? |
| 03:48.54 | kavit | flenders: this is a test setup |
| 03:48.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.09 | kavit | flenders: production has a VLAN 802.1q |
| 03:49.16 | asteriskguy | kavit, I'm very interested in the your clustering solution. Can you send me the info |
| 03:49.21 | flenders | ah ok |
| 03:49.57 | kavit | asteriskguy: dude there is no solution at the moment...i will let you know in a month or two.. i am planning on doing 3 different kinds of solutions to see what works best |
| 03:50.03 | asteriskguy | we have a hard time implementing 802.1q. We have about 250 phones in the office and VLAN just die when we power up that many phones at once |
| 03:50.25 | asteriskguy | cool, definitely be interested. |
| 03:50.35 | kavit | asteriskguy: I have been in a setup with 3500 handsets |
| 03:50.45 | kavit | asteriskguy: 802.1q ed |
| 03:50.53 | *** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:51.01 | kavit | asteriskguy: just need to throw cash at it and buy good hardware |
| 03:51.05 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 03:51.07 | kavit | this was all cisco |
| 03:51.09 | asteriskguy | whoa, bigger then our implementation, we're going to about 2000 |
| 03:51.11 | dfgas | omg |
| 03:51.23 | dfgas | now i can't call into my box again |
| 03:51.32 | asteriskguy | yeah, we have 3 cisco PoE switches per floor |
| 03:51.57 | kavit | i am just trying to clear my head of the Cisco CallManager and IPCC clustering and the way they do it..it is a bit wasteful |
| 03:52.09 | *** join/#asterisk bmg505 (n=leon@196.209.181.176) |
| 03:52.23 | JT | asteriskguy: something wrong with your irc client? |
| 03:52.36 | kavit | asteriskguy: thats weird... have you tried to contact the TAC? |
| 03:53.08 | asteriskguy | oh no....sorry I didn't see it |
| 03:53.56 | kavit | now I need to figure out how to not drop the audio when a machine fails over and I will be happy |
| 03:55.19 | kavit | any ideas? |
| 03:55.41 | asteriskguy | TAC? Actually, I'm not the one who implemented it. Our network engineer did |
| 03:55.53 | JT | kavit: rewrite asterisk |
| 03:55.59 | kavit | asteriskguy: TAC = cisco technical assistance centre |
| 03:56.06 | kavit | JT: no :( |
| 03:56.19 | Sweeper | Taco Assistance Center |
| 03:56.31 | JT | kavit: maintaining active calls is serious telco level redundancy |
| 03:56.42 | JT | seems pointless on x86 consumer hardware anyway |
| 03:56.44 | kavit | JT: yeah |
| 03:56.57 | kavit | JT: i have seen it done with PRI |
| 03:57.01 | asteriskguy | I'd be happy if I can just restore service with in seconds or minutes. |
| 03:57.01 | Sweeper | kavit: openser, and rtp redirect~ |
| 03:57.20 | Sweeper | maybe rtp proxy as well |
| 03:57.21 | JT | kavit: you need 2 systems parallel processing everything |
| 03:57.35 | kavit | JT: i thought that would be the case |
| 03:57.37 | Sweeper | although that won't really solve the problem |
| 03:57.45 | Sweeper | because the openser server could go down... |
| 03:57.48 | SwK | you know they do make lock-step CPU servers that solve all that problem |
| 03:57.51 | Sweeper | but maybe you could do some voodoo |
| 03:57.59 | kavit | hrm |
| 03:58.02 | SwK | so if one cpu goes out the other just takes over |
| 03:58.16 | kavit | Cisco refused to support it with callmanager |
| 03:58.17 | JT | i'm aware |
| 03:58.17 | JT | telcos use them |
| 03:58.28 | SwK | yep |
| 03:58.30 | JT | but that only protects against cpu failure |
| 03:58.34 | Sweeper | is cpu death really that common? :/ |
| 03:58.54 | SwK | jt: actually it protects again CPU, ram and several other failures |
| 03:59.10 | Sweeper | I think you'll find RAID controller death, and RAM death to be the biggest killers |
| 03:59.12 | kavit | Sweeper: disaster recovery isnt about it actually happening.... it is about the possibility of it happening |
| 03:59.12 | SwK | however real telco equipment just detects a failure and moves it to a different media path |
| 03:59.15 | Sweeper | well, power first, but that's easy |
| 03:59.21 | JT | SwK: sure, but i assume the systems are right next to each other, what if the place catches fire? |
| 03:59.45 | Sweeper | kavit: how often is the cpu gonna die and it not be the whole server? |
| 03:59.51 | SwK | jt: theres full redundany, and then theres reasonable redundancy |
| 04:00.12 | Sweeper | SwK: multi-home/multi-NOC is well within reasonable |
| 04:00.14 | kavit | Sweeper: sure but people have money to burn... so why not |
| 04:00.29 | SwK | its like security... sure I can make build a system thats completely impossible to hack, but it would be completely useless |
| 04:00.31 | JT | Sweeper: on some telco equipment, 1 cpu card handles tens of thousands of concurrent calls, they can't afford it to go down |
| 04:00.47 | Sweeper | kavit: because you might as well spend that money on something that will actually be effective? |
| 04:00.52 | kavit | I have worked with Nortel VPS/MPS and Oscar Systems |
| 04:00.55 | SwK | sweeper: do you realize what a single DS3 port on real telco hardware costs? |
| 04:01.02 | kavit | Sweeper: they work very well |
| 04:01.05 | JT | kavit: the financial loss from a major telco switch being down can be over $50k/minute |
| 04:01.12 | Sweeper | SwK: we're talking about asterisk here :P |
| 04:01.12 | SwK | yep |
| 04:01.21 | SwK | asterisk != carrier grade |
| 04:01.26 | JT | yeah |
| 04:01.31 | SwK | asterisk is a PBX |
| 04:01.46 | Sweeper | but we're discussing making asterisk moar reliable, are we not? |
| 04:01.53 | SwK | I dunno I just wandered in |
| 04:01.58 | JT | and telephony in general |
| 04:02.25 | kavit | JT: i am well aware of that... infact the company I was contracting for got slapped a financial penalty because an employee blundered out on site at an exchange |
| 04:02.39 | JT | heh |
| 04:02.44 | SwK | ouck |
| 04:02.56 | Sweeper | notice he "was" contracting for them >.> |
| 04:02.58 | JT | ex employee |
| 04:03.08 | kavit | haha |
| 04:03.12 | JT | play with gear that he wasn't mean to? |
| 04:03.20 | kavit | nah they didn't fire him... he got blamed for it |
| 04:03.33 | kavit | but it wasn't really his fault.. he got away with a rap on the wrist |
| 04:04.10 | kavit | all he did was fix it using stuff from outside the MOP |
| 04:04.14 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 04:04.30 | kavit | and it actually caused a problem elsewhere |
| 04:04.39 | voltagex | Channel SIP/mnf-081e7110 was never answered. - when using a callfile I get this and the extension never gets connected.... |
| 04:04.53 | kavit | heh i know it sounds ambigious but I dont know if I can reveal anymore bloody contracts |
| 04:05.00 | kavit | and NDAs |
| 04:05.26 | Sweeper | JT: it was the polyester leisure suit that did it >.> |
| 04:06.02 | kavit | but yeah, you get a Memorandum of Procedure approved and you go to the exchange and do ONLY THAT, if not they take your balls |
| 04:06.16 | Sweeper | that, and not using the static discharge bracelet >.> |
| 04:06.19 | JT | kavit: coming to cebit? |
| 04:06.35 | kavit | JT: maybe.... when is it? |
| 04:06.39 | JT | 1-3 |
| 04:06.43 | kavit | i didn't even know |
| 04:06.51 | kavit | do you need tickets? |
| 04:06.53 | voltagex | why would a callfile fail to work, if there are no syntax errors and the extension is correct? |
| 04:06.56 | JT | yes |
| 04:07.06 | JT | unless you want to pay tewnty something dollars |
| 04:07.13 | kavit | heh thats not too bad |
| 04:07.27 | kavit | i may come there... depending on work |
| 04:07.42 | JT | there's a VoIP section this year |
| 04:07.47 | kavit | they called me and said $10,000 for a small stall |
| 04:07.49 | JT | and Open Source, and a few others |
| 04:07.51 | kavit | i said no |
| 04:07.59 | kavit | i will come there and network for free |
| 04:08.09 | JT | apparently it's a lot cheaper if you get it in the week before it opens :P |
| 04:08.19 | kavit | yeah |
| 04:08.34 | kavit | I should print pamphlets and stand outside |
| 04:08.39 | kavit | handing them out |
| 04:08.51 | JT | heh |
| 04:09.31 | kavit | but seriously I have 3 good projects going that should be finished in the next 3 months |
| 04:09.42 | kavit | and then I can bring good stuff to the market |
| 04:09.49 | kavit | good OPEN SOURCE stuff |
| 04:10.00 | JT | :) |
| 04:10.02 | voltagex | Is there a way to get even more verbose output from a callfile? |
| 04:10.16 | JT | mark spencer will be doing a fe talks there |
| 04:10.39 | kavit | JT: should I pie him in the face? |
| 04:10.45 | JT | lol |
| 04:10.46 | kavit | heh like bill gates |
| 04:10.58 | kavit | it might be good publicity for asterisk |
| 04:11.18 | JT | the pic they have on his cebit profile... looks so geeky |
| 04:11.19 | asteriskguy | that's the second time I heard someone said something about Mark Spencer compare to Bill Gates |
| 04:11.35 | kavit | i am going to open solaris meeting as well |
| 04:11.42 | kavit | Simon Phipps is in sydney |
| 04:12.14 | kavit | plus Sun have offered pizza and beer |
| 04:12.43 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 04:13.30 | kavit | I found something cool... http://ultrapossum.org/ |
| 04:13.50 | kavit | now I can store the cluster config in LDAP and use it to add and remove nodes |
| 04:13.55 | blitzrage | I found something better... (not for kids, those easily offended by boobs): http://www.fullur.is/drasl/063.wmv |
| 04:14.42 | kavit | so when a new node is added it gets the config off ultrapossum and bam! |
| 04:15.53 | kavit | <PROTECTED> |
| 04:16.05 | kavit | blitzrage: thats what real ninjas should do |
| 04:16.30 | flenders | hahahaha |
| 04:17.22 | asteriskguy | what distro do you guys have asterisk running under? and how do you feel about it? |
| 04:17.58 | kavit | asteriskguy: i use ubuntu-server or gentoo |
| 04:18.02 | Sweeper | ~distro |
| 04:18.03 | jbot | methinks distro is significa distribution (english) |
| 04:18.12 | Sweeper | ~distribution |
| 04:18.13 | jbot | [distribution] a collection of goodies built around a kernel, also a whole lot of infrastructure and people. |
| 04:18.20 | Sweeper | gah |
| 04:18.27 | kavit | asteriskguy: i have never been a big fan of red hatty stuff |
| 04:18.28 | Sweeper | there's a factoid in there somewhere |
| 04:18.31 | *** part/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 04:18.33 | kavit | since rh 6.2 |
| 04:18.40 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:18.42 | asteriskguy | oh why? |
| 04:18.44 | Sweeper | I like gentoo, or slackware |
| 04:18.55 | asteriskguy | hey CunningPike, what's up? |
| 04:18.55 | kavit | because of a lpd/lpr exploit they had |
| 04:18.56 | *** join/#asterisk jmacz (n=jmacz@201.244.170.241) |
| 04:19.05 | asteriskguy | oh |
| 04:19.13 | kavit | i think thats just an excuse |
| 04:19.21 | asteriskguy | hmm....so you like ubuntu better? |
| 04:19.23 | voltagex | all I get is a failed message from the callfile, where do I start to debug this? |
| 04:19.27 | asteriskguy | redhat = $$ |
| 04:19.39 | kavit | asteriskguy: yeah ubuntu is alright |
| 04:20.10 | asteriskguy | that's debian based right? |
| 04:20.18 | asteriskguy | <-- linux noob |
| 04:20.21 | kavit | asteriskguy: if customers don't mind paying and want RH.. i work with it as well but I dont have to like it |
| 04:20.30 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:20.31 | kavit | yeah asteriskguy it is based on debian |
| 04:21.21 | asteriskguy | what are the major differences between deb and rh? |
| 04:21.28 | kavit | but people like HP only write their uitls for RH and Novell |
| 04:21.41 | kavit | asteriskguy: heh where do I start.... |
| 04:21.42 | CunningPike | Hi asteriskguy |
| 04:22.25 | JT | .deb and .rpm |
| 04:22.32 | drfreeze | Is there a better way to get Asterisk to start on reboot other than adding safe_asterisk to /etc/rc.local? |
| 04:22.34 | asteriskguy | just give me a few? Our network engineer seems to prefer RH, seems more like their support |
| 04:22.51 | kavit | asteriskguy: http://toykeeper.net/soapbox/debian-redhat/ |
| 04:23.20 | CunningPike | asteriskguy: Just got in from a 12 hour SANS course |
| 04:23.33 | asteriskguy | CunningPike, do you know how to get fax to email to work with HylaFAX+ & IAXmodem + asterisk? |
| 04:23.42 | asteriskguy | oh wow....intense? |
| 04:24.12 | CunningPike | asteriskguy: No clue, sorry - all we've done is stick a couple of faxes on SPA-3000 ATAs and they seem to work |
| 04:24.24 | CunningPike | asteriskguy: A little intense, yes |
| 04:24.28 | kavit | yeah thats what I do usually as well |
| 04:24.49 | kavit | blitzrage: what is the purpose of the cluster? |
| 04:24.57 | blitzrage | kavit: to handle thousands of calls |
| 04:24.58 | asteriskguy | I took the CEH course, it's interesting |
| 04:25.07 | kavit | blitzrage: asterisk cluster? |
| 04:25.14 | blitzrage | kavit: of course, I'm in #asterisk |
| 04:25.43 | kavit | blitzrage: sweet... is there a way to not lose audio when machines die or fail over? |
| 04:26.23 | kavit | blitzrage: i am trying to write an easy frontend/admin/clustering solution around asterisk |
| 04:27.20 | asteriskguy | oh, we used IAXy from Digium |
| 04:27.42 | asteriskguy | but it's not cutting it, 1 fax machine / floor |
| 04:27.53 | asteriskguy | so I'm looking into fax -> email w/asterisk |
| 04:28.03 | asteriskguy | no a whole lot of documentations out there though |
| 04:28.29 | blitzrage | kavit: yah -- don't put the media through Asterisk |
| 04:28.42 | blitzrage | if the media goes through asterisk, you're going to lose the call |
| 04:28.54 | kavit | blitzrage: thats what I thought... |
| 04:29.10 | asteriskguy | so canreinvite=yes? |
| 04:29.28 | blitzrage | I use directrtpsetup=yes |
| 04:29.37 | kavit | plus a good mediaproxy |
| 04:29.55 | blitzrage | I just use OpenSER and Asterisk |
| 04:29.59 | asteriskguy | directrtpsetup=yes? where is this at? |
| 04:30.03 | blitzrage | sip.conf |
| 04:30.05 | blitzrage | in 1.4 |
| 04:30.17 | asteriskguy | oh dang it. I have 1.3 ABE |
| 04:30.25 | kavit | blitzrage: ah I am doing more of a 1 on 1 hot standby thing for now... for SMEs |
| 04:30.31 | asteriskguy | blitzrage, how hard is it to setup OpenSER? |
| 04:30.36 | blitzrage | not easy |
| 04:30.43 | blitzrage | unless you're good at coding |
| 04:30.45 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.189) |
| 04:30.45 | kavit | I like OpenSER |
| 04:30.50 | *** join/#asterisk last1 (n=dood@modemcable215.11-57-74.mc.videotron.ca) |
| 04:30.52 | asteriskguy | not good at coding |
| 04:30.54 | blitzrage | its fine once you understand |
| 04:30.59 | last1 | anyone with a 7970 ? |
| 04:31.02 | blitzrage | OpenSER is harder than Asterisk |
| 04:31.15 | kavit | what do you guys think about sipx ? |
| 04:31.17 | NormanAthol | i have 2 contexts office and service if some one from teh service wants to call an ext is office how to i add that to the dial plan |
| 04:31.19 | blitzrage | although I've been using Asterisk for 5 years... so I may be biased |
| 04:31.26 | blitzrage | kavit: never used it -- Asterisk does everything I need |
| 04:31.42 | asteriskguy | OpenSER requires mySQL right? |
| 04:31.42 | JT | kavit: has a very dead irc channel |
| 04:31.47 | kavit | i worked with sipx for a bit and it was a bloody pain to setup... unless you buy it from pingtel |
| 04:32.01 | blitzrage | asteriskguy: it doesn't require a DB at all if you want to run it in memory only mode |
| 04:32.24 | kavit | blitzrage: pg rules... |
| 04:32.33 | asteriskguy | oh....hmm...more reading for me I guess |
| 04:32.36 | JT | postgresql or memory is the only way to go :P |
| 04:32.44 | kavit | asteriskguy: where are you located? |
| 04:32.50 | JT | my servers have a no mysql policy :D |
| 04:32.52 | asteriskguy | irvine, CA |
| 04:32.54 | asteriskguy | US |
| 04:33.03 | kavit | ah ok |
| 04:33.28 | last1 | anyone knows the XML command for the 7970 to enable: Display On When Incoming Call ? |
| 04:33.33 | asteriskguy | from your conversation with JT earlier I'm assuming you're out of state? |
| 04:33.44 | kavit | asteriskguy: i am in Australia |
| 04:33.57 | JT | state? heh, more than out of California |
| 04:33.59 | asteriskguy | :) cool |
| 04:34.02 | kavit | or as they like to say... 'stralya |
| 04:34.08 | JT | although i guess it's only 13000km of water |
| 04:34.11 | kavit | haha |
| 04:34.18 | asteriskguy | I meant out of the States |
| 04:34.18 | NormanAthol | nice ot know i am not the only aussie spending his day off on the net |
| 04:34.18 | voltagex | is there any way to provide a normal dialtone to a remote caller, say for callback? |
| 04:34.32 | asteriskguy | lol |
| 04:34.35 | voltagex | NormanAthol: I am too |
| 04:34.40 | NormanAthol | ok good |
| 04:34.48 | kavit | NormanAthol: I am supposed to be working |
| 04:34.54 | kavit | i got to sleep in though |
| 04:34.56 | NormanAthol | lol |
| 04:34.59 | asteriskguy | oh wow. I would love to visit AUS |
| 04:35.01 | JT | the Aussies outnumber the yanks talking atm |
| 04:35.19 | NormanAthol | yeah i need sleep had a problem last night wne to sleep woke up and had to fix it like 1 hor later |
| 04:35.26 | kavit | asteriskguy: You can come on a Asterisk course here... i wont charge you a lot... |
| 04:35.28 | kavit | :P |
| 04:35.31 | jql | even google maps doesn't tell you to swim to australia |
| 04:35.41 | JT | heh |
| 04:35.43 | asteriskguy | :) what course do you offer kavit? |
| 04:36.12 | kavit | asteriskguy: none as of yet... but we just started offering courses today |
| 04:36.17 | kavit | ;) |
| 04:36.22 | JT | beer appreciation ;) |
| 04:36.22 | jql | voltagex: lookup DISA |
| 04:36.40 | asteriskguy | lol...ok. I took the Digium's Bootcamp. Enough info to get me by at my workplace |
| 04:37.05 | kavit | JT: that would be a good course for these yanks... trying to tell me america makes decent beer |
| 04:37.22 | JT | i don't know if there are any asterisk courses in Australia |
| 04:37.23 | voltagex | jql: thanks |
| 04:38.38 | asteriskguy | but any course out there that offers advanced subjects would be of interest to me |
| 04:38.38 | voltagex | kavit: budweiser, eww |
| 04:38.53 | asteriskguy | I used to like budweiser, but fell in love with Ashahi and Heineiken |
| 04:38.58 | JT | kavit: is one of the training aids a cricket bat to whack them over the head when they try to drink an american beer? |
| 04:39.06 | kavit | hahah yeah |
| 04:39.17 | kavit | and they dont like Ales |
| 04:39.30 | kavit | seriously whats wrong with a good old stout |
| 04:39.36 | JT | i'm not a drinker, so it all doesn't make a difference to me :P |
| 04:39.55 | jql | more for me |
| 04:40.07 | kavit | neither am I... i just like to berate foreigners |
| 04:40.09 | kavit | :P |
| 04:40.15 | kavit | aussie as |
| 04:40.32 | asteriskguy | lol |
| 04:40.58 | asteriskguy | alright guys, take it easy, it's bed time for me |
| 04:41.00 | kavit | i got a lot of shit when I first came to australia |
| 04:41.05 | kavit | laters asteriskguy |
| 04:42.38 | kavit | this ultrapossum.org is cool... does replication as well |
| 04:42.50 | voltagex | wow nice, DISA works well |
| 04:42.57 | voltagex | any way to get a non-us dialtone? |
| 04:43.28 | jql | oh, it might be a conf file somewhere to specify the frequency |
| 04:43.28 | JT | change the tonezone in indications.conf |
| 04:43.33 | jql | yeah, that one |
| 04:44.41 | voltagex | heh, I could have a french dialtone |
| 04:45.23 | kavit | voltagex: it would probably whinge about cheese or something like that |
| 04:45.25 | kavit | :\ |
| 04:45.32 | *** join/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 04:45.38 | voltagex | but I am le tired |
| 04:45.50 | kavit | SACRE BLEU |
| 04:45.50 | voltagex | well have a nap, zen fire ze missiles! |
| 04:47.45 | kavit | but one thing australia needs is more paying customers |
| 04:47.45 | drfreeze | Any zaptel/TDM experts up? |
| 04:47.55 | danp | i |
| 04:47.58 | kavit | who want features from VoIP, not just cheap phone calls |
| 04:48.04 | danp | whoops, sorry. internet hiccup |
| 04:48.16 | drfreeze | I've built 1.4 and can't seem to get asterisk to see zap channels |
| 04:48.25 | tzafrir | drfreeze, ask up |
| 04:48.59 | drfreeze | tzafrir: first off, I expect asterisk -r to recognize the zap show command, but it doesn't |
| 04:49.09 | tzafrir | do you see the channels in /proc/zaptel/* ? |
| 04:49.18 | drfreeze | lemme check |
| 04:49.35 | tzafrir | hmm... chan_zap is not loaded (or not built at all) |
| 04:49.43 | tzafrir | try: load chan_zap.so |
| 04:49.50 | tzafrir | in the asterisk cli |
| 04:50.05 | drfreeze | tzafrir: Yes, http://pastie.textmate.org/56363 |
| 04:51.09 | drfreeze | tzafrir: loaded chan_zap.so |
| 04:51.44 | tzafrir | do you now get anything from 'zap show channels' ? |
| 04:52.05 | drfreeze | tzafrir: No such module 'chan_zap.so' |
| 04:52.21 | voltagex | yep, that change to indications worked |
| 04:52.37 | tzafrir | so chan_zap wasn't built when you built asterisk |
| 04:52.44 | drfreeze | tzafrir: ok |
| 04:52.57 | tzafrir | in the asterisk source dir: make menuselect |
| 04:53.13 | drfreeze | tzafrir: so I need to build and install zaptel before building asterisk? |
| 04:53.19 | tzafrir | if it is just deselected, select and rebuild |
| 04:53.30 | tzafrir | generally, yes |
| 04:54.59 | drfreeze | tzafrir: chan_zap is already selected |
| 04:56.00 | tzafrir | if you re-run make, does it build it? |
| 04:57.57 | drfreeze | tzafrir: I'll check |
| 04:58.57 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 05:00.11 | drfreeze | tzafrir: nope, doesn't make it |
| 05:00.23 | drfreeze | chan_zap.so is not to be found in the asterisk directory |
| 05:00.59 | drfreeze | there's a chan_zap.c in channels/ |
| 05:02.02 | tzafrir | I some reports that claim it takes a 'make clean' or even 'make distclean' to fix that. But distclean is a big hammer |
| 05:02.11 | tzafrir | I can't think of anything else right now |
| 05:03.03 | drfreeze | tzafrir: should I try that with zaptel first? |
| 05:03.59 | tzafrir | are you sure you see chan_zap enabled? '[*]' ? |
| 05:04.11 | drfreeze | yes |
| 05:04.18 | tzafrir | (as opposed to 'XXX' which is "grayed-out") |
| 05:04.29 | drfreeze | yes, enabled with [*] |
| 05:04.51 | drfreeze | zaptel did fail to build BTW |
| 05:04.53 | tzafrir | try disabling it, saving, then enabling it and saving again |
| 05:05.01 | drfreeze | I self patched it |
| 05:05.17 | drfreeze | tzafrir: ah, the old Windows trick. :) |
| 05:05.50 | drfreeze | it just made chan_zap.o |
| 05:08.07 | jql | I believe the .so is in a hidden .libs directory, would it not be? |
| 05:08.20 | *** part/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 05:10.59 | drfreeze | what does this mean: |
| 05:10.59 | drfreeze | <PROTECTED> |
| 05:10.59 | drfreeze | <PROTECTED> |
| 05:11.23 | dfgas | is skype allowed to go into asterisk yet? |
| 05:11.30 | JT | it mean it compiles then runs the linker |
| 05:12.13 | drfreeze | tzafrir: yeehaw. zap now shows up. Did a make distclean |
| 05:16.14 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 05:16.41 | drfreeze | tzafrir: thanks! |
| 05:17.04 | drfreeze | tzafrir: load automatically after reboot? |
| 05:23.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 05:23.50 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 05:24.58 | dfgas | are there any other cool free services for asterisk besides iaxtel, sipbroker, or voxalot? |
| 05:30.14 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:33.30 | *** join/#asterisk Hoondie (n=andre@ppp241-234.lns3.syd6.internode.on.net) |
| 05:34.11 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 05:36.18 | Hoondie | Hey people.. Question, i have an extension that you can see at > http://www.pastebin.ca/457113 <... The call is answered, but I don't hear any ringing tone as the end user.. |
| 05:38.01 | Hoondie | Anyone able to help out? |
| 05:39.31 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
| 05:41.55 | Keltus | are there any providers that can convert my toll free 1-800 number to sip and call my IP address directly? |
| 05:45.17 | Hoondie | And if it helps regarding my question above, the ringing is heard internally.. just not when calling my phone number externally (phone number is provided by my ISP, forwarded to my asterisk box) |
| 05:46.57 | *** join/#asterisk xxi (i=foobar@cpe-70-112-123-132.austin.res.rr.com) |
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| 06:04.14 | *** join/#asterisk ams1701 (n=ams1701@202.189.249.206) |
