| 00:01.16 | leifmadsen | the easiest way to tell the Polycom how to connect to a server, is via option 66 on the dhcp server |
| 00:01.29 | leifmadsen | and with that tidbit of info, I'm away from the computer for the night |
| 00:01.46 | Micc | leifmadsen, yeah, but its on a bunch of different networks we don't have dhcp servers that can do that on. |
| 00:02.06 | Micc | bbourdage, we have the provisioning server up still. |
| 00:02.12 | Micc | But we are decomissionging it. |
| 00:02.14 | bbourdage | Micc, set a dhcp forwarder, |
| 00:02.21 | Micc | So we want to change it to another ip. |
| 00:02.46 | Micc | can the provisioning method and url be set in the cfg file? |
| 00:03.22 | bbourdage | yes |
| 00:03.37 | Micc | and override whatever is in the phone locally |
| 00:04.10 | bbourdage | be carefull with that, you will change all settings |
| 00:05.19 | bbourdage | just look at a Polycom config file, reg.1.server.1.address="x.x.x.x" , and replace that for now |
| 00:05.23 | bbourdage | Also check any MWI |
| 00:10.11 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 00:25.27 | *** part/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 00:25.53 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:25.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:42.07 | *** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com) |
| 00:44.33 | Micc | bbourdage, that just changes the registration server. I need to change the provisioning server where it checks for the cfg file. |
| 00:45.24 | Micc | I know how to make the phones register to a new asterisk server. Thats easy. |
| 00:46.20 | bbourdage | Micc, then use dhcp option 66 |
| 00:48.24 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:48.38 | Micc | I would, but that would be more work than remoting into each network and just changing the setting in each phone. |
| 00:49.18 | p3nguin | I think if there is a setting in the phone, it is going to override anything else anyway. |
| 00:49.57 | Micc | p3nguin, yes, I agree. I was just hoping there might be some kind of back door for this since I'm sure others have come across this before. |
| 00:50.56 | Micc | I'll see if I can take over the old IP with one of my servers. It's in the same data center, so they might be able to do that for me if they are decomissioning the whole server. |
| 00:51.08 | p3nguin | I think the idea is to change it only in the config and never set it on the phone. Change it in the config, reset the phone, now it pulls from the new server. |
| 00:51.39 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:52.08 | bbourdage | Micc, Ok, I do not see that it would be more work ? |
| 00:53.37 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:55.17 | bbourdage | Micc:How many phones, we have a solution that will do this automatically, but it takes a little setup |
| 01:03.35 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 01:09.51 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 01:13.09 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 01:27.47 | Micc | bbourdage, its about 50 phones, but all on different networks. |
| 01:27.50 | Micc | all around the country |
| 01:28.36 | bbourdage | Ok, for 50, I would provision, if they have different button assignments then for sure. |
| 01:30.12 | Micc | they are already on auto provision but the ftp server is set with a static ip address, which I was going to change, but I don't need to now. They aren't going to get rid of the machine for a while. |
| 01:30.19 | Micc | So I can still use it if I need to for a while. |
| 01:30.33 | Micc | so I don't need to worry about it anymore. |
| 01:31.39 | Micc | Now on to fun stuff. Asterisk 1.8.9.2 has been up and running about 2 weeks now with no problems. Same server running 1.6.2.20 was dead locking every day. |
| 01:32.13 | Micc | I'd kind of like to try 1.8.10 but I don't want to fix something thats not broken, but I'd like to have the udptl fix. |
| 01:32.52 | Micc | I don't really have time to apply my patches yet anyways. So I may just give it a week or two and see how it works for others. |
| 01:35.41 | Micc | You are not authorized to access this page. <--- when I clicked on asterisk available now 1.8.11.0-rc2 from asterisk.org |
| 01:35.48 | Micc | I don't even remember seeing an rc1 |
| 01:45.32 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 01:47.49 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:47.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:00.34 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 02:18.55 | GGD | question is there an archive of presentations from the last astricon? |
| 02:37.23 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 02:38.42 | *** join/#asterisk lordvadr (~lordvadr@2604:8800:117:4:224:d7ff:fe8e:eca8) |
| 02:39.42 | lordvadr | Hello. I've got an older asterisk 1.4 with a BUNCH of subscriptions to other extensions (BLF keys), and notifyringing=no isn't working (trying to cut down on the sip traffic). Am I doing something wrong; is this a known bug; anybody have any input? |
| 02:39.49 | lordvadr | I'm sorry, in sip.conf. |
| 02:40.28 | *** join/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 02:46.02 | *** join/#asterisk rue_house (~rue@h24-207-19-104.cst.dccnet.com) |
| 02:46.16 | rue_house | how can I kill a dahdi channel that isn't hung up? |
| 02:49.09 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 02:55.48 | volga629 | I am trying set konference through misc app, but have syntax problem in dial plan |
| 02:55.53 | volga629 | http://fpaste.org/SM6M/ |
| 02:56.10 | *** part/#asterisk rue_house (~rue@h24-207-19-104.cst.dccnet.com) |
| 02:56.11 | volga629 | this extension_custom.conf part |
| 02:57.22 | volga629 | and this the asterisk log http://fpaste.org/OTNE/ |
| 03:01.13 | volga629 | this line exten => s,n,ExecIF($["${FLAGS:${i}:1}" = "i"]|Set|announcenorev=1) |
| 03:01.45 | p3nguin | It's wrong. Fix it. |
| 03:02.06 | volga629 | asterisk version Asterisk 1.8.9.2 |
| 03:02.38 | p3nguin | exten => s,n,ExecIf($["${FLAGS:${i}:1}" = "i"]?Set(announcenorev=1)); <---- should be like this. |
| 03:03.42 | volga629 | I see extra | |
| 03:03.50 | p3nguin | What? |
| 03:06.10 | volga629 | Excellent thank you thank you it working I says that I was wrong in extra | |
| 03:06.39 | p3nguin | Using | is asterisk 1.2 syntax. |
| 03:07.40 | volga629 | It create dynamic conference room how possible add that it will add announcement of dynamically created room |
| 03:07.54 | volga629 | number |
| 03:12.50 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:17.22 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 03:20.50 | volga629 | ERROR[7984]: app_voicemail.c:2415 inboxcount2: Couldn't find mailbox 107 in context default |
| 03:22.43 | volga629 | I have another problem this IMAP voicemail storage I setup one test user on IMAP server zimbra and I see new voice messages delivered into folder, but error some place can't find |
| 03:30.59 | volga629 | http://fpaste.org/5JGr/ |
| 03:31.42 | volga629 | this IMAP config and logs when leave message and when check the message |
| 03:32.49 | p3nguin | The mail box configuration is in IMAP? |
| 03:33.10 | p3nguin | I've never used IMAP for Asterisk voice mail, so I don't know how it is configured without going to look it up for myself. |
| 03:34.30 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:34.44 | volga629 | yes |
| 03:35.02 | volga629 | pbx*CLI> module show like voice |
| 03:35.02 | volga629 | Module Description Use Count |
| 03:35.02 | volga629 | app_voicemail.so Comedian Mail (Voicemail System) with IM 0 |
