| 00:06.33 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 00:09.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.176) |
| 00:22.27 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:22.27 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:35.51 | *** join/#asterisk Ionic (~ionic@2001:41d0:1:6276:fa1e:dfff:fedd:1f7) |
| 00:39.09 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 00:50.38 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 00:52.18 | *** join/#asterisk iq (~iq@cab10-39.1scom.net) |
| 00:56.48 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 01:01.58 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 01:08.08 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 01:15.18 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 01:25.19 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 01:27.36 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 01:40.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.208) |
| 02:13.17 | *** join/#asterisk andrewyager (~andrewyag@114.141.100.226) |
| 02:14.19 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 02:29.58 | *** join/#asterisk mintos (~mvaliyav@14.96.185.229) |
| 02:42.53 | igcewieling | OT: Anyone happen to know of a program which will find duplicate images in a group of 20,000 images? |
| 02:44.00 | WIMPy | The best I can think of spontaneousely would be md5sum, sort and uniq. |
| 02:45.03 | igcewieling | WIMPy: I have a window program which will find "similar" images, but it tends to choke at about 8,000 images |
| 02:45.24 | igcewieling | s/window/windows |
| 02:55.22 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:59.02 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 03:11.01 | *** join/#asterisk gufi (~quassel@cpe-24-25-214-252.san.res.rr.com) |
| 03:11.14 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 03:29.43 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 03:31.54 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 04:07.11 | floren | hi everyone :) |
| 04:08.55 | floren | quick question about asterisk sounds for voicemail. what sound format should i install, wav or ulaw? |
| 04:09.07 | floren | running asterisk on centos 6 x86_64 |
| 04:09.35 | ChannelZ | ulaw probably |
| 04:09.45 | floren | thanks ChannelZ |
| 04:10.18 | floren | i'm allowing only ulaw into sip.conf, i wanted to make sure about that part |
| 04:11.05 | floren | allow = !all,ulaw |
| 04:11.06 | ChannelZ | yeah save yourself the transcoding power |
| 04:11.22 | floren | thanks :) |
| 04:11.38 | floren | how are you tonight ChannelZ |
| 04:11.41 | ChannelZ | it's negligble |
| 04:11.55 | ChannelZ | not bad, you? |
| 04:12.24 | floren | worked a bit more on my fancy phone directory and also finished a set of rpm's for asterisk core sounds |
| 04:13.04 | floren | i'm pulling the google contacts from my account and format it nice for my crisco phone |
| 04:13.38 | floren | i'm going to share the script on asterisk forums, pretty sure others will want it |
| 04:13.42 | ChannelZ | fun |
| 04:13.56 | floren | then i will probably tackle the visual voicemail in cisco |
| 04:16.04 | floren | here it is an example: http://youtu.be/Xbm4SgkLCqY?t=2m59s |
| 04:16.51 | floren | those phones are so nice, to bad they dont play well with asterisk :( |
| 04:23.59 | floren | right now i dial into voip.ms voicemail: exten => _*XX,1,Dial(SIP/${EXTEN}@voipms,60) |
| 04:24.24 | floren | i presume i'm going to need a separate context for the local voicemail? |
| 04:24.39 | floren | i'm going to drop the voip.ms voicemail |
| 04:40.44 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.183) |
| 04:41.25 | *** join/#asterisk smirker (~x@101.162.216.118) |
| 04:43.33 | ChannelZ | a dialplan context? no |
| 04:47.49 | floren | i'm going to play with it tomorrow, to tired tonight, i'll read the voicemail part into asterisk definite guide |
| 04:48.16 | floren | all i have to do basically is add 2-3 lines to the inbound context |
| 04:49.04 | floren | anyways, i'll check it in depth tomorrow, now i'm in brainless mode.. to tired my friend |
| 04:55.11 | *** join/#asterisk smirker (~x@101.162.216.118) |
| 05:01.25 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 05:15.46 | *** join/#asterisk cmendes0101| (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 05:22.00 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 05:41.08 | *** join/#asterisk russum (~russum@91.112.86.202) |
| 05:41.43 | *** part/#asterisk russum (~russum@91.112.86.202) |
| 05:59.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.53) |
| 06:07.26 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 06:07.54 | *** join/#asterisk dash__ (~dash@unaffiliated/dash-/x-7576607) |
| 06:13.24 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:19.46 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 06:21.07 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
| 06:27.46 | *** join/#asterisk russum (~russum@91.112.86.202) |
| 06:27.59 | *** part/#asterisk russum (~russum@91.112.86.202) |
| 06:53.12 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-tzkallueewchizgq) |
| 07:08.15 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 07:13.38 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-fcglxqrvurhgnqse) |
| 07:18.13 | *** join/#asterisk andrewyager (~andrewyag@114.141.100.226) |
| 07:27.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.53) |
| 07:30.36 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:48.38 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:08.14 | *** join/#asterisk Rumbles (~Rumbles@host213-123-247-112.in-addr.btopenworld.com) |
| 08:08.46 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 08:13.13 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2539:9816:523c:1f6) |