| 06:04.41 | ams1701 | hello |
| 06:05.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:06.39 | voltagex | is there a reason asterisk wouldn't receive # or * tones? |
| 06:07.20 | FuriousGeorge | a lot of ATAs reserve them |
| 06:07.29 | voltagex | damn |
| 06:07.40 | FuriousGeorge | and i believe asterisk itself will not accept # in the dialstring |
| 06:08.08 | voltagex | only need * to dial sipbroker numbers |
| 06:08.13 | FuriousGeorge | i use * but not # |
| 06:08.57 | voltagex | yeah...no idea how to disable it |
| 06:10.00 | *** join/#asterisk rmayorga (n=rmayorga@168.243.73.11) |
| 06:10.02 | Keltus | are there any providers that can convert my toll free 1-800 number to sip and call my IP address directly? |
| 06:10.11 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 06:11.04 | Hoondie | Another question people.. I want to have a setup like this.. Someone calls, it rings as normal.. While it's ringing, we can enter a PIN or something, it will then allow me to have a local dialtone using the DISA command |
| 06:11.55 | voltagex | FuriousGeorge: the annoying way to do it is to have an IVR type thing to dial out to sipbroker |
| 06:14.16 | FuriousGeorge | voltagex: does sound annoying |
| 06:14.33 | voltagex | FuriousGeorge: broadcom ATA, any ideas? |
| 06:14.35 | *** join/#asterisk dj-fu (i=aj@unaffiliated/dj-fu) |
| 06:14.59 | FuriousGeorge | voltagex: you wanna use * or you wanna use * and # |
| 06:15.09 | voltagex | FuriousGeorge: * only |
| 06:15.11 | FuriousGeorge | Hoondie: that's possible |
| 06:15.26 | FuriousGeorge | voltagex: doesnt the thing let you set you *XX functions? |
| 06:16.00 | voltagex | FuriousGeorge: what thing, Asterisk or the ATA? |
| 06:16.00 | FuriousGeorge | i guess the only way would be to disable them all together so the thing doesnt intercept them |
| 06:16.04 | FuriousGeorge | ata |
| 06:16.05 | Hoondie | FuriousGeorge: This is where I am stuck at the moment, how do i get the phone to ring, and be able to accept input from the end user at the same time? |
| 06:16.18 | voltagex | FuriousGeorge: nowhere in the ATA settings :( |
| 06:16.44 | FuriousGeorge | voltagex: at this point i would contact the company |
| 06:17.10 | voltagex | FuriousGeorge: and I'm going to be FuriousVoltage |
| 06:17.39 | voltagex | FuriousGeorge: this router (integrated ATA) has been replaced multiple times, I still have crashes and this kindof stuff |
| 06:18.02 | FuriousGeorge | Hoondie: not as easy as i thought when i first read it b/c you want it to ring "as normal" |
| 06:18.06 | FuriousGeorge | but here is a guess |
| 06:19.01 | FuriousGeorge | when the call comes in put it in a queue. i believe there is a way to make the queue music a ring instead of music |
| 06:19.24 | FuriousGeorge | nm that |
| 06:19.53 | FuriousGeorge | ok answer the call, and use page |
| 06:20.23 | *** join/#asterisk grEvenX (n=even@ti500720a080-4710.bb.online.no) |
| 06:20.39 | FuriousGeorge | page an extension that auto answers and gives you disa, and page a queue that gives you ringing instead of moh |
| 06:21.14 | FuriousGeorge | voltagex: i feel your pain. i find certain telephony hardware problematic, im sure no one wants to hear me rehash all my woes |
| 06:21.24 | voltagex | me neither |
| 06:21.45 | voltagex | any australians here ever get a refund out of Netcomm? |
| 06:22.19 | Hoondie | try the retailer your got it from, better chance of getting a refund |
| 06:22.56 | voltagex | no, they don't give refunds after 30 days |
| 06:23.05 | voltagex | see, I only started using the ata a few weeks ago |
| 06:23.15 | voltagex | but I've had the router for 7 months |
| 06:23.26 | Hoondie | voltagex: Good luck |
| 06:23.32 | voltagex | month 2, replacement 1, month 6, replacement 2 |
| 06:24.28 | Hoondie | FuriousGeorge: That way seems complex.. Could I do something like this, call comes in, wait x seconds for the code, if no code has come in, dial the extension of the end user |
| 06:27.16 | *** join/#asterisk |ryan| (n=foo@c-67-188-22-216.hsd1.ca.comcast.net) |
| 06:27.44 | |ryan| | is there any way to dial and require that the dialed party enter a PIN to have the call be connected? |
| 06:28.49 | *** join/#asterisk zeeesh (n=zeeesh@2-237-154-202.wol.net.pk) |
| 06:28.50 | zeeesh | hi |
| 06:29.03 | Frosh | is this using softphone, not a real phone? |
| 06:37.37 | |ryan| | either |
| 06:37.52 | |ryan| | in this case it's a phone hooked up to an ATA |
| 06:37.52 | *** part/#asterisk dhakatel (n=root@58.65.224.5) |
| 06:45.31 | voltagex | |ryan|: like a "secure line" or more like a collect call? |
| 06:46.38 | |ryan| | like |
| 06:46.46 | |ryan| | I want to set up an extension |
| 06:47.11 | |ryan| | so that people can dial it, and if someone without the pin picks up, it won't be connected. |
| 06:47.25 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
| 06:48.02 | voltagex | hrrm |
| 06:48.18 | voltagex | is this dialling asterisk to asterisk or asterisk to any phone? |
| 06:48.32 | voltagex | you'd have to have some kind of PBX at the other end |
| 06:48.59 | voltagex | that's what I think but my knowledge of this is limited |
| 06:53.29 | voltagex | |ryan| http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Authenticate |
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| 06:58.02 | sooli | Hi ! MpPlayer work fine bu t MOH not |
| 06:58.23 | sooli | i use the latest version of asterisk and asterisk-addon |
| 07:01.20 | Hoondie | Anyone know where a Asterisk 1.4 IVR guide is? |
| 07:01.50 | jql | there's a guide? |
| 07:02.03 | voltagex | voip-info.org maybe |
| 07:02.09 | voltagex | or the book |
| 07:02.09 | voltagex | ~book |
| 07:02.19 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:02.56 | sooli | anyidea about a MOH problem ? |
| 07:03.30 | Hoondie | This for 1.4? I'm having issues with 1.2 examples on 1.4 |
| 07:14.03 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136017.home.otenet.gr) |
| 07:39.00 | *** join/#asterisk zeeesh (i=zeeesh@13-237-154-202.wol.net.pk) |
| 07:39.02 | zeeesh | hi |
| 07:48.15 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:50.27 | vgster | morning does anyone else use suse linux and are having a problem with zap 1.2.17? |
| 07:51.43 | vgster | 1.2.16 is fine but 1.2.17 seems to have a problem http://pastebin.ca/457193 |
| 07:54.52 | nasls_lsa | I use suse , but I don't use zap .. |
| 07:55.15 | *** join/#asterisk hfb (n=hfb@adsl-75-39-161-90.dsl.irvnca.sbcglobal.net) [NETSPLIT VICTIM] |
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| 07:55.52 | vgster | ive gone back to 1.2.16 and its fine |
| 07:56.04 | vgster | suse 10.0 |
| 07:57.15 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
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| 08:02.04 | pbr503 | is there a way to dial out from within an application via some sort of code? |
| 08:02.53 | pbr503 | i am trying to dial out and add a caller with app_conference |
| 08:03.07 | pbr503 | features.conf seems to not be the right route? |
| 08:04.03 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
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| 08:04.04 | *** join/#asterisk hal2k (n=am@2002:5470:9fd9:0:0:0:0:1) [NETSPLIT VICTIM] |
| 08:04.21 | pbr503 | no ideas? |
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| 08:16.26 | *** join/#asterisk duki (n=duki@host-85-27-49-12.brutele.be) |
| 08:16.34 | duki | hello |
| 08:17.46 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 08:17.49 | pbr503 | hi |
| 08:17.52 | pbr503 | no one is awake |
| 08:17.53 | pbr503 | or helping |
| 08:18.18 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 08:18.32 | mattwj2005 | hey anyone want to go to work for me? |
| 08:18.48 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:19.26 | mattwj2005 | I am joking of course....but my vacation is coming to end |
| 08:19.36 | mattwj2005 | at least I had a nice 10 days off :D |
| 08:20.27 | duki | I download, compiled and install the 1.4.3 version of asteriks. But when I cannot compile the 1.4.0 version of asteriks-addons. There is an error. |
| 08:20.27 | duki | error src/chan_h323.c:250: error: too few arguments to function 'ast_channel_alloc' |
| 08:20.27 | duki | Thanks for any help or suggestion. |
| 08:21.03 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 08:21.31 | mattwj2005 | hmmm |
| 08:21.35 | mattwj2005 | what OS? |
| 08:22.01 | mattwj2005 | have you tired Google? |
| 08:23.25 | Uatec_ | hey, has anybody tried a digium s101i analog to digital converter, aka IAXY? |
| 08:23.25 | Uatec_ | how do you connect to it? |
| 08:23.25 | Uatec_ | the installation manual just says "start doing X" |
| 08:23.25 | Uatec_ | but now how to connect to it in the first place |
| 08:23.42 | mattwj2005 | is IAXY a softphone? |
| 08:24.08 | Uatec_ | no |
| 08:24.12 | Uatec_ | it's an analog to digital converter |
| 08:24.34 | mattwj2005 | http://www.voip-info.org/wiki-IAXy |
| 08:24.38 | mattwj2005 | :) |
| 08:25.17 | duki | hi, Yes I tied google and found the same bug fixed in 2006 after applying a patch. But all this is old, because asteriks changes. |
| 08:25.17 | duki | My os is debian lenny |
| 08:25.17 | duki | I tried ... |
| 08:26.01 | mattwj2005 | is this a fxo or fxs? |
| 08:27.32 | mattwj2005 | so basically an IAXy allows you to receive calls from the analog phone network right? |
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| 08:29.28 | Uatec_ | oh yes |
| 08:29.34 | Uatec_ | the voip-info wiki |
| 08:29.35 | Uatec_ | :) |
| 08:29.36 | Uatec_ | thanks |
| 08:29.51 | Uatec_ | mattwj2005 |
| 08:30.04 | mattwj2005 | your welcome :) |
| 08:30.10 | Uatec_ | what it does is allow a single analog phone be connected to a digital network |
| 08:30.13 | Uatec_ | using the IAX protocol |
| 08:30.32 | mattwj2005 | sounds awesome :D |
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| 08:36.11 | tzafrir | duki, 1.4.3 has just been submitted into Sid |
| 08:36.32 | tzafrir | As it is a security fix, maybe it will get a faster track into Lenny |
| 08:36.33 | Uatec_ | it's ok, mattwj2005 |
| 08:36.37 | Uatec_ | but the device is rather rubbish |
| 08:37.04 | mattwj2005 | sounds frastrating |
| 08:37.09 | mattwj2005 | I am getting a landline! |
| 08:37.11 | mattwj2005 | :) |
| 08:37.59 | mattwj2005 | here is a nice movie about Linux....a friend of mine found it on Google Video |
| 08:38.00 | mattwj2005 | http://video.google.com/videoplay?docid=7707585592627775409&hl=en |
| 08:39.26 | Uatec_ | i'll look at it when i get home |
| 08:39.26 | Uatec_ | we're getting 10 more snom 190s |
| 08:39.26 | Uatec_ | however we've got a BT 4600 conference phone |
| 08:39.27 | Uatec_ | which is analog |
| 08:39.27 | mattwj2005 | it is a movie called Revolution OS |
| 08:39.27 | Uatec_ | and not worth finding a digital replacement |
| 08:39.27 | *** part/#asterisk lukketto (n=lukketto@host30-193-dynamic.7-87-r.retail.telecomitalia.it) |
| 08:39.27 | mattwj2005 | I hope this isn't a copyright violation :( |
| 08:39.37 | Uatec_ | what, the video? |
| 08:40.26 | mattwj2005 | "Revolution OS" |
| 08:40.55 | mattwj2005 | http://www.revolution-os.com/ |
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| 08:43.08 | Beave | It's okay.. |
| 08:43.22 | Uatec_ | ah |
| 08:44.13 | mattwj2005 | I guess we could e-mail Linus T. |
| 08:44.17 | mattwj2005 | he stars in it! |
| 08:44.28 | Uatec_ | why would it be in violation of copyright? |
| 08:44.50 | mattwj2005 | I am not sure what the license is on the video |
| 08:44.56 | mattwj2005 | I know they sell the DVDs |
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| 08:46.16 | mattwj2005 | it is encryption free at least |
| 08:46.57 | mattwj2005 | anyways back to the topic |
| 08:47.03 | mattwj2005 | how is the IAXy going? |
| 08:48.45 | Uatec_ | i don't know |
| 08:48.52 | Uatec_ | the conference room is in use |
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| 08:48.59 | pbr503 | this is bullshit |
| 08:49.03 | Uatec_ | so i can't play with the phone and start ringing it |
| 08:49.06 | pbr503 | asterisk doesnt have operator codes? |
| 08:49.11 | Uatec_ | what a happy morning it is pbr503 |
| 08:49.40 | Uatec_ | what do you mean pbr503? |
| 08:49.40 | mattwj2005 | well that device supports ulaw |
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| 08:49.41 | mattwj2005 | it doesn't look like many other codecs |
| 08:49.43 | Uatec_ | that doesn't matter |
| 08:49.45 | Uatec_ | ulaw is fine |
| 08:49.55 | Uatec_ | we've got a gigabit network |
| 08:50.00 | pbr503 | i would like to be able to press a control code (ie: *9) to dial out a person in the middle of a module |
| 08:50.24 | Uatec_ | pbr503, write it in to the dial plan |
| 08:50.39 | mattwj2005 | extensions.conf |
| 08:50.53 | pbr503 | if it's in the middle of a module you cant dial another extension while another one is already in use |
| 08:50.55 | mattwj2005 | would be a good place to start to look :) |
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| 08:52.36 | mattwj2005 | you could use something like |
| 08:52.39 | Uatec_ | by a module do you mean an application? |
| 08:52.50 | mattwj2005 | exten => *9xxxx |
| 08:53.07 | mattwj2005 | that isn't even a full line by the way |
| 08:53.09 | mattwj2005 | :) |
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| 08:55.52 | mattwj2005 | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
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| 09:01.34 | Uatec_ | wow |
| 09:01.36 | Uatec_ | a full nmap scan sends 2meg of data |
| 09:01.37 | Uatec_ | that's alot |
| 09:01.42 | Uatec_ | considering it's not actually doing anything |
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| 09:08.45 | sbingner | Uatec_, um... nmap is doing something... but the amount of data will vary based on the type of scan and the number of open ports |
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| 09:11.04 | DarKnesS_WolF | i have a problem i have http://www.teltonika.lt/en/pages/view/?id=346 taltonika GSM gateway connected to asterisk via TDM400P FXO port ... i can make calls very normal but when i do receve calls the devices fwd the call to asterisk but if the mobile part hangs up the device keep giving asterisk tone so it don't get disconnected and if i pick up the asterisk call i find the device dialing tone .. how can i tell asterisk if the tone coming from |
| 09:12.45 | pbr503 | you guys are fucking idiots |
| 09:12.58 | DarKnesS_WolF | pbr503: who are u talking to ? |
| 09:13.07 | pbr503 | *9 should BREAK OUT of whatever is currently happening |
| 09:13.13 | pbr503 | hence what I said about a module |
| 09:13.16 | pbr503 | or app |
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| 09:15.52 | pbr503 | well specifically matt |
| 09:16.01 | mattwj2005 | yes? |
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| 09:16.17 | pbr503 | <mattwj2005> exten => *9xxxx |
| 09:16.17 | pbr503 | <mattwj2005> that isn't even a full line by the way |
| 09:16.17 | pbr503 | <mattwj2005> :) |
| 09:16.19 | pbr503 | wtf is that? |
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| 09:16.34 | pbr503 | that is of absolutely no use |
| 09:16.40 | pbr503 | did you even read what i said? |
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| 09:18.54 | mattwj2005 | at the time I was suggesting that if you want to separate extensions from other extensions |
| 09:19.32 | mattwj2005 | for example you could use |
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| 09:20.21 | mattwj2005 | exten => _*9,1,Dial(SIP/{Exten:1}) |
| 09:20.21 | pbr503 | that is fine for separating extensions |
| 09:20.22 | mattwj2005 | exten => _*9,2,Hangup |
| 09:21.07 | mattwj2005 | sorry if those commands are not 100 % correct |
| 09:21.07 | pbr503 | that is elementary shit |
| 09:21.08 | pbr503 | i need a control code to break out of whatever process is happening |
| 09:21.15 | pbr503 | exten => 6969,1,Answer() |
| 09:21.15 | pbr503 | exten => 6969,n,Playback(conf-enteringno) |
| 09:21.15 | pbr503 | exten => 6969,n,SayDigits(${EXTEN}) |
| 09:21.15 | pbr503 | exten => 6969,n,Playback(beep) |
| 09:21.24 | pbr503 | exten => 6969,n,Conference(6969) |
| 09:21.24 | pbr503 | exten => 6969,n,Hangup |
| 09:21.35 | pbr503 | so a user enters that conference |
| 09:21.43 | mattwj2005 | okay |
| 09:21.51 | mattwj2005 | lets call that step 1 |
| 09:21.55 | mattwj2005 | next part? |
| 09:22.07 | sooli | Hi! I have some troubles with MOH => no sound ... but background and mp3player work fine ... anu idea ? I have the latest version of asterisk and asterisk-addon |
| 09:22.08 | pbr503 | is there a way where they can press *9 and it will go to a separate context to dial out and add a user to the conference? |
| 09:22.13 | mkl1525 | Hi, just installed * 1.4 - did a call to 095 and used NoOp(${CALLERIDNUM}) and NoOp(${CALLERID}) to show the id of the calling party but both are empty although it worked with 1.2 version, so were there changes in the variable access? variables docu in /doc seems nearly equal to 1.2 - any hints? |
| 09:23.35 | Uatec_ | would a conference not have it's own exti codes? |
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| 09:23.44 | pbr503 | meetme does |
| 09:23.53 | Uatec_ | sooli, what application? |
| 09:24.08 | pbr503 | i have to work with app_conference since i cannot use ztdummy since my VPS is using Xen kernel |
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| 09:24.38 | mattwj2005 | I am sorry pbr503 I am not sure |
| 09:24.59 | pbr503 | cool matt, thanks anyway |
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| 09:25.41 | mattwj2005 | your welcome |
| 09:25.44 | mattwj2005 | for me Asterisk is a toy....a hobby |
| 09:26.11 | mattwj2005 | not a profession :) |
| 09:26.11 | liversmudge | morning all ... anyone awake? |
| 09:26.54 | mattwj2005 | aye |
| 09:26.54 | mattwj2005 | *I |
| 09:27.43 | liversmudge | more to the point any one awake who knows something about callerid .. and how to hide it adhoc? |
| 09:28.26 | mattwj2005 | do you have the default make sample files? |
| 09:28.31 | mattwj2005 | if so.... |
| 09:28.38 | mattwj2005 | look in the iax.conf configuration |
| 09:28.49 | mattwj2005 | under the [guest] section |
| 09:29.25 | mattwj2005 | that works for the iax trunks |
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| 09:29.50 | HarryR | I don't think so, their all sleepy heads in here |
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| 09:31.48 | liversmudge | oops clicked the wrong button |
| 09:32.31 | liversmudge | setting callerid .. adhoc to withhold the number ... any ideas? |
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| 09:34.30 | sooli | Uatec_: I want to use Music On Hold with my asterisk server but I have no sounds |
| 09:34.42 | hi365 | how do i set the delay bafor zap answers a line |
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| 09:36.03 | sbingner | sooli, step1: acquire sound |
| 09:36.05 | mattwj2005 | liversmudge |
| 09:36.09 | mattwj2005 | [guest] |
| 09:36.09 | mattwj2005 | type=user |
| 09:36.10 | mattwj2005 | context=default |
| 09:36.10 | mattwj2005 | callerid="Guest IAX User |
| 09:36.21 | liversmudge | ji |
| 09:36.21 | mattwj2005 | please note the the last line |
| 09:36.23 | liversmudge | hi |
| 09:36.57 | mattwj2005 | there should be a " at the en sorry |
| 09:37.04 | mattwj2005 | *end |
| 09:38.10 | Uatec_ | ah, not got round to doing music on hold yet |
| 09:38.13 | Uatec_ | sorry |
| 09:38.24 | liversmudge | ok , I do something similar, I put UNAVAILABLE as the caller id, but when this is passwed out via pri I get the pri hunt caller id |
| 09:38.42 | liversmudge | if I set a callerid .. then the pri sets it to what I want |
| 09:39.02 | liversmudge | if I leave it blank or none numerical it adopts the "base" hunt caller id |
| 09:39.20 | liversmudge | driving me up the wall |
| 09:39.34 | mattwj2005 | your using a T1? |
| 09:39.35 | sbingner | liversmudge, I don't think you can set a name over most telco circuits, just a number... or is it not a PRI to a telco? |
| 09:39.40 | mattwj2005 | aka pri? |
| 09:39.51 | mattwj2005 | I have never done voip at that level :( |
| 09:40.10 | liversmudge | Thats ok , sbingner Im not too bothered about the name .. |
| 09:40.18 | liversmudge | its the withholding of the number I want |
| 09:41.08 | DarKnesS_WolF | how can i know the busy / dial tone signal level that i'm running on asterisk ? |
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| 09:41.51 | liversmudge | I can set the callerpres in zapata.conf to unavailable and this then sends an unavailable dow the pri |
| 09:44.24 | liversmudge | Hi Matt : yes 4 E1 |
| 09:44.34 | liversmudge | Matt UK version of T1 |
| 09:44.45 | mattwj2005 | ~ 8 Mbps :| |
| 09:44.57 | mattwj2005 | wow that is a lot of information! |
| 09:45.00 | mattwj2005 | :) |
| 09:45.46 | liversmudge | yeah .. we take quite a few in an out bound calls |
| 09:45.56 | liversmudge | and resell the service |
| 09:46.00 | mattwj2005 | defintely |
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| 09:46.02 | mattwj2005 | nice |
| 09:46.16 | mattwj2005 | in the US I have had pretty good with ULAW codecs |
| 09:46.24 | liversmudge | you would think so .. but its so competitive you make bugger all money |
| 09:46.34 | mattwj2005 | I can my dad using it and an Internet connection |
| 09:47.08 | liversmudge | yeah broadband in the uk came alive for voip call swith the introduction of 8Mbit lines as they have a 512k upload |
| 09:47.28 | Uatec_ | what codec gives the best sound quality? |
| 09:47.31 | liversmudge | so a resonable router running a qos setup make a good few calls and still enough data for web browsing |
| 09:47.44 | liversmudge | g711 (or ulaw/alaw) |
| 09:47.59 | Uatec_ | liversmudge, i have 4 ISDN 2 lines :) |
| 09:48.02 | Uatec_ | cool |
| 09:48.13 | Uatec_ | i wonder why i'm getting such bad sound quality |
| 09:48.26 | liversmudge | well calls routed to PSTN networks through your ISDN cards are g711 |
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| 09:48.33 | Uatec_ | if i had sound on my PC i could listen to my call recordings, that would be helpful |
| 09:48.44 | liversmudge | thats a good starting place |
| 09:48.54 | Uatec_ | i have sound |
| 09:48.55 | Uatec_ | no speakers |
| 09:49.09 | Uatec_ | hmm |
| 09:49.15 | Uatec_ | onboard speakers on my screen |
| 09:49.16 | Uatec_ | excellent |
| 09:49.17 | Uatec_ | brb |
| 09:50.00 | *** join/#asterisk lwh (n=lwh192@rdsl-0593.tor.pathcom.com) |
| 09:52.44 | *** join/#asterisk manopulus (n=manopulu@213.197.181.154) |
| 09:53.54 | manopulus | hello, can someone help, i do not understand the regexps and overall i am not programmer. i have text: sip:NUM@hostname.xxx, i want to extract NUM (or name). with Set(sip_account=$["${SIPURI}" : "([a-zA-Z0-9]\+)\:"]) i receiving word sip from URI... how to do the rest? |
| 09:54.58 | HarryR | manopulus, you could use something like: [^\:]\:([0-9]+)@ |
| 09:55.21 | Hmmhesays | regexps make my brain melt |
| 09:55.36 | sbingner | regex = cool |
| 09:55.47 | HarryR | or for the domain name, [^\:]\:[^@]@(.+) |
| 09:56.04 | HarryR | uh, [^\:]+\:([0-9]+)@ and [^\:]+\:[^@]+@(.+) |
| 09:56.46 | manopulus | regex is cool, but i am not guru :) |
| 09:58.52 | manopulus | HarryR: last one return domain and port, i.e. 10.0.0.57:5061 ;) |
| 09:59.40 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 10:01.03 | mattwj2005 | well work is calling my name |
| 10:01.08 | mattwj2005 | I need to answer it :) |
| 10:01.14 | mattwj2005 | have a great day everyone! |
| 10:01.53 | *** join/#asterisk dawizard (n=dawizard@mimas.xios.be) |
| 10:02.30 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 10:04.00 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 10:04.33 | Uatec_ | manopulus, i have a cool little regex builder |
| 10:04.36 | Uatec_ | let me find a link for you |
| 10:05.24 | Uatec_ | damn |
| 10:05.26 | Uatec_ | it doesn't have details |
| 10:05.32 | Uatec_ | it's just called "Regex Builder" |
| 10:05.33 | Uatec_ | sorry |
| 10:06.41 | *** join/#asterisk neokairo (n=dflorell@ip-177.net-80-236-80.issy.rev.numericable.fr) |
| 10:12.14 | mkl1525 | my problem is that NoOp() doesn't show any * variables, using Set() and NoOp() I get the value of variables dialplan example http://pastebin.ca/457301 - any suggestions where the error could be? |
| 10:12.15 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 10:13.37 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 10:18.12 | Uatec_ | i'm getting an error message when i restart asterisk |