| 03:35.02 | volga629 | 1 modules loaded |
| 03:47.22 | *** join/#asterisk radic (~radic@dslb-094-216-236-046.pools.arcor-ip.net) |
| 03:51.54 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 03:53.45 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 04:10.35 | *** join/#asterisk kobaz (~kobaz@cpe-67-240-79-210.nycap.res.rr.com) |
| 04:13.24 | kobaz | so i'm having a fun problem with trying to get a qualify=yes sip peer to be reachable |
| 04:13.36 | kobaz | asterisk is sending out invalid from headers in an OPTIONS message |
| 04:13.38 | kobaz | http://pastebin.com/Mjh8VYdC |
| 04:13.52 | kobaz | From: "Unavailable <0>" <sip:Unavailable <0>@10.3.1.58>;tag=as1cccb04e |
| 04:14.22 | kobaz | the endpoint (twinkle) is dropping the sip message because of the invalidness |
| 04:15.30 | volga629 | yes problem I am missing app_voicemail_imapstorage.c |
| 04:58.04 | *** join/#asterisk wheeler_32 (~wheeler_3@macauth-160-245.resnet.mtu.edu) |
| 05:26.47 | [TK]D-Fender | kobaz: Got actual SIP debug from * itself? |
| 05:34.44 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 05:35.36 | *** join/#asterisk luke0512 (~eric@HSI-KBW-091-089-189-101.hsi2.kabel-badenwuerttemberg.de) |
| 05:36.42 | luke0512 | hello |
| 05:37.07 | luke0512 | maybe today someone knows something, so i ask again |
| 05:37.09 | luke0512 | does someone know a howto for configuring CCD/Billion/Asuscom 2BD0 with hfc-s chip? i want to use asterisk behind a hardware pbx on the internal S0 |
| 05:37.41 | luke0512 | running asterisk 1.8 on centos6 |
| 05:37.52 | luke0512 | kernel 2.6.32 |
| 05:38.04 | *** join/#asterisk Vince-0 (~AndChat@196.215.188.51) |
| 05:41.59 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 05:45.17 | *** part/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 05:52.20 | *** join/#asterisk hfb (~hfb@cpe-98-151-249-95.socal.res.rr.com) |
| 05:53.04 | *** join/#asterisk reenigne_esrever (~reenignee@static-50-53-151-158.bvtn.or.frontiernet.net) |
| 05:54.08 | reenigne_esrever | hello, I just wanted to jump on and see if anyone was thinking of attending the Digium classes in Vegas in April? I'm also curious if anyone has been to any of the classes provided by digium and had any opinions about the quality of the class. |
| 05:55.58 | reenigne_esrever | quiet room tonight |
| 06:01.55 | Henchman21 | never been |
| 06:02.33 | Henchman21 | i'd be gambling not learning |
| 06:02.37 | reenigne_esrever | thanks Henchman for responding was beginning to wonder if I was alone in here |
| 06:03.02 | reenigne_esrever | Of course there will be gambling and drinking, it is Vegas after all |
| 06:04.14 | volga629 | hasvoicemail = yes is this valid option for user ? |
| 06:08.36 | [TK]D-Fender | volga629: What does the sample config say? |
| 06:09.45 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 06:11.14 | volga629 | than config file per user |
| 06:13.20 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:14.36 | volga629 | I am just trying understand why get error about IMAP, on default content |
| 06:14.40 | volga629 | <PROTECTED> |
| 06:16.45 | kaldemar | volga629: what does your voicemail.conf look like? |
| 06:18.50 | volga629 | http://fpaste.org/0n7I/ |
| 06:19.47 | luke0512 | well as i see, in here there are only us people? |
| 06:20.32 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-yrcbeibxvtyyolyf) |
| 06:20.47 | kaldemar | us people? |
| 06:22.11 | kaldemar | volga629: something is looking for 101 in a context called "device". yours is called "default" unless you have something in those included files. |
| 06:22.27 | luke0512 | user from UnitedStates |
| 06:22.53 | kaldemar | luke0512: there are people outside the US. :) |
| 06:23.04 | luke0512 | ok :) |
| 06:23.45 | luke0512 | i only read about tonight and when theres evening it could not be europe ;-) |
| 06:27.26 | volga629 | <kaldemar> mailbox parameter look like wrong let me check config |
| 06:30.15 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 06:34.19 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:35.56 | phix | luke0512: I know stuff |
| 06:37.04 | phix | [TK]D-Fender: Hey!!! how are you this evening? |
| 06:37.29 | [TK]D-Fender | Meh |
| 06:37.57 | phix | heh, thnx for your assistance the other day! I figured it all out |
| 06:38.35 | phix | the problem was my router had ingress filtering on it (wouldn't allow port forwarding or connections to external IP from an internal IP) |
| 06:38.42 | phix | even though I disabled the firewall it still kept that |
| 06:39.07 | phix | so the extern_ip and host was being applied tothe internal snom handsets, and so wasn't hearing anything as the router was dropping it |
| 06:41.29 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:43.22 | [TK]D-Fender | I never hear about smart devices doing anything other that STUPID things. |
| 06:43.28 | [TK]D-Fender | than* |
| 06:48.41 | Mattybear | Quick question: Can T38 operate over IAX2 ? |
| 06:48.52 | [TK]D-Fender | no |
| 06:49.11 | Mattybear | Cool, thanks |
| 06:50.48 | volga629 | IMAP Warning: Unknown message data: 2 FETCH what is mean |
| 07:04.04 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 07:09.14 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 07:10.49 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 07:19.06 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:19.08 | schmidts | good morning |
| 07:19.51 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:25.25 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 07:31.52 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:40.04 | luke0512 | hmmm...seems that nobody knows a howto for my prob |
| 07:40.39 | *** join/#asterisk chasing`Sol (~cS@197.132.36.67) |
| 07:46.32 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:46.39 | v0lZy | hi |
| 07:51.06 | jacc0 | morning |
| 07:51.40 | jacc0 | Set(TIMEOUT(absolute)=300) |
| 07:51.51 | jacc0 | will disconnect the line after 5min. ? |
| 07:53.22 | jacc0 | because I had some discussion about this some weeks ago |
| 07:54.06 | jacc0 | it seems to work |
| 07:54.09 | jacc0 | :) |
| 07:54.35 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:59.28 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:59.46 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:59.52 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 08:00.43 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:08.18 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 08:14.41 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 08:15.58 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:34.19 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 08:34.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:36.44 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:53.54 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 08:54.56 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 08:57.00 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 08:57.43 | *** join/#asterisk datarecal (~data@loxely.illusivecreations.com) |
| 08:59.37 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:00.39 | aberrios | is it possible the function CURL would cause a soft hangup if the page it requested returned nothing? |
| 09:07.49 | aberrios | Having a strange probblem where after a Set(CURL etc.. the channel hangs up... I'll try n pastebin some debug info |
| 09:11.06 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:12.47 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 09:13.46 | aberrios | http://pastebin.com/RPmeWtfZ |
| 09:13.48 | aberrios | any ideas? |
| 09:14.21 | aberrios | ill pastbin dialplan too |