| 08:19.15 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 08:22.27 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 08:23.16 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:24.36 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 08:32.36 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 08:47.02 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:47.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.231) |
| 08:50.58 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:51.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:04.19 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 09:04.37 | danfromuk | Hi, can anyone recommend a sip softphone that supports paging? |
| 09:11.21 | apb1963 | danfromuk: what exactly does it need to support paging? Auto answer?? |
| 09:12.27 | kleszcz | danfromuk: 3cx ? |
| 09:16.22 | apb1963 | danfromuk: Yeah, a quick google seems to imply you simply need autoanswer. phonerlite has that. |
| 09:36.44 | *** join/#asterisk aruntomar (~Thunderbi@49.248.157.43) |
| 09:42.11 | *** join/#asterisk ghost75 (~trechber@188.105.16.29) |
| 09:42.51 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 09:52.29 | *** join/#asterisk RZero (~chatzilla@85.118.159.250) |
| 09:54.24 | RZero | Hi all I am running Asterisk 1.8 and looking for a way to disable DTMF on the fly when receiving an inbound call from DAHDI, I have tried set(MUTEAUDIO(in)=on) but that doesn't seem to do what I want. |
| 10:15.11 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 10:45.00 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 10:45.38 | Greenlight | Is there anything I should be aware of when trying to get the SIP response for a call terminating via ${HASH(SIP_CAUSE,${CDR(dstchannel)})} ? |
| 10:45.42 | Greenlight | Seems always to be empty. |
| 10:47.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.231) |
| 10:50.09 | *** join/#asterisk razu (~razu@2001:ad0:1:1:202:b3ff:fe36:921c) |
| 10:53.24 | *** join/#asterisk aruntomar (~Thunderbi@49.248.157.43) |
| 10:55.30 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 10:59.57 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 11:14.11 | *** join/#asterisk CeBe (~CeBe@port-92-206-83-5.dynamic.qsc.de) |
| 11:14.47 | *** join/#asterisk Rumbles (~Rumbles@host213-123-247-112.in-addr.btopenworld.com) |
| 11:15.42 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 11:19.10 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:20.31 | *** join/#asterisk petn-randall (~petn-rand@borma.rocketjump.eu) |
| 11:21.26 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 11:25.53 | petn-randall | Hi, I've recently upgraded from 1.6 to 1.8, and after that the ability to receive faxes broke. Relevant config is http://paste.debian.net/16756/, and the log messages are http://paste.debian.net/16757/. Could anyone point me in the right direction? |
| 11:25.55 | *** join/#asterisk brian98 (~brian98@unaffiliated/brian98) |
| 11:29.57 | petn-randall | The other SIP peer that sends the fax is a beronet card, which has T.38 enabled. It's also enable in asterisk. |
| 11:31.55 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 11:35.30 | CeBe | petn-randall: any NAT involved? |
| 11:36.10 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 11:37.18 | petn-randall | CeBe: No, asterisk and the beronet card are on the same subnet. |
| 11:38.52 | CeBe | okay, than I have no idea. Had a similar problem with nat while upgrading because "canreinvite" parameter has been renamed to "directmedia" |
| 11:43.55 | *** join/#asterisk jmls1 (~julian@77.107.171.82) |
| 11:47.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.16) |
| 11:49.02 | Greenlight | Anyone got a working example of getting the SIP hangup cause using ${HASH(SIP_CAUSE,${CDR(dstchannel)})} |
| 11:51.42 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:52.02 | igcewieling | have you tried ${CHANNEL} instead of ${CDR(dstchannel)} |
| 11:52.34 | Greenlight | Nope - I thought it had to be the destination channel - I'll give it a shot though |
| 11:53.16 | igcewieling | If that doesn't work, check UPGRADE.txt, I think SIP cause stuff changed in Ast 10 or 11 |
| 11:53.35 | Greenlight | Yea, still nothing. Will check it out, thanks! :) |
| 11:53.53 | igcewieling | Greenlight: normal HANGUPCAUSE is available on the calling channel so I assume the sip cause would be too |
| 11:54.04 | igcewieling | no problem |
| 11:55.23 | Greenlight | Seems it may be been disabled by default now because of performance issues. Damn :( |
| 11:56.57 | igcewieling | Greenlight: only in 1.8 |
| 11:57.16 | igcewieling | in 10 and 11 that was all reworked to take care of the performance issues |
| 11:57.27 | igcewieling | thugh it may still not be enabled by default |
| 11:58.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.16) |
| 12:03.19 | *** join/#asterisk runtux (~runtux@91-119-209-194.dynamic.xdsl-line.inode.at) |
| 12:08.07 | *** join/#asterisk Draecos (~Draecos@220-253-166-184.dyn.iinet.net.au) |
| 12:12.15 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 12:16.10 | RZero | Hi all I am running Asterisk 1.8 and looking for a way to disable DTMF on the fly when receiving an inbound call from DAHDI, I have tried set(MUTEAUDIO(in)=on) but that doesn't seem to do what I want. |
| 12:19.44 | *** join/#asterisk kannan (~chatzilla@112.79.44.88) |
| 12:22.35 | WIMPy | What does "disable DTMF" mean? |
| 12:22.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:23.35 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 12:23.39 | RZero | sorry I mean disable/ignore DTMF dection I have an IVR that has Ivona tts running on it, when some one presses any key on the phone it skips the TTS playback |
| 12:24.33 | WIMPy | Then change your dialplan to a way that doesn't abourt the playback. |
| 12:24.38 | WIMPy | i.e. Playback(). |
| 12:25.21 | RZero | Ivona doesn't work like that :( |
| 12:26.00 | igcewieling | Then you have a problem. |
| 12:26.07 | RZero | Yup |
| 12:26.23 | igcewieling | The only thing I can think of is to restart the IVR if DTMF is received |