| 10:18.16 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 10:18.19 | Uatec_ | how can i find out where the message comes from |
| 10:18.25 | Uatec_ | i'm just getting Error in state 3 |
| 10:19.21 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 10:19.39 | *** join/#asterisk sooli (n=sooli@AMontpellier-156-1-122-17.w90-14.abo.wanadoo.fr) |
| 10:20.16 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 10:20.58 | Mahmoud | is there any phone dialing sound in asterisk's package? |
| 10:26.11 | Uatec_ | beep isn't it? |
| 10:26.17 | Uatec_ | as in that's a dial tone |
| 10:27.35 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-190-47.nycmny.east.verizon.net) |
| 10:27.43 | Mahmoud | will check it.. |
| 10:28.55 | Mahmoud | not similar |
| 10:29.53 | creativx | have you set proper locale |
| 10:30.00 | creativx | different countries have different beepers |
| 10:30.37 | Hmmhesays | haha |
| 10:31.24 | Mahmoud | it's not what i'm trying to do |
| 10:31.46 | creativx | what are you trying to do then |
| 10:31.58 | Mahmoud | i want to set extentions in my house, so whenever i dial from outside i dial the extention and get redirected to the proper phone/laptop |
| 10:32.05 | Mahmoud | the same time, i want to make it transparent |
| 10:32.16 | Mahmoud | so, when any one rings, he just hears the phone ringing, while it's a background music |
| 10:32.17 | *** join/#asterisk xexexe (n=xexexe@cpe-91733.ip.primehome.com) |
| 10:32.24 | Mahmoud | during the background music, i dial extentions ;) |
| 10:32.32 | *** part/#asterisk xexexe (n=xexexe@cpe-91733.ip.primehome.com) |
| 10:32.40 | Mahmoud | for others, it just sounds as normal dial tone |
| 10:32.51 | Mahmoud | the background lasts for 5 seconds |
| 10:33.04 | Mahmoud | after 5 seconds, it goes to my home's main phone |
| 10:33.51 | creativx | then use the proper args in dial() to allow the caller to enter digits, play the backgrund music and set up extensions that you can redirect yourself to |
| 10:34.18 | Mahmoud | can i tell dial to just send dial tone, and not ring the phone? |
| 10:34.24 | Uatec_ | mahmoud, try Ringing |
| 10:34.31 | Uatec_ | then WaitExten(10) (i think) |
| 10:34.46 | creativx | or waitexten |
| 10:35.07 | creativx | if you use "d" with dial it will wait for a 1-digit number while waiting for the call to be answered |
| 10:35.27 | creativx | and it jumps to that number in the current context |
| 10:35.50 | Mahmoud | only one digit? what if it was 3? |
| 10:36.20 | creativx | len(3) = 1 |
| 10:36.24 | creativx | :) |
| 10:36.27 | creativx | use waitexten() |
| 10:36.40 | creativx | show application WaitExten |
| 10:37.38 | Mahmoud | does waitexten provide phone ringing the same time? |
| 10:38.26 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 10:39.28 | creativx | do 1, answer |
| 10:39.30 | creativx | then 2, ringing |
| 10:39.32 | creativx | then 3, waitexten |
| 10:39.41 | creativx | if it doesnt provide ringing :) |
| 10:41.19 | creativx | yup you must use ringing first |
| 10:41.26 | creativx | i just tested |
| 10:41.37 | Mahmoud | same here, didn't ring it self |
| 10:48.40 | DarKnesS_WolF | hello Mahmoud :-) |
| 10:48.43 | DarKnesS_WolF | Mahmoud: min masr ? |
| 10:48.52 | Mahmoud | min jahanam |
| 10:49.01 | DarKnesS_WolF | Mahmoud: agd3 nas :-) |
| 10:49.09 | Mahmoud | jahanam agd3 nas? |
| 10:49.21 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 10:49.25 | DarKnesS_WolF | all ppl are agd3 nas :-) |
| 10:49.30 | Mahmoud | ana min UAE |
| 10:49.43 | Uatec_ | Question to people familiar with the snom 190: When i get the MWI lights up, i press vmail and it tries to dial 'asterisk'. That's not the extension of my voicemail. how can i point it to the right number? |
| 10:49.59 | DarKnesS_WolF | Mahmoud: i think u can use dial,r |
| 10:50.18 | DarKnesS_WolF | so while ur dialing the calling will hear a rining tone |
| 10:50.39 | DarKnesS_WolF | Uatec_: in the options u'll find a mailbox number |
| 10:50.43 | DarKnesS_WolF | u add the vmail box |
| 10:50.52 | DarKnesS_WolF | i used this here on my snom's 300 320 |
| 10:51.26 | Uatec_ | DarKnesS_WolF, there's the mailbox number, but that's the number of the mailbox for this phone, not for the voicemail application |
| 10:52.09 | DarKnesS_WolF | now i have a problem i have a gsm gateway connected as exnetion " phone port in this devices connected to FXO port in asterisk " i can dial out very good .. but when i get gsm calls if the gsm mobile hands up the devices don't hang up and still giving the asterisk a dial tone so the asterisl feels that the call still on an idea ? |
| 10:52.34 | DarKnesS_WolF | Uatec_: no u don't add the mailbox number u add the voicemailmain number |
| 10:52.53 | DarKnesS_WolF | so when u dial message u'll go to ur voicemailmain app. |
| 10:56.25 | Uatec_ | ok, DarKnesS_WolF, i've set that to my mailbox number |
| 10:56.31 | Uatec_ | but it's still trying to dial 'asterisk' |
| 10:57.35 | DarKnesS_WolF | hmmmm |
| 10:57.46 | DarKnesS_WolF | Uatec_: strange never had such problems with snome. |
| 10:58.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:58.55 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 11:01.23 | Uatec_ | werid |
| 11:02.21 | DarKnesS_WolF | the device still giving tone to asterisk :-s |
| 11:03.32 | Uatec_ | i've got two lines coming in to my b410p device, but if they dial in on one number, i want to ignore that number, because there's a fax machine on it |
| 11:03.52 | Uatec_ | how can i make asterisk ignore that number? |
| 11:04.02 | Uatec_ | i don't have an extension for it at the moment |
| 11:09.50 | *** join/#asterisk swyrus (n=woder@demon-gw.denbridgemarine.com) |
| 11:10.59 | swyrus | Hi, can anyone help me to configure my asterisk sip port? |
| 11:11.32 | Uatec_ | swyrus, google? |
| 11:11.34 | *** join/#asterisk saravia (n=jovannot@69.79.144.206) |
| 11:11.34 | Uatec_ | look on there |
| 11:11.39 | Uatec_ | read the documentation |
| 11:11.44 | Uatec_ | then, if you have any problems, come and ask |
| 11:12.38 | swyrus | have tried many web sites. setting seem right. Am just missing a config somewhere |
| 11:15.29 | Uatec_ | well what's the problem? |
| 11:15.37 | Uatec_ | what shows up when you type: sip show peers |
| 11:16.25 | swyrus | Name/username Host Dyn Nat ACL Port Status |
| 11:16.25 | swyrus | 0 sip peers [0 online , 0 offline] |
| 11:16.38 | Uatec_ | in that case, you've not defined anything in your sip.conf |
| 11:16.53 | swyrus | I think it might be a port issuse as I cant telnet to 5060 |
| 11:16.53 | Nugget | telnet is eeeeeeevil! |
| 11:17.16 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:18.04 | swyrus | here is my sip.conf |
| 11:18.15 | swyrus | [general] |
| 11:18.15 | swyrus | port=5060 |
| 11:18.15 | swyrus | bindaddr=0.0.0.0 |
| 11:18.15 | swyrus | disallow=all |
| 11:18.15 | swyrus | allow=ulaw |
| 11:18.15 | swyrus | context=default |
| 11:18.17 | swyrus | srvlookup=yes |
| 11:18.19 | swyrus | [usr1] |
| 11:18.21 | swyrus | type=friend |
| 11:18.23 | swyrus | username=1234 |
| 11:18.25 | swyrus | secret=paul |
| 11:18.27 | swyrus | callerid=Paul<1234> |
| 11:18.29 | swyrus | host=dynamic |
| 11:18.31 | swyrus | allow=all |
| 11:18.33 | swyrus | context=from-internal |
| 11:18.34 | cpm | please use a pastebin |
| 11:18.42 | cpm | dont' flood the channel |
| 11:18.47 | saravia | you cant telnet to 5060 because sip port is udp/5060, if you use telnet you are aswking by tcp/5060 which is wrong |
| 11:19.02 | swyrus | sorry cpm a bit new here |
| 11:19.09 | cpm | no worries |
| 11:19.31 | cpm | http://rafb.net/paste/ |
| 11:19.46 | cpm | then post the url |
| 11:19.50 | cpm | it's easier on everyone |
| 11:21.37 | *** join/#asterisk zePh7r (n=nome@53-175.dial.nortenet.pt) |
| 11:21.43 | *** part/#asterisk zePh7r (n=nome@53-175.dial.nortenet.pt) |
| 11:22.10 | swyrus | http://rafb.net/p/DqYNQB17.html |
| 11:27.49 | Hoondie | Hey people.. Does anyone know why the Ringing command, the WaitExten commands don't work unless i do a playback command before any of them? |
| 11:31.00 | DarKnesS_WolF | what is better to connect a gsm gateway to asterisk " as extension ? or as a trunk " ? |
| 11:33.15 | *** join/#asterisk [ProB]CrazyMan (n=CrazyMan@pD9E91C0E.dip0.t-ipconnect.de) |
| 11:34.32 | DarKnesS_WolF | Hoondie: try option r for dial command this will make the caller get a ringing tone even the pther phone still not ringing |
| 11:36.37 | Hoondie | DarKnesS_WolF: Yea, that doesn't work unless I have used the playback command also |
| 11:38.42 | *** part/#asterisk phyber (i=phyber@irssi.co.uk) |
| 11:41.28 | Uatec_ | does anybody know how i can get a list of the audio file formats that my particular asterisk configuration supports? |
| 11:42.17 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-168-234.dhcp.southerncoastalcable.net) |
| 11:43.33 | DarKnesS_WolF | Hoondie: no it dont need that u put it in the dial command |
| 11:44.29 | Hoondie | DarKnesS_WolF: It's not ment to be like that, but that's whats happening, obviously it's some sort of bug |
| 11:46.32 | liversmudge | Uatec : show translations |
| 11:46.45 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-128-158.buff.east.verizon.net) |
| 11:50.31 | creativx | Hoondie: answer -> ringing -> waitexten(10) -> hangup worked like a charm here |
| 11:51.32 | Hoondie | creativx: That doesn't work for me either |
| 11:52.17 | Hoondie | creativx: It won't play the ringing, it also won't accept input from waitexten, just goes to timeout |
| 11:56.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:57.31 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:04.55 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
| 12:10.01 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 12:11.54 | nasls_lsa | hi |
| 12:21.11 | *** join/#asterisk Mavvie (n=edwin@ppp246-56.lns3.syd6.internode.on.net) |
| 12:23.43 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 12:23.45 | *** join/#asterisk Ast001 (n=uros@77-105-50-3.adsl-1.sezampro.yu) |
| 12:24.02 | Ast001 | Hello |
| 12:24.51 | Ast001 | Can this /proc/interrupts be a problem because of which my * system suffer from music on hold cutting |
| 12:24.55 | Ast001 | <PROTECTED> |
| 12:25.38 | Ast001 | seems like my LAN card is sharing interrupt with INTELICH5 |
| 12:25.41 | zeeesh | i hv peer 100 at asterisk machine"A" and peer 200 at asterisk machine "B".. how can both peer will comunicate by using sip through xpro ???? |
| 12:27.13 | Ast001 | i wonder wich 90 things are using zaptel |
| 12:27.17 | Ast001 | zaptel 195528 90 zttranscode,wcte11xp |
| 12:27.30 | Ast001 | is output of lsmod | grep zaptel |
| 12:31.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:34.03 | *** join/#asterisk JoanaDoe (n=yvonne@193.174.25.23) |
| 12:35.17 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 12:36.38 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 12:37.11 | *** join/#asterisk sashion (n=djbdsf@196.211.16.202) |
| 12:39.05 | sashion | anyone wanna explain this? |
| 12:39.05 | sashion | chan_zap.c:8140 zt_pri_error: !! Got a UA, but i'm in state 1 |
| 12:39.05 | FuriousGeorge | speaking of zaptel holy cow. ive been experiencing deadlocks lately, so i tried upgrading to 1.2.17/1.2.18 zaptel last night |
| 12:39.06 | FuriousGeorge | what a disaster |
| 12:40.24 | sashion | well seems we both in the same boat |
| 12:40.24 | FuriousGeorge | i didnt test the system before i left b/c it was so late. huge mistake. they get me up this morning because EVERY call would lock the entire system, not just asterisk. the guy at the location seemed to describe a kernel panic |
| 12:40.34 | sashion | I get that error.. then ALL my calls drop and asterisk happily says "Primary D-Channel on span 1 up" |
| 12:41.09 | FuriousGeorge | looking up at the topic i see that the upgrades were released last night |
| 12:41.15 | sashion | shaw... atleast my system stays up, but client gets infuriated about the calls |
| 12:41.39 | sashion | another ugrade.. |
| 12:41.46 | sashion | zaptel 1.4.2 |
| 12:42.00 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666864.dsl.bell.ca) |
| 12:42.32 | sashion | hmmm great... its like playing russian roullete with a barrel of fish and a barrel cylinda of 300 shots, but only 1 bullet :) |
| 12:42.50 | FuriousGeorge | so i downgraded back to the configuration that only deadlocked or crashed the server once a week |
| 12:43.15 | sashion | lol thats promising |
| 12:43.37 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 12:43.37 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 12:43.44 | FuriousGeorge | ironically, i have another almost identical computer that runs flawlessly. i think im gonna copy this ones root partition over that one |
| 12:43.52 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 12:44.42 | sashion | i have the same situation.. 2 other callcenters running 10x more capacity, but never a single problem on those systems |
| 12:44.47 | sashion | once again, identical setups... |
| 12:44.52 | sashion | barring hardware |
| 12:45.18 | *** join/#asterisk sashion (i=synergy@wbs-41-208-217-247.wbs.co.za) |
| 12:49.08 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:49.08 | FuriousGeorge | even my hardware is the same. one mb has onboard audio. thats the only difference (and one has 2 tdm400p (the naughty one) and the other has 1) |
| 12:49.10 | FuriousGeorge | different ram |
| 12:49.12 | FuriousGeorge | same speed |
| 12:49.12 | sashion | ghost in the machines :) |
| 12:49.13 | FuriousGeorge | ive made that joke so many times over the last two months that it has ceased being funny :( |
| 12:49.15 | FuriousGeorge | and i have a pretty good sense of humor |
| 12:49.15 | sashion | hmmm... have you tried miming the joke ? |
| 12:49.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:49.18 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 12:50.09 | sashion | oh well... guess I'll be upgrading to 1.4.3 tonight in a desperate hope to free my slaved soul from debugging |
| 12:50.09 | FuriousGeorge | ive never met a trusted voice that recommended 1.4.x over 1.2.x |
| 12:50.09 | tzanger | heh |
| 12:50.09 | tzanger | I wonder what fun I'll encounger... going from svn trunk from February to today |
| 12:50.10 | puzzled | hi |
| 12:50.15 | sashion | everyone rants about 1.4.x |
| 12:50.23 | sashion | threaded call allocation |
| 12:50.24 | FuriousGeorge | in terms of stability? |
| 12:50.26 | sashion | SLA... bla bla |
| 12:50.32 | nasls_lsa | people help: U use a grandream phone. When I wall from that it works fine , when I get calls I can't hear anything :( |
| 12:50.35 | FuriousGeorge | or features |
| 12:50.39 | sashion | i have 2 stable sites running 1.4.x... |
| 12:50.47 | sashion | but this one client.. their building is curseD! |
| 12:50.49 | sashion | I swear |
| 12:51.15 | FuriousGeorge | my main culprit is the tdm400ps |
| 12:51.18 | nasls_lsa | sorry , wrong |
| 12:51.23 | FuriousGeorge | at last, my top suspect |
| 12:51.27 | FuriousGeorge | *at least |
| 12:52.03 | nasls_lsa | when I dial from grandstream 2000 ( SIP Phone ) I can't hear and they can't hear me .. when I get calls all fine .. |
| 12:52.19 | sashion | lol |
| 12:52.23 | sashion | CSI for Asterisk |
| 12:52.41 | sashion | so mister TDM400PS, where were you on the night of.... |
| 12:53.38 | FuriousGeorge | even after i upgraded and every call would crash the system, the two or three times they picked up, heard my voice, and didnt need to reboot, i had one or both tdm400p disabled |
| 12:54.43 | sashion | hmm I hate analogue.. |
| 12:54.47 | sashion | always causes issues :) |
| 12:55.44 | SuPrSluG | nasls_lsa:sounds like nat issue |
| 12:55.46 | FuriousGeorge | im being convinced by myself and others that a channel bank would be the way to go |
| 12:56.10 | nasls_lsa | I read about that BUT : asterisk and SIP phone are in the same LAN |
| 12:56.28 | nasls_lsa | which connected via 2 swiches ... |
| 12:56.54 | FuriousGeorge | nasls_lsa: codec issue |
| 12:57.20 | FuriousGeorge | when you call out from the problematic direction one end is picking a codec the other doesnt understand |
| 12:57.22 | FuriousGeorge | just a guess |
| 12:57.35 | Hoondie | nasls_lsa: where are the people located that you are dialing?/ |
| 12:57.40 | FuriousGeorge | u trying to use g729? |
| 12:57.49 | nasls_lsa | I think that too , cause once the phone says it uses PCMU codec and all works fine , and the other says GSM and can't hear / they can't hear me |
| 12:58.01 | nasls_lsa | internal phones , |
| 12:58.11 | FuriousGeorge | same subnet too? |
| 12:58.14 | nasls_lsa | yes |
| 12:58.27 | nasls_lsa | using a grandstream Analog getway .. |
| 12:58.30 | FuriousGeorge | maybe one of your ends doesnt do gsm? |
| 12:58.44 | FuriousGeorge | since everything should do pcmu |
| 12:59.00 | FuriousGeorge | anyway set both to pcmu |
| 12:59.08 | FuriousGeorge | ~pcmu is ulaw right? |
| 12:59.14 | jbot | okay, FuriousGeorge |
| 12:59.14 | FuriousGeorge | ~pcmu |
| 12:59.15 | jbot | extra, extra, read all about it, pcmu is ulaw right? |
| 12:59.16 | nasls_lsa | I think so |
| 12:59.23 | FuriousGeorge | fOrk |
| 12:59.25 | nasls_lsa | thank you for your help .. |
| 12:59.34 | FuriousGeorge | ill fix |
| 12:59.40 | *** part/#asterisk Ast001 (n=uros@77-105-50-3.adsl-1.sezampro.yu) |
| 13:00.41 | *** join/#asterisk awk (i=awk@inet-corp.com) |
| 13:01.53 | awk | out of freepbx/asterisk hardcore console users :) what does 1 say is better freepbx or asterisk gui ? |
| 13:02.08 | nasls_lsa | well , it worked for my internal phones , prety nice |
| 13:02.24 | nasls_lsa | but now I have problem when I get phonecalls from my mISDN |
| 13:03.29 | FuriousGeorge | ~pcmu is a logarithmic pulse-code modulation (PCM) sample for signals of voice frequencies, sampled at the rate of 8000 samples/second. a.k.a. g.711 or u-law, pronounced myou-law |
| 13:03.31 | jbot | ...but pcmu is already something else... |
| 13:03.52 | FuriousGeorge | jbot: No, pcmu is a logarithmic pulse-code modulation (PCM) sample for signals of voice frequencies, sampled at the rate of 8000 samples/second. a.k.a. g.711 or u-law, pronounced myou-law |
| 13:03.54 | jbot | okay, FuriousGeorge |
| 13:04.01 | FuriousGeorge | damn straight |
| 13:07.17 | Hoondie | FuriousGeorge: Have you got any idea why i have to use the playback command before the ringing or waitexten commands will work? |
| 13:07.32 | *** join/#asterisk ccvp (n=axe@66.0.46.210) |
| 13:07.34 | ccvp | hello fellow internet addicts - are we all looking forward to another long & glorious day of irc? yet again, eat, shit, wake up, rinse/repeat/recycle |
| 13:07.57 | last1 | I also have a few bj's thrown in in that cycle |
| 13:11.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.24 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:15.13 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 13:15.50 | *** join/#asterisk simplexio (n=sim@ogw.oamk.fi) |
| 13:15.56 | FuriousGeorge | Hoondie: no idea |
| 13:16.12 | last1 | anyone owns a 7970 ? |
| 13:16.33 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
| 13:16.34 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:16.44 | simplexio | anyidea where i find Dialplan command "Backround" code ? |
| 13:17.07 | Hoondie | FuriousGeorge: Thanks anyway |
| 13:18.40 | simplexio | i need play sound from agi connection and be able to start dial while music is playing. |
| 13:19.21 | *** part/#asterisk ccvp (n=axe@66.0.46.210) |
| 13:23.22 | sashion | simplexio: check app_background.c |
| 13:23.47 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-71-125-244-192.nycmny.east.verizon.net) |
| 13:23.58 | VJFROMGT | can asterisk do silent supression? |
| 13:24.37 | zeeesh | i hv peer 100 at asterisk machine"A" and peer 200 at asterisk machine "B".. how can both peer will comunicate by using sip through xpro ???? |
| 13:26.30 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 13:27.26 | SuPrSluG | zeeesh:what are u using to connet box A and box B? |
| 13:27.43 | VJFROMGT | iax2 |
| 13:28.00 | VJFROMGT | http://keystoneit.wordpress.com/2006/08/16/trunking-two-trixbox-servers/ |
| 13:28.10 | VJFROMGT | that link is for you zeeesh |
| 13:32.08 | [TK]D-Fender | VJFROMGT: No. |
| 13:32.14 | blaylock | what does it mean when ther are 1's and 0's filling up the Tx and Rx fields on a single span on a TE412p card? |
| 13:32.40 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 13:34.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:34.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:35.13 | VJFROMGT | tk ,, i am interconnecting 2 boxes, and have limited bandwidth (128kbps), I need to have at least 7 channels ,, what would be the best way to conserve bandwidth |
| 13:35.26 | VJFROMGT | so far i am using g729 with iax2, anything else? |
| 13:35.28 | [TK]D-Fender | blaylock: funny... I don't remeber those dards having a display on them..... |
| 13:35.38 | blaylock | zttool |
| 13:36.03 | [TK]D-Fender | VJFROMGT: thats it unless you can change the IAX2 voice frame size. |
| 13:36.10 | illsci | morning |
| 13:36.18 | [TK]D-Fender | blaylock: Typically that shows you which channels are in use. |
| 13:36.31 | blaylock | hmm |
| 13:37.05 | blaylock | i keep getting HDLC notices on the D channels as well |
| 13:37.25 | blaylock | bad FCS or HDLC Abort |
| 13:37.34 | blaylock | but not all the time |
| 13:37.51 | VJFROMGT | what are the bads of chainging IAX2 voice frame size |
| 13:38.27 | blaylock | only seems to happen when a dail up modem tries to connect to our dail up service |
| 13:38.41 | [TK]D-Fender | blaylock: pastebin "cat /proc/interrupts" and "dmesg" |
| 13:42.16 | *** join/#asterisk joe_bag_odonuts (n=erne@dsl027-164-165.atl1.dsl.speakeasy.net) |
| 13:42.55 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 13:43.08 | Polis_ttt | god evening |
| 13:43.16 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 13:43.42 | blaylock | pastebin? |
| 13:43.47 | blaylock | sorry, ive not used that before |
| 13:44.31 | *** join/#asterisk mkl1525 (n=mkl1525@pD9532A8F.dip0.t-ipconnect.de) |
| 13:44.48 | blaylock | [TK]D-Fender, do want me to priv mesg you and paste it in? |
| 13:45.04 | joe_bag_odonuts | I'm having a problem dialing out with Local Channel. I'm using the Manager API. What's funny is that I can dial a registered sip user on the asterisk server and I can dial from a registered sip user (device) to a number on the pstn using my termination providers server. I can send details if you'd like. |
| 13:45.10 | *** join/#asterisk s-ndh-c (n=michi@85.93.11.18) |
| 13:45.41 | s-ndh-c | hey guys, would it be possible to implement a trunk between my existing pbx and asterisk via a standard passive isdn card? just for testing purposes? |
| 13:45.55 | s-ndh-c | an e1 interface card costs ways too much for a test |
| 13:46.01 | joe_bag_odonuts | Here are the manager variables. |
| 13:46.01 | joe_bag_odonuts | Variables in Manager |
| 13:46.02 | joe_bag_odonuts | <PROTECTED> |
| 13:46.02 | joe_bag_odonuts | <PROTECTED> |
| 13:46.02 | joe_bag_odonuts | <PROTECTED> |
| 13:46.02 | joe_bag_odonuts | <PROTECTED> |
| 13:46.04 | joe_bag_odonuts | <PROTECTED> |
| 13:46.06 | joe_bag_odonuts | <PROTECTED> |
| 13:46.08 | tzanger | STOP FLOODING!! |
| 13:46.08 | joe_bag_odonuts | <PROTECTED> |
| 13:46.20 | s-ndh-c | joe_bag_odonuts: heard of nopaste yert? |
| 13:46.22 | puzzled | joe_bag_odonuts: put your output up on http://pastebin.ca and copy the link here to prevent flooding |
| 13:46.24 | s-ndh-c | *yet* |
| 13:46.40 | tzanger | I asked him nicely not to flood in -dev |
| 13:46.59 | creativx | being nice never helped |
| 13:47.29 | s-ndh-c | is that possible? or is chan misdn or simmilar not thought for that? |
| 13:48.03 | *** join/#asterisk prevail20 (n=prevail2@wan-hts-26.highertech.net) |
| 13:48.14 | blaylock | [TK]D-Fender, nevermind, i found the site |
| 13:48.46 | puzzled | s-ndh-c: I think you can do that with chan_misdn and chan_capi. Get a cheap HFC-S chipset based card and try |
| 13:49.07 | creativx | cpm :> |
| 13:49.08 | puzzled | s-ndh-c: chan_capi requires a more expensive Eicon Diva Server card. |
| 13:49.54 | s-ndh-c | puzzled: i just want to use that to make sip calls to the pbx and vice versa |
| 13:50.21 | s-ndh-c | how would i need to setup that thing? |
| 13:50.38 | puzzled | s-ndh-c: you can't do sip calls over an isdn link :) |
| 13:51.12 | s-ndh-c | puzzled: i want to call from sip to phones on the pbx and vice versa |