| 09:14.53 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 09:16.27 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 09:17.59 | aberrios | http://pastebin.com/pADxcYXZ |
| 09:18.07 | aberrios | ^ dialplan + Debug output |
| 09:22.57 | jacc0 | aberrios: isn't the call hung-up with softhangup(${CHANNEL}) from some other channel? or via AMI/CLI |
| 09:23.35 | Dovid | [Mar 6 16:36:17] DEBUG[11534] pbx.c: Extension s, priority 7 returned normally even though call was hung up |
| 09:23.39 | Dovid | seems like something is hanging it up |
| 09:24.13 | aberrios | jacc0, Dovid. I'll have a check around but I can't possibly think what would be doing that |
| 09:25.16 | jacc0 | what type of phone are you using? sip ? cisco? |
| 09:25.28 | aberrios | Polycom Soundpoint IP 331 SIP |
| 09:25.33 | jacc0 | maybe it has a max ringtime set |
| 09:25.41 | aberrios | they're not ringing at that point |
| 09:25.53 | jacc0 | okay, but there is no answer() |
| 09:25.59 | aberrios | aha! |
| 09:26.10 | aberrios | burrdurrrr <hits head> |
| 09:26.51 | Dovid | look in the full log. it will show a lot more than cli (if u r using cli) also try usind debug 15 so u see much more |
| 09:27.02 | aberrios | Dovid, that was from the full log. |
| 09:27.16 | aberrios | think i had debug @ 10 at that point |
| 09:32.53 | aberrios | jacc0, at the risk of counting chickens.... i think that's fixed it.. thanks for the extra pair of eyes. |
| 09:37.10 | Dovid | it's showing the call hung up. maybe add iax2 debug to see where the hangup happens and who initiates it? |
| 09:37.18 | Dovid | also try verbose and debug @ the max of 15 |
| 09:38.04 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 09:39.05 | Dovid | it's not saying who hung up and why which is what we need to know |
| 09:40.22 | aberrios | Dovid, I think as jacc0 pointed it out the channel was never Answer() 'd properly |
| 09:49.44 | v0lZy | hey guys |
| 09:49.48 | v0lZy | how do i debug voicemail? |
| 09:53.06 | *** join/#asterisk gain_ (~gain@mail.ufficyo.com) |
| 09:57.30 | v0lZy | never mind |
| 10:03.23 | *** join/#asterisk jozza (~chatzilla@BSN-77-186-224.dsl.siol.net) |
| 10:03.36 | jozza | hi what does AST_CONTROL_SRCUPDATE mean |
| 10:04.15 | jozza | i see this that a sip client requested special control 20 |
| 10:05.37 | jozza | this is sent from caller and then passed to the callee |
| 10:08.38 | jozza | requested special control 20, passing it to SIP |
| 10:10.23 | jozza | anyone? |
| 10:16.31 | *** join/#asterisk ziz212 (3df5a830@gateway/web/freenode/ip.61.245.168.48) |
| 10:17.32 | *** part/#asterisk ziz212 (3df5a830@gateway/web/freenode/ip.61.245.168.48) |
| 10:26.18 | v0lZy | anyone have any ideas how to disable email notifications in asterisk? |
| 10:32.15 | kaldemar | v0lZy: don't define a mail address or define mailcmd to be some dummy script that doesn't do anything. |
| 10:32.47 | v0lZy | so if i get it correctly |
| 10:32.53 | v0lZy | its gonna run the script |
| 10:32.55 | v0lZy | regardless? |
| 10:35.33 | kaldemar | regardless of what? |
| 10:35.52 | v0lZy | regardless of weather there's an email address or not |
| 10:37.24 | v0lZy | or weather the script exists or not |
| 10:38.34 | kaldemar | why do you think so? |
| 10:40.20 | *** join/#asterisk mtbf (~ewilded@n0life.pl) |
| 10:46.36 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:46.44 | v0lZy | well.. cause my webui is prompting me to input something |
| 10:46.53 | v0lZy | and doesnt let me leave it blank |
| 10:47.05 | v0lZy | and u also said i should either make the email address blank |
| 10:47.18 | v0lZy | or define mailcmd to be some dummy script |
| 10:47.28 | v0lZy | that would still be running it, it just wouldnt do anything, right? |
| 10:47.43 | v0lZy | i was hoping i can do it on a per user basis.. some users receiving email, some not |
| 10:47.52 | v0lZy | so i guess ill be deleting the email addresses |
| 10:48.32 | mtbf | Hi. Yesterday I've encountered some weird SPIT related issue (asterisk 1.4.42), clients were claiming they started to receive spam calls, I took a look into messages log and the only suspected thing I found out was this http://pastie.org/3540182 . What is weird to me is the fact that IP (85.17.x.x) appeared in the logs for the first time in the supplied pasted part. There were no extension and password enumeration attempts before and so on.. what could this be t |
| 10:49.43 | kaldemar | v0lZy: i talk about asterisk and voicemail.conf. if you want to know what do to with your GUI, ask the people who made it or support it. i.e. not me. |
| 10:50.02 | v0lZy | yeah thats why im asking |
| 10:50.08 | v0lZy | my gui allows me to edit config files too |
| 10:50.21 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:51.18 | kaldemar | mtbf: your message got cut at "could this be t" |
| 10:52.26 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 10:52.34 | kaldemar | mtbf: do you have anonymous guest calls rejected (allowguest=no in sip.conf)? |
| 10:53.23 | mtbf | kaldemar: oh, it was just 'hen?' part from 'then?' that was cut off, thanks. |
| 10:55.12 | mtbf | It's set to default, which appears to be 'yes', thank you very much, I hope this will do the trick and I'll learn something about guest calls. |
| 10:56.41 | aberrios | mtbf, very important it is off on boxes that are exposed to the net on port 5060. Toll fraud can hurt. A lot. |
| 10:57.20 | kaldemar | mtbf: also read this if you have not: http://svn.digium.com/svn/asterisk/tags/10.2.0/README-SERIOUSLY.bestpractices.txt |
| 10:59.38 | mtbf | Ok, thanks a lot, guys. |
| 11:17.34 | *** join/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 11:17.39 | hurdman | hi |
| 11:18.08 | *** join/#asterisk albertoandrade (~albertoan@177.16.137.198) |
| 11:18.30 | hurdman | I'm back with my TE410P: RECEIVE slip POSITIVE on span 1 and 2 and 3 and 4 |
| 11:19.01 | hurdman | i ever get that |
| 11:19.35 | hurdman | now, i have change my conf, i provide the clock on span 1 , 2 , 3 and 4 to my others server |
| 11:19.47 | hurdman | and i get the clock from providers from the over span |
| 11:20.10 | hurdman | but i ever get this error ONLY on spans 1 , 2 , 3 and 4 |
| 11:20.47 | hurdman | is-it an hardware problem ? |
| 11:21.42 | kaldemar | how is the other server configured? |
| 11:22.49 | hurdman | they get the clock from their span |
| 11:23.06 | hurdman | kaldemar: i had the same error when my providers was on span 1 , 2 ,3 and 4 |
| 11:23.10 | hurdman | :/ |
| 11:34.34 | gajini | Hi, please help to install dahdi-2.1.0.2 on opensuse-11.3. I got compilation error |
| 11:51.18 | hurdman | gajini: pastbin your error ? |
| 11:52.29 | gajini | hurdman- The error i got is http://pastebin.com/suiYnqWX |
| 11:57.19 | hurdman | gajini: how do you make your dahdi ? |
| 11:58.59 | gajini | hurdman- i using tor3e module with dahdi, i give, make; make MODULES_EXTRA="tor3e" install commands |
| 12:05.05 | *** part/#asterisk gajini (~root@61.12.17.171) |
| 12:11.45 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 12:23.51 | puzzled | gajini: why don't you use the latest dahdi which is 2.6.0? |
| 12:26.16 | puzzled | oh it's a Govarion driver. that company has been out of business for years. good luck with that |
| 12:35.46 | kaldemar | hurdman: is your problematic card sharing an interrupt with another device that has activity? |
| 12:43.51 | hurdman | kaldemar: it seems , i have made a mistake, i haven't seen that into my dmesg, the te410p module said always span 1 / 2 / 3 / 4 but with different pci adress .... |