| 12:26.38 | WIMPy | Google tells me Ivona is a TTS engine. How is that related to DTMF recognition? |
| 12:28.27 | RZero | WIMPy we use Ivona TTS so customers can create intro's on the fly so some require the message to be played in full before moving onto the IVR but pressing a key on the phone skips it |
| 12:29.03 | igcewieling | This sounds like something to talk to Ivona about |
| 12:29.07 | WIMPy | But that's how YOU use it. |
| 12:29.40 | WIMPy | Or maybe we haven't heard th full story, yet. |
| 12:31.08 | RZero | Call flow is Inbound DAHDI -> Fastagi -> pass back to dialplan play Ivona message -> enter IVR |
| 12:31.45 | WIMPy | So it's the way you use the Ivona output. |
| 12:33.59 | RZero | looks that way I was hoping there was away I could ignore the DTMF in Asterisk |
| 12:34.21 | WIMPy | Don't tell it do do something about it. |
| 12:39.32 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:39.32 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:52.29 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 12:57.13 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:57.56 | *** join/#asterisk TimeRider (~steve@97e2fc92.skybroadband.com) |
| 12:58.19 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 12:59.12 | kannan | hello, we have an issue with background noise (like a hissing sound). IP Phones are grandstream 2100 HD, pstn connectivity via e1 PRI and another by SIP sevice. the system is running fine for over a yar with no issues thus far. internal calls are ine. the problem happens when 3 or more phones join a third party conference room. |
| 13:00.09 | igcewieling | sounds like a bandwidth issue |
| 13:00.14 | WIMPy | MeetMe or ConfBridge? |
| 13:00.26 | WIMPy | Or CPU time. |
| 13:00.33 | Greenlight | Possibly timing issue |
| 13:00.52 | WIMPy | Or just an olg bug. |
| 13:01.05 | WIMPy | old |
| 13:01.07 | igcewieling | Greenlight: Maybe a PRI clock source issue, but not likely a timing issue with the conference |
| 13:01.19 | WIMPy | Who exactely will hear that noise? |
| 13:01.46 | Greenlight | Ahh 3rd party conerence room. Still the issue could be there. |
| 13:01.50 | *** join/#asterisk vlad_starkov (~vlad_star@62.141.80.122) |
| 13:02.02 | igcewieling | Greenlight: Asterisk does not require timing for MeetMe. |
| 13:02.15 | igcewieling | Asterisk requires dahdi to MIX the audio, not for timing the packets |
| 13:02.26 | WIMPy | Hmm |
| 13:02.27 | Greenlight | Sure it requires timing for MeetMe |
| 13:02.33 | igcewieling | no, it does not. |
| 13:02.37 | Greenlight | Yes. It does. |
| 13:02.38 | igcewieling | It requires mixing for meetme |
| 13:03.01 | Greenlight | dahdi needs timing |
| 13:03.09 | igcewieling | huh? |
| 13:03.14 | Greenlight | To mix |
| 13:03.19 | Katty | morning |
| 13:03.29 | Greenlight | It'll use the hardware timing or emulated |
| 13:03.37 | Greenlight | BUt it still needs accurate timing or you'll get issues |
| 13:03.56 | Greenlight | There's even an application "dahdi_test" |
| 13:04.01 | Greenlight | TO measure timing accuracy |
| 13:04.02 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:04.04 | [TK]D-Fender | Greenlight: No |
| 13:04.07 | kannan | hmm, if it could be a third party issue, we should be able to rule that out |
| 13:04.21 | [TK]D-Fender | Greenlight: It is only the mixing engine, not used for timing in Meetme |
| 13:04.25 | kannan | i will try and setup a conference room an re test |
| 13:04.48 | igcewieling | kannan: why not simply call into the external conferencing service from a non-asterisk system like a POTS fax line |
| 13:04.57 | igcewieling | well, a couple of pots lines |
| 13:05.05 | kannan | the dahdi_test is at an average of 99.996 |
| 13:05.12 | Greenlight | So, with poor timing, how come the MeetMe conference will suck? There are LOADS of articles online |
| 13:05.14 | kannan | igceweiling, ok |
| 13:05.37 | Greenlight | I don't understand how you can claim a MeetMe conference doesn't reuqire a timing source |
| 13:05.46 | *** join/#asterisk dilibau (~paranoic@82.137.11.75) |
| 13:06.06 | igcewieling | Greenlight: Because that is what Digium told us. |
| 13:06.21 | Greenlight | Link? |
| 13:06.30 | igcewieling | Greenlight: DAHDI timing used to be required for lots of audio stuff, as of 1.6 that isn't the case anymore. |
| 13:06.33 | igcewieling | file! |
| 13:06.41 | dilibau | someone can help me? agi script doesn't work well |
| 13:06.54 | Greenlight | I'm not talking about dahdi timing here, as used by asterisk |
| 13:06.55 | igcewieling | still a little early for Qwell or file or malcomd to be around |
| 13:07.02 | Greenlight | I'm talking about meetme mixing timing |
| 13:07.19 | Greenlight | A MeetMe conference still requires accurate timing |
| 13:07.28 | Greenlight | As would *any* conference |
| 13:07.36 | WIMPy | would be VERY surprised if MeetMe wouldn't require dahdi timing. |
| 13:07.40 | igcewieling | It does, and that timing can come from any of Asterisk's supported timing source. |
| 13:07.53 | Greenlight | facepalms |
| 13:08.07 | WIMPy | dahdi getting timing from Asterisk indtead of vice versa? |
| 13:08.12 | igcewieling | Wait until Digium wakes up and we can ask them |
| 13:08.23 | Greenlight | It requires accurate timing, regardless of the source |
| 13:08.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:08.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:08.46 | igcewieling | leifmadsen: Does MeetMe use DAHDI timing or any Asterisk timing source? |
| 13:08.48 | Greenlight | I did *think* that MeetMe always used dahdi's timing source, but if you say it uses timerfd or the others fine. |
| 13:08.57 | leifmadsen | igcewieling: DAHDI only |
| 13:09.07 | Greenlight | See... |
| 13:09.11 | leifmadsen | only ConfBridge() can use additional timing sources |
| 13:09.18 | igcewieling | leifmadsen: are you sure? I thought DAHDI was only used for mixing |
| 13:09.34 | Greenlight | Even ConfBrdige still requires accurate timing |