| 13:51.24 | s-ndh-c | that doesnt work? |
| 13:51.42 | joe_bag_odonuts | Sorry about the flood the details of the manager variables, the sip.conf and extensions.conf as well as some of the output from the console can be found here. |
| 13:51.42 | joe_bag_odonuts | http://pastebin.ca/457563 |
| 13:51.45 | puzzled | s-ndh-c: dunno how you set it up. Grab mISDN, mISDNuser, install them, read the docs and ask on the misdn mailinglist at http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk |
| 13:52.10 | s-ndh-c | puzzled: will try that thx |
| 13:52.11 | s-ndh-c | :) |
| 13:52.15 | puzzled | s-ndh-c: from sip phone to asterisk via isdn link to pbx should be possible |
| 13:52.47 | s-ndh-c | puzzled: yeah thats what i want for testing, if that works we will buy a e1 card for that |
| 13:52.49 | [TK]D-Fender | joe_bag_odonuts: your local channel does not match the context |
| 13:53.26 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:55.03 | blaylock | [TK]D-Fender, pastebin isnt working for some reason |
| 13:55.13 | blaylock | [TK]D-Fender, ill priv mesg you |
| 13:55.15 | [TK]D-Fender | blaylock: www.pastebin.ca |
| 13:55.17 | joe_bag_odonuts | Are you concerned about the /n at the end of the channel. That's there to prevent the termination of the channel. I can't tell you where I found that right now. But I tried it with and without the /n. It doesn't work either way. In addition since the console output says the code was run, it must match the context. |
| 13:55.21 | blaylock | ok |
| 13:55.23 | prevail20 | would someone be able to help me figure out how to get a TE212 card to work with a T1 Channel bank? I am currently Red alarms |
| 13:55.44 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:55.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:56.46 | [TK]D-Fender | joe_bag_odonuts: I didn't see your output when I wrote that. I jsut noticed when you changed the number, the call file length was wrong |
| 13:57.01 | joe_bag_odonuts | In addition the code runs fine when the Dial is (SIP/registered sip device) |
| 13:57.27 | joe_bag_odonuts | It's just the call to my provider that doesn't run. |
| 13:58.42 | joe_bag_odonuts | I've seen a few discussions of this problem out on the web but no solutions. Every solution that has been proposed doesn't deal with SIP. It's always IAX or ZAP. The authors of those solutions say this should work. Perhaps it does but not with SIP. |
| 13:58.56 | [TK]D-Fender | joe_bag_odonuts: try calling out to your provider NORMALLY, without call files |
| 13:59.11 | joe_bag_odonuts | I do and can without any problem |
| 14:00.13 | *** join/#asterisk chris_asterisk` (n=test@217.205.167.170) |
| 14:00.27 | chris_asterisk` | hey guys, if i do a sip show peers, it shows my sipgate account is registered |
| 14:00.35 | chris_asterisk` | yet on the site, it says the status is offline |
| 14:00.38 | chris_asterisk` | which is wrong? |
| 14:00.43 | [TK]D-Fender | joe_bag_odonuts: it'd be nice to see what that MACRO is doing.... |
| 14:00.48 | chris_asterisk` | does it take time to register with the sipgate account? |
| 14:01.24 | joe_bag_odonuts | Coming up. I'll post the macro and the results from the macro in just abit. |
| 14:02.00 | [TK]D-Fender | joe_bag_odonuts: Also I don't see the remote side even answering the dial... |
| 14:02.03 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:02.44 | Polis_ttt | have anyone used vicidial? i have som problem in my extensions.conf-file i think, can't call in or out whit the example-file, but can make my own but it doen't work whit vicidial :( |
| 14:03.46 | simplexio | sabakas1: im using 1.4.2 and grep -i back * in apps/ dosent give me any file |
| 14:04.30 | chris_asterisk` | anyone? :) |
| 14:05.37 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:06.24 | [TK]D-Fender | chris_asterisk`: does it WORK?? |
| 14:08.41 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 14:09.01 | blitzrage | this may be off topic, but I'm gonna ask it anyways :) What bluetooth headsets does everyone like? I'm looking for something small, that can charge from a USB cable, and just sits in the ear (no loop on the outside) |
| 14:09.16 | blitzrage | I guess I can make this on topic by saying I'll be using it with chan_cellphone :) |
| 14:09.24 | mkl1525 | Hi, when an agent enters a queue after a caller is already in retry time doesn't seem to work (only when no agent was in queue then it seems to work) - any hints if this is a bug or a configuration issue? |
| 14:10.29 | joe_bag_odonuts | Here's the link to the macro and the console output http://pastebin.ca/457595 |
| 14:10.47 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-239.hsd1.mi.comcast.net) |
| 14:10.56 | joe_bag_odonuts | Notice the macro is running as if the call had been made. Only no call occurs |
| 14:11.21 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 14:11.23 | Chris-NB | hi |
| 14:11.38 | unik-rados | do you think that Asterisk can be a good solution for a voicemail system in an enterprise environment? 6000+ users? |
| 14:12.08 | blitzrage | if you cluster it, probably |
| 14:12.11 | [TK]D-Fender | joe_bag_odonuts: Well your first pastbin seems to have stopped at the point where the first call is answered..... whats the actual problem at this point? |
| 14:12.12 | unik-rados | current system we have and another system we're looking at is just not good enough. |
| 14:12.14 | Chris-NB | anyone discovered that behaviour: If I call from a sip client outside via a sip provider, asterisk does net recognize if I hang up befor the first ringtone |
| 14:12.36 | Chris-NB | so the call remains and doesn't get disconnected |
| 14:12.44 | blitzrage | unik-rados: you might want to look at minivoicemail as that will let you build your own voicemail system in the dialplan, but no ODBC or IMAP access with it |
| 14:12.54 | *** join/#asterisk JoanaDoe (n=JoanaDoe@193.174.25.23) |
| 14:12.59 | unik-rados | blitzrage: thanks. this would be in an Avaya environment. does anyone have any experience with it? are there any limitations I need to watch out for? |
| 14:12.59 | mosty | Chris-NB: is there NAT in the picture there? |
| 14:13.06 | Chris-NB | mosty, jep. |
| 14:13.21 | Chris-NB | I thought abaout that, causing that |
| 14:13.26 | blaylock | [TK]D-Fender, http://www.pastebin.ca/457597 |
| 14:14.07 | mosty | Chris-NB: do you have correct nat setting on the server side and externhost setting on the client side? |
| 14:14.11 | blitzrage | unik-rados: well, you need to try out the voicemail system first to see if the standard voicemail will do what you want -- either standard voicemail can let you use IMAP and ODBC (but not customizable), or you can use minivoicemail but not attach to IMAP or ODBC. There was also a patch in the bug tracker I remember that let you change some of the prompts and stuff in standard voicemail |
| 14:14.22 | chris_asterisk` | mosty : nat = yes |
| 14:14.27 | chris_asterisk` | externhost? |
| 14:14.39 | chris_asterisk` | the SIP trunk supposedly is connecting, according to asterisk |
| 14:14.51 | Chris-NB | mosty, I've defined extern IP and localnet on the asterisk side |
| 14:14.53 | blitzrage | externhost is used when your asterisk is behind nat |
| 14:14.56 | Chris-NB | and defined nat=yes |
| 14:15.07 | blitzrage | nat=yes is only useful when the OTHER end is behind NAT |
| 14:15.08 | Chris-NB | but what do you mean with externhost on the client side? |
| 14:15.11 | mosty | Chris-NB: is asterisk the server side or the client side? |
| 14:15.28 | Chris-NB | asterisk is server |
| 14:15.35 | unik-rados | blitzrage: got it. I'll give it a shot. but as far as Avaya specific stuff, are there any limitations? I heard that you can't set a message light on for example, when you get a voicemail. do you know if it's true? |
| 14:15.58 | blitzrage | no idea -- I don't use crappy PBX systems :) |
| 14:16.07 | mosty | Chris-NB: what do you mean then when you say a sip client calls via a sip provider? in that case isn't asterisk acting as sip client? |
| 14:16.34 | nasls_lsa | I have problem when I get an mISDN call to my asterisk , and answer it using SIP device , grandstream 2000 .. can't hear anything :( |
| 14:16.34 | unik-rados | blitzrage: :) |
| 14:16.34 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:16.51 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 14:18.08 | Chris-NB | a sip client (phone) calls via my local asterisk out into pstn via a sipprovider |
| 14:18.26 | Chris-NB | so my local asterisk is server vor my client and ... .peer (?) to the sip provider |
| 14:18.35 | joe_bag_odonuts | [TK]D-Fender: The actual problem is that no call is being made or completed. Asterisk is proceeding as if the call has occured at that a bridge between the called number is established. However, the number actually being called (my cell phone) never rings, I never answer it, and my provider never sees the call in its logs. |
| 14:19.49 | [TK]D-Fender | joe_bag_odonuts: Well typically if you don't answer your cell, your *VM* kicks in, or the cell company gives up and probably plays a message to the caller, technically answering the line. |
| 14:20.47 | mosty | Chris-NB: ok then nat=yes should not be set on your asterisk server, but externip or externhost or whatever it's called these days should be set |
| 14:21.34 | Chris-NB | mosty, Jep, I've allready set that. but ... it's not working correctly : / |
| 14:21.39 | BSD_Tech | /clea |
| 14:21.53 | Chris-NB | mosty, If I hang up, the call doesn't get closed |
| 14:21.56 | joe_bag_odonuts | [TK]D-Fender: I can't answer a call when the cell phone doesn't ring. If I use a registered sip device (the phone on my laptop) and call my cell. The call goes through just fine. This occurs in another context but essentially its the same dial command, just no macros. |
| 14:22.17 | mosty | Chris-NB: do you allow the sip phone to reinvite? try disabling that |
| 14:22.19 | Chris-NB | (If I hang up before the first ringtone) |
| 14:22.48 | Chris-NB | mosty, canreinvite=no |
| 14:22.54 | joe_bag_odonuts | [TK]-Fender: My provider tells me it doesn't see the call even arriving at its server. |
| 14:24.01 | joe_bag_odonuts | [TK]-Fender: I haven't done a TCP dump on my end to see if it is actually going out yet. |
| 14:24.53 | dfgas | stupid question, is there a way to use an internal modem with asterisk, heh |
| 14:25.16 | [TK]D-Fender | joe_bag_odonuts: this looks a bit circular. You have an "answer at the start of your exten and as your call file is calling FROM that exten TO that exten, its calling your cell TWICE. looks damn irregular. |
| 14:25.29 | [TK]D-Fender | dfgas: For what? |
| 14:25.56 | [TK]D-Fender | dfgas: It might help your server pass certain weight minimums for shipping for instance.... |
| 14:26.01 | dfgas | <PROTECTED> |
| 14:26.02 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:26.11 | dfgas | lol |
| 14:26.41 | [TK]D-Fender | dfgas: Seperately it might be used to keep some printed WIKI pages listed supported hardware from flaying around due to a slight breeze in your room... |
| 14:28.11 | irule | I found that in the sample dialplan I dial nine, and have no sound at all, and then I should keed dialing out to zap, is there a method to give the user a dial tone after 9? |
| 14:28.13 | joe_bag_odonuts | [TK]-Fender: Actually. I have a canreinvite=yes on the sip device [provider-conference]. That's what's causing what looks like a double call. However, you'll notice that the the second call is on a different channel (2) as opposed to (1). The Local channel (as I understand it) creates two channels. The first one makes the call the second one gets the call on the re-invite and falls through the rest of the dialplan. |
| 14:28.57 | [TK]D-Fender | irule: "ignorepat => 9" in your phone's context. |
| 14:29.10 | irule | thanks <[TK]D-Fender> |
| 14:29.47 | [TK]D-Fender | joe_bag_odonuts: No, thats not why. BOTH legs of the call call your cell. both sides auto-answer as step 1, GARUANTEEING that it calls you twice. read your logic again |
| 14:30.16 | joe_bag_odonuts | [TK]-Fender: If I change the canreinvite to no. Only one call shows up. |
| 14:31.03 | [TK]D-Fender | joe_bag_odonuts: look at what that exten in that macro DOES. its calling you once, and instantly calling out to you again as the OTHER sied of the call. |
| 14:31.28 | joe_bag_odonuts | [TK]-Fender:In any event it, appears no call is ever being made. |
| 14:32.03 | [TK]D-Fender | actually... "Exten: dialoutconf" <- what the ehck is THIS?! |
| 14:32.28 | [TK]D-Fender | joe_bag_odonuts: I may have missed something... that exten does not look valid. |
| 14:33.17 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 14:33.24 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 14:33.50 | joe_bag_odonuts | TK-Fender: Its valid. If I force the dial to something like Dial(SIP/registereddevice....) Everything works fine. It's just the call to the provider that is causing problems. |
| 14:34.44 | joe_bag_odonuts | TK-Fender in other words and internal extension can recieve the call and everything runs like a top. |
| 14:35.32 | [TK]D-Fender | joe_bag_odonuts: Doesn't look like it should work. the called side answers immediately, but the exten its supposed to drop it into doesn't match what you showed for the context - Exten: dialoutconf is not valid |
| 14:36.38 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 14:37.08 | mkl1525 | trying to install cdr_mysql, have asterisk-addons, libmysql dev files ready configure sees them but make doesn't build the module - any hints what to do |
| 14:37.54 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 14:39.15 | joe_bag_odonuts | TK-Fender: Just ran it again with a local extension (registered sip device) It worked fine. I'll post the consol output for that in just a second. |
| 14:39.52 | irule | is it possible to match 04433-4561-1895 but dial 04533-4561-1895? |
| 14:39.56 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 14:40.00 | Greek-Boy | hello |
| 14:40.09 | Greek-Boy | what is the best fax solution for asterisk? astfax? |
| 14:41.02 | irule | well rather match 044XX-XXXX-XXX but dial out 045XX-XXXX-XXX |
| 14:41.22 | [TK]D-Fender | joe_bag_odonuts: I'm sure that if you replace one end of that call with a SIP device instead of a local channel it WILL work. thats the poijnt. the locall channel does an ANSWER IMMEDIATELY! |
| 14:41.35 | [TK]D-Fender | irule: Yes. |
| 14:41.45 | drako | hwo you can set a bigger space for voicemail |
| 14:42.06 | irule | [TK]D-Fender well can I least get a hint on what to google for? heh |
| 14:42.10 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:42.22 | flujan | hi guys... I am having a lot of problems with asterisk queue. |
| 14:42.28 | [TK]D-Fender | ~variables |
| 14:42.51 | jbot | Docs on how to use variables in * can be found in doc/README.variables in your * source folder or http://www.voip-info.org/wiki-Asterisk+variables |
| 14:42.51 | *** part/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 14:42.51 | flujan | I am using the roundrobin strategy in the queue and x-lite as a softphone. |
| 14:43.05 | irule | thanks |
| 14:43.06 | [TK]D-Fender | irule: THERE |
| 14:43.14 | flujan | The problems is that asterisk try to delivery a queued call to a extension, if it is (In Use) |
| 14:43.18 | Greek-Boy | does asterisk 1.2.18 come with spandsp. |
| 14:43.25 | Qwell[] | Greek-Boy: no |
| 14:43.27 | flujan | so, I am having a big delay to delivery the calls... |
| 14:43.30 | Qwell[] | No version of asterisk does. |
| 14:43.30 | [TK]D-Fender | Greek-Boy: Of course not. |
| 14:43.42 | [TK]D-Fender | Greek-Boy: Nor will it ever. |
| 14:43.42 | drako | Greek-Boy, dunno but i can't compile spandsp on .17 |
| 14:43.58 | drako | Greek-Boy, it ends with error on compile time |
| 14:44.02 | nasls_lsa | has anyone problems of hearing people when they call him to ISDN phone line ? |
| 14:44.09 | *** join/#asterisk mkl1525 (n=mkl1525@i59F75B25.versanet.de) |
| 14:44.11 | flujan | Asterisk rings all the queue Members, not only the ones that are (not in use) this is the correct behavior? |
| 14:44.11 | mosty | flujan: asterisk queues suck. the only way to fix that is to use queue members that only allow one concurrent call (via sip.conf) |
| 14:44.37 | flujan | Can I configure the queue to use only the Not In Use queue Members? |
| 14:44.43 | flujan | mosty, how Can i do that? |
| 14:45.06 | flujan | configure in the sip.conf only one concurrent call? |
| 14:45.19 | mosty | flujan: yes |
| 14:45.25 | Hmmhesays | you can configure your endpoint to only accept one call also |
| 14:45.51 | flujan | Hmmhesays, x-lite supports it? |
| 14:46.00 | flujan | mosty, which option I shall set in sip,conf? |
| 14:46.23 | mosty | flujan: look up sip.conf in the wiki, i don't remember the exact details |
| 14:46.29 | drako | my voicemails get full too fast, how i can assign them more space/numbers ? |
| 14:46.41 | Hmmhesays | ride the lightning is a good song |
| 14:46.55 | Hmmhesays | you're inboxes get more than 100 messages? |
| 14:47.06 | drako | hmm-home, yes |
| 14:47.08 | Qwell[] | drako: maxmsgs in voicemail.conf |
| 14:47.10 | drako | Hmmhesays, yes |
| 14:47.27 | drako | Qwell, thank you |
| 14:48.48 | joe_bag_odonuts | TK-Fender: Here's the console output from the local Dial. Notice the Dial is to SIP/ernie http://pastebin.ca/457656 |
| 14:49.24 | joe_bag_odonuts | So the context is valid. |
| 14:54.17 | joe_bag_odonuts | TK-Fender: I've seen some users with a dial to ZAP/........ use an L option in the Dial command. I can't seem to find any explanation of what he L option does or what the (I assume time out variables) do anywhere. Do you know? |
| 14:54.48 | nasls_lsa | has anyone problems of hearing people when they call him to ISDN phone line -> asterisk -> SIP phone ( grandstream 2000 )? |
| 14:55.30 | [TK]D-Fender | joe_bag_odonuts: Ernie is still being double called, and ernie is answering. |
| 14:56.07 | [TK]D-Fender | joe_bag_odonuts: You do not want that macro effectively calling itself. |
| 14:56.42 | [TK]D-Fender | s/macro/exten/ |
| 14:56.49 | joe_bag_odonuts | Where do you see a double call. The macro doesn't call itself. It's only called once |
| 14:57.42 | [TK]D-Fender | joe_bag_odonuts: do you not see the second call out to ernie in there? |
| 14:59.10 | joe_bag_odonuts | Nope. I see one execution of Dial. 3 rings and an answer. |
| 15:00.09 | [TK]D-Fender | joe_bag_odonuts: http://pastebin.ca/457656 Lines 9 & 15 |
| 15:02.17 | *** part/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 15:02.24 | joe_bag_odonuts | Ok I see it now. I know how this is occurring. I initially was using another channel def to answer. That didnt' work either. I'll change it back. and show you again. Give me a minute. |
| 15:02.40 | nasls_lsa | has anyone problems of hearing people when they call him to ISDN phone line -> asterisk -> SIP phone ( grandstream 2000 )? any codec problems perhaps ? |
| 15:03.21 | dfgas | k, i am using xlite on asterisk, what is the best codec to use, cause what ever i am using atm is choppy |
| 15:04.03 | dfgas | i don't care about quality |
| 15:04.04 | [TK]D-Fender | joe_bag_odonuts: yes. that is bad... its like trying to dump a peron into that conf TWICE. |
| 15:04.11 | mosty | dfgas: gsm |
| 15:04.12 | *** join/#asterisk eBody (n=icechat5@209-163-204-250.static.twtelecom.net) |
| 15:07.19 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 15:07.20 | *** join/#asterisk rmayorga (i=rmyorg@168.243.89.17) |
| 15:10.18 | BSD_Tech | whats the command on linux to get it to send emails out |
| 15:10.30 | mosty | mail |
| 15:10.37 | BSD_Tech | to force the spool |
| 15:10.47 | [TK]D-Fender | joe_bag_odonuts: Your approach to this is quite off.... |
| 15:10.49 | mosty | depends on the smtpd |
| 15:11.01 | BSD_Tech | believe its just sendmail |
| 15:11.25 | eBody | i'm using a tdm2400 and sometimes when i call out the call will get connected to a call coming in; anyway to prevent this?? |
| 15:11.36 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:11.38 | [TK]D-Fender | eBody: No |
| 15:11.50 | Polis_ttt | is there anyway to debug a extensions.conf-file? |
| 15:12.03 | [TK]D-Fender | eBody: thts the curse of Analog. you can call out before the first right registers and becomes reserved |
| 15:12.47 | [TK]D-Fender | Polis_ttt: If you're asking for a validator, then no. it is an interpreted language and as things are determined in run-time this is not viable |
| 15:12.53 | eBody | so ur saying it can or CAN'T be prevented?? |
| 15:13.21 | [TK]D-Fender | eBody: can't |
| 15:14.04 | eBody | thanks Fender |
| 15:14.21 | mosty | [TK]D-Fender: there could be at least a partial validator, imo it's viable to build a useful tool |
| 15:14.24 | eBody | is this also a problem on older analog PBX |
| 15:14.26 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:14.53 | [TK]D-Fender | eBody: GotoIf($[${myvar}=hello]?6) <- this will work if the var isn't EMPTY, but will bomb otherwise. as this is evaluated in run-time and based on the syntax that vars and apps/functions use, there is no way to tell. |
| 15:15.14 | [TK]D-Fender | Polis_ttt: rather : GotoIf($[${myvar}=hello]?6) <- this will work if the var isn't EMPTY, but will bomb otherwise. as this is evaluated in run-time and based on the syntax that vars and apps/functions use, there is no way to tell |
| 15:15.18 | *** join/#asterisk Fieldy (i=yqhnVEBv@gentoo/contributor/Fieldy) |
| 15:15.35 | Polis_ttt | [tk]d-fender: i got a extensionfile that works for out-calls, but not when i call in, then i just get a "that option is invalid" so i wanna find where the problem is |
| 15:15.51 | joe_bag_odonuts | TK]D-Fender: Here is the post with the channel changed so that only one call is made. It still works with the local dial. http://pastebin.ca/457684 |
| 15:15.59 | mosty | [TK]D-Fender: obviously, but that doesn't mean a static analysis util couldnt help spot other kinds of bugs |
| 15:16.02 | [TK]D-Fender | eBody: Yup. Analog sucks. Don't like it, get a PRI. And if you have enough channels to validate a TDM2400 you probably should have considered a partial PRI |
| 15:16.38 | aptura | TK pri is limited to 700 feet? |
| 15:16.46 | [TK]D-Fender | joe_bag_odonuts: looks a lot better, but your approach is still off. |
| 15:16.57 | [TK]D-Fender | aptura: No, where did you get this idea? |
| 15:17.05 | joe_bag_odonuts | How is it off. |
| 15:17.32 | Sweeper | PRI's run over T1, aptura |
| 15:17.40 | [TK]D-Fender | joe_bag_odonuts: You just want to dial out and then run that AGI & go to conference right? Well calling a macro inside of dial is not the way. |
| 15:18.04 | aptura | Pri's electrical characteristic range is how long |
| 15:18.17 | Sweeper | same as a t1 \o\ |
| 15:18.34 | *** join/#asterisk serotonin|work (i=ryan@mail.tankprofiler.com) |
| 15:18.44 | Sweeper | since pri is a signalling thing, not a physical layer |
| 15:18.51 | [TK]D-Fender | joe_bag_odonuts: You should have 1 leg call out your provider. ALONE. thats IT. IF they answer it should dump them into a context that does the stuff in your agi. |
| 15:19.48 | [TK]D-Fender | There... PRI is now a physical layer! You can even mount a server on your "PRI"! ;) |
| 15:19.56 | Sweeper | TK ruins furniture D: |
| 15:20.29 | [TK]D-Fender | Sweeper: I'm friggen' Picasso! I just increased it value tenfold by laying my mark upon it! |
| 15:20.42 | mosty | i'm having trouble with MAX_INCLUDE_LEVEL being exceeded in my extensions.conf, is that a recursive level? i haven't been able to spot any circular include's |
| 15:20.43 | Sweeper | roight |
| 15:21.13 | aptura | yes but signaling can have different ranges based on frequency and impedence. The higher freqs that are injected in a given cable the shorter the range do to impedence restrictions unless of course the cable is replaced such as hardline. BTW I would mind having 200 feet of hardline :) |
| 15:21.31 | joe_bag_odonuts | TK-Fender:I don't understand. If I just do a call out and do noting but play a greeting, The call still doesn't go through. |
| 15:21.39 | aptura | Anyway what is the impedence of T1 cable anyway? |
| 15:21.42 | [TK]D-Fender | mosty: maybe you don't have any? ;) |
| 15:21.47 | Sweeper | t1...cable? |
| 15:22.12 | Sweeper | it's facking phone wire :v |
| 15:22.21 | aptura | its still cable |
| 15:22.29 | aptura | and has impedence. |