| 12:44.28 | hurdman | so the error should be into my "client" conf or something like that |
| 12:46.17 | *** part/#asterisk gajini (~root@61.12.17.171) |
| 12:47.18 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 12:53.05 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 12:57.13 | WIMPy | Hi hurdman. Still on the timing issue? |
| 12:57.28 | hurdman | WIMPy: yes , ever |
| 12:57.45 | hurdman | i have split my different providers/client into 3 cards |
| 12:57.46 | WIMPy | Looks like I didn't get the full picture last time. How many connections do you have? |
| 12:57.51 | hurdman | remove the timing cables |
| 12:57.55 | hurdman | but ever the same issue |
| 12:58.17 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 12:58.20 | cusco | hi |
| 12:58.22 | cusco | quick question |
| 12:58.34 | *** join/#asterisk C-S-B (~C-S-B-Wor@fwe.office.community.net.uk) |
| 12:58.40 | cusco | whats the most reliable way to read dtmf in 1.6 |
| 12:58.46 | hurdman | WIMPy: i have 3 cards with 4 port |
| 12:58.48 | cusco | dtmf is comming from dahdi in alaw |
| 12:59.05 | WIMPy | Hmm. So the cards syncronize even without the cable? |
| 12:59.17 | hurdman | it semms yes |
| 12:59.26 | kaldemar | cusco: be more elaborate on "from dahdi" |
| 13:00.15 | cusco | kaldemar: actually I have a asterisk box separated with digium PRI cards, and another asterisk gets the call via sip. In this second asterisk I'm reading dtmf |
| 13:00.25 | cusco | I can read them but if I press keys too quickly it skips some |
| 13:01.21 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:01.23 | hurdman | cusco: perhaps you can use a sip info packet to signalling dtmf |
| 13:02.22 | kaldemar | cusco: your explanation of the setup is all over the place. so two asterisk boxes, connected with PRI, right? one of them gets a call via SIP (from where and what?) and you want it's DTMF to be relayed over PRI. how are you verifying the DTMF's in the receiving asterisk box? |
| 13:02.38 | C-S-B | Hello, I hope you can help. i've been lumped with resolving an issue on a legacy asterisk install. There are 2 servers connected by iax. the phones connect using sip. when I call the number I get hold music or a couple of rings then the call drops. http://pastebin.com/fxWPEGJq |
| 13:02.50 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 13:03.24 | kaldemar | cusco: actually, what you should do is put "dtmf" on the console line in logger conf in both boxes and look at both CLI's when you press the buttons to see what is the problematic leg. |
| 13:03.26 | cusco | pstn -> pri box --sip--> main asterisk |
| 13:03.49 | cusco | k I'll check |
| 13:06.10 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:10.15 | *** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it) |
| 13:12.14 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.57.251) |
| 13:13.18 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:31.17 | Dovid | is there anyone from Russia here? |
| 13:34.27 | hurdman | re |
| 13:34.39 | hurdman | kaldemar: perhaps i have someting strange |
| 13:34.56 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.113.214) |
| 13:34.56 | hurdman | kaldemar: if i make " cat /proc/dahdi/* | grep -i span " |
| 13:35.35 | hurdman | kaldemar: on the server that's fail , i haven't any "ClockSource" line |
| 13:36.47 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:37.02 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fafdihsqfpsbfibs) |
| 13:37.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:39.17 | WIMPy | hurdman: None is quite little. |
| 13:39.29 | WIMPy | Have you tried Digiums Support? |
| 13:41.19 | hurdman | WIMPy: i try google-support before :=) |
| 13:44.21 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:44.28 | *** join/#asterisk jetlag (~jetlag@pool-71-168-250-69.cmdnnj.east.verizon.net) |
| 13:50.25 | *** join/#asterisk Li7h (~dps@71-81-21-14.static.gwnt.ga.charter.com) |
| 13:55.37 | *** part/#asterisk C-S-B (~C-S-B-Wor@fwe.office.community.net.uk) |
| 13:59.18 | volga629 | kaldemar: Some thing mess around with mailbox it says 101@default and vm content default, but still have the error. I wonder if it is because IMAP is not enabled for every extension ? |
| 14:05.49 | volga629 | Couldn't find mailbox 101 in context default |
| 14:06.10 | *** join/#asterisk serafie1 (~erin@nat/digium/x-zqqtmclsfuqzoyio) |
| 14:16.10 | *** join/#asterisk mhogan (~MatthewHo@63.97.74.114) |
| 14:19.00 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:19.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:20.02 | mhogan | morning, I have a sangoma a108 in an asterisk server. the card is no longer in use and I want to remove it. Can someone tell me what I need to do so I dont freak asterisk out? |
| 14:21.36 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 14:28.25 | WIMPy | mhogan: Nothing. |
| 14:28.38 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 14:28.43 | *** join/#asterisk Srini (~Srinivasa@115.115.80.121) |
| 14:28.54 | WIMPy | If the configured channels don;t match the existing ones, I guess chan_dahdi will even refuse to load. |
| 14:28.54 | Srini | Hi Room |
| 14:28.56 | jacc0 | what about timing? |
| 14:29.06 | jacc0 | modprobe adhdi-dummy ? |
| 14:29.13 | jacc0 | *dahdi-dummy |
| 14:29.34 | mhogan | considered renaming the dadhi module so it wouldn't load |
| 14:29.37 | jacc0 | depanding on the dahdi version |
| 14:29.41 | jacc0 | remove the drivers and modprobe dahdi-dummy ande remove womera/wanpipe modules&configfiles from asterisk |
| 14:29.44 | WIMPy | That has been included in to the main module. |
| 14:29.55 | Srini | While trying to configure a SIP carrier, I am getting an error: add_realm_authentication: Format for authentication entry is user[:secret]@realm, Bur I have the registration string formed the right way! |
| 14:30.42 | WIMPy | Sure, you can remove the drivers. But if you use MeetMe, don't remove dahdi. |
| 14:30.59 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-jmvkayfvpxsoljwu) |
| 14:31.47 | jacc0 | yeah. yo should keep dahdi and modprobe dahdi-dummy for timing |
| 14:32.09 | jacc0 | (asterisk internel timing modules suck/crash) |
| 14:32.32 | mhogan | ok, thanks. will give it a go. |
| 14:32.54 | kaldemar | no need to modprobe dahdi_dummy in recent dahdi versions. |
| 14:32.58 | jacc0 | true |
| 14:37.14 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:37.16 | [TK]D-Fender | mhogan, ou want DAHDI. Just disable your channels |
| 14:38.05 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 14:38.18 | Rico29 | hi |
| 14:39.34 | Rico29 | I'm trying to get the channels name when I do an incoming call, from a cellphone to a sip peer. in this case I have two channels : one from my provider to my *, and one from my * to my phone |
| 14:40.04 | Rico29 | how can I retreive these channels name during a call ? if ther is a way to do it |
| 14:42.31 | kaldemar | Rico29: where are you trying to retrieve the channel names from? AMI? dialplan? something else? |
| 14:42.46 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 14:43.30 | Rico29 | kaldemar, actually I'm tryng from anywhere |
| 14:43.47 | jrose_atDigium | From dialplan: ${CHANNEL} |
| 14:43.59 | Rico29 | I just saw something interesting here : ${BRIDGEPEER} |
| 14:44.16 | kaldemar | i won't certainly give you a lecture on all the possible screnarios and means. how do you want to do it? |
| 14:45.25 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 14:46.10 | Rico29 | what I want to do : i.e : a user is in conversation with a cellhpone (via a sip trunk provider). Hé clicks on a web page button, which tells the manager ton call another cellphone. When the cellphone is answered, I bridge the two channels : the channel from my sip provider and the new channel created by the manager |