| 13:09.42 | *** join/#asterisk xnice (xnice@85.13.198.219) |
| 13:09.45 | Greenlight | Doesn't matter on the source - it's still a factor |
| 13:09.46 | leifmadsen | igcewieling: I think it's both |
| 13:10.41 | Greenlight | If I'm using MeetMe I check timing with "dahdi_test". If I'm using ConfBridge I check with asterisk CLI "timing test" |
| 13:10.55 | igcewieling | leifmadsen: I hate it when you disagree with me. I can't just write you off as an idiot, I have to actually listen. |
| 13:11.07 | igcewieling | 8-| |
| 13:11.23 | Greenlight | wonders if he was the one getting written off as an "idiot" |
| 13:11.27 | runtux | RZero: still here? I've written an asterisk application that can turn off DTMF detection on a channel |
| 13:11.38 | WIMPy | wonders what that means for Greenlight and himself. |
| 13:11.41 | igcewieling | Greenlight: "almost idiot" 8-) |
| 13:11.46 | leifmadsen | igcewieling: heh, could always check the code for certain :) |
| 13:11.47 | Greenlight | lol |
| 13:12.04 | igcewieling | Greenlight: but you are more than enough non-idiot for me to confirm with someone from Digium later in the day |
| 13:12.19 | Greenlight | I *guess* that's something |
| 13:12.28 | Greenlight | If no a back-handed compliment :) |
| 13:12.31 | Greenlight | *not |
| 13:12.38 | igcewieling | leifmadsen: I spent years in therapy after the last time I looked at the code. |
| 13:12.46 | leifmadsen | fair play |
| 13:13.06 | igcewieling | Greenlight: We are all a "little bit idiot". |
| 13:13.08 | RZero | runtux I am still here |
| 13:13.33 | Greenlight | igcewieling: Yea, on that one I'd agree |
| 13:13.37 | igcewieling | Generally if I wish you the best of luck,that is code for "you are an idiot" |
| 13:13.49 | xnice | hello...i have a sip device at house to enable me using the landline while i am outside/through my cellphone sip software, but when i need to make free calls i first have to call the home sip device xxxx then it will open the landline to enter the number iwant to call "manually" is there a way to just enter for example the home sip device+the number i want to call together as i dont memorize |
| 13:13.49 | xnice | all my contacts numbers? something maybe like xxxx*cellphonenumber ? |
| 13:13.50 | igcewieling | 8-) |
| 13:15.08 | igcewieling | Greenlight: it can be difficult to figure out if the issue is idiot or ESL. The ESL people I'll do my best to help if they are not also idiots |
| 13:15.56 | Greenlight | ESL ? |
| 13:16.07 | igcewieling | English as a second language |
| 13:16.10 | Greenlight | Oh |
| 13:16.46 | Greenlight | You can usually tell when someone's tried to solve an issue, and put some thought in. I'm much more inclinded to try and help those people (if I'm able) |
| 13:17.33 | igcewieling | WIMPy: we often disagree, but you are NOT an idiot. |
| 13:17.44 | igcewieling | most people here are not idiots |
| 13:18.39 | xnice | hello |
| 13:18.56 | Greenlight | xnice: I can't work out what questions you're tryhing to ask |
| 13:19.03 | Greenlight | You want to dial two numbers at once? |
| 13:19.18 | xnice | no |
| 13:19.32 | Greenlight | And where does Asterisk fit into your question |
| 13:19.34 | igcewieling | xnice: You may want to read the Asterisk book, the tasks you need to do to accomplish your goal should be covered in the book. |
| 13:19.49 | igcewieling | The specific goal likely is not. |
| 13:20.06 | xnice | i dont want to dial first the sip device number then wait 3 seconds then again i dial the number i am trying to reach |
| 13:20.40 | Greenlight | There are examples of just this |
| 13:20.54 | xnice | what they call this ? so that i can search for it |
| 13:21.25 | leifmadsen | xnice: check the flags to the Dial() command |
| 13:21.43 | Greenlight | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial <-- Example 3 |
| 13:21.51 | Greenlight | (As much as I hate to refernece voipinfo) |
| 13:21.53 | leifmadsen | it has the ability to let you dial stuff as you connect to the other end automatically, so you can just setupa star code or whatever you want to trigger a specific string |
| 13:22.07 | leifmadsen | look it up in the AstDB or something, and create an interface to be able to add and modify the values that can be dialed |
| 13:22.13 | leifmadsen | *1, *2, etc |
| 13:22.38 | xnice | so you mean i must first save the numbers in the sip device at home ? |
| 13:25.09 | leifmadsen | in your asterisk box... |
| 13:25.13 | leifmadsen | it will then dial them for you |
| 13:25.39 | xnice | isnt there a way to just do it in my sip client without touching the server ? |
| 13:25.40 | leifmadsen | or you can enter whatever you want, save it to a channel variable, and pass it to the Dial() application and flag to dial for you once the other end (SIP ATA) has answered |
| 13:25.40 | WIMPy | igcewieling: It's a a question of definition. It you go back towards it's origins, I'm pretty certainly an idiot :-) |
| 13:25.42 | Cuzner | Greenlight: I don't think voip-info has changed the look of the site in a decade |
| 13:25.48 | leifmadsen | xnice: probably not |
| 13:26.01 | leifmadsen | there is SOME modification required on the server based on what you requested |
| 13:26.09 | WIMPy | Cuzner: Or the contents? |
| 13:26.15 | Cuzner | and 3/4 of the links off voip-info don't work anymore, it's understandable that you don't like to link to it |
| 13:26.18 | xnice | leifmadsen i see |
| 13:26.19 | Cuzner | WIMPy: right |
| 13:26.30 | Cuzner | but a decade ago, it was super useful :) |
| 13:26.51 | xnice | thanks a lot ppls |
| 13:27.00 | xnice | i appreciate the help |
| 13:27.07 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 13:27.08 | [TK]D-Fender | xnice: You can either Dial() it with D() for in-call DTMF,or pass it the number in the actual SIP dial to the device |
| 13:27.14 | WIMPy | 5 years ago it was still very useful, but you had to be aware that it wasn't up to date. |