| 15:22.41 | [TK]D-Fender | joe_bag_odonuts: pastebin your new code & callfile. extens & macros all... |
| 15:22.43 | Sweeper | so? |
| 15:22.47 | *** join/#asterisk JackRippr (n=vic@pool-141-158-214-97.alt.east.verizon.net) |
| 15:23.19 | [TK]D-Fender | aptura: T1 is good for what.. about 1500 feet from demarc before neding a repeater IIRC. |
| 15:23.27 | aptura | k |
| 15:23.29 | JackRippr | hi |
| 15:23.36 | mosty | [TK]D-Fender: i would prefer to be sure though ;) a static config analysis checker could help if there was one |
| 15:23.43 | Sweeper | [TK]D-Fender: but zmg signalling will invert the impedance of the frequency! |
| 15:24.05 | [TK]D-Fender | Besides who in their right mind would put their smart-jack so damn far away from their server? Even if it is 100feet or something, thats what LBO is for. |
| 15:24.25 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
| 15:24.25 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 15:24.27 | aptura | excuse me? you cannot invert the inductive ractince or capacitive impedence of a wire :) |
| 15:24.31 | [TK]D-Fender | mosty: ..... sure... and can you get ME some of that crack you're on? ;) |
| 15:25.17 | *** join/#asterisk ucfmethod (n=ucfmetho@office.eyestreet.com) |
| 15:26.12 | liversmudge | mm |
| 15:27.58 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:27.58 | mosty | [TK]D-Fender: well it's something the asterisk parser should be able to figure out itself |
| 15:28.08 | joe_bag_odonuts | TK-Fender: I'm not using a call file. I'm using the manager API. I've already posted everything,but I'll do it again. The variables will be listed at the top. |
| 15:29.27 | aptura | BTW is there some dial plan bug that causes a dial plan to point to a different outbound carrier then what is stipulated in the dial plan? I am running 1.2.7 |
| 15:29.57 | aptura | For the oddest reason my 1800 dial plan is pointing to trunkld. |
| 15:30.13 | aptura | that is, it is refering to it when trunk is in the dial plan. |
| 15:30.36 | *** join/#asterisk svenna (n=svenna@p548d0e5b.dip0.t-ipconnect.de) |
| 15:30.56 | JackRippr | Why are you running such an old version of Asterisk? |
| 15:31.14 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 15:35.11 | last1 | anyone has a working 7970 with SIP ? |
| 15:35.24 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 15:35.46 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 15:35.56 | Polis_ttt | i looked in my logg-file, and could find this message when i get incomming calls: |
| 15:35.56 | Polis_ttt | "Executing DeadAGI("SIP/0850007954-081a5118", "agi://127.0.0.1:4577/call_log") in new stack Apr 25 17:15:59 WARNING[15276]: res_agi.c:210 launch_netscript: Connect to 'agi://127.0.0.1:4577/call_log' failed: Connection refused" |
| 15:36.18 | *** part/#asterisk prevail20 (n=prevail2@wan-hts-26.highertech.net) |
| 15:36.46 | Polis_ttt | can it have something whit that i don't hade php fileupload activated, or what is "agi://"? i looked at google, but google didn't know :( |
| 15:37.02 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 15:37.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:37.15 | putzz | google does know it |
| 15:37.16 | putzz | lol |
| 15:37.26 | Polis_ttt | ;) |
| 15:38.18 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 15:39.25 | ManxPower | I'm starting to think that perhaps Digium should shutdown the Asterisk mailing lists and IRC channels. Pretty much every question asked the past week or so has been answered in the mailing list archives. |
| 15:40.33 | ManxPower | Qwell: I made some progress with encoding media. Apparently most DVDs have errors in the audio track that DVD players (software and hardware) deal with just fine, but WME barfs on. |
| 15:40.37 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:40.44 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:42.02 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:42.20 | syzygyBSD | is there a way to write out the currently loaded dialplan to a file? |
| 15:42.42 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) |
| 15:42.55 | synthetiq | is there some way to get asterisk working with toll bypass? |
| 15:43.05 | synthetiq | with the carrier just sending a 183 back |
| 15:44.11 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 15:44.44 | ManxPower | synthetiq: Asterisk works just fine with toll bypass. |
| 15:44.55 | ManxPower | In fact toll bypass is just nothing more than a phone call. |
| 15:45.18 | synthetiq | well we just get ring no answer |
| 15:45.42 | mosty | hmm, asterisk won't let me #include 10 or more files from extensions.conf |
| 15:45.55 | mosty | there are no loops, the #included files don't even exist |
| 15:45.56 | ManxPower | synthetiq: you have some issue not related to toll bypass |
| 15:46.06 | synthetiq | from multiple carriers... |
| 15:46.10 | joe_bag_odonuts | [TK]D-Fender: I've uploaded everything here http://pastebin.ca/457798 |
| 15:46.16 | synthetiq | on many asterisk machine |
| 15:46.17 | synthetiq | s |
| 15:46.22 | synthetiq | that are runnign 1.4 |
| 15:46.32 | ManxPower | synthetiq: perhaps you did something stupid like use "r" on the Dial line? |
| 15:46.43 | synthetiq | yes |
| 15:47.00 | synthetiq | if we dont use it , it just hangs up |
| 15:47.27 | ManxPower | The only time you want to use "r" on the Dial line is when you want to OVERRIDE the tones the telco is sending and replace them with a ringing sound. |
| 15:47.34 | synthetiq | (well we get a voice thats says the number is invalid) |
| 15:47.38 | ManxPower | synthetiq: perhaps if you paste an actual Dial line. |
| 15:47.57 | synthetiq | i did this with the r and without the r |
| 15:48.09 | ManxPower | Or even better, paste the Dial line as shown on the CLI |
| 15:48.36 | synthetiq | exten => _011.,1,dial(SIP/${EXTEN:3}@carrier1) |
| 15:49.07 | ManxPower | synthetiq: how exactly is the "toll bypass"? |
| 15:49.27 | synthetiq | as implemented by the carrier |
| 15:49.37 | synthetiq | as in the customer is not charged for ivr sessions |
| 15:50.03 | ManxPower | So your REAL problem is that you are unable to use the IVR |
| 15:50.25 | synthetiq | no audio is established for the one way ivr sessions |
| 15:50.37 | synthetiq | witth the itnernational the customer picks up not ivr |
| 15:50.39 | ManxPower | What DOES your carrier require for an international dialing prefix anyway? |
| 15:50.49 | synthetiq | e.164 |
| 15:51.21 | ManxPower | synthetiq: What carrier requires country code+ phone number with no international prefix? |
| 15:51.35 | synthetiq | l3, gblx, verizon |
| 15:51.37 | ManxPower | Since you are stripping off the 011 in that Dial line |
| 15:52.04 | ManxPower | synthetiq: I highly doubt Verizon will let you dial an international call without the 011 |
| 15:52.11 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-40-188.socal.res.rr.com) |
| 15:52.27 | synthetiq | for sip they do =] |
| 15:52.42 | ManxPower | synthetiq: what happens if you don't strip off the 011? |
| 15:53.53 | ManxPower | I heard Verizon is sueing themselves for patent infringement. |
| 15:53.54 | synthetiq | error 404 not found =] |
| 15:54.45 | Qwell[] | ManxPower: wasn't there a company that sued themselves once? |
| 15:54.45 | ManxPower | synthetiq: Is your Asterisk server on a public IP address? |
| 15:54.46 | Qwell[] | I seem to recall something like that... through buyouts, etc |
| 15:54.46 | ManxPower | Qwell[]: SCO? |
| 15:54.46 | ManxPower | synthetiq: Are your phones IP or analog? |
| 15:54.46 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:54.46 | ManxPower | If they are IP, do you have canreinvite=no set for them? |
| 15:55.31 | Qwell[] | ManxPower: possible |
| 15:55.31 | synthetiq | ip |
| 15:55.33 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 15:55.34 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 15:55.34 | ManxPower | (10:53:42) ManxPower: synthetiq: Is your Asterisk server on a public IP address? |
| 15:56.12 | synthetiq | yes |
| 15:56.30 | ManxPower | That only leaves 1 question unanswered |
| 15:56.31 | synthetiq | and can reinvite is set to no |
| 15:56.40 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 15:56.45 | ManxPower | what codecs are you allow='ing? |
| 15:56.56 | *** join/#asterisk Superbaloo (n=Superbal@home.superbaloo.net) |
| 15:57.36 | synthetiq | good ole g729 and ulaw |
| 15:57.51 | ManxPower | synthetiq: you have g729 licenses from Diguium? |
| 15:57.56 | synthetiq | yeps |
| 15:58.13 | ManxPower | synthetiq: do you have a /etc/asterisk/indications.conf ? |
| 15:58.20 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source PBX -=- Asterisk 1.2.18, 1.4.3 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 15:58.46 | synthetiq | yep |
| 15:59.40 | ManxPower | synthetiq: I don't really have any more suggestions other than making sure your iptables/ipchains/firewall is allowing the RTP audio ports thru |
| 16:01.58 | synthetiq | if i remember working with the carriers its that after the 183 asterisk doesnt send an ack back |
| 16:02.16 | *** join/#asterisk Bazy (n=bazy@eclipse.upcnet.ro) |
| 16:02.50 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) [NETSPLIT VICTIM] |
| 16:03.13 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 16:03.59 | [TK]D-Fender | mosty: GotoIf($[${myvar}=hello]?6) <- this will work if the var isn't EMPTY, but will bomb otherwise. as this is evaluated in run-time and based on the syntax that vars and apps/functions use, there is no way to tell |
| 16:04.35 | dfgas | how do i change what codec astrisk is using? |
| 16:07.37 | *** join/#asterisk wai22 (n=waisky2@244.34.95.219.cbj02-home.tm.net.my) |
| 16:08.39 | wai22 | hi, anyone has solution for this "Ring requested on channel 0/11 already in use on span 1. Hanging up owner" issue ? |
| 16:10.11 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:10.32 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 16:12.23 | ManxPower | dfgas: usually by reading the docs |
| 16:12.31 | synthetiq | ok the 011 issue was that a 0 was after the country code for some reason |
| 16:12.52 | synthetiq | (works in the pstn) |
| 16:12.54 | ManxPower | Well there is 30 mins of my life I 'll never get back |
| 16:13.34 | plasmid | [TK]D-Fender, what kind of VOIP phone would you recommend for home office use? Just a basic (nothing too fancy i suppose) phone that can handle 2-5 incoming calls. |
| 16:13.53 | Sweeper | plasmid: polycom! |
| 16:13.55 | synthetiq | snom |
| 16:13.58 | Sweeper | :P |
| 16:14.08 | Qwell[] | [TK]D-Fender: I told him to ask you, because...you rock :p |
| 16:15.19 | plasmid | ip501? |
| 16:15.41 | synthetiq | snom 360 snom 320 |
| 16:15.48 | synthetiq | snom > polycom |
| 16:15.50 | synthetiq | =D |
| 16:16.24 | Qwell[] | ~phones |
| 16:16.27 | jbot | [phones] http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 16:16.28 | Qwell[] | the bot has spoken |
| 16:16.42 | plasmid | heheh.. |
| 16:16.53 | mvanbaak | I see the grandsucks still did not make it to that list |
| 16:16.58 | synthetiq | wtf |
| 16:17.00 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-93.dsl1.mdl.ny.frontiernet.net) |
| 16:17.27 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136017.home.otenet.gr) |
| 16:17.30 | mvanbaak | ~gs |
| 16:17.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:17.32 | synthetiq | i think that order is complete reverse |
| 16:17.55 | nasls_lsa | jbot I think I agree :P |
| 16:17.56 | jbot | You think I agree :P? |
| 16:19.10 | ManxPower | synthetiq: you are wrong, of course |
| 16:20.00 | ManxPower | For the most part all the phones listed are pretty good from all the reports that I've seen. |
| 16:20.54 | *** join/#asterisk Hack_ (n=root@exchange.ieqsa.com.pe) |
| 16:21.17 | plasmid | polycom 301... there.. nothing fancy... but does the job well. |
| 16:21.23 | Hack_ | hi |
| 16:22.39 | plasmid | ahh.. wait.. that phone does not power itself over RJ45. Hmm. |
| 16:22.49 | Hack_ | asterisk appliance support PRI? |
| 16:22.52 | Qwell[] | plasmid: Do you have a POE switch? |
| 16:22.56 | Qwell[] | Hack_: no |
| 16:23.17 | illsci | asterisk applliance == trixbox? |
| 16:23.18 | plasmid | Qwell, POE = point of entry? |
| 16:23.25 | ManxPower | illsci: no |
| 16:23.26 | illsci | power over ethernet |
| 16:23.26 | mosty | [TK]D-Fender: deja-vu? or did you do that accidentally? |
| 16:23.27 | Qwell[] | illsci: no, one sec, I'll link |
| 16:23.42 | *** join/#asterisk bmg505 (n=leon@196.209.180.187) |
| 16:23.44 | Qwell[] | illsci: http://www.digium.com/en/products/hardware/asteriskappliance.php |
| 16:23.51 | SuPrSluG | do the polycom ip500 have poe capability? |
| 16:24.01 | Qwell[] | plasmid: You need a power over ethernet switch to power a POE phone |
| 16:24.08 | plasmid | Qwell, checking if my LInksys SD2008 has POE. |
| 16:24.21 | ManxPower | They all have PoE support. Some of them need a special cable, however. |
| 16:24.22 | Qwell[] | plasmid: Unless you paid > $500 for it...it probably doesn't. :) |
| 16:24.42 | Hack_ | mmm |
| 16:24.49 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 16:24.50 | ManxPower | 30x and 50x need a special polycom cable for PoE |
| 16:24.52 | plasmid | Qwell, reading documentation on my switch. Maybe it will have POE or not. |
| 16:24.55 | ManxPower | I don't know about the others |
| 16:24.58 | Hack_ | Español? |
| 16:25.06 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:25.21 | illsci | ouuu.... thats pretty |
| 16:25.27 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:25.43 | Qwell[] | illsci: very |
| 16:25.46 | mosty | i'd rate snom above linksys spa-9XX |
| 16:26.11 | ManxPower | mosty: SNOM has a history of firmware...oddities |
| 16:26.13 | illsci | is the asterisk gui thats on that appliance opensource? |
| 16:26.37 | syzygyBSD | is it possible to write out the currently loaded dialplan to a file? |
| 16:26.39 | Qwell[] | illsci: yes and no |
| 16:26.52 | Qwell[] | illsci: It *is* available under the GPL, but the one it ships with will not be, I don't believe. |
| 16:26.57 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 16:27.04 | plasmid | Qwell, from the looks of it the answer is most probably "no" http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416900513&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=0051395358B01 |
| 16:27.20 | Hack_ | I have 1 PRI and what need for connect with asterisk? |
| 16:27.21 | mosty | ManxPower: only in beta releases, which is expected |
| 16:27.23 | ManxPower | syzygyBSD: no. |
| 16:27.41 | ManxPower | Hack_: any of the Digium (or compatable) T-1/E-1 cards |
| 16:27.46 | syzygyBSD | :( I didn't think so, just overwrote extensions.conf and didn't have the most recent backed up |
| 16:28.22 | Hack_ | ok |
| 16:28.27 | *** part/#asterisk Hack_ (n=root@exchange.ieqsa.com.pe) |
| 16:28.29 | ManxPower | syzygyBSD: you can have it save the current dialplan (to extensions.conf) but only if you have writeprotect=no and static=no I believe |
| 16:28.43 | ManxPower | syzygyBSD: there was a massive duscission about this on the mailinglist this week |
| 16:29.05 | syzygyBSD | thanks |
| 16:29.26 | ManxPower | syzygyBSD: asterisk -rx "show dialplan" > gawd_im_stupid.txt |
| 16:29.55 | syzygyBSD | :P we have already recreated the last 4 weeks of changes, but in the future that will help |
| 16:30.09 | ManxPower | syzygyBSD: at least that gives you the current dialplan, even if it is in a format you can't use directly |
| 16:30.37 | syzygyBSD | meh, a couple of awk and sed statements and it is |
| 16:30.54 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 16:31.09 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-d2218dd25c0f023f) |
| 16:31.09 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:31.57 | *** join/#asterisk joe-f- (n=jf@adsl-75-4-37-53.dsl.emhril.sbcglobal.net) |
| 16:32.31 | illsci | that southern ivr is great :) |
| 16:33.48 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 16:33.49 | kippi | hey |
| 16:33.56 | kippi | How can I setup voice recording ? |
| 16:34.41 | mosty | kippi: write an ivr |
| 16:35.01 | kippi | is there not one built in? |
| 16:35.22 | ManxPower | kippi: no |
| 16:35.30 | mosty | the building blocks are |
| 16:35.44 | ManxPower | well yes "Record" but that is not terribly useful without an IVR |
| 16:35.52 | kippi | whats the one that asterisk@home or whatever it is called |
| 16:36.12 | kippi | uses |
| 16:36.30 | ManxPower | kippi: huh? |
| 16:36.44 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 16:36.45 | kippi | asterisk@home has recording built into it |
| 16:37.04 | ManxPower | kippi: Yes. That is an IVR using Record() |
| 16:37.11 | ManxPower | just like you would build yourself. |
| 16:37.14 | kippi | ah |
| 16:37.17 | ManxPower | ~trixbox |
| 16:37.27 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 16:37.28 | kippi | so could just nick that one |
| 16:37.56 | ManxPower | kippi: not easily Trixbox has the most complicated and downright insane config I have ever seen other than the Windows Registry |
| 16:38.05 | kippi | lol |
| 16:38.06 | kippi | great |
| 16:38.10 | kippi | i'lkl have a play |
| 16:38.52 | *** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net) |
| 16:39.05 | ManxPower | I've heard that the US military uses the threat of forcing someone to figure put the t |
| 16:39.40 | ManxPower | <PROTECTED> |
| 16:39.50 | *** join/#asterisk tausq (n=randolph@64.81.244.94) |
| 16:39.50 | ManxPower | *shudder* |
| 16:39.56 | alexns | having random call breakup... tdm400 & polycom 501s . Any Suggestions ?? |
| 16:40.11 | ManxPower | alexns: IRQ conflicts |
| 16:40.18 | ManxPower | the polycom and asterisk are on the same lan? |
| 16:40.32 | alexns | manxpower: same lan |
| 16:40.34 | mosty | what do you mean by breakups? |
| 16:41.00 | alexns | manxpower: i was thinking irq myself. lan has about 10 people with some light smb usage |
| 16:41.03 | ManxPower | I'll bet the Polycom wants custody of the kids. |
| 16:42.45 | alexns | manxpower: tried using 2 tdm400s instead of tdm800 and got bad feedback on the phones |
| 16:42.58 | ManxPower | alexns: perhaps you should check to see if there are IRQ conflicts |
| 16:43.30 | alexns | stupid question but fastest way to do it ? |
| 16:43.39 | illsci | is the asteriskgui by digium on the digium site anywhere? |
| 16:43.58 | alexns | cat /proc/interrupts ? |
| 16:44.47 | alexns | ManxPower: ha 5: 197989876 XT-PIC libata, VIA8237, eth1, ehci_hcd:usb1, uhci_hcd:usb4, uhci_hcd:usb5, wctdm |
| 16:45.18 | tausq | when i'm doing a voicemail recording, is there a way to properly record dtmf tones sent by the caller? it seems that the dtmf tones get "eaten" by asterisk |
| 16:45.47 | anthony] | how do I enable the transfer feature on xlite? |
| 16:46.07 | dfgas | this is messed up, if i call an 800 over fwd from asterisk i get all circuits are busy, however if i connect straight to fwd through my softphone i get the 800 number i am trying to call |
| 16:46.13 | dfgas | any ideas? |
| 16:47.22 | alexns | ManxPower: when using zttest i get 99% average afer running for a few minutes.. no interrupt errors either but there is alot of shit on irq 5 |
| 16:47.48 | uwe | dfgas, i didnt understand the fwd thing, but could it be related to the contexts you are dropped in ? they would be diffrent ,no ? |
| 16:48.12 | dfgas | hmmm |
| 16:49.34 | dfgas | uwe: well with fwd you have to dial *18009241000 and if you do it should call them but i get all circuits are busy from asterisk instead. now lets say i forgot the * then it connects to fwd and tells me i need to proceed with a * |
| 16:49.45 | SuPrSluG | dfgas:is fwd registered w/ asterisk? |
| 16:49.58 | dfgas | yah |
| 16:50.07 | dfgas | all fwd works but that part |
| 16:50.34 | SuPrSluG | dfgas:have you included the phones in that context? |
| 16:50.37 | anthony] | anyone here use xlite? |
| 16:51.18 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 16:51.56 | dfgas | ? |
| 16:53.51 | SuPrSluG | the phone you use is in a context. u use the include=> statement to use, say an outgoing context. |
| 16:53.52 | dfgas | Superbaloo: what do you mean? |
| 16:54.29 | tausq | anthony]: the "lite" version of xten doesn't support that feature |
| 16:54.37 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:55.09 | stubert | anyone having problems with zaptel 1.4.2? |
| 16:55.37 | tausq | what kind of problems? |
| 16:55.47 | stubert | kernel panics... |
| 16:55.59 | dfgas | -- Called fwd/*18009241000 |
| 16:55.59 | dfgas | <PROTECTED> |
| 16:55.59 | dfgas | <PROTECTED> |
| 16:56.17 | tausq | stubert: no(t yet) :) |
| 16:56.17 | anthony] | tausq do you know how to get video working? |
| 16:56.18 | Deeewayne | stubert: there's an update for this |
| 16:56.30 | stubert | Deeewayne: in the trunk? |
| 16:56.46 | tausq | anthony], don't know, i was just searching google :) |
| 16:57.19 | *** part/#asterisk JackRippr (n=vic@pool-141-158-214-97.alt.east.verizon.net) |
| 16:57.52 | Deeewayne | use 1.4.2.1 from the digium ftp site |
| 16:58.04 | stubert | heh... thanks... |
| 17:00.38 | tausq | speaking of zaptel, i had written a simple program that just opens the zaptel channel, sets it to LINEAR mode, and then writes out the data i read back into a WAV file. it can record fine most of the time, but the SNR is quite low. is there some tricks to this? |
| 17:01.15 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 17:01.26 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:01.27 | tausq | i've been looking at the asterisk code but haven't found anything yet |
| 17:01.43 | SuPrSluG | dfgas:canreinvite=no? |
| 17:02.13 | dfgas | SuPrSluG: yes |
| 17:02.23 | anthony] | tausq found anything? |
| 17:02.59 | SuPrSluG | disallow=all then allow=ulaw allow=gsm or whatever |
| 17:03.36 | thekidrio | Is there a compatibility list that I can either add too or request an addition? I have been able to get the TDM400p to work in 3 out of 3 Dell PowerEdge 2550's |
| 17:03.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:03.49 | tausq | anthony], i'm not sure what you are looking for. i've only used xlite for simple voice SIP stuff |
| 17:04.42 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:05.06 | Keltus | anyone have a recommended provider for tollfreenumber -> IP/asterisk? |
| 17:05.16 | anthony] | i want to use the video but its not letting me ;/ |
| 17:05.19 | SuPrSluG | for the sip codecs |
| 17:05.57 | [TK]D-Fender | mosty: Yeah, I repasted, becaue YOU are asking questions that have just been answered. Don't think for a second you deserve my riting it by HADN for you ;) |
| 17:06.24 | [TK]D-Fender | plasmid: Polycom > all. IP 301 = nearly pointless today. |
| 17:07.01 | dfgas | SuPrSluG: you are awsome |
| 17:07.08 | dfgas | lol |
| 17:07.09 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:07.12 | [TK]D-Fender | plasmid: for 2-5 & using PoE, you might want to sonsider the IP 320. if not, the IP 501, and if you think you MIGHT go PoE later/soon after, IP 430. |
| 17:07.27 | dfgas | allow=ulaw was it |
| 17:08.03 | tzanger | fun fu nfun |
| 17:08.30 | tzanger | [TK]D-Fender: there's a 5-series phone now that has the miniborwser IIRC |
| 17:08.36 | tzanger | I forget what it is offhand but it was pretty sweet |
| 17:08.38 | Qwell[] | 550? |
| 17:08.53 | tzanger | that may be it |
| 17:10.01 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 17:11.57 | SuPrSluG | anyone play w/ the nokia phones and asterisk? |
| 17:12.05 | *** part/#asterisk stubert (i=stu@techtools.actusa.net) |
| 17:12.10 | HarryR | we've had the nokia phones working in the office |
| 17:12.36 | SuPrSluG | reliable? |
| 17:12.48 | HarryR | the E61 works pretty well |
| 17:13.06 | HarryR | (like 3 or 4 people here use it all the time) |
| 17:13.08 | SuPrSluG | how about in hotspots? |
| 17:13.42 | HarryR | never tried it, but in most the connection wouldn't be reliable enough at busy times |
| 17:14.28 | HarryR | (e.g. we've got QOS for voip calls in the office, but at starbucks it's supposed to be kinda sporadic if you've got more than 1 or 2 other people) |
| 17:15.18 | ManxPower | dfgas: "cat /proc/interrupts" If there is more than one device on the IRQ for the Digium card then you have a problem and must fix that so that the Digium card is on an IRQ all by itself. |
| 17:15.49 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 17:17.21 | SuPrSluG | HarryR, how's the range? |
| 17:18.07 | HarryR | our office is like 40ft long with loads of wireless points (i'm probably going to get serious brain cancer from all the wireless gear here)... |