| 14:46.43 | Rico29 | now my user can leave his work, the conversation is on his cellphone |
| 14:46.55 | Rico29 | and the over user didn't see anything |
| 14:47.22 | Rico29 | did I explained correctly ? |
| 14:47.37 | jrose_atDigium | So this is some kind of gui via AMI? |
| 14:48.22 | Rico29 | jrose_atDigium, > GUI is not the problem |
| 14:48.31 | Rico29 | the problem is to get the channels name |
| 14:49.35 | Rico29 | ok |
| 14:49.37 | jrose_atDigium | Well, the funny thing about getting variables from a channel via AMI is that you need to know the channel's name to send the command in the first place... |
| 14:49.45 | [TK]D-Fender | Rico29, Read the AMI command list. There is one that lists channels. |
| 14:49.57 | Rico29 | i've just find a way : |
| 14:50.37 | Rico29 | first approach : asterisk -rx "core show channel SIP/105-00037b13" | grep BRIDGEPEER returns the provider side channel name (BRIDGEPEER=SIP/trunk-00037b03) |
| 14:51.27 | kaldemar | Rico29: CoreShowChannels lists all channels and even a field that tells what the channel is bridged to. |
| 14:51.40 | Rico29 | ok |
| 14:51.47 | Rico29 | trying right now |
| 14:52.57 | Rico29 | ok |
| 14:53.37 | Rico29 | too bad there is no way to pass a regex for this command in AMI |
| 14:54.06 | Rico29 | but It will do the trick |
| 14:55.16 | *** join/#asterisk rue_house (~rue@h24-207-19-104.cst.dccnet.com) |
| 14:55.28 | rue_house | is visual dialplan available for linux? |
| 14:55.30 | Rico29 | thanks all |
| 14:55.45 | rue_house | has anyone ever written a dialplan for doing voicemail? |
| 14:56.54 | kaldemar | ~ask |
| 14:56.55 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:57.23 | rue_house | sorry were my questions not clear? |
| 14:57.58 | rue_house | is the program "visual dialplan" available for the linux operating system to the knowledge of anyone here and listening and answering? |
| 14:59.08 | rue_house | is it to the knowledge of the people listening ans answering questions if anyone documented the re-implementation of voicemail in an extension configuration format for the purpose of customization? |
| 15:00.35 | *** join/#asterisk serafie1 (~erin@nat/digium/x-czzhcenbaxieyicp) |
| 15:01.03 | kaldemar | rue_house: for the visual dialplan question, ask the apstel people. for the voicemail question, be specific on what you want. |
| 15:02.04 | kaldemar | rue_house: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-Voicemail.html |
| 15:03.28 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 15:05.52 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.113.214) |
| 15:07.21 | *** join/#asterisk jkroon (~jkroon@197.173.189.135) |
| 15:10.21 | rue_house | I'd like a copy of extention programming to do voicemail, I wasn't aware the whole thing was in C till last night, the C is so gooped up with other languages and systems that, it was my oppinion, that if I was gonna modify it I'd have to give it a complete overhaul |
| 15:10.44 | mhogan | [TK]D-Fender: can I just rename chan_dahdi.so in modules? to disable the channels? |
| 15:10.59 | rue_house | how about just dont load the module? |
| 15:12.19 | *** join/#asterisk af_ (~getsmart@78.134.18.236) |
| 15:12.42 | kaldemar | rue_house: what on earth are you talking about? extensions have nothing to do with C. |
| 15:13.10 | [TK]D-Fender | mhogan, No. change your CONFIGS |
| 15:13.24 | rue_house | I'm asking if anyone known of has implemented voicemail.c in extension programming |
| 15:13.55 | [TK]D-Fender | rue_house, If the answer is "yes", then what's your next question? |
| 15:14.28 | rue_house | where is a copy :) |
| 15:14.39 | [TK]D-Fender | rue_house, Write it yourself :p |
| 15:14.42 | kaldemar | rue_house: search for Minivm if you want flexibility for voicemail. |
| 15:15.15 | rue_house | there are a few things about voicemail.c s operation that bug a lot of people |
| 15:17.34 | rue_house | seeing the source for the voicemail last night made up for not seeing a horror movie in the last 3 years tho.... but C just gets like that |
| 15:17.54 | *** join/#asterisk PhiBrick (~phil@212.80.245.102) |
| 15:19.34 | *** join/#asterisk jmartinez916 (~jmartinez@mail.fronteraproduce.com) |
| 15:21.31 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:25.17 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 15:32.07 | *** part/#asterisk mhogan (~MatthewHo@63.97.74.114) |
| 15:34.51 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:34.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:38.42 | anonymouz666 | why the 1.8.11 started with rc2 instead rc1? |
| 15:38.49 | anonymouz666 | something changed? |
| 15:40.27 | jacc0 | on asterisk.org click `Asterisk 1.8.11.0-rc2 Now Available` -> You are not authorized to access this page. |
| 15:41.13 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:43.29 | malcolmd | anonymouz666: there was a 1.8.11.0-rc1 tag (http://svnview.digium.com/svn/asterisk/tags/1.8.11.0-rc1/). tarball was skipped |
| 15:44.17 | malcolmd | jacc0: try again |
| 15:47.11 | *** join/#asterisk navaismo (~navaismo@189.230.118.84) |
| 15:48.07 | jacc0 | it works now :0 |
| 15:48.43 | *** part/#asterisk Srini (~Srinivasa@115.115.80.121) |
| 15:49.04 | Qwell | anonymouz666: I said Monday it would be dropped on the floor when you asked. :( |
| 15:52.55 | anonymouz666 | very good fixes on rc2 |
| 15:59.13 | *** join/#asterisk blindkangaroo (~Bryan_Ham@dsl-173-248-245-65.acanac.net) |
| 15:59.56 | blindkangaroo | need help with a Cisco SPA504g with attendant console |
| 16:00.24 | blindkangaroo | caller ID is not showing when transferring calls |
| 16:00.58 | [TK]D-Fender | bliSo you see nothing at all? |
| 16:01.07 | [TK]D-Fender | blindkangaroo, So you see nothing at all? |
| 16:01.20 | blindkangaroo | no it shows the reception extension |
| 16:01.28 | blindkangaroo | I need it to show the original caller ID |
| 16:01.57 | sp00kz | does it if you blind forward? |
| 16:02.06 | blindkangaroo | hmmm |
| 16:02.08 | blindkangaroo | not sure |
| 16:02.11 | blindkangaroo | haven't tried |
| 16:02.46 | blindkangaroo | I will try it now |
| 16:06.20 | blindkangaroo | so NOT using the attendant console the callerID is passed along |
| 16:06.40 | blindkangaroo | so hitting <xfer>extension<xfer> |
| 16:07.00 | sp00kz | that's because you're making a call from internal -> internal, and then splicing a previous call into it |
| 16:07.02 | blindkangaroo | but hitting <xfer><button on console> does NOT |
| 16:07.10 | blindkangaroo | okay |
| 16:07.12 | blindkangaroo | makes sense |
| 16:07.36 | blindkangaroo | so how can I do the same thing with the attendant console |
| 16:07.56 | sp00kz | I don't know the answer to that, sorry =( |
| 16:08.06 | [TK]D-Fender | blindkangaroo, Read your phone's manual. You should have an option to specify a blind transfer BEFORE pressing the button |
| 16:08.19 | [TK]D-Fender | blindkangaroo, It should be apparent on the soft-keys |
| 16:08.24 | blindkangaroo | so I'm looking for blind transfer .... |
| 16:09.12 | blindkangaroo | is anyone familiar with the SPA504G? |
| 16:09.18 | sp00kz | the option best suited to show the external caller-id to the internal user who got the call 2nd |
| 16:09.33 | sp00kz | may be to not transfer the call initially, and make the call from user->user to tell them whos calling |
| 16:09.35 | sp00kz | hang up |
| 16:09.37 | sp00kz | and then blind it |
| 16:10.24 | blindkangaroo | okay that's the way we were doing it before I installed the attendant console |