| 13:28.31 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 13:28.53 | Cuzner | meh, it still is pretty useful, just outdated. |
| 13:30.48 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mcqrayziwwmxjgls) |
| 13:30.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:32.14 | Greenlight | I agree, it often has examples, which the wiki sometimes lacks |
| 13:32.46 | xnice | <[TK]D-Fender> sorry as i am newbie in this, do you mean i can do something in the sip client or i must do it first in the sip device ? |
| 13:32.49 | kannan | With regars to background noise on our asterisk system, some updates, which have me more stumped than before. The noise issue happens only with speaker phone mode on grandstream gxp 2100 model, though this was not the case before. it was all fine until yesterday. |
| 13:33.15 | Greenlight | kannan: So the issue is the handset then |
| 13:33.33 | danfromuk | Hi, Why would LEN be returning 19 when the screen has 20 characters in it? |
| 13:33.33 | Cuzner | kannan: yep, mic too sensitive |
| 13:33.35 | Greenlight | *or* some other issue as well which isn't presenting itself today |
| 13:33.53 | kannan | but, its on all phones, of this model, all of a sudden. |
| 13:34.04 | Greenlight | danfromuk: Huh? What do you mean, do you have an example ? |
| 13:34.54 | Greenlight | kannan: So a normal internal call, you activate speakphone and get this "noise" ? |
| 13:35.08 | kannan | they have been is use in thi mode for months now. could it be some netwrk based issue. i am un able to get perission to power recycle poe switch |
| 13:35.28 | kannan | Greenlight , let me check |
| 13:36.30 | [TK]D-Fender | xnice: This is you dialing your device from Asterisk |
| 13:37.01 | danfromuk | Greenlight: heres the code and CLI output. http://pastebin.com/fsPkpYqQ You see that a CDR(accountcode) is set to a string, and then LEN of CDR(accountcode) returns 19 instead of 20. |
| 13:37.58 | xnice | <[TK]D-Fender> so i can just enter a something in my sip client and save it for example sip device number + the number i want to call, xxxx+12345678 ? |
| 13:38.04 | xnice | <[TK]D-Fender> something like this ? |
| 13:38.43 | Greenlight | danfromuk: Is there perhaps a limit on the field length? |
| 13:38.49 | [TK]D-Fender | xnice: You would dial it in whatever pattern YOU set up for your exten in asterisk |
| 13:38.51 | *** join/#asterisk jsarrel (~jsarrel@24-158-61-198.static.hckr.nc.charter.com) |
| 13:39.02 | Greenlight | Can you NoOP the CRR(accountcode) and see what its set to |
| 13:39.31 | *** join/#asterisk jsarrel (~jsarrel@24-158-61-198.static.hckr.nc.charter.com) |
| 13:40.07 | [TK]D-Fender | <PROTECTED> |
| 13:40.15 | [TK]D-Fender | Seems to indicate the last char never made it |
| 13:40.35 | Greenlight | Indeed |
| 13:40.43 | [TK]D-Fender | otherwise the last char should have been "s" |
| 13:40.54 | [TK]D-Fender | Perhaps ... FIELD LENGTH limit.... |
| 13:40.58 | Greenlight | mmhmm |
| 13:41.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:41.44 | danfromuk | The first line that sets cdr(accountcode) to sunlocksolutions_101, then a NoOP shows sunlocksolution_10, chopping off the last 1 |
| 13:42.00 | danfromuk | So the final line chops off the s |
| 13:42.13 | Greenlight | Yea, you've got a limit on the lenfth of that field, as we said |
| 13:42.26 | danfromuk | thanks for pointing it out. |
| 13:43.02 | danfromuk | Is there an IndexOf command in Asterisk to check the location of a character within a string? |
| 13:43.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:43.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.48 | leifmadsen | danfromuk: that is getting pretty specific... sounds more like an AGI() or SHELL() thing |
| 13:44.21 | danfromuk | leifmadsen: found it. Func CUT will do the trick. |
| 13:44.21 | WIMPy | A combination of CUT and LEN? |
| 13:44.59 | leifmadsen | I guess I was thinking of something that wasn't a loop looking at each char :\ |
| 13:46.32 | *** join/#asterisk vlad_starkov (~vlad_star@62.141.80.122) |
| 13:47.12 | WIMPy | ~askozia |
| 13:47.33 | WIMPy | Oh, unknown. |
| 13:48.46 | Katty | Qwell: SO |
| 13:48.51 | Katty | Qwell: how was your german chocolate cake? |
| 13:50.42 | *** join/#asterisk dtx (~dtx@cdf-imaging.com) |
| 13:58.17 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:58.17 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:16.15 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 14:16.51 | sereal-work | Does anyone know of any models or manufactures making hardphones that support IPv6 out of the the box? |
| 14:17.23 | sereal-work | Do the digium ip phones support it? |
| 14:20.21 | file | there are very few hardphones that support IPv6 |
| 14:20.26 | file | Digium phones are currently not one of them |
| 14:20.43 | sereal-work | Do you know of any specific models that do support v6? |
| 14:21.22 | file | some Snom and some... uh... can't remember... |
| 14:21.34 | file | lemme try to jog my memory |
| 14:21.50 | file | Yealink |
| 14:22.09 | file | although how well IPv6 works I dunno... I did some quick reading and it seemed to vary between firmware quite a bit |
| 14:22.40 | Chainsaw | My Polycoms here don't do it. |
| 14:23.21 | sereal-work | I would both like to support manufactures making v6 phones and to do some research on it. |
| 14:23.36 | sereal-work | none of the phones I have access to at the moment do IPv6 |
| 14:24.07 | file | I know the Bria softphone on PC also supports IPv6 |
| 14:26.57 | *** join/#asterisk kannan (~chatzilla@112.79.40.134) |
| 14:27.08 | sereal-work | Yealink seems to be the guys to go with |
| 14:27.18 | sereal-work | apparently all their phones support v6 |
| 14:29.35 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 14:45.36 | *** join/#asterisk cjm_ (~cjm@tclc.org) |
| 14:46.02 | WIMPy | "Some Snoms"? Don't they all do V6? |