| 17:18.17 | HarryR | so suprisingly there's always good signal |
| 17:21.03 | [TK]D-Fender | tzanger: IP 320/330/430/501/550/600/601/650 |
| 17:21.28 | tzanger | 501 does not have minibrowser |
| 17:22.30 | [TK]D-Fender | tzanger: Yes, it DOES |
| 17:22.35 | [TK]D-Fender | :) |
| 17:23.06 | [TK]D-Fender | tzanger: That "Services" button is a beautiful thing... |
| 17:23.19 | tzanger | you're fucking shitting me |
| 17:23.39 | [TK]D-Fender | tzanger: I am not... your shit is hard to swallow ;) |
| 17:27.31 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
| 17:28.52 | Polis_ttt | a lillte problem, i can't shut down asterisk. i tryed whit kill pid, but it starts up directly after, what shall i do? |
| 17:29.20 | tausq | running the safe_asterisk script? |
| 17:29.30 | [TK]D-Fender | tausq: Obviously |
| 17:29.47 | [TK]D-Fender | Polis_ttt: Kill the safe_asterisk script first. |
| 17:30.05 | Polis_ttt | where? |
| 17:30.46 | [TK]D-Fender | Polis_ttt: "ps-A" |
| 17:30.52 | *** join/#asterisk joe_bag_odonuts (n=erne@dsl027-164-165.atl1.dsl.speakeasy.net) |
| 17:31.58 | Polis_ttt | command not found :( but i'v tryed to runt ps -u root|grep asterisk, and then kill it, if that's what you mean |
| 17:32.18 | Polis_ttt | ps -a |
| 17:32.27 | Polis_ttt | ups:) |
| 17:33.01 | tausq | on most distros you can do /etc/init.d/asterisk stop |
| 17:33.05 | Polis_ttt | if i runt that "ps -a" i get 7 hits |
| 17:33.09 | tausq | if you installed a package, that is |
| 17:33.17 | Polis_ttt | tausq: uses ubunto, so it isn't there :( |
| 17:33.34 | tausq | er.... |
| 17:33.49 | tausq | i'm running asterisk on ubuntu, and it sure is there :) |
| 17:34.01 | Polis_ttt | tausq: bash: /etc/init.d/asterisk: No such file or directory |
| 17:36.26 | joe_bag_odonuts | [TK]D-Fender: I finally got this thing to work. I changed the context in extensions.conf. I'll post the variables and the context. I didn't change the macros, they work just fine. I got a big clue on how to do this from here http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message |
| 17:36.37 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 17:38.22 | [TK]D-Fender | joe_bag_odonuts: I already know wuite well how to do it, thanks :) |
| 17:38.38 | *** join/#asterisk Fieldy (i=rpI9vcDt@gentoo/contributor/Fieldy) |
| 17:38.46 | [TK]D-Fender | joe_bag_odonuts: Was a question of if you'd figure it out yourself or require outside consulting or not :) |
| 17:39.46 | joe-f- | has anyone actually setup asterisk with NAT? |
| 17:40.27 | joe-f- | no doc has worked.. i've looked at all open ports when i've removed nat, and forwarded those correctly.. i have NAT working for my webserver.. but cant for * |
| 17:40.45 | [TK]D-Fender | ~sipnat |
| 17:40.49 | jbot | i heard sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:40.59 | joe-f- | k |
| 17:41.07 | [TK]D-Fender | joe-f-: Yes, I've worked each combination of phone/* behind NAT jsut fine. |
| 17:41.16 | joe-f- | then i suck |
| 17:41.56 | joe-f- | atleast i know it works though when i remove NAT.. i'll get it working soon |
| 17:42.01 | joe-f- | thx |
| 17:42.03 | flujan | hi guys... I am using asterisk queues... But it is not working well... |
| 17:42.07 | [TK]D-Fender | joe-f-: shouldn't take you long. |
| 17:42.23 | flujan | Sometimes I have a lot of users on the Not in Use and the calls are not being delivered... |
| 17:42.38 | flujan | does asterisk queues runs well for a production environment? |
| 17:43.42 | [TK]D-Fender | flujan: Show us your config, CLI output etc. Your description is nearly worthless. |
| 17:45.08 | *** join/#asterisk my007ms (i=my007ms@196.218.96.53) |
| 17:45.23 | syzygyBSD | flujan: yes queues run great! |
| 17:46.04 | syzygyBSD | my guess based on the wonderful description you didn't give, is you need to update your ring strategy. May I suggest ringall or rrmemory? |
| 17:46.34 | flujan | hi [TK]D-Fender, here it goes... |
| 17:46.35 | flujan | http://pastie.caboo.se/56493 |
| 17:47.04 | [TK]D-Fender | flujan: "strategy=rrmemory" <--- |
| 17:47.05 | *** join/#asterisk waltz (n=walterbr@bas1-toronto44-1177949543.dsl.bell.ca) |
| 17:47.21 | flujan | synthetiq, I am using roundrobin strategy... ringall works but it is crashing the softphone due to concurrency problems... |
| 17:47.22 | flujan | :( |
| 17:47.41 | joe_bag_odonuts | TK-Fender Thanks for all of your help. It pushed me in the right direction here is the code |
| 17:47.41 | joe_bag_odonuts | http://pastebin.ca/457991 |
| 17:47.43 | [TK]D-Fender | flujan: SEE ABOVE |
| 17:48.26 | jm|laptop | hmm |
| 17:48.44 | syzygyBSD | [TK]D-Fender: oh.. who answered his question before he asked it... |
| 17:48.48 | jm|laptop | will using queues mean my 7912s don't show a "missed call" when it was picked up on another extension? |
| 17:48.55 | [TK]D-Fender | joe_bag_odonuts: Thats one way... I would STILL use a local channel instead of dialing the SIP peer, but make a new custom exten that ONLY dials the peer from there w/o answer in front and your tartger should STILL be another context. |
| 17:49.28 | [TK]D-Fender | joe_bag_odonuts: And you may want to fix your code so its 1.2 compliant and 1.4 WORKABLE ;) |
| 17:50.05 | [TK]D-Fender | syzygyBSD: Yeah, and who's smart enough to sit silent and let him incriminate himself before wasting time on potentially useless suggestings? ;) |
| 17:50.05 | syzygyBSD | flujan: use rrmemory |
| 17:50.06 | flujan | I will try rrmemory and see the results.. :d |
| 17:50.10 | flujan | thanks guys! |
| 17:50.22 | syzygyBSD | [TK]D-Fender: I never said I was smart... |
| 17:50.49 | [TK]D-Fender | syzygyBSD: Ah... the sound of no-one disagreeing with you! ;) |
| 17:51.01 | [TK]D-Fender | pwned |
| 17:51.12 | syzygyBSD | well, nobody claimed I wasn't smart either... |
| 17:52.11 | syzygyBSD | I just had an issue with the leastrecent strategy that was similar to his issue, so I made a suggestion |
| 17:53.34 | jm|laptop | I don't think this is going to help me :( |
| 17:53.48 | syzygyBSD | whats your problem jm|laptop |
| 17:53.55 | joe-f- | [TK]D-Fender: wow, i guess i had my sip.conf configured wrong.. my externip and and localnet were goofed |
| 17:54.01 | joe-f- | its a workin' |
| 17:54.28 | jm|laptop | syzygyBSD: I have a couple of Cisco 7912s - in my rather basic dialplan I do "Dial(SIP/1&SIP/2)" |
| 17:54.36 | jm|laptop | if 1 picks up 2 shows a "missed call" |
| 17:54.43 | jm|laptop | could I get around this with queues? |
| 17:54.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:55.02 | syzygyBSD | no |
| 17:55.06 | jm|laptop | bummer |
| 17:55.10 | jm|laptop | what about with Skinny? |
| 17:55.18 | syzygyBSD | don't know about that... |
| 17:55.21 | jm|laptop | k |
| 17:55.33 | jm|laptop | don't think I want to play with firmware again anyway |
| 17:55.36 | jm|laptop | thanks syzygyBSD |
| 17:55.41 | syzygyBSD | well, you could get around it with queues that didn't call back, but I wouldn't suggest it |
| 17:55.50 | jm|laptop | o.O |
| 17:55.53 | syzygyBSD | much more annoying to have a line always off hook |
| 17:56.11 | jm|laptop | hmm |
| 17:56.22 | syzygyBSD | bad suggestion, ignore it |
| 17:56.28 | jm|laptop | k |
| 17:57.00 | *** join/#asterisk boch (n=fran@190.48.237.194) |
| 17:57.09 | [TK]D-Fender | jm|laptop: No way, no how. |
| 17:57.26 | boch | dudes, how can i convert a .wav file to .ulaw ? |
| 17:58.24 | [TK]D-Fender | boch: sox |
| 17:58.57 | [TK]D-Fender | boch : if you need help, google the WIKI for it. |
| 17:59.40 | boch | found it, thanks |
| 18:00.45 | *** join/#asterisk Goodjoke (n=Goodjoke@74.202.86.23) |
| 18:01.26 | Goodjoke | any recommendations on a good way to automate a full backup? (I am thinking of full OS with offsite capabilities)...wondering if anyone has tried something like acronis |
| 18:01.39 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-112-47.dsl.irvnca.pacbell.net) |
| 18:01.40 | mosty | jm|laptop: tell the phone to not display missed calls if you don't want them to |
| 18:01.52 | Qwell[] | Goodjoke: What's wrong with rsync? |
| 18:02.35 | *** join/#asterisk moprilo (n=jjohn@190.10.0.120) |
| 18:02.38 | Goodjoke | @Qwell:dunno..haven't tried it (not that much of a linux guy, unfortunately) |
| 18:02.39 | syzygyBSD | or dd |
| 18:02.41 | syzygyBSD | :) |
| 18:03.04 | syzygyBSD | rsync is a better choice than dd |
| 18:03.33 | syzygyBSD | in fact within the last hour I setup backups of 4 asterisk boxes with rsync... |
| 18:04.08 | [TK]D-Fender | Goodjoke: amanda perhaps |
| 18:05.56 | moprilo | hi guys.. I have a digium card, and when I do a bridge between lines, asterisk hangs, and doesn't disconnect on hangup. |
| 18:06.16 | moprilo | and didn't do that before.. on a previous asterisk installation :S |
| 18:06.34 | syzygyBSD | zap lines I assume? |
| 18:06.52 | syzygyBSD | when you upgraded did you update libpri and zapata? |
| 18:07.09 | *** join/#asterisk masked (i=masked@shell.iinet.net.au) |
| 18:08.14 | moprilo | zap lines yes |
| 18:08.19 | moprilo | zapata.. |
| 18:08.24 | moprilo | i don't use libpri |
| 18:08.33 | moprilo | its 2 analog digium lines |
| 18:09.00 | moprilo | i'm try'n to find what it is.. could it be the TX RX gain? |
| 18:09.39 | [TK]D-Fender | moprilo: No. If it hangs, it isn't an option. Its either a feature, or a bug. |
| 18:09.49 | syzygyBSD | lol.. feature |
| 18:10.08 | syzygyBSD | I think it is a library issue |
| 18:10.32 | syzygyBSD | what version of asterisk are you using? |
| 18:10.41 | moprilo | 1.4 |
| 18:10.59 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 18:11.19 | moprilo | 1.4.3 i think |
| 18:11.26 | moprilo | i installed it about a week ago |
| 18:12.24 | [TK]D-Fender | moprilo: That'd be 1.4.2 then. |
| 18:13.10 | moprilo | ok..1.4.2. :) |
| 18:13.25 | Goodjoke | any reccomendations of phones that natively support autoanswer/intercom? and have multiple speed dial buttons? |
| 18:13.43 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-246-217-44.washdc.fios.verizon.net) |
| 18:13.49 | SomeOne1 | with H323 |
| 18:13.55 | SomeOne1 | are two ports opened for RTP |
| 18:13.56 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 18:14.05 | SomeOne1 | like one for receiving audio and one for sending audio |
| 18:14.16 | SomeOne1 | or is just one RTP port used for both receiving / sending |
| 18:14.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:14.34 | [[blah]asfd | I am getting a make error when I do a make install in asterisk-addons-1.4.1 http://pastebin.ca/458047. Could anyone suggest how to correct this? |
| 18:16.40 | [TK]D-Fender | Goodjoke: Polycom, # of buttons depending on model & expansion modules. |
| 18:16.53 | [TK]D-Fender | Goodjoke: Aastra is a runner-up |
| 18:17.02 | Goodjoke | i have the polycom 501 |
| 18:17.54 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:19.10 | syzygyBSD | [[blah]asfd: do you have asterisk 1.4 installed? |
| 18:19.12 | [TK]D-Fender | Goodjoke: Congratulations. Great phone. |
| 18:19.31 | [[blah]asfd | syzygyBSD: yes, and have launced it successfully. |
| 18:19.34 | SuPrSluG | did you compile asterisk w/ chan_ooh323.so if deb install libopenh323-dev then you'll be able to build the driver w/ make menuselect |
| 18:19.44 | Goodjoke | fender: not enough buttons.. and i haven'tdelved into it much, but the auto answer doesn't appear to be native |
| 18:21.15 | [TK]D-Fender | Goodjoke: Sure it it's native. on a single reg you can have 2 buttons for speed-dials. on a 601, 5, more by adding attendant modules. |
| 18:21.17 | [[blah]asfd | from the asterisk-1.4.3 directory, menuselect i see this: XXX 5. chan_h323 means i cant install it right? |
| 18:21.49 | SuPrSluG | install libopenh323-dev first |
| 18:22.12 | SuPrSluG | then the XXX should go away |
| 18:22.28 | [TK]D-Fender | Goodjoke: Aastra's support quite a lot on their soft-keys, but the call-handling , handset quality, and audio are all inferior. |
| 18:22.35 | [[blah]asfd | ok... is that file available on the ftp site? |
| 18:23.24 | SuPrSluG | i use debian so i can just apt-get it. depends on your distro choice |
| 18:23.32 | [[blah]asfd | redhat |
| 18:23.34 | Goodjoke | Fender: wondering what anyone thought of the linksys PHB1100..whether or not that would do the job |
| 18:23.55 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 18:23.57 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a0df0de395a6eb4c) |
| 18:24.26 | SuPrSluG | they usually end in devel look for it in yum |
| 18:24.28 | joe-f | what's the status on dynamic meetme conferencing? the .tar.gz is dead - http://www.voip-info.org/wiki/view/Asterisk+Dynamic+Conferences |
| 18:24.46 | joe-f | is it possible to control conferencing in a db? |
| 18:26.14 | [[blah]asfd | what if i dont plan on using h323.. how can i make it so that asterisk addons does not look for it? |
| 18:27.21 | [TK]D-Fender | Goodjoke: a 1-line phone, and I see no support of SIP listed. I would touch it. |
| 18:27.25 | [TK]D-Fender | Goodjoke: Polycom / Aastra. Take your pick. |
| 18:27.45 | [[blah]asfd | can i just remove it from the .PHONY: line? |
| 18:28.23 | [[blah]asfd | nope, now that gives me a menuselect error |
| 18:29.27 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:29.44 | Goodjoke | fender: thanks |
| 18:30.04 | [TK]D-Fender | woudn't* |
| 18:30.09 | j0 | i've added console => notice,warning,error,dtmf to logger.conf, but it doesn't seem to be showing any dtmf presses... any ideas? |
| 18:30.37 | SomeOne1 | with H323 |
| 18:30.39 | SomeOne1 | are two ports opened for RTP |
| 18:30.40 | SomeOne1 | like one for receiving audio and one for sending audio |
| 18:30.41 | SomeOne1 | or is just one RTP port used for both receiving / sending |
| 18:30.48 | SomeOne1 | audio |
| 18:31.53 | [TK]D-Fender | j0: "set debug 10" |
| 18:32.09 | flujan | hi guys... |
| 18:32.12 | flujan | hi [TK]D-Fender |
| 18:32.18 | flujan | even with the rrmemory strategy |
| 18:32.29 | flujan | and the limitation to the users to 1 call per channel |
| 18:33.00 | flujan | I am still having problems |
| 18:33.14 | flujan | some users appear in the show queue command as Busy |
| 18:33.21 | flujan | but they aren't busy at all. |
| 18:33.21 | SomeOne1 | [TK]D-Fender, you probably know the answer to my question |
| 18:33.37 | [TK]D-Fender | SomeOne1: 1 port per CALL. |
| 18:33.57 | j0 | [TK]D-Fender: should i be seeing the dtmf tones as i press them anywhere during a call? |
| 18:34.04 | SomeOne1 | [TK]D-Fender: :) |
| 18:34.31 | [TK]D-Fender | flujan: Oh, they show as busy and they aren't? That usually happens when they use a SIP transfer to someone else. they are considered to be on that call until it terminates. |
| 18:34.53 | flujan | [TK]D-Fender, any ideas why the users appear busy? I already closed and openned the softphone, sip reload on cli... |
| 18:34.55 | [TK]D-Fender | j0: I believe so. At least any time * is supposed to CARE about DTMF. |
| 18:35.20 | flujan | [TK]D-Fender, how can I avoid it? |
| 18:35.29 | j0 | [TK]D-Fender: even with set debug 10, and running asterisk -rv(lots of vvs :) doesn't show anything |
| 18:35.29 | flujan | [TK]D-Fender, they are not transfering the call. |
| 18:35.38 | [TK]D-Fender | flujan: Stop doing SIP transfers, and use the * DTMF transfer |
| 18:36.00 | [TK]D-Fender | flujan: You sure your agents aren't passing off caller to OTHER inside people to furter assist them? |
| 18:36.17 | [TK]D-Fender | flujan: This is the only thing I know of for your scenario... |
| 18:36.31 | [TK]D-Fender | j0: and about the * CARING about DTMF part? |
| 18:37.29 | j0 | [TK]D-Fender: i'm pushing buttons both during an ivr and during calls |
| 18:38.28 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 18:39.57 | SuPrSluG | [[blah]asfd:use make menuselect and uncomment h323 |
| 18:40.33 | [[blah]asfd | did a make distclean and started over... that fixed my issue... thank you all |
| 18:41.24 | flujan | flujan, dammed... I am having a lot of problems with asterisk queues... When I find that I solved one... another appear... |
| 18:41.40 | flujan | I do not have the persistenmembers=yes on queues.conf |
| 18:41.46 | flujan | do you think it could be the problems? |
| 18:42.03 | flujan | and the calls are bein put in the queue by a agi script... |
| 18:42.07 | flujan | could it be the problem? |
| 18:42.57 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:45.09 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 18:48.04 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 18:48.20 | Mahmoud | is there anything special in Asterisk about dialing * or #? |
| 18:48.47 | Mahmoud | for example: dial(SIP/foo/*100#) |
| 18:49.08 | Mahmoud | i want to dial "*100#".. is there anything special in Asterisk that doesn't allow me to dial the number as it is? |
| 18:49.20 | Mahmoud | such as some reserved characters? |
| 18:49.27 | j0 | Mahmoud: that should work.. |
| 18:49.35 | Mahmoud | hmm so it's probably with SPA |
| 18:49.51 | Mahmoud | i can't dial any number in this format :/ |
| 18:50.00 | j0 | well the # at the end might be a problem |
| 18:50.10 | Mahmoud | may be, or even the * |
| 18:50.25 | j0 | the * should work, i use that for some things... asterisk uses that for a number of features too |
| 18:50.37 | Mahmoud | you use * with your SPA? |
| 18:51.05 | j0 | i can use it with my linksys pap2 |
| 18:51.38 | Mahmoud | hmm mine looks the same as yours |
| 18:52.02 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 18:52.04 | j0 | i'd lose the # on the end.. my sip phones use the # to 'send' a call |
| 18:52.28 | Mahmoud | SPA uses # to send a call as well |
| 18:52.52 | *** join/#asterisk dhorner_ (n=dhorner@pool-71-127-121-77.aubnin.fios.verizon.net) |
| 18:53.42 | Mahmoud | any way to send #? |
| 18:54.10 | j0 | Mahmoud: why do you need to have a # on the end? |
| 18:54.19 | bkw_ | j0, he's in the dialplan it seens |
| 18:54.22 | bkw_ | NOT using a phone |
| 18:54.26 | bkw_ | and * and # are not reserved in any way |
| 18:54.45 | bkw_ | Mahmoud, what happens when you try to dial that? |
| 18:54.51 | Mahmoud | j0, my phone ISP needs these numbers for some special services |
| 18:55.00 | bkw_ | and why are you dialing SIP/foo/*100#? is foo a phone that is registered inbound to asterisk? |
| 18:55.09 | Mahmoud | bkw_, i recieve special tones, similar to busy tone |
| 18:55.15 | bkw_ | try SIP/*100#@foo |
| 18:55.35 | Mahmoud | wat i do is SIP/account_name/*100# |
| 18:55.43 | bkw_ | don't do that |
| 18:55.48 | bkw_ | SIP/number@foo |
| 18:55.48 | Mahmoud | any difference? |
| 18:55.53 | bkw_ | with foo being the entry in sip.conf |
| 18:56.08 | Mahmoud | yeah, i heard both formats are the same.. but i'll try it |
| 18:56.22 | Mahmoud | any difference between two formats? |
| 18:56.32 | bkw_ | they are so not the same.. allowing multiple ways to dial the same thing causes problems confusion . |
| 18:56.41 | bkw_ | I have never had them dial the same at all EVER |
| 18:56.49 | *** join/#asterisk jpalmer (n=jpalmer@about/windows/regular/jpalmer) |
| 18:57.00 | [TK]D-Fender | bkw_: Actually you DO want to do it as "SIP/peerentery/number". Typo's cause resolution errors, can be hijacked, etc, and I've seen some cases where it won't work PERIOD if you use @ |
| 18:57.15 | bkw_ | [TK]D-Fender, then asterisk is broken |
| 18:57.24 | [TK]D-Fender | bkw_: And this is NEW to you? ;) |
| 18:57.39 | bkw_ | [TK]D-Fender, nope.. but having him dial the way I know works. |
| 18:57.57 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 18:58.55 | [TK]D-Fender | j0: Sounds like you are using the wrong DTMF mode on your usint. For Linksys you should be using INFO on both sides. |
| 18:59.36 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 19:00.02 | j0 | [TK]D-Fender: the problems are between asterisk and my sip provider.. i had been using rfc2833 but just tried using info with no luck |
| 19:00.30 | flujan | [TK]D-Fender, I get this information from cli |
| 19:00.30 | flujan | http://pastie.caboo.se/56519 |
| 19:00.34 | j0 | it seems to lose about 20% of my dtmf tones |
| 19:00.44 | flujan | maybe this is related to the "busy channel |
| 19:00.46 | flujan | s? |
| 19:00.55 | bkw_ | [TK]D-Fender, what linksys uses info? |
| 19:01.32 | [TK]D-Fender | bkw_: All their ATA's, phones, etc |
| 19:01.36 | bkw_ | NOT |
| 19:01.39 | bkw_ | 2833 works fine |
| 19:01.45 | [TK]D-Fender | bkw_: Everyone I've seen. |
| 19:01.55 | bkw_ | I have never ones had to us INFO for DTMF |
| 19:02.06 | bkw_ | I do know 2833 in asterisk is broken yet again :P |
| 19:02.12 | [TK]D-Fender | bkw_: I'll take your word for it, but I will still stand by my assessment that the mode moismatch si to blame. |
| 19:02.28 | bkw_ | you shouldn't have to set it.. that is what the SDP is for. |
| 19:02.41 | [TK]D-Fender | bkw_: What a wonderfully synchronous series of events this all is ;) kinda cosmic... |
| 19:02.42 | bkw_ | I set mine to AVT or Auto |
| 19:03.03 | bkw_ | and they just work |
| 19:03.22 | [TK]D-Fender | bkw_: And all of my gear "just works" as well. |
| 19:04.32 | *** join/#asterisk keith80403 (n=keith804@64-17-89-46.us.warpdriveonline.com) |
| 19:04.49 | *** join/#asterisk Bazy (n=bazy@89.137.178.124) |
| 19:04.56 | Qwell[] | ~vsc |
| 19:04.59 | jbot | vsc is, like, Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 19:05.18 | bkw_ | SIP info is great because it doesn't say the numbers arrive in order.. so if digits do arrive out of order.. tuff shit |
| 19:07.04 | J4k3 | SIP seems to be based on extreme quantities of faith in your network |
| 19:07.32 | bkw_ | not really.. |
| 19:07.46 | bkw_ | SIP is just signalling |
| 19:10.53 | J4k3 | well, sometimes things go 'kaboom' for me |
| 19:11.02 | J4k3 | like my wifi phone doesn't get incoming calls, yet its all registered and happy. |
| 19:11.10 | J4k3 | but I really suspect its crap software issues |
| 19:11.21 | J4k3 | most likely on the handset |
| 19:11.33 | J4k3 | and I am using trixbox.... |
| 19:11.34 | J4k3 | haha |
| 19:13.55 | [TK]D-Fender | bkw_: You just keep forgetting to load chan_fluxcapacitor.so on your systems to account for packet order mismatch! |
| 19:15.00 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 19:15.32 | *** join/#asterisk gpowers (n=chatzill@c-68-41-247-78.hsd1.mi.comcast.net) |
| 19:17.13 | *** join/#asterisk svenna_ (n=svenna@p548d0f7b.dip0.t-ipconnect.de) |
| 19:19.25 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 19:20.00 | flujan | [TK]D-Fender, I removed the call-limit=1 from the sip.conf stopped and started asterisk again |
| 19:20.09 | flujan | now I do not have busy locked channels... |
| 19:20.20 | flujan | but the strategy rrmemory aren't good yeat. |
| 19:20.35 | flujan | it does not work only with the not in use channels |
| 19:20.39 | flujan | try each channel |
| 19:20.50 | flujan | do you have sugestions to improve that? |
| 19:20.51 | joe-f | anyone used Web-MeetMe? |
| 19:25.03 | plasmid | to make things easier on my end I need to register a domain (the name is not taken..luck i guess). Google shows 100's of registrars...which are "recommended"? |
| 19:25.21 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 19:26.11 | champster | Somewhere meetme is disconnecting the callers, how can I troubleshhot it? |
| 19:29.29 | champster | I do see this in the log when it occurs: app_meetme.c: Cmdline: 5488|K |
| 19:29.40 | champster | I do not recall ever seeing that before |
| 19:30.42 | flujan | hi guys, someone using asterisk queues in a production environment? |
| 19:31.03 | flujan | I am having a lot of problems... queues are well distributed only in the ringall strategy |
| 19:31.15 | flujan | rrmemory does not work well with it. |
| 19:31.17 | *** join/#asterisk flashnet (n=flashnet@80-192-140-197.cable.ubr07.wiga.blueyonder.co.uk) |
| 19:32.45 | *** join/#asterisk ManOfMilk (n=CpnPlnet@70-56-26-12.eugn.qwest.net) |
| 19:33.28 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 19:33.31 | j0 | has anyone managed to get asterisknow running properly? i've just spent an hour screwing with it, and it seems hopeless.. am i just wasting my time with a beta product? |
| 19:34.44 | [[blah]asfd | I just upgraded sip.ld version on my polycom 301s and now they buttons are very slow to respond. anyone experience this before? is this common? I had this on the last version i tried to upgrade to. when upgrading versions, i only need to replace the sip.ld files right? |
| 19:34.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-210-69.red.bezeqint.net) |