| 16:10.33 | blindkangaroo | so is this an Asterisk configuration |
| 16:10.36 | blindkangaroo | or phone config? |
| 16:10.41 | sp00kz | user training |
| 16:10.44 | sp00kz | ;) |
| 16:11.00 | [TK]D-Fender | Not "config". Basic phone usage |
| 16:11.06 | blindkangaroo | but if I it <xfer>+button on console |
| 16:11.11 | [TK]D-Fender | LOOK at the screen |
| 16:11.13 | blindkangaroo | it transfers the call right away |
| 16:11.21 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 16:11.51 | sp00kz | answer inbound call; put on hold; call end-user; hang up with end user; resume inbound call, xfer to end-user |
| 16:13.11 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 16:14.48 | blindkangaroo | management want's to see the callerID come through |
| 16:14.56 | blindkangaroo | it's not a matter of user training |
| 16:15.08 | blindkangaroo | it's a matter of the man who signs my paycheque |
| 16:15.11 | blindkangaroo | :) |
| 16:15.20 | blindkangaroo | so is there a way? |
| 16:15.26 | blindkangaroo | do I check my phone config? |
| 16:16.18 | *** join/#asterisk timahvo1 (~rogue@197.177.44.153) |
| 16:18.24 | [TK]D-Fender | blindkangaroo, No, this is basic usage on the phone itself |
| 16:19.41 | blindkangaroo | okay I think I see what's happening |
| 16:20.05 | blindkangaroo | really what I want the reception to do is a blind transfer |
| 16:20.12 | blindkangaroo | so I'll just configure the phone that way |
| 16:20.33 | sp00kz | what we've described is a manual, two step proccess |
| 16:20.57 | blindkangaroo | yea |
| 16:20.58 | sp00kz | if you configure the phone to blind automatically, what's the point of someone else answering the phone in the first place |
| 16:21.07 | blindkangaroo | but really I need a one step process |
| 16:22.33 | [TK]D-Fender | bliThere is a quick setting to default it to blind |
| 16:22.39 | [TK]D-Fender | blindkangaroo, There is a quick setting to default it to blind |
| 16:22.42 | blindkangaroo | the person answering the phone is really only there to categorize the call and transfer to the correct department. letting the caller choose where to call and placing them in a queue is something that management doesn't want |
| 16:23.42 | sp00kz | just note that doing the auto-blind will prevent the receptionist from talking to the end-user before connecting the inbound call =) |
| 16:24.01 | blindkangaroo | yes ... that is what I want |
| 16:24.20 | blindkangaroo | she's just there to classify the call and take a message if the person is on the phone |
| 16:24.34 | blindkangaroo | we all sit a max of 60 feet from the reception |
| 16:28.51 | blindkangaroo | if I may ask an unrelated question .... |
| 16:29.11 | blindkangaroo | at what point should I stop using a hosted asterisk system and bring my own PBX in house? |
| 16:29.46 | ChannelZ | at the point where it's more expensive/less convenient for you to have it away |
| 16:30.11 | chuckf | blindkangaroo: when you can manage it yourself and you feel confident in your infrastructure's redundancy |
| 16:30.30 | blindkangaroo | hmmm, we've been using a hosted PBX for a few years now |
| 16:30.44 | *** join/#asterisk teratoma (~teratoma@i.dont.get.mad.i.get.stabby.net) |
| 16:30.47 | blindkangaroo | I'm really a network/systems guy not telco |
| 16:30.53 | [TK]D-Fender | blindkangaroo, Why did you choose that in the first place and what do you think the difference is? |
| 16:31.04 | blindkangaroo | I didn't choose it |
| 16:31.11 | blindkangaroo | I've only been here 4 months |
| 16:31.21 | blindkangaroo | this was the previous admins choice |
| 16:31.23 | [TK]D-Fender | Why di THEy choose it then. |
| 16:31.25 | blindkangaroo | not that it is a bad one |
| 16:31.51 | ChannelZ | Damn they! |
| 16:31.51 | blindkangaroo | they choose it becacuse they were told we're using VoIP phone lines so there is not Long distance |
| 16:32.05 | blindkangaroo | I know I can get the same thing installed here |
| 16:32.12 | blindkangaroo | just trying to figure the cost |
| 16:32.37 | blindkangaroo | like how much does a "VoIP" phone line (what I can tell is a SIP trunk) cost |
| 16:32.55 | blindkangaroo | because between two companies we've used I'm not happy with the service |
| 16:33.05 | blindkangaroo | I'd rather have my own PBX with a support contract |
| 16:35.51 | [TK]D-Fender | blindkangaroo, les.net , voip.ms are both pretty decent and offer Canadian DID's |
| 16:36.53 | *** join/#asterisk PhiBrick (~phil@212.80.245.102) |
| 16:39.57 | *** join/#asterisk blindkangaroo (~Bryan_Ham@dsl-173-248-201-159.acanac.net) |
| 16:40.01 | blindkangaroo | sorry ... having network problems |
| 16:40.31 | [TK]D-Fender | blindkangaroo, les.net , voip.ms are both pretty decent and offer Canadian DID's |
| 16:40.34 | blindkangaroo | I missed anything anyone send for the last few minutes |
| 16:40.46 | *** join/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
| 16:41.01 | blindkangaroo | so is what I want a SIP trunk for outgoing calls? |
| 16:42.36 | Flyingbull | Good day everyone, quick question. I'm working on my asterisk system, and out of the blue my queues stop functioning. When I try to unload or reload the queue module, it errors out saying it can't load. Can someone point me to the right direction fo find the problem. I was working on the exentions and the sip configurations when this happened, so I put them back to their orignal state |
| 16:42.37 | Flyingbull | and I'm still having problems. Any Ideas? |
| 16:43.35 | Flyingbull | I've also tried reloading the asterisk install to see if that would fix it, but nothing. I'm thinking the problem isn't in the queues, but somewhere else, and it happens to be affecting the queues. |
| 16:44.09 | [TK]D-Fender | bliFirst, ditch the word "trunk" When you want inbound, those come in on DID's |
| 16:44.10 | [TK]D-Fender | ~did |
| 16:44.10 | infobot | i guess did is Direct Inward Dialing, or just a phone number |
| 16:44.31 | blindkangaroo | yea a DID is incoming |
| 16:44.42 | blindkangaroo | so what is outgoing? |
| 16:45.03 | [TK]D-Fender | blindkangaroo, You may pay per-channel, per minute, etc all based on your plan and usage. Outbound is the same. Some services are flat-rate, others metered. Some limited in how many simultaneous calls you have, others not. |
| 16:45.16 | [TK]D-Fender | outgoing is just "termination" and doesn't have an implied "number" really |
| 16:45.39 | blindkangaroo | okay this is all new to me |
| 16:46.01 | blindkangaroo | the last two PBX's I was running one was centrex lines and the other was a T1 |
| 16:46.02 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 16:46.15 | blindkangaroo | so I'm not even familiar with what get's installed |
| 16:46.28 | chuckf | blindkangaroo: it sounds to me that you're not ready to bring the pbx inhouse |
| 16:46.50 | blindkangaroo | checkf: probably not.... but I want to get informed |
| 16:46.52 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 16:47.03 | [TK]D-Fender | blindkangaroo, Read THE BOOK for a little bit just to get caught up on some basic concepts : |
| 16:47.07 | [TK]D-Fender | ~book |
| 16:47.07 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:47.47 | blindkangaroo | this will cover the basics of the telco side of things also? |
| 16:47.54 | [TK]D-Fender | a chunk of it |
| 16:48.39 | blindkangaroo | Fender: good! thanks .... I need to know where to start |
| 16:48.53 | blindkangaroo | this is new tech for me and I really am a systems guys not a telco guy |
| 16:50.12 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-xeexsceyjmnzcfim) |