| 14:48.56 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:04.54 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 15:10.39 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:22.40 | cjm_ | Hi Folks, My asterisk service is listening on ports 5038:TCP, 5060:UDP and 4569:UDP. 5060:TCP is sip and 4569:UDP is IAX. What is 5038? |
| 15:23.17 | file | Asterisk Manager Interface |
| 15:23.29 | cjm_ | file, Thanks. |
| 15:29.12 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 15:30.37 | rrittgarn | Trying to accomplish the following and not sure the best way... Customer wants to be able to blind transfer to an extension, and if the extension doesn't answer, ring back the person who did the transfer. Am I over thinking this or is there something I'm misisng? |
| 15:31.08 | [TK]D-Fender | How is that over-thinking? |
| 15:31.12 | [TK]D-Fender | That is just stating what you want |
| 15:31.31 | [TK]D-Fender | Now if you gave some psycho implementation you had in mind for it... that'd be another matter :) |
| 15:32.05 | rrittgarn | yeah... i figured I would start with what i want and get pointers from there. I'm sure it can be done, just not sure how... if only the return function were magical |
| 15:32.08 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:32.50 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:33.06 | [TK]D-Fender | Feel free to go do it then... |
| 15:33.57 | [TK]D-Fender | You could set a variabl on that channel before you call the device in the first place... and it'll still be there. And you can also use that other rather specifically named asterisk standard channel variable... you should go read that list... |
| 15:34.25 | *** join/#asterisk vlad_starkov (~vlad_star@93.191.18.82) |
| 15:38.04 | cjm_ | Hi Folks, anybody know of a command line sip client? I know that sounds silly, but I need to make sure I have my firewall working properly and that means sip from the public. |
| 15:38.12 | *** join/#asterisk nix8n82 (~AndChat27@151.sub-97-48-3.myvzw.com) |
| 15:38.14 | mjordan | pjsua |
| 15:38.31 | mjordan | http://www.pjsip.org/pjsua.htm |
| 15:38.53 | mjordan | cjm_: ^^^ |
| 15:39.30 | cjm_ | mjordan, Perfect! Better than perfect! Thanks for the help. |
| 15:40.26 | *** join/#asterisk larrymo (c6bee63b@gateway/web/freenode/ip.198.190.230.59) |
| 15:41.04 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 15:42.21 | *** join/#asterisk cneb3000 (~cneb3000@2.221.241.85) |
| 15:42.48 | larrymo | Im building a website for a restaurant and im allowing them to do online ordering. I want on a successful order over the web, to generate an automated phonecall that says the items the person purchased but also has features like press 1 to accept order, press 2 to decline order.. which would then alert my website by running a file. What do I use to get this system working? Asterisk? I can't find web hosting with this I d |
| 15:43.23 | larrymo | I know asterisk will allow me to generate an automated call file and drop it in an outbound box... this sounds tempting. |
| 15:43.31 | Cuzner | cjm_: telnet works in a pinch too :) |
| 15:43.31 | Nugget | telnet is eeeeeeevil! |
| 15:43.39 | Cuzner | shush |
| 15:43.57 | rrittgarn | @Cuzner that's TCP though isn't it? OP wanted to do UDP SIP testing |
| 15:45.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.52) |
| 15:45.04 | larrymo | anyone? |
| 15:45.05 | cjm_ | Cuzner, That was my first attempt. I was pretty sure it would. But I keep getting "Connection refused." and that made me suspect I have a problem. |
| 15:45.11 | rrittgarn | @larrymo using your favorite flavor of web language you can generate calls via a socket connection to the AMI, or via the AJAM interface via web, or call files if its on the same network. Lots of options |
| 15:45.54 | larrymo | rrottgarn: Think there is a 3rd party provider with this setup? Hosting it myself right now is not an option. |
| 15:46.05 | larrymo | rrittgarn* |
| 15:46.15 | igcewieling | larrymo: Asterisk is a toolkit which allows you to build a PBX. You can trigger a call from Asteirsk using .call files, and point them to an IVR, but everything else is outside the scope of Asterisk |
| 15:46.41 | igcewieling | larrymo: you may want to read the first part of the Asterisk Book, which should give you a good idea of what Asterisk can do. |
| 15:46.42 | igcewieling | ~book |
| 15:46.42 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:47.27 | igcewieling | larrymo: I'm not aware of any companies building custom Asterisk applications, but I'm sure they exist. |
| 15:47.41 | igcewieling | Likely outside your budget though. |
| 15:49.51 | larrymo | right now I can alert a restaurant of an order through fax, SMS, email and a custom application on their computer. The ability to do it with voice would be nice using text to speech. |
| 15:50.14 | Greenlight | Sounds fairly easy to do |
| 15:50.57 | Greenlight | Just make an AMI call to originate a call with your required parameters |
| 15:57.08 | *** join/#asterisk guitarHester (~hester@nat/digium/x-xgutagkuffepjtzo) |
| 16:02.43 | *** join/#asterisk slidesinger (~slidesing@c-69-141-23-216.hsd1.nj.comcast.net) |
| 16:04.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:10.08 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 16:11.34 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:13.41 | gtTuna | igcewieling, have done some more testing today with that hangup issue |
| 16:14.12 | gtTuna | user can make outgoing calls fine w/o it hanging up. i tried making an internal call, bypassing any external SIP providers / PSTN, and it still hungup |
| 16:16.55 | *** join/#asterisk Protux (~Protux@abo-57-129-68.bdx.modulonet.fr) |
| 16:19.49 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 16:19.55 | igcewieling | gtTuna: You are in Spain, right? |
| 16:22.08 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-edzxeyxmsasqqsaf) |
| 16:23.23 | vlad_starkov | Question: Using features.conf I successfully make atxfer. But when I make bxfer the call transfers correctly but there is one-way audio. Is is possible that the problem is not in the router? |