| 19:35.07 | *** join/#asterisk knobenheimer (n=knoben@c-71-205-184-144.hsd1.mi.comcast.net) |
| 19:35.17 | [TK]D-Fender | [[blah]asfd: You may need to rebuild your configs depending on where you are migrating from and to |
| 19:35.57 | flujan | [TK]D-Fender, sorry for bothering you... but yesterday you suggested me to use hardphone instead of softphones to asterisk queues... |
| 19:36.08 | flujan | [TK]D-Fender, did you already have such problems? |
| 19:36.14 | [[blah]asfd | from 1.0.3 to 2.0.3 rev b |
| 19:37.52 | [[blah]asfd | [TK]D-Fender: Which config? the <mac>,cfg file? or some other? |
| 19:39.26 | Frosh | How do I see the price range for verizon Hosted IP Centrex? |
| 19:40.32 | [TK]D-Fender | j0: Likely, but others have gotten it working fine. Its not a "end-to-end" as FreePBX, which is in many ways a good thing |
| 19:40.46 | [TK]D-Fender | [[blah]asfd: sip.cfg , phoneXXX.cfg, the works |
| 19:41.02 | [TK]D-Fender | [[blah]asfd: There was no 1.0.3 ..... |
| 19:41.04 | [[blah]asfd | ok, thanks... will try that. |
| 19:42.08 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 19:43.22 | MaartenB | anyone knowing a good password strategy for securing multiple passwords? |
| 19:43.30 | MaartenB | oops... servers... :) |
| 19:44.59 | [[blah]asfd | [TK]D-Fender: 1.0.something_old ;-) |
| 19:45.13 | [TK]D-Fender | [[blah]asfd: uh-huh :) |
| 19:45.14 | froguz | how can i see the last time a sip user (peer) registered |
| 19:45.38 | [TK]D-Fender | froguz: You've got the source... start coding :) |
| 19:46.20 | [TK]D-Fender | MaartenB: http://keepass.info/ |
| 19:46.49 | froguz | i don't feel capable to do that =( |
| 19:47.32 | froguz | i'm going to suggest that feature on the dev list |
| 19:47.41 | [TK]D-Fender | froguz: Yes, begging/bribery can be effective... |
| 19:49.15 | froguz | hahahahaha i'll try anyway |
| 19:49.23 | MaartenB | [TK]D-Fender, thanks! |
| 19:54.26 | *** join/#asterisk Aji-Dahaka (n=kjelderg@fluffy.dname.net) |
| 19:54.41 | Aji-Dahaka | what bot are you using that interpretes REs? |
| 19:54.48 | Aji-Dahaka | s/RE/regular expression/ |
| 19:57.25 | *** join/#asterisk type0 (i=type0@167-144-42-72.gci.net) |
| 19:57.35 | type0 | anyone have a favorite for a DSX jackfield? |
| 19:58.18 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 19:59.39 | *** join/#asterisk ChongLi (n=chongli@gades.famouscontent.com) |
| 19:59.54 | ChongLi | wow |
| 19:59.58 | ChongLi | lots of people in here |
| 20:01.07 | clyrrad | Does this make sense to anyone? I have a SPA-941 4 Line phone with 4 Providers (4 Seperate DID's) - all 1 of the 4 I have a puzzeling issue. When I am talking the other person can hear me clear as a bell, but when i talk to them I sould very choppy and distorted. This only happens on 1 of my 4 DID's. The other 3 work perfect both ways. When I put the call on hold the line sounds perfect, but as soon as I take |
| 20:01.07 | clyrrad | it off hold it goes choppy again. Any ideas? |
| 20:01.40 | clyrrad | sorry I mean on 1 of the 4 I have the issue... |
| 20:03.07 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a99bac058a06269f) |
| 20:04.29 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 20:04.30 | clyrrad | hrm... that didnt make sense did it - LOL - lets try that again... |
| 20:04.57 | clyrrad | I have a SPA-941 4 Line phone with 4 Providers (4 Seperate DID's) - 1 of the 4 I have a puzzeling issue. When I am talking I can hear the other person clear as a bell, but when i talk to them I sould very choppy and distorted. This only happens on 1 of my 4 DID's. |
| 20:06.38 | bkw_ | what codec? |
| 20:06.59 | clyrrad | ulaw |
| 20:07.18 | bkw_ | what size packets are you sending the provider? Default is 30ms |
| 20:07.24 | bkw_ | on those sipuras |
| 20:07.35 | clyrrad | I have not changed any of those settings... |
| 20:07.43 | clyrrad | so my guess is it would be the same |
| 20:07.46 | bkw_ | you may want to lower it to 0.020 |
| 20:07.57 | clyrrad | on the phones correct? |
| 20:08.05 | bkw_ | do the phones go thru asterisk? |
| 20:08.10 | clyrrad | yep |
| 20:08.17 | bkw_ | na asterisk smooths that out |
| 20:08.20 | clyrrad | phones to asterisk - then asterisk to carriers |
| 20:08.21 | bkw_ | who is the provider? |
| 20:08.28 | clyrrad | the one with the issue? |
| 20:08.32 | bkw_ | yes |
| 20:08.35 | clyrrad | is Voicenetwork |
| 20:08.52 | bkw_ | does it sound out of order or just jittery audio? |
| 20:09.04 | clyrrad | like jittery / choppy voice |
| 20:09.17 | bkw_ | have you got reinvite on? |
| 20:09.18 | clyrrad | sounds like parts of the words are missing |
| 20:09.35 | bkw_ | canreinvite=no? on all phones and in general? |
| 20:09.44 | clyrrad | canreinvite=no |
| 20:09.53 | clyrrad | yes |
| 20:10.25 | bkw_ | try getting a packet dump and watching it |
| 20:10.26 | clyrrad | and its only this 1 Carrier DID that has the issue - and only when I talk, but if you were talking to me over the DID I would hear you perfectly |
| 20:10.52 | clyrrad | its almost like when your ISP not giving you enough upload bandwith |
| 20:11.08 | bkw_ | try a lower codec |
| 20:11.09 | clyrrad | but this is not the case.... as the other 3 carrier DID work perfect inbound and outbound |
| 20:11.11 | bkw_ | like g726 |
| 20:11.23 | clyrrad | ok let me try that |
| 20:12.34 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:13.20 | clyrrad | bkw_: same thing with g726 |
| 20:14.31 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:15.17 | clyrrad | really strange - cant seem to make sense out of this |
| 20:17.36 | clyrrad | anyone have any idea what can be the issue? |
| 20:17.53 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:18.06 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 20:18.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:18.46 | froguz | clyrrad, silence supresion disabled? |
| 20:19.29 | clyrrad | in the phone? |
| 20:20.34 | froguz | aha |
| 20:22.09 | carrar | man it's been at last 2 hours, when are they gonna release more updates!! |
| 20:22.41 | *** join/#asterisk _DAW (n=chatzill@adsl-222-56-130.msy.bellsouth.net) |
| 20:23.01 | *** join/#asterisk AMUG (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 20:23.18 | _DAW | Anyone using Audiocodes MP-XXX gateways with asterisk? |
| 20:24.44 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 20:24.49 | danp | _DAW: i think i almost did. i did some research on it |
| 20:25.19 | Aji-Dahaka | what sort of bot is jbot? |
| 20:25.43 | _DAW | I really like the MP-118 I have, however the strangest thing is happening. After a perfect 10 minutes the gateway sends a 408 and disconnects.. |
| 20:25.44 | chefrs | Anyone know why my PRI T1 w/Asterisk is sending out caller id as restricted? |
| 20:26.39 | chefrs | Im passing the data but it still for some reason at the CPE is saying that we're restricting presentation. |
| 20:33.07 | clyrrad | froguz: im checking for that setting in the phone |
| 20:34.29 | *** join/#asterisk karlhaines (n=karl@74.94.250.252) |
| 20:34.50 | *** part/#asterisk karlhaines (n=karl@74.94.250.252) |
| 20:35.00 | ChongLi | this Asterisk TFOTP pdf doesn't say anything about how to set up voip |
| 20:35.01 | ChongLi | hmm |
| 20:35.05 | froguz | clyrrad, under admin login, go to Line1 |
| 20:35.27 | froguz | there's a combobox saying Silence Supp Enable: |
| 20:35.37 | froguz | you should set up that as NO |
| 20:35.43 | clyrrad | ok gotta get the admin password for this phone and I will check |
| 20:35.50 | froguz | ok |
| 20:35.54 | clyrrad | but interesting if thats the issue.... |
| 20:36.01 | clyrrad | why the other 3 DID's dont have the issue |
| 20:36.09 | clyrrad | I know all line settings 1 to 4 are the same |
| 20:36.17 | clyrrad | but I will double check |
| 20:36.36 | froguz | maybe the other lines doesn't have silence suppresion enabled |
| 20:37.35 | froguz | you should also try switching DIDs on phone line's |
| 20:40.08 | clyrrad | froguz: ok im in to admin |
| 20:40.11 | clyrrad | lookign for that option now |
| 20:40.29 | clyrrad | indeed its set to NO |
| 20:41.53 | *** join/#asterisk rrocha (n=ruyrocha@200.139.123.131.adsl.gvt.net.br) |
| 20:42.43 | clyrrad | froguz: I also just moved the DID to a new key on the phone same issue with the choppy sound |
| 20:43.26 | froguz | have you tried with a softphone or another phone? |
| 20:43.57 | clyrrad | froguz: yes I have tried with 2 other known working phones |
| 20:44.08 | froguz | same problem? |
| 20:44.12 | *** join/#asterisk xkev (i=kevin@orbit.xmission.com) |
| 20:44.16 | clyrrad | froguz: the phone its self is good - yes same problem |
| 20:44.32 | clyrrad | froguz: I am wondering if there is some setting i can change / tweak |
| 20:44.47 | clyrrad | froguz: I have evend tried to change coded from ulaw to g726 |
| 20:45.01 | irule | http://pastebin.ca/458345 this is my sources.list and I dont have termcap available, what am I missing? thanks |
| 20:45.05 | clyrrad | same choppyness, but only one way - only on me talking, the other end sounds perfect to me |
| 20:46.27 | froguz | are the sip.conf settings the same for every sip trunk |
| 20:47.25 | clyrrad | nope each carrier had their own example for sip.conf |
| 20:47.45 | champster | No sound files are playing properly from macro while using Dial M(macro) |
| 20:47.51 | champster | Is this a known issue? |
| 20:48.10 | clyrrad | froguz: the interesting thing is........if I put you on hold, the music will stream with no choppyness |
| 20:48.11 | champster | Sometimes, I will hear a chirp, but no proper sound |
| 20:48.35 | clyrrad | froguz: but as soon as I pickup the phone again and the music on hold stops - the chopyness starts again |
| 20:48.50 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 20:50.56 | froguz | mmmm... maybe Tx gain is too loud for VoiceNetwork settings |
| 20:51.47 | clyrrad | im not sure - but I sure would like to know so I can fix it |
| 20:52.51 | froguz | FXS Port Output Gain: look for that setting on advanced regional settings |
| 20:52.59 | froguz | it should be -3. try with -5 or -6 |
| 20:53.41 | clyrrad | Under Regional Tab? |
| 20:53.42 | clyrrad | I dont see it |
| 20:54.03 | froguz | admin->advanced->regional |
| 20:54.11 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 20:54.25 | jm|laptop | ~sipbroker |
| 20:54.38 | clyrrad | froguz: Did you mean under the phone tab? |
| 20:54.43 | froguz | start searching backwards |
| 20:54.55 | clyrrad | Handset input gain? |
| 20:54.58 | Brandon_W | sippoker? I'll see your OPTIONS and raise you a 200 OK? |
| 20:55.20 | Brandon_W | and that's why I'm not a comedian.... |
| 20:55.28 | froguz | output |
| 20:56.16 | clyrrad | froguz: the only options I have for "Gain" are under the "Phone" tab |
| 20:57.32 | jm|laptop | ~sipbroker.com |
| 20:57.40 | froguz | maybe, i'm looking on a PAP2 GUI... it's very similar to the 941 |
| 20:58.12 | froguz | set the output gain to a lower level |
| 20:58.13 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-94853.home.otenet.gr) |
| 20:58.21 | clyrrad | so i have option for Handset Gain and Speakerphone Input Gain |
| 20:58.24 | clyrrad | is that what you refer to? |
| 20:58.48 | champster | No sound files are playing properly from macro while using Dial M(macro) |
| 20:58.54 | champster | Is this a known issue? |
| 20:58.56 | champster | Sometimes, I will hear a chirp, but no proper sound |
| 20:59.26 | [[blah]asfd | if i have a phone that drops its network connection while on a call the server still shows its channel active and never hangs up. Is there a way to make the server kill these phantomed channels? |
| 21:01.55 | clyrrad | froguz: is that what you were refering to? |
| 21:02.58 | froguz | is handset gain for input or output? |
| 21:03.27 | *** join/#asterisk ncampion (i=chatzill@nat/ibm/x-77901105e722cff6) |
| 21:04.24 | clyrrad | froguz: FXS Port shoudl only apply to a PAP2 devive not an 941 as a 941 is a VOIP IP Phone |
| 21:05.11 | clyrrad | froguz: they are all for input |
| 21:05.11 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 21:05.12 | clyrrad | froguz: I dont see any output gain options under any tab |
| 21:05.12 | froguz | i know, but. in term of software. 941 is just a pap2 with handset |
| 21:05.29 | clyrrad | froguz: there are no output gain options here that I can see under any tab |
| 21:05.37 | ChongLi | I can't find any good open source SIP softphone |
| 21:05.54 | ncampion | Does anyone know if using an MD5'd password is possible with the management interface? |
| 21:06.01 | ChongLi | does anyone know of a good one? |
| 21:06.43 | *** join/#asterisk dfgas (n=dfgas@adsl-75-51-45-62.dsl.milwwi.sbcglobal.net) |
| 21:07.33 | froguz | clyrrad, try using a softphone and turn the volume down. if MOH sounds good, there's no reason to get choppy voice, except for the MOH recorded level that is very low. |
| 21:07.50 | FuriousGeorge | hey all, how is it possible to have an extension which calls an agi (or something) in order to run a command in my shell? |
| 21:08.00 | dfgas | i am am trying to connect a remote softphone over the internet to my asterisk box, but its not registering, it just times out, i have dmz set to the computer too and still the same |
| 21:08.06 | dfgas | any ideas |
| 21:09.26 | errr | I have a user who is having problems with an aastra 55i phone, when you try to call them it says they are on DND we have dialed *79 to disable the DND but it still says they are DND. Is there a way I can change their status from the asterisk cli? |
| 21:09.45 | clyrrad | froguz: I can adjust the volume on the IP phone too |
| 21:10.25 | clyrrad | froguz: you think volume would make chopy sound quality though? |
| 21:12.14 | clyrrad | froguz: I just set Handset Input Gain: to -6 but still the same problem |
| 21:12.40 | clyrrad | froguz: im wondering if MOH sreams with a different codec or something and thats the reason MOH works but sould quality is choppy?? |
| 21:13.03 | *** join/#asterisk afrosheen (n=cj@207.71.49.164) |
| 21:13.21 | afrosheen | so who has used Asterisk Business Edition |
| 21:14.37 | *** join/#asterisk |ryan| (n=foo@c-67-188-22-216.hsd1.ca.comcast.net) |
| 21:15.26 | *** join/#asterisk zoranoth (n=gla@c-68-47-42-4.hsd1.ga.comcast.net) |
| 21:15.38 | afrosheen | not real lively in here these days is it |
| 21:15.58 | froguz | i think you should contact your ITSP to get proper codec settings |
| 21:16.47 | zoranoth | hi - i'm kindof new to asterisk but had a quick question |
| 21:17.19 | clyrrad | froguz: so you think its codec related? |
| 21:17.25 | zoranoth | does it do text to speech internally - do would I need something like festival to ad on? |
| 21:18.02 | froguz | maybe |
| 21:18.19 | froguz | probably you ITSP is working with a different frame size |
| 21:18.24 | afrosheen | zoranoth, you need festival, but thankfully the integration is easy |
| 21:18.54 | clyrrad | froguz: and that affect the one way audio as im seeing here? |
| 21:20.16 | froguz | it should affect on both ways |
| 21:20.50 | froguz | have you tried rtp enable debug |
| 21:21.08 | froguz | sorry: rtp debug [peer] |
| 21:22.02 | *** join/#asterisk CVirus (n=GoD@62.135.96.151) |
| 21:22.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:22.45 | froguz | rtp debug ip [your_itsp_ip] and make a call |
| 21:23.57 | clyrrad | gawd the ITSP is down now |
| 21:24.17 | clyrrad | I wish there was an easier way to get DID's for Barrie Ontario Canada |
| 21:24.26 | clyrrad | this is the only carrier I found that can provide these DID's |
| 21:24.33 | clyrrad | but I can never seem to reach anyone |
| 21:24.37 | *** part/#asterisk zim (n=chatzill@zimonline.demon.co.uk) |
| 21:25.15 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 21:25.27 | ncampion | still looking for some help setting up md5 secrets for the management interface, anyone tried this? |
| 21:25.32 | *** join/#asterisk shadou (i=aj@unaffiliated/dj-fu) |
| 21:26.24 | afrosheen | ncampion, can't say that I have, no |
| 21:27.32 | clyrrad | froguz: thanks for your help - dont look like this will get resolved in here - you have any idea where I can buy DID's for Barrie? |
| 21:27.59 | _DAW | Where's Barrie? |
| 21:27.59 | clyrrad | Ontario Canada |
| 21:28.09 | clyrrad | I found only one provider that has DID's there |
| 21:28.10 | froguz | clyrrad, try debuging RTP packets, maybe you're having dropped packets |
| 21:28.15 | clyrrad | im looking for another one |
| 21:28.18 | _DAW | Ahh.. Sorry. I only have number in US. |
| 21:28.20 | clyrrad | froguz: ok I will try now |
| 21:28.26 | froguz | also look on rtp.conf |
| 21:28.36 | froguz | disble any checksum |
| 21:29.01 | clyrrad | ther eis no checksum entry in rtp.conf |
| 21:29.12 | clyrrad | i mean there is - but its commented out |
| 21:29.28 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 21:29.39 | froguz | maybe default value is yes |
| 21:29.51 | froguz | well i have to go |
| 21:29.54 | froguz | good luck |
| 21:30.11 | flujan | guys, it is possible to track a call on a queue? |
| 21:30.31 | flujan | I want to see to which extension asterisk is trying to deliver the call. |
| 21:31.05 | Mercestes | show queues |
| 21:31.11 | Mercestes | and it should show "ringing" in the CLI |
| 21:31.32 | clyrrad | Done anyone know of a Carrier that can Provide DID's In Barrie Ontario Canada? In the 705 Area Code? |
| 21:35.12 | ChongLi | I'm installing asterisk on OS X |
| 21:35.22 | ChongLi | it is bugging me about wget |
| 21:35.37 | ChongLi | I have wget installed in /opt/local/bin/wget |
| 21:35.48 | russellb | is it in your PATH ? |
| 21:35.51 | ChongLi | and /opt/local/bin is part of my PATH |
| 21:35.52 | ChongLi | yes |
| 21:36.49 | russellb | re-run the configure script after you insstalled it? |
| 21:37.02 | ChongLi | trying that now |
| 21:37.34 | LeddyHM | logout and login |
| 21:38.18 | ChongLi | rerunning the config script worked |
| 21:38.18 | ChongLi | thanks |
| 21:38.51 | russellb | np |
| 21:39.52 | Keltus | just to confirm, DID is when you bypass local phone lines, and just have a PHONENUMBER -> IP -> ASTERISK set up right? |
| 21:40.11 | Keltus | I am new to the whole voip thing and all the acronyms are a bit confusing |
| 21:40.59 | afrosheen | Keltus, DID doesn't have to involve an IP, it can just be NUMBER -> T1/ANALOG INTERFACE -> ASTERISK |
| 21:41.02 | LeddyHM | Direct Inward Dial |
| 21:41.23 | russellb | DID == phone number |
| 21:41.29 | Keltus | hmm.. okay |
| 21:41.40 | russellb | ~did |
| 21:41.47 | jbot | i guess did is Direct Inward Dialing |
| 21:41.47 | russellb | ~die |
| 21:41.50 | jbot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
| 21:41.56 | LeddyHM | yes it is |
| 21:42.19 | LeddyHM | what you do with the did is up to you |
| 21:42.21 | *** join/#asterisk VioByte (n=afreeman@unallocated-63-210-246-146.cvg1.bytehosting.com) |
| 21:42.27 | Keltus | so if I'm shopping for a DID, and I purchase one, is there a field where I would put my IP address? |
| 21:42.42 | Keltus | it's up to me, but within the limitations of my service provider |
| 21:42.55 | *** part/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 21:42.58 | VioByte | anyone know if theres a way to send a call out IAX2 to a number with an extension? |
| 21:43.04 | LeddyHM | you aren't talking apples to apples |
| 21:46.41 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 21:51.33 | DrukenHME | anyone want a headache?? free of charge.... |
| 21:51.43 | SwK | I charge for headaches |
| 21:52.03 | DrukenHME | SwK: i'd be willing to give you mine for nuttin |
| 21:52.14 | illsci | Hey.... there's people from digium in here right |
| 21:52.20 | SwK | I was refering to when people give them to me |
| 21:52.26 | DrukenHME | :) |
| 21:52.30 | SwK | illsci sometimes |
| 21:52.43 | illsci | is the remotely exploitable bug they just posted limited to chan_sip in the affected versions? |
| 21:53.02 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:53.31 | CVirus | I'll design a system that consists of a telephone line connected to an FXO interface in an Asterisk server and the asterisk server is connected to an ATA that has 2 network jacks and 4 telephone jacks connected to 4 analog phones ... If I want to extend the system to 8 phones ... Do I connect another ATA to the first one using a network cable ? |
| 21:53.33 | CVirus | will this work ? |
| 21:54.06 | BSD_Tech | clear |
| 21:54.34 | [TK]D-Fender | CVirus, NO. |
| 21:54.35 | SwK | CVirus, use a hub and plug the asteriskbox and 2 ata's into that hub (or switch...) its not like the switches are that expensive these days |
| 21:54.54 | [TK]D-Fender | CVirus, The ATA's with 2 jacks were meant to be used like a residential router. |
| 21:55.12 | CVirus | SwK: got that |
| 21:55.13 | jm|laptop | what should I be checking for in show applications to make sure I can use EnumLookup ? |
| 21:55.17 | *** join/#asterisk Stridernzl (n=neville@125-239-160-237.jetstream.xtra.co.nz) |
| 21:55.18 | [TK]D-Fender | CVirus, you will royally screw your networking if your daisy--chain them like that |
| 21:55.29 | CVirus | [TK]D-Fender: thanks alot |
| 21:55.49 | jm|laptop | "pbx_extension_helper: No application 'EnumLookup' for extension" |
| 21:55.50 | illsci | i guess the key is to have 't38_udptl' set to 'no' |
| 21:56.32 | jm|laptop | oh |
| 21:56.36 | jm|laptop | it's deprecated |
| 21:57.04 | Stridernzl | [TK]D-Fender: Any news re SPanDSP ? |
| 21:57.36 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
| 21:58.44 | afrosheen | anyone ever buy or install Asterisk Business Edition? |
| 21:58.51 | [TK]D-Fender | Stridernzl, nope , shortly hope to be in touch with the guy who was going to point me in the right direction |
| 22:00.24 | Stridernzl | [TK]D-fender: Heres Hoping, .. Its kind of starting to drag on now :( .. Can email me if any news , I hate to sit on you :) |
| 22:01.14 | clyrrad | Done anyone know of a Carrier that can Provide DID's In Barrie Ontario Canada? In the 705 Area Code? |
| 22:01.48 | Stridernzl | TK-Defender: on a more positive note everything that we have got seems to be working just fine ..! :) |
| 22:02.17 | [TK]D-Fender | Stridernzl, excellent to hear |
| 22:03.18 | dfgas | i am am trying to connect a remote softphone over the internet to my asterisk box, but its not registering, it just times out, i have dmz set to the computer too and still the same |
| 22:03.35 | Stridernzl | TK-Defender : actually can you answer the follwing ... is it possible to tak and inbound call .. part it .. play a message to user then forward thier call out to another number ? |
| 22:04.18 | Sweeper | 'part' it? |
| 22:04.22 | [TK]D-Fender | Stridernzl, PM |
| 22:04.48 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:05.55 | LeddyHM | ya, wtf is part it |
| 22:07.41 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 22:08.05 | LeddyHM | katana's are bad....mkay |
| 22:09.10 | *** join/#asterisk Ezway (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 22:09.21 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
| 22:11.46 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 22:11.55 | [TK]D-Fender | LeddyHM, Only if you're on the wrong side :D |
| 22:11.56 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 22:13.10 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 22:13.45 | [TK]D-Fender | LeddyHM, this new one had a POB thats just sick. very nimble blade.... |
| 22:14.29 | LeddyHM | bad bad |
| 22:14.35 | LeddyHM | I know what you should be doing :) |
| 22:14.54 | *** join/#asterisk rmayorga_ (i=rmyorg@168.243.89.17) |
| 22:14.57 | *** join/#asterisk dfgas (n=dfgas@adsl-75-44-47-177.dsl.milwwi.sbcglobal.net) |
| 22:15.02 | dfgas | ? rtp |
| 22:16.56 | *** join/#asterisk toombaloomba (n=phph@do.you.like.my.frippers.com) |
| 22:17.21 | Vec | If I want to write variables to a mySQL database would Realtime be the correct way to go about it ? It seems realtime is more for storing the entire config in the database rather than reading and writing user specific variables to tables etc ? |
| 22:17.29 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 22:17.46 | LeddyHM | maybe agi |
| 22:18.23 | [TK]D-Fender | Vec, realtime is for sip users, dialplan, etc. If you just want to store & grab for IVR usage, ect, then AGI or app_odbc |
| 22:18.42 | LeddyHM | damn I'm smart |
| 22:20.00 | [TK]D-Fender | LeddyHM, thats right.... you're damned.... |
| 22:20.31 | LeddyHM | hey now |
| 22:20.33 | [TK]D-Fender | :D |
| 22:20.34 | Vec | [TK]D-Fender : thanks |