| 16:50.26 | blindkangaroo | thank-you all very much for your help today |
| 16:50.32 | blindkangaroo | this is why I love open source |
| 16:53.08 | *** part/#asterisk blindkangaroo (~Bryan_Ham@dsl-173-248-201-159.acanac.net) |
| 16:53.49 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-375536.home.otenet.gr) |
| 16:56.53 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 16:58.49 | Flyingbull | so any ideas? |
| 16:59.13 | Qwell | Flyingbull: You haven't shown us the error. |
| 16:59.23 | Qwell | "it can't load" isn't an error that exists anywhere in Asterisk, or app_queue |
| 17:00.42 | Flyingbull | Ok, well that is all it says really. I'm going to try to find the logs and see if it will show me more detail, but all I'm getting is "fails to load" from the module load command. |
| 17:02.07 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 17:02.17 | raden | where has naikorvek been ? |
| 17:02.59 | p3nguin | AWOL |
| 17:03.02 | Qwell | -NickServ- Last seen : Mar 06 00:52:06 2012 (1 day, 16:10:50 ago) |
| 17:03.08 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 17:04.28 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 17:06.25 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 17:18.55 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a006b.pool.mediaWays.net) |
| 17:19.22 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:39.15 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 17:47.25 | *** join/#asterisk cyborg-one (1000@212-178-0-68.broadband.tenet.odessa.ua) |
| 17:49.48 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-vnyyjdsgvrinlxbg) |
| 17:51.50 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 18:16.10 | *** join/#asterisk kjs (kjs@fedora/kjs) |
| 18:16.33 | kjs | Are the Asterisk 10 centos 6 RPM's in the repo yet? |
| 18:17.20 | Qwell | no |
| 18:17.32 | Qwell | There are no RPMs for CentOS 6 yet. |
| 18:18.15 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 18:20.08 | *** join/#asterisk teratoma (~teratoma@i.dont.get.mad.i.get.stabby.net) |
| 18:23.11 | bmoraca_work | anyone here use callwithus? |
| 18:23.17 | bmoraca_work | i want to use them for CNAM and LRN lookups |
| 18:35.37 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 18:36.09 | *** join/#asterisk NOSaturn (~nosaturn@207.14.59.254) |
| 18:47.19 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 18:47.28 | russellb | EPEL 6 has rpms |
| 18:47.31 | russellb | of Asterisk 1.8 probably |
| 18:47.32 | russellb | looks |
| 18:47.54 | russellb | yes, 1.8 |
| 18:48.10 | russellb | kjs: ^ |
| 18:51.06 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 18:54.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:54.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:57.01 | kjs | russellb: thanks |
| 18:58.25 | Flyingbull | exit |
| 18:58.29 | *** part/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
| 19:04.50 | davidduffy | hi all,want to do asterisk pattern match in dialplan to exclude "& and alphabets" ,any ideas? |
| 19:05.20 | volga629 | is anybody know about this warning on IMAP volicemail storage IMAP Warning: Unexpected tagged response: send LITERAL data |
| 19:05.37 | [TK]D-Fender | davidduffy, there is no "exclude", there is only "allow". |
| 19:06.03 | [TK]D-Fender | davidduffy, And if the length is variable then you will have to allow more than you want, match it, and then validate in processing as to whether to continue or not. |
| 19:07.18 | davidduffy | [TK]D-Fender :just want to weed out & and alphabets |
| 19:07.38 | [TK]D-Fender | davidduffy, You've already got your answer... |
| 19:08.00 | davidduffy | do u have a diaplan example? |
| 19:08.24 | davidduffy | don't really know how to implement that |
| 19:10.02 | davidduffy | [TK]D-Fender:do u have a dialplan example to point me in the right direction? |
| 19:12.56 | *** join/#asterisk jamicque (~jamicque@80.50.125.74) |
| 19:14.48 | [TK]D-Fender | davidduffy, "core show function REGEX" |
| 19:15.58 | davidduffy | ok |
| 19:19.29 | davidduffy | have to google the proper format and what goes where |
| 19:19.57 | davidduffy | btw any pattern match dialplan examples to weed out "& and alphabets"? pls pm me |
| 19:21.32 | navaismo | like exten => XXX,1,APP?? |
| 19:22.39 | navaismo | s/XXX/_XXX/ |
| 19:24.27 | davidduffy | like exten => _X.,1,Set(EXTEN=(0-9)) |
| 19:24.28 | p3nguin | You can filter extensions. |
| 19:24.42 | davidduffy | how do I do that? |
| 19:25.10 | p3nguin | core show function FILTER |
| 19:25.29 | navaismo | X matches any digit from 0-9 |
| 19:25.51 | [TK]D-Fender | Careful that that strips it in-line... it doesn't say "it's not valid"... you cuold get it in pieces illegally that way |
| 19:25.58 | [TK]D-Fender | 123FRED456 = 123456 |
| 19:26.01 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 19:26.28 | *** part/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 19:26.39 | [TK]D-Fender | There is a different between refusing non-numeric VS stripping so only numeric remains. BAD things happen in poor security models |
| 19:26.51 | p3nguin | Refer here for an example of use: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Security_id245121.html |
| 19:27.37 | davidduffy | tried that with a2billing.php but it did not work |
| 19:27.58 | davidduffy | could still dial non numeric on my softphone |
| 19:28.58 | [TK]D-Fender | davidduffy, Go make a pile of X, XX, XXX, XXXX, XXXX, etc patterns then up to you max limit |
| 19:31.06 | *** join/#asterisk jetlag (~jetlag@pool-71-168-244-149.cmdnnj.east.verizon.net) |
| 19:31.09 | davidduffy | exten => _X.,1,Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})}) then exten => SAFE_EXTEN,2,AGI(a2billing.php,1) |
| 19:31.19 | davidduffy | the proper format |
| 19:31.26 | davidduffy | is what I am looking for |
| 19:33.42 | davidduffy | just want that logic in a working diaplan |
| 19:34.31 | [TK]D-Fender | davidduffy, Just do as I showed above instead and save yourself the grief |
| 19:35.40 | davidduffy | [TK]D-Fender: the dialed numbers vary from 10 -19 digits,not quite practical |
| 19:36.00 | [TK]D-Fender | what 10 more lines? Big deal.. |
| 19:36.05 | *** join/#asterisk freeedrich| (friedrich@2a01:4f8:130:2023:1:151::) |
| 19:36.15 | [TK]D-Fender | I already showed ways that filtering is crap |
| 19:37.41 | davidduffy | want to apply the logic to a macro as I can't do 10-15 lines in all of the 9 contexts in the dialplan |
| 19:38.45 | navaismo | if you can pass arguments to your php scrip then pass ${SAFE_EXTEN} |
| 19:38.48 | [TK]D-Fender | I fail to see how this impacts any macros... |
| 19:39.14 | [TK]D-Fender | Macro's only have "s" |
| 19:43.41 | navaismo | maybe ....Macro(mymacro,${SAFE_EXTEN}) then in the macro: s,1,AGI(myagiscript,${ARG1}) :P |
| 19:44.40 | davidduffy | ok will try that |
| 19:45.25 | navaismo | not my code |
| 19:47.04 | davidduffy | failing that will do a special context with piles of X's then pass the other contexts to it with a Goto cmd |
| 19:47.11 | davidduffy | well thanks all |
| 19:47.26 | davidduffy | got lines of X's to write |
| 19:49.27 | [TK]D-Fender | 10 apparently |
| 19:50.27 | [TK]D-Fender | Total coding time... maybe a minute? |
| 19:52.12 | davidduffy | lol |
| 19:52.16 | davidduffy | hope so |
| 20:05.39 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:11.44 | *** join/#asterisk krotos (~d3v1l@host230-100-dynamic.15-87-r.retail.telecomitalia.it) |
| 20:12.12 | krotos | hi guy :) |
| 20:12.33 | krotos | there is way to use a standard v92 modem pci card with asterisk |
| 20:12.41 | krotos | for receiving fax from pstn line ? |