| 16:28.38 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-393104.home.otenet.gr) |
| 16:30.21 | igcewieling | almost all one-way or no-way audio issues are NAT or firewall issues. |
| 16:30.27 | igcewieling | or direct media issues |
| 16:32.20 | vlad_starkov | igcewieling: I got this error in CLI after bxfer complete "[2013-07-18 20:29:53] WARNING[11867][C-00009513]: translate.c:343 framein: no samples for ulawtolin" |
| 16:32.32 | vlad_starkov | Ast 11.3 |
| 16:33.21 | *** join/#asterisk datachomper (~rmeyerrie@nat/digium/x-kqzpowdgyxypryug) |
| 16:33.27 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:33.52 | Greenlight | I see that "no samples" error from time to time - think it's a red herring |
| 16:33.59 | *** join/#asterisk cmendes0101| (~cmendes01@72.1.46.254) |
| 16:34.25 | gtTuna | igcewieling, OH, USA |
| 16:34.44 | vlad_starkov | Greenlight: Ok |
| 16:45.45 | igcewieling | gtTuna: pastebin the cli output if a failed call without SIP debug, and one with sip debug enabled. preferrably sip debug enabled only for the relevant peers |
| 16:46.17 | igcewieling | I don't have time to troubleshoot at the moment, but someone here might and they'll need the info. |
| 16:47.50 | *** part/#asterisk larrymo (c6bee63b@gateway/web/freenode/ip.198.190.230.59) |
| 16:51.35 | *** join/#asterisk miztic (~gerard@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 16:53.24 | *** join/#asterisk nix8n82 (~AndChat27@151.sub-97-48-3.myvzw.com) |
| 17:02.16 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 17:19.10 | *** join/#asterisk peter (~peter@x2f00811.dyn.telefonica.de) |
| 17:19.27 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 17:28.58 | *** join/#asterisk Rumbles (~Rumbles@31.205.54.123) |
| 17:29.15 | *** join/#asterisk random_cat (~random_ca@gateway/tor-sasl/randomcat/x-49498005) |
| 17:34.01 | *** join/#asterisk nix8n82 (~AndChat27@151.sub-97-48-3.myvzw.com) |
| 17:37.34 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:50.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.245) |
| 18:17.59 | cjm_ | Hi Folks. Echo Test works when I call "sip:*43@Asterisk.TCLC.org", but NOT in for either "sip:*43@sip.TCLC.org" or "sip:*43@10.1.1.13", and Asterisk.TCLC.org, sip.TCLC.org and 10.1.1.13 are all the same as far as ping is concerned. Any advice about this mystery? |
| 18:22.35 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:24.05 | igcewieling | cjm_: did you do something stupid like use bindaddr option? |
| 18:24.45 | cjm_ | igcewieling, I've been know to do stupis things, but I don't think I did anything stupid today... Let me check my notes... Hmm.. Nope. NOthing stupid today. |
| 18:24.48 | cjm_ | (-: |
| 18:25.25 | igcewieling | cjm_: you are not using bindaddr? Maybe some localnet, externip, or directmedia issue? |
| 18:25.26 | cjm_ | igcewieling, I'm wondering if Apache Virtuyal hosting has anything to do with this? |
| 18:25.41 | igcewieling | cjm_: I have no idea what you are talking about |
| 18:26.14 | cjm_ | Asterisk, sip and 10,.1.1.13 are all the same machine. However, http://Asterisk.TCLC.org gets a different site than http://sip.TCLC.org... |
| 18:28.58 | cjm_ | igcewieling, Oops... You don't know what I am talking about because it is FreePBX that responds to the HTTP calls..., but it is a clue. Asterisk doesn't have any sort of virtual host capability, but he apparently does revers dns lookups or something. |
| 18:29.38 | igcewieling | FreePBX and you are asking on Asterisk. You are dead to me. |
| 18:29.45 | cjm_ | In any case, calls to ...@Asterisk.TCLC.org work but other equivalent addresses don't. |
| 18:29.54 | cjm_ | igcewieling, Ha! Sorry, Dude. My mistake. |
| 18:29.57 | *** join/#asterisk lvlolvlo_ (~lvlolvlo@unaffiliated/lvlolvlo) |
| 18:31.07 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 18:31.32 | cjm_ | igcewieling, Sorry about the FreePBX insult, but I am really asking an Asterisk question. Equivalent host IP designations -- Asterisk.TCLC.org, sip.TCLC.org and 10.1.1.13 give me different results. |
| 18:32.09 | cjm_ | igcewieling, Asterisk.TCLC.org works but the others don't. Any thoughts? |
| 18:32.12 | igcewieling | cjm_: pastebin the sipdebug of the failed call. |
| 18:32.31 | igcewieling | it is unlikely an asterisk issue, likely a networking issue |
| 18:32.57 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:33.14 | igcewieling | or a softphone issue, I assume you are using softphones |
| 18:38.19 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:41.50 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 18:43.24 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 18:43.35 | cjm_ | igcewieling, I agree, but I don't know why the distinction between three equivalent host designations. Standby for pastebin... |
| 18:44.12 | igcewieling | two, not three. the 10.1.1.13 address is different from the address of the other two |
| 18:46.24 | [TK]D-Fender | This isn't even an asterisk question.. this is a DNS question |
| 18:47.27 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 18:49.26 | cjm_ | igcewieling, Publicly, yes. But my LAN DNS translates to local IP addresses. |
| 18:49.32 | cjm_ | http://pastebin.com/5aq33tYW |
| 18:50.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 18:51.51 | cjm_ | [TK]D-Fender, Maybe... Asterisk shouldn't care if I use an A record or a CNAME record or an IP, right? But there seems to be a difference and I don't know what is causing the difference. |
| 18:51.53 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:52.21 | cjm_ | [TK]D-Fender, And asterisk is making the distinction... |
| 18:52.47 | [TK]D-Fender | That isn't SIP debug |
| 18:53.00 | [TK]D-Fender | I don't see linux CLI confriming what things should resolve as either |
| 18:53.46 | cjm_ | [TK]D-Fender, My appologies... Where do I find SIP debug? Is that a CLI request? |