| 22:20.37 | [TK]D-Fender | pwned |
| 22:21.32 | LeddyHM | is that how you treat all your paying customers? |
| 22:21.38 | LeddyHM | or just the pita ones? |
| 22:22.26 | JT | you pay him? |
| 22:23.03 | [TK]D-Fender | heh, comic relief :) |
| 22:23.10 | LeddyHM | not for what you think *wink* |
| 22:23.29 | Vec | [TK]D-Fender : maybe u can help me: I have a call coming in via a DID Zap channel, I am then forwarding out via the ZAP using exten => incomingNumber,1,Dial(Zap/outgoingNum), it works as expected, except in the CDR database it does not have the outgoingNum, in the dst field, it is just blank, I have tried ResetCDR and ForkCDR ? |
| 22:23.55 | [TK]D-Fender | but not the kind you're thinking of..... |
| 22:24.35 | [TK]D-Fender | Vec, that is not forwarding, and should not create another CRD IIRC. |
| 22:24.38 | toombaloomba | where can I find the changelog for all those new versions in topic |
| 22:24.47 | Qwell[] | toombaloomba: in the tarball |
| 22:24.50 | toombaloomba | ah |
| 22:24.52 | toombaloomba | ok |
| 22:24.55 | toombaloomba | was looking ont he web :) |
| 22:24.55 | [TK]D-Fender | toombaloomba, in the same FTP folder that hosts them |
| 22:25.02 | Qwell[] | there too |
| 22:25.49 | toombaloomba | cool thanks |
| 22:26.19 | Vec | [TK]D-Fender : How do I divert (forward) the call correctly ? |
| 22:26.54 | Vec | I want asterisk to see the Dial(blah) as a seperate call |
| 22:28.36 | [TK]D-Fender | Vec, You could prehaps us Transfer and pass the call through chan_sip by calling yourself on an internally auth'd call.... |
| 22:34.27 | ChongLi | wow... looking at these sample conf files |
| 22:34.33 | ChongLi | pretty overwhelming |
| 22:34.47 | JT | they're not that hard |
| 22:35.05 | toombaloomba | hmm getting make: *** No rule to make target `xpp/xpp_usb.ko', needed by `install-modules'. Stop. on make install on zaptel 1.4.2.1 |
| 22:35.07 | ChongLi | there's so many options |
| 22:35.08 | CVirus | [TK]D-Fender: SwK: Is it better to use SIP or IAX for my system ? |
| 22:35.32 | ChongLi | I just want to set up SIP and do a test call to myself |
| 22:35.34 | [TK]D-Fender | CVirus, for WHAT is the question... |
| 22:35.53 | CVirus | [TK]D-Fender: I'll design a system that consists of a telephone line connected to an FXO interface in an Asterisk server and the asterisk server is connected to an ATA that has 2 network jacks and 4 telephone jacks connected to 4 analog phones |
| 22:36.19 | JT | an expensive ATA |
| 22:36.25 | [TK]D-Fender | CVirus, And where would IAX or SIP fit into that equation? |
| 22:36.54 | CVirus | [TK]D-Fender: it wouldn't ? |
| 22:37.04 | CVirus | [TK]D-Fender: I'm still an asterisk newbie .. sorry |
| 22:37.24 | CVirus | [TK]D-Fender: what protocol will asterisk use to route the incoming calls ? |
| 22:37.27 | JT | it would, to the ATA |
| 22:37.38 | JT | most ATAs will no speak IAX, so SIP |
| 22:37.47 | JT | but the rest of it is internal to asterisk |
| 22:37.53 | CVirus | I see |
| 22:38.00 | CVirus | JT: why would I need an expensive ATA ? |
| 22:38.02 | [TK]D-Fender | CVirus, you need to think what kind of languages your prospective HARDWARE can use, and THEN think about which is BEST for you. |
| 22:38.05 | JT | CVirus: VoIP protocols are only relevant if using VoIP |
| 22:38.21 | JT | CVirus: an ATA with 4 X FXS and 2 X Ethernet is expensive if it exists |
| 22:38.34 | CVirus | 1 ethernet sorry |
| 22:38.47 | JT | even so, 4 FXS is getting into business grade |
| 22:38.52 | CVirus | hmm |
| 22:38.56 | [TK]D-Fender | CVirus, Do you have specific equipment in mind? Do share... |
| 22:39.11 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 22:39.13 | CVirus | [TK]D-Fender: haven't thought about it yet ... searching for the cheapest though |
| 22:39.30 | [TK]D-Fender | ~ygwypf |
| 22:39.32 | jbot | from memory, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 22:39.34 | CVirus | [TK]D-Fender: I might be using SIP phones instead of ATA's too |
| 22:39.35 | CVirus | [TK]D-Fender: excuse me ? |
| 22:39.38 | JT | that would be smarter |
| 22:39.43 | JT | using sip phones |
| 22:39.50 | CVirus | ah |
| 22:39.56 | ChongLi | I just did a "dialplan show" |
| 22:40.06 | ChongLi | and I got tons of stuff in here |
| 22:40.16 | ChongLi | all of this sample stuff |
| 22:40.24 | CVirus | JT: would that provide a cheaper solution ? |
| 22:40.41 | [TK]D-Fender | CVirus, stop thinking CHEAP, and start thinking bang-for-buck |
| 22:40.47 | JT | CVirus: it would provider a superior solution, not necessarily cheaper |
| 22:40.59 | [TK]D-Fender | CVirus, You want 1 line, 4 unique phones? |
| 22:41.22 | clyrrad | Done anyone know of a Carrier that can Provide DID's In Barrie Ontario Canada? In the 705 Area Code? If not someone know how / where I can look for such a provider? I tried voip-info with no luck |
| 22:41.34 | CVirus | [TK]D-Fender: yes |
| 22:41.37 | [TK]D-Fender | clyrrad, Checked out les.net yet? |
| 22:41.52 | clyrrad | [TK]D-Fender: yes - they dont provide for Barrie |
| 22:42.02 | [TK]D-Fender | CVirus, Do you already OWN the analog phones you might use with an ATA if that is part of the solution we might suggest? |
| 22:42.05 | ChongLi | CDN ON Barrie$3.50$0.011 |
| 22:42.18 | [TK]D-Fender | clyrrad, Yeah, its nota common area... this might be pretty hard. |
| 22:42.18 | ChongLi | $3.50 for a DID |
| 22:42.19 | clyrrad | ChongLi: where did you get this from? |
| 22:42.27 | ChongLi | http://les.net/products/product_ipdidcanada.php |
| 22:42.37 | CVirus | [TK]D-Fender: no |
| 22:42.40 | clyrrad | [TK]D-Fender: yea but they have 135 thousand residents in Barrie |
| 22:42.54 | CVirus | [TK]D-Fender: I'm thinking of either analog phones, SIP phones, soft phones |
| 22:42.54 | clyrrad | ChongLi: I spoke with them before they discontinue Barrie |
| 22:43.11 | [TK]D-Fender | clyrrad, hmmmm 135k should be significant enough for SOMEBODY to offer it.... |
| 22:43.34 | [TK]D-Fender | clyrrad, but i suspect this might not be easy. |
| 22:43.58 | clyrrad | [TK]D-Fender: yea you would think so - I been searching for months......... I cant find anyone except Voice network - but this guy is never available |
| 22:43.58 | ChongLi | clyrrad: why'd they discontinue service in barrie? |
| 22:44.02 | ChongLi | they still list it on their site |
| 22:44.04 | clyrrad | ChongLi: he said too expensive |
| 22:44.07 | [TK]D-Fender | CVirus, next, where are you located? |
| 22:44.22 | CVirus | [TK]D-Fender: Egypt |
| 22:44.24 | clyrrad | ChongLi: yea - he said he needs to remove it from his website but he still hasent |
| 22:45.09 | clyrrad | I wish there was a way to get into Barrie - aparantly Vonage is there - not sure how they are doing it |
| 22:45.23 | ChongLi | vonage is proprietary |
| 22:45.25 | clyrrad | when you go to the Vonage site and enter a DID from Barrie it sais you can port it |
| 22:45.26 | ChongLi | isn't it? |
| 22:45.49 | clyrrad | ChongLi: I am wondering how they can get into Barrie if others cant / arnt |
| 22:45.51 | [TK]D-Fender | CVirus, Ok, I don't know how pricing works out there. In North America I'd suggest you get 4 Polycom IP 320's ($95 ea), a 4-port Poe switch (4 non PoE) +/- $135, and a TDM400P w/ 1FXO |
| 22:45.58 | ChongLi | I am looking for ways to get rid of PSTN crap |
| 22:46.03 | ChongLi | and go with voip only |
| 22:46.12 | ChongLi | PSTN service is such a ripoff |
| 22:46.17 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 22:46.19 | clyrrad | thats forsure |
| 22:46.25 | kuku5 | Can you park to different extensions ? |
| 22:46.50 | clyrrad | [TK]D-Fender: any idea how Vonage can get in there to port and register new DID's while others are not? |
| 22:46.57 | [TK]D-Fender | CVirus, If you want to be a complete cheap-ass, 1 x SPA-3102, 2x SPA-2102 + a switch. Mind you to acheive sheap-ass state you'd pair it up with a really sucky phone. for each port |
| 22:47.15 | THX2000 | $$ |
| 22:47.26 | [TK]D-Fender | kuku5, Rephrase.... your meaning is unclear and wording awkward |
| 22:47.30 | kuku5 | Refrase: can I park a call to a different context. It seems there is only one "parking lot" |
| 22:47.38 | Brandon_W | clyrrad - depends on where you are porting to |
| 22:47.42 | Brandon_W | the company that is |
| 22:47.58 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:47.58 | [TK]D-Fender | kuku5, *'s parking has only 1 lot. If you don't like that, get app_valetparking. |
| 22:48.01 | JT | clyrrad: vonage have this thing called "money" - that's my guess |
| 22:48.04 | clyrrad | Brandon_W: what do you mean? I am trying to find a compnay that can port these |
| 22:48.11 | Brandon_W | that company may not have the agreements in place to take calls from that area |
| 22:48.20 | kuku5 | app_valetparking .... where od i get that |
| 22:48.39 | [TK]D-Fender | kuku5, last I checked pbxfreeware.org |
| 22:48.40 | LeddyHM | google |
| 22:49.20 | *** join/#asterisk SomethingISODD (n=dan@XPLR-TS-10-VAN-72-45-68-29.barrettxplore.com) |
| 22:49.46 | SomethingISODD | hello all question, could one use asterisk to route a did to another server to use it for dial up internet access |
| 22:49.56 | kuku5 | ok - thanks guys |
| 22:50.15 | JT | what?! dialup? |
| 22:50.19 | [TK]D-Fender | SomethingISODD, Modem over VoIP? Are you on crack? Do you rquire a dealer? |
| 22:50.24 | clyrrad | Brandon_W: Well I wish a half decent carrier would get into there soon.... |
| 22:50.27 | JT | strong drugs |
| 22:50.45 | SomethingISODD | [TK]D-Fender do you figure it would work?? |
| 22:50.55 | SomethingISODD | there is a guy i know that says he`s doing it but its hard to believe |
| 22:50.55 | JT | SomethingISODD: what do you bloody think? |
| 22:50.57 | Brandon_W | I agree, I have the same issue where my parents are from |
| 22:51.15 | clyrrad | from Barrie too? |
| 22:51.45 | Brandon_W | no, Western Maryland |
| 22:51.47 | Brandon_W | up in the sticks |
| 22:52.46 | clyrrad | I see.... so what does a company like Vonage need to do to get into these Areas? Do they need physical equipment in Barrie? IE... How can you get these DID's to your service if you were willing to pay for them? |
| 22:53.48 | clyrrad | Becase I dont know of any Datacenters in Barrie that could house equipment reliably for VOIP - so there must be some other way to get the DID's into your Datacenter |
| 22:54.05 | Brandon_W | I'm not a 100% sure, may have to do with interconnect agreements between Joe-VoIP-Company and the local Carrier you would be porting from. Whoever owns the NPA-NXX block in LERG |
| 22:54.32 | anthony] | Haha |
| 22:54.33 | clyrrad | Brandon_W: from what I understand thats Bell Canada |
| 22:54.38 | anthony] | Modem Over IP |
| 22:54.43 | anthony] | I need to quote that :) |
| 22:54.56 | JT | no |
| 22:55.03 | anthony] | why :( |
| 22:55.09 | JT | Modem over VoIP over Internet |
| 22:55.17 | JT | that's what you need to quote |
| 22:55.22 | JT | as it's the comedy of errors |
| 22:55.28 | anthony] | Hh |
| 22:55.31 | anthony] | heh* |
| 22:57.10 | JT | BananAway: please stop that |
| 22:57.17 | JT | BananAway: disable annoying away announces |
| 23:10.20 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 23:10.21 | clyrrad | Does anyone here use Voice Network for DID's? |
| 23:10.21 | clyrrad | Better question is - if anyone here DOES use DID's from them.... have you found a successful way to get ahold of anyone? |
| 23:10.39 | [TK]D-Fender | clyrrad, if you have to ask that, run, run fast, run NOW. |
| 23:10.45 | clyrrad | yea no kidding LOL |
| 23:10.46 | clyrrad | im trying to |
| 23:10.59 | clyrrad | Im trying to find a place to get DID's for Barrie - they are the ONLY one I can find |
| 23:11.57 | clyrrad | with VOIP telco's I have found there are so many awful carriers - carriers with no sense of how to provide even minimal customer support |
| 23:12.09 | clyrrad | and the good ones arnt in the areas you need :s |
| 23:13.03 | ChongLi | hi, I got a sip call going |
| 23:13.07 | ChongLi | but I'm not getting any audio |
| 23:13.26 | ChongLi | using softphones on 2 computers |
| 23:13.32 | ChongLi | I can get them to call each other |
| 23:13.37 | ChongLi | but no audio |
| 23:13.55 | Brandon_W | firewalls? NAT? evil Gnome with a hatchet? |
| 23:14.02 | ChongLi | this is on a LAN |
| 23:14.06 | ChongLi | no firewalls or nat |
| 23:14.22 | Brandon_W | Evil Gnome FTW! |
| 23:14.43 | Brandon_W | fire up wireshark, see if you are getting RTP |
| 23:14.53 | Brandon_W | software firewalls on the computers perhaps? |
| 23:14.55 | Brandon_W | never know... |
| 23:15.00 | ChongLi | no software firewalls |
| 23:15.31 | Brandon_W | wireshark (ethereal) is gonna be the place to start |
| 23:15.45 | Brandon_W | capture SIP Messaging between the two, see where the SDP info is pointing |
| 23:15.47 | ChongLi | guess I'll have to go install that |
| 23:15.52 | Brandon_W | just to make sure IP and PORT stuff is right |
| 23:16.18 | Brandon_W | oh |
| 23:16.19 | Brandon_W | also |
| 23:16.21 | Brandon_W | mmm |
| 23:16.43 | Brandon_W | if using asterisk you can tcpdump it or do sip debug peer |
| 23:16.43 | JT | or we could start off easier and you could describe your setup and configuration |
| 23:16.50 | Brandon_W | yeah, that too |
| 23:16.58 | ChongLi | ok |
| 23:16.58 | Brandon_W | heh |
| 23:17.01 | Brandon_W | sorry |
| 23:17.09 | ChongLi | I have 2 friends in my sip.confg |
| 23:17.11 | ChongLi | conf |
| 23:17.19 | ChongLi | 2000 and 2001 |
| 23:17.54 | ChongLi | basically I'm following this |
| 23:17.55 | ChongLi | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 |
| 23:18.43 | *** join/#asterisk Caplain (n=shayne@pool-71-184-33-177.bstnma.east.verizon.net) |
| 23:18.56 | Caplain | whats a good software phone? |
| 23:19.10 | JT | ChongLi: what soft phone are you using? |
| 23:19.16 | ChongLi | wengophone |
| 23:19.22 | LeddyHM | xlite |
| 23:19.25 | JT | Caplain: none are very good... iax2 or sip? |
| 23:19.37 | ChongLi | xlite is proprietary :( |
| 23:20.22 | JT | and there is absolutely no open source softphone that comes close to "good" |
| 23:20.22 | JT | ekiga is probably the best |
| 23:20.22 | JT | but still |
| 23:20.22 | Brandon_W | I wouldn't call xlite proprietary, it just costs $$$ |
| 23:20.22 | JT | all softphones are varying degrees of shit |
| 23:20.30 | ChongLi | Brandon_W: where can I download the source code to xlite? |
| 23:20.37 | JT | xlite is closed source, not sure whether technically that's proprietary |
| 23:20.47 | ChongLi | that's proprietary enough to me |
| 23:21.00 | JT | ChongLi: proprietary usually refers to compliance with open standards |
| 23:21.00 | Brandon_W | heh |
| 23:21.11 | Brandon_W | Skype is proprietary |
| 23:21.14 | JT | yes |
| 23:21.22 | ChongLi | anyway, I don't plan to use a softphone forever, I am just evaluating it with asterix before I buy any hardware |
| 23:21.39 | ChongLi | asterisk rather |
| 23:21.39 | ChongLi | hah |
| 23:21.58 | ChongLi | I'm doing this to replace skype |
| 23:22.13 | Caplain | thanks ChongLi |
| 23:22.24 | ChongLi | Caplain: thanks? |
| 23:22.24 | JT | try ekiga or idefisk |
| 23:22.37 | Caplain | xlite |
| 23:22.43 | *** join/#asterisk pbr503 (i=devilsda@enygma.org) |
| 23:22.55 | pbr503 | anyone know how to do an emergy code break? |
| 23:22.58 | LeddyHM | or xlite |
| 23:22.58 | pbr503 | emergency even |
| 23:23.11 | JT | xlite works but unfortunately only handles 2 calls, and one account, which is very limiting |
| 23:23.21 | JT | pbr503: what the hell are you talking about? :) |
| 23:23.25 | LeddyHM | great for testing |
| 23:23.27 | LeddyHM | ;) |
| 23:23.34 | ChongLi | neither ekiga nor idefisk run on OS X |
| 23:24.05 | ChongLi | hmm |
| 23:24.05 | JT | idefisk runs on OSX last i checked |
| 23:24.05 | pbr503 | JT; say you are in the middle of checking voicemail. have a special code definied (ie: *9) to break out of voicemail and say drop you to a dialtone |
| 23:24.17 | JT | ekiga, i'm not sure, in theory you could probably compile it on osx |
| 23:26.55 | pbr503 | closest i've found is features.conf but that is apparently only for transferring calls |
| 23:26.55 | Brandon_W | idefisk has an OSX download from what I see on their site |
| 23:26.55 | JT | pbr503: oh, and we were all meant to know what that is? :) |
| 23:26.55 | ChongLi | I see now |
| 23:26.55 | pbr503 | sorry. ;) |
| 23:26.55 | ChongLi | the initial page I got said windows only, but this other page shows OS X |
| 23:26.55 | ChongLi | http://www.asteriskguru.com/idefisk/ |
| 23:26.56 | pbr503 | or better question has any successfully gotten asterisk to interface with grandcentral ? |
| 23:26.56 | JT | pbr503: it sounds like a pointless exercise, and something that would be difficult to do |
| 23:26.56 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 23:26.56 | pbr503 | JT; well the real-world situation i would like it for is the conference module (app_conference.so) does not seem to have the functionality to manually add people to the conference |
| 23:26.56 | pbr503 | so i was thinking if i could program in a code (*9) that would allow me to enter a phone number to call out via IAX2 and add the person to the conference |
| 23:26.56 | JT | i know meetme can add people to the conference |
| 23:27.09 | pbr503 | zaptel/meetme is not an option since my VPS has a Xen kernel installed which is incompatible with ztdummy |
| 23:27.15 | pbr503 | i knew you were going to go there. ;) |
| 23:27.37 | Brandon_W | ChongLi - so about your no-audio issues. The phones register with asterisk fine? They can call to eachother? The lines pick up and you just can't hear either party? |
| 23:27.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:27.42 | JT | eww, running asterisk on vps |
| 23:27.48 | ChongLi | Brandon_W: right |
| 23:27.56 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 23:28.00 | ChongLi | I am trying idefisk |
| 23:28.05 | pbr503 | yea.. i know. :( |
| 23:28.19 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-94853.home.otenet.gr) |
| 23:28.32 | Brandon_W | ChongLi - I hate to ask, and not that I think anyone is dumb, but just in case someone missed it, are the computer microphones unmuted? :) |
| 23:28.45 | ChongLi | the mics are working fine |
| 23:28.55 | pbr503 | okay. forget that. what about the grandcentral question? ;) |
| 23:29.16 | JT | pbr503: i thought it was one question |
| 23:29.34 | JT | ChongLi: try using Playback or Echo before trying to make a call to another endpoint |
| 23:29.34 | pbr503 | <pbr503> or better question has any successfully gotten asterisk to interface |
| 23:29.34 | pbr503 | <PROTECTED> |
| 23:29.52 | JT | pbr503: that's be great if i knew what grandcentral was |
| 23:30.19 | Brandon_W | ChongLi: *43 |
| 23:30.20 | pbr503 | www.grandcentral.com |
| 23:30.29 | ChongLi | Brandon_W: *43? |
| 23:30.31 | ChongLi | dial that? |
| 23:30.34 | Brandon_W | yeah |
| 23:30.34 | VioByte | sounds like a train station |
| 23:30.38 | Brandon_W | asterisk echo test |
| 23:32.10 | Brandon_W | as JT suggested |
| 23:32.11 | pbr503 | you can pick a number almost anywhere in the US and have that number forward to any phone number you have |
| 23:32.11 | ChongLi | damn this client sucks heh |
| 23:32.11 | ChongLi | it crashes now |
| 23:32.11 | JT | Brandon_W: only useful if you have the sample dialplan activated, which is not advised |
| 23:32.11 | pbr503 | i setup a local number here to forward to my PSTN number in iowa and it works. it runs through asterisk. but grandcentral makes you press a number to answer the phone call |
| 23:32.11 | ChongLi | *43 gets an error |
| 23:32.20 | JT | ChongLi: just add an extension of your choice that Answers a call then does Echo |
| 23:32.30 | JT | and another with Playback instead of Echo |
| 23:33.45 | ChongLi | exten => 1999,1,Echo |
| 23:33.46 | ChongLi | exten => 1998,1,Playback |
| 23:33.49 | ChongLi | like that? |
| 23:33.51 | JT | no |
| 23:33.58 | JT | priority numbers must increment |
| 23:34.29 | JT | or 1 for the first priority, and n for subsequent priorities |
| 23:35.02 | JT | argh |
| 23:35.10 | JT | another error also |
| 23:35.17 | ChongLi | yeah I'm a total nub hehe |
| 23:35.22 | JT | please read the book on dialplan stuff |
| 23:35.44 | JT | ~thebook |
| 23:36.03 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:36.03 | JT | what you need to do is |
| 23:36.03 | JT | exten => 1999,1,Answer |
| 23:36.07 | JT | exten => 1999,n,Echo |
| 23:36.09 | JT | exten => 1998,1,Answer |
| 23:36.20 | JT | exten => 1998,n,Playback(tt-monkeys) |
| 23:36.27 | ChongLi | wow |
| 23:36.30 | ChongLi | I'm getting audio |
| 23:36.43 | JT | nice |
| 23:36.51 | JT | Echo? |
| 23:36.54 | ChongLi | a nice feedback loop |
| 23:37.01 | Brandon_W | speaking of O'Reilly books they got a second edition of Asterisk: tFoT comeing out soon; and also "Asterisk Cookbook" |
| 23:37.14 | JT | Brandon_W: it's overdue i think |
| 23:37.18 | Brandon_W | very |
| 23:37.48 | ChongLi | audio is choppy |
| 23:37.50 | ChongLi | but it works |
| 23:38.01 | JT | hmm |
| 23:38.12 | ChongLi | over gigabit ethernet no less... |
| 23:38.12 | JT | choppy, might be the softphone |
| 23:38.26 | ChongLi | yeah I probably should trash this one |
| 23:38.37 | ChongLi | it crashes |
| 23:38.43 | JT | it might have silence supression/voice activity detection turned on |
| 23:38.52 | VioByte | soft phones suck |
| 23:39.06 | ChongLi | ok, I have idefisk now |
| 23:39.15 | VioByte | :) |
| 23:39.17 | JT | you need to setup IAX2 then |
| 23:39.27 | JT | Polycom > * :) |
| 23:39.41 | JT | ? |
| 23:39.50 | Brandon_W | ugh, they were a nightmare back when I had to support them |
| 23:39.55 | JT | uhuh |
| 23:39.55 | Brandon_W | MGCP setup though |
| 23:39.59 | JT | well |
| 23:40.01 | Brandon_W | maybe they are better with SIP |
| 23:40.01 | JT | there you go |
| 23:40.08 | Brandon_W | I've been meaning to dust one off and try it |
| 23:40.10 | JT | mgcp is lame :P |
| 23:40.18 | Mercestes | mgcp is masochistic |
| 23:40.19 | Brandon_W | not gonna argue with you |
| 23:40.36 | Mercestes | masochistic geeks controlling phones. That's what MGCP stands for. |
| 23:40.43 | Brandon_W | again, no arguments; but ya gotta support what you gotta support |
| 23:41.07 | [TK]D-Fender | Brandon_W, http://internetarguing.ytmnd.com/ |
| 23:41.20 | mcab | Brandon_W: what callserver did you have to use MGCP for? |
| 23:41.37 | Brandon_W | Fender - ah yes, I love that picture |
| 23:41.39 | Brandon_W | good site |
| 23:42.11 | Mercestes | ROFLMAO |
| 23:42.19 | Brandon_W | mcab - we were using a sylantro server |
| 23:42.19 | [TK]D-Fender | Brandon_W, and sorry.. with SIP firmware I have yet to see a phone that polycom's call handling capabilities doesn't destroy. |
| 23:42.28 | Brandon_W | hahaha |
| 23:42.49 | [TK]D-Fender | Everyone is entitled to my own opinion ;) |
| 23:42.59 | mcab | Brandon_W: you could've used used SIP then :-D |
| 23:43.16 | [TK]D-Fender | Coulda had a V-8! |
| 23:43.24 | Brandon_W | mcab - not my call, I just work for a living |
| 23:43.41 | mcab | Brandon_W: yah, I understand that... :-7 |
| 23:44.17 | ChongLi | ok |
| 23:44.24 | ChongLi | I got idefisk echoing |
| 23:44.27 | ChongLi | this is cool |
| 23:44.30 | ChongLi | sound quality is good |
| 23:46.50 | JT | ChongLi: nothing compared to the sound quality from a good sip hardphone :) |
| 23:47.06 | JT | the difference between idefisk and polycom, for example, is night and day |
| 23:47.35 | ChongLi | it depends on the mic too though |
| 23:47.56 | ChongLi | get a really good studio mic for example |
| 23:47.58 | JT | received audio also |
| 23:48.08 | JT | idefisk repeats every little bit of background noise |
| 23:53.14 | Brandon_W | so what's the easiest way to change over to SIP firmware for a polycom |
| 23:53.39 | Brandon_W | besides a hammer and/or explosives |
| 23:53.50 | Brandon_W | as tempted as I may be |
| 23:54.03 | mcab | Brandon_W: wipe the filesystem, point phone at bootserver with SIP firmware |
| 23:58.59 | ChongLi | where can I buy a good hard phone? |
| 23:58.59 | VioByte | ebay :) |
| 23:58.59 | Brandon_W | I found it eeeeeeeeeeeeeeebaaaaay |
| 23:58.59 | VioByte | or look around on google's froogle |
| 23:59.31 | VioByte | http://www.google.com/products |
| 23:59.34 | ChongLi | should I go with SIP or IAX2? |
| 23:59.35 | VioByte | use to be froogle |
| 23:59.40 | *** join/#asterisk ron_lautmann (n=ron@208.106.33.66) |
| 23:59.48 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 23:59.54 | VioByte | i'd go with sip but thats me :) |