| 20:16.49 | [TK]D-Fender | krotos, Asterisk will hav nothing to do with that modem |
| 20:17.07 | [TK]D-Fender | And there is no such thing as "standard" |
| 20:20.00 | hurdman | krotos: to receive fax, you can just have a modem and a soft like http://www.hylafax.org/content/Main_Page |
| 20:26.21 | krotos | mmm, hylafax is not my preferite faxing system..:( so spandsp or res_fax_digium and a digium pstn card i think is the right choice |
| 20:26.29 | krotos | for sleeping without problem :9 |
| 20:28.10 | *** join/#asterisk ChrisInSydneyToo (~Chris@60-242-81-231.tpgi.com.au) |
| 20:29.29 | [TK]D-Fender | And * isn't a "faxing system" |
| 20:29.43 | [TK]D-Fender | It can call a single app to send or recieve but has no other control mechanism |
| 20:30.19 | krotos | right :) |
| 20:30.45 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 20:31.49 | *** join/#asterisk jks (jks@193.189.93.254) |
| 20:35.54 | *** join/#asterisk cr_ellis (~C@188-220-60-86.zone11.bethere.co.uk) |
| 20:36.22 | cr_ellis | hi, quick question to anyone around.... |
| 20:36.52 | cr_ellis | having a bit of an issue getting backgrounddetect to detect talking and wondered if there's a volume threshold somewhere |
| 20:38.19 | hurdman | cr_ellis: is-it not for detect a digit ? |
| 20:39.58 | cr_ellis | is it tone only? Thought it was any non-silence pattern conforming to min,max,silence pattern? |
| 20:40.20 | cr_ellis | is there anything equivalent for detecting speech? |
| 20:41.12 | cr_ellis | I have been following http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect |
| 20:41.27 | cr_ellis | Plays a background sound, jumping to an extension on receipt of DTMF tones or to the "talk" extension if it detects talking. |
| 20:45.37 | hurdman | http://forums.digium.com/viewtopic.php?f=13&t=79008 |
| 20:45.43 | hurdman | cr_ellis: for you |
| 20:47.28 | cr_ellis | yeah i saw that one. Is it possible to use it for incoming calls? I basically want to be able to record messages, but also check that someone talked and to auto-end it if they don't bother pushing # |
| 20:48.01 | [TK]D-Fender | cr_ellis, then that app isn't what you want. |
| 20:48.55 | [TK]D-Fender | cr_ellis, http://www.voip-info.org/wiki/index.php?page_id=140 |
| 20:49.57 | cr_ellis | the regular 'record' doesn't tell me if someone actually recorded something |
| 20:50.20 | cr_ellis | that's why i've been trying to do it with monitor() and backgrounddetect |
| 20:51.35 | [TK]D-Fender | cr_ellis, You're thinking too hard ... and the wrong way. |
| 20:51.36 | cr_ellis | if someone goes silent without recording I need to have a way to loop back and ask them to try again |
| 20:52.11 | [TK]D-Fender | cr_ellis, Attache a silence limit. After the recording is done caculate the LENGTH of it and backtrack the silence max. That should tell you |
| 20:53.14 | cr_ellis | i can query the length of recording within the IVR? |
| 20:55.37 | [TK]D-Fender | Probably not. Go call an external command |
| 20:58.51 | jaytee | anyone here configured a Polycom 650 with Expansion Module with Asterisk? Not much info on voip-info.org |
| 21:03.15 | [TK]D-Fender | jaytee, Generally nothing to configure for the module. Directory spills onto it naturally |
| 21:06.36 | jaytee | [TK]D-Fender, thanks. Sounds too easy... :-) |
| 21:17.11 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:22.16 | *** part/#asterisk jmartinez916 (~jmartinez@mail.fronteraproduce.com) |
| 21:23.58 | [TK]D-Fender | checkout time, back later.... |
| 21:27.00 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 21:37.08 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 21:37.53 | DrDamnit | core show hints shows an extension (1522@local_stations) has a hint registered and 1 watch, but the status is idle even when the user is on the phone. What am I missing here? |
| 21:45.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-pwzuuavoyjdqechc) |
| 21:56.15 | KavanS | extension not found msg, means that it's a local issue, not provider related? |
| 21:58.14 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 22:01.09 | *** join/#asterisk Micc (~Mic@c-24-19-33-189.hsd1.wa.comcast.net) |
| 22:01.42 | Micc | does asterisk have support for time zones for the time functions like gotoiftime? Like can I set a time zone then it'll offset my times automatically? |
| 22:12.03 | *** join/#asterisk DapWerk (~cthorn@firewall.drgutah.com) |
| 22:13.02 | DapWerk | Anyone familiar with what causes calls to disconnect if the caller puts themselves on mute? |
| 22:14.02 | DapWerk | is it a system setting or a phone setting? |
| 22:15.21 | sp00kz | silence detection perhaps |
| 22:16.43 | *** join/#asterisk gunnarar (~gunnarar@96.49.96.81) |
| 22:16.56 | gunnarar | can someone help me with echo issues? I'm using freepbx 2.10 and Digium TDM800P w Echo Canceller (that doesn't seem to do anything) |
| 22:17.07 | gunnarar | asterisk 1.8 |
| 22:20.41 | Henchman21 | anyone use a sipura ATA? |
| 22:21.59 | Henchman21 | im trying to dialout its fxo but it instantly drops me, but if i call the line from outside it picks up and send the dtmf that was supposed to be sent from the first outbound call |
| 22:22.14 | *** part/#asterisk DapWerk (~cthorn@firewall.drgutah.com) |
| 22:22.22 | Henchman21 | its the strangest thing in the world i cant figure out whats wrong |
| 22:23.20 | volga629 | install asterisk 10 on centos missing sqlite3 is need build from source because can't find rpm ? |
| 22:25.39 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 22:29.41 | gunnarar | so I guess to make it simple: if I have a hardware module on my pots card, should I enable anything on my server to help with sound? |
| 22:29.57 | gunnarar | or could that be causing my issue? |
| 22:37.54 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 22:42.27 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-kahdiwrlavcpxjil) |
| 22:43.26 | fprior | Hi, can you help me to understand this SIP trace (http://pastebin.com/8qF3wk2T) ? Is a call from asterisk to ATA spa400. ATA is answering with an 503 code ? |
| 22:47.40 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 23:06.35 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:18.33 | *** join/#asterisk davidcsi (~davidcsi@46.Red-79-147-163.dynamicIP.rima-tde.net) |
| 23:27.59 | volga629 | segfault at 3 ip 00000003 sp bfac14bc error 4 in libodbcinst.so.1.0.0[101000+12000] |
| 23:29.45 | volga629 | is any body so this behavior ? |
| 23:33.35 | phix | volga629: I so not this behavior |
| 23:40.04 | volga629 | what it is |
| 23:40.14 | volga629 | what is mean |
| 23:40.25 | davidcsi | hello guys, i have a NAT problem I can't seem to be able to fix... 1 |
| 23:40.59 | davidcsi | i have an asterisk <-> voip_switch <-> customer |
| 23:41.37 | volga629 | unixODBC-devel-2.2.11-7.1.i386 look like this lib |
| 23:42.58 | davidcsi | Asterisk in behind NAT... customer makes a call that goes to asterisk, everything works fine, then asterisk sends the 200 OK, and the customer responds with the ACK, but the voip_switch tries to route ONLY the ACK to the private IP... i have configured the externip, localnet, everything... any ideas¿ |
| 23:55.22 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fafdihsqfpsbfibs) |
| 23:56.09 | ChrisInSydney | davidcsi: Hi. Does outbound work ? (I am assuming that you have nat=yes) |
| 23:56.52 | ChrisInSydney | davidcsi: What sort of router. (I am assuming you have turned off any SIP ALG) |
| 23:59.51 | volga629 | here another one asterisk[8121]: segfault at 3 ip 00000003 sp bf97a81c error 4 in libtiff.so.3.8.2[101000+57000] |