| 18:53.51 | [TK]D-Fender | ... |
| 18:53.59 | [TK]D-Fender | [2013-07-18 11:36:45]sip:*43@Asterisk.TCLC.org succeeds |
| 18:54.01 | [TK]D-Fender | asterisk.tclc.org == 10.1.1.13 |
| 18:54.08 | [TK]D-Fender | next, your CAPITALS don't even match |
| 18:54.14 | [TK]D-Fender | -1 for inconsistency |
| 18:54.25 | [TK]D-Fender | and show proof from CLI first |
| 18:54.28 | cjm_ | Ha! Yes. But DNS is not case sensitive... |
| 18:54.37 | [TK]D-Fender | Still not caring yet |
| 18:54.47 | [TK]D-Fender | "sip set debug on" <--- |
| 18:55.01 | cjm_ | O.K., I know how to fire up CLI. Got it... |
| 18:55.47 | cjm_ | [TK]D-Fender, Done. Where will I collect the output? |
| 18:55.58 | [TK]D-Fender | is a PASTEBIN |
| 18:56.30 | cjm_ | [TK]D-Fender, Yes, I'll put what I find in a pastebin, but which log do I seek? |
| 18:57.00 | [TK]D-Fender | the CLI OUTPUT |
| 19:00.10 | cjm_ | [TK]D-Fender, How do I direct the output of CLI to a file? |
| 19:04.51 | Cuzner | copypasta |
| 19:05.34 | Cuzner | if you want to log the specific output of a CLI command you can "asterisk -rx '<command>' >> outputfile" |
| 19:05.55 | [TK]D-Fender | I tried to copy pasta once... I got a yeast infection... |
| 19:09.00 | *** join/#asterisk Rumbles (~Rumbles@cpc25-shef11-2-0-cust142.17-1.cable.virginmedia.com) |
| 19:09.06 | wolrah | What might cause a queue to think it's been answered when it hasn't? |
| 19:09.20 | wolrah | I have a queue for my support line that rings out to two cell phones if no one's in the office |
| 19:09.30 | wolrah | it has answer confirmation obviously |
| 19:09.55 | wolrah | but if I miss the first ring, more often than not attempting to answer on the second try results in the "this call has been answered elsewhere" message |
| 19:10.05 | wolrah | and then five seconds later it rings through to my phone again |
| 19:10.11 | wolrah | and repeat until the caller gives up |
| 19:10.33 | [TK]D-Fender | Show us actual call debug |
| 19:10.38 | [TK]D-Fender | ~pb |
| 19:10.38 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:10.39 | [TK]D-Fender | ^^^ |
| 19:11.27 | wolrah | I'm trying to find the starting point, unfortunately in my attempts to answer the call just now I made some call flow changes so the log is full of garbage from that |
| 19:12.28 | [TK]D-Fender | go place a new call then |
| 19:14.00 | *** join/#asterisk vlad_starkov (~vlad_star@213.152.135.29) |
| 19:16.21 | *** join/#asterisk peter (~peter@x2f00811.dyn.telefonica.de) |
| 19:17.48 | wolrah | got it, this is from the point the call hits the queue through the first "answered elsewhere" response: |
| 19:17.49 | wolrah | http://pastebin.com/Ld9vTb8s |
| 19:18.14 | wolrah | there was nothing else going on with the system at the time, so it should be all from this |
| 19:20.36 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.15) |
| 19:22.54 | [TK]D-Fender | [2013-07-18 14:58:05] <- placed |
| 19:23.01 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 19:23.06 | [TK]D-Fender | [2013-07-18 14:58:32] VERBOSE[4901] app_dial.c: -- SIP/Traceroutes-000027f6 answered Local/3303049337@from-queue-0a13;2 |
| 19:23.07 | [TK]D-Fender | answered |
| 19:23.17 | [TK]D-Fender | [2013-07-18 14:58:34] VERBOSE[4902] file.c: -- <SIP/Traceroutes-000027f8> Playing 'incoming-call-1-accept-2-decline.slin' (language 'en') |
| 19:23.18 | [TK]D-Fender | [2013-07-18 14:58:37] VERBOSE[4898] app_queue.c: -- Nobody picked up in 30000 ms |
| 19:23.19 | eirirs_ | dials [TK]D-Fender |
| 19:23.26 | [TK]D-Fender | Starts to play the prompts and DIES because it took too long to answer |
| 19:24.06 | wolrah | ok, it still wasn't answered though. |
| 19:24.22 | [TK]D-Fender | It was |
| 19:24.29 | wolrah | no, it went to voicemail |
| 19:24.31 | wolrah | on that phon |
| 19:24.38 | wolrah | that's the whole point of answer confirmation |
| 19:24.42 | wolrah | something picked up the line |
| 19:24.48 | [TK]D-Fender | Well the line technically answered |
| 19:24.50 | wolrah | but the call is not answered until someone presses 1 |
| 19:24.52 | [TK]D-Fender | Ad did not confirm |
| 19:25.03 | [TK]D-Fender | So what else are we supposed to see in there |
| 19:25.05 | wolrah | right, i guess that defeats the pirpose |
| 19:25.09 | wolrah | *purpose |
| 19:25.15 | [TK]D-Fender | I don't see "answered elsewhere" |
| 19:25.29 | [TK]D-Fender | Or anything else resembling an error |
| 19:25.37 | wolrah | the whole point of answer confirmation is so if one of the cells is out of service calls don't just end up going to the voicemail |
| 19:25.57 | [TK]D-Fender | That's fine |
| 19:26.14 | [TK]D-Fender | What failure am I supposed to be seeing in there? |
| 19:26.18 | wolrah | look at the last few lines, that's where I pick up and press 1 to try to grab the actual call |
| 19:26.35 | wolrah | it then has Allison tell me politely to f off, effectively |
| 19:26.47 | wolrah | but no actual human has yet answered the call |
| 19:27.23 | cjm_ | igcewieling, [TK]D-Fender Failed Call to IP address SIP debug: http://pastebin.com/Nz8pGqQr |
| 19:27.32 | wolrah | until a human presses 1, that call is NOT answered |
| 19:27.58 | wolrah | or well I guess not a human, but if one of my techs puts the DTMF for "1" in their VM greeting I'm going to beat them |
| 19:28.19 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 19:28.50 | wolrah | the failure is that answer confirmation is apparently not working properly, since it seems the queue counts anything hearing the confirmation prompt as "answered" |
| 19:30.23 | *** join/#asterisk peter_ (~peter@x2f00811.dyn.telefonica.de) |
| 19:31.28 | [TK]D-Fender | wolrah: I'm still not quite seeing it... |
| 19:32.12 | [TK]D-Fender | I see an "incoming call no longer available... |
| 19:32.38 | [TK]D-Fender | because the spawned channel may still exist.. but your queue timeout has killed its parent |
| 19:34.22 | [TK]D-Fender | Actually... |
| 19:34.33 | [TK]D-Fender | [2013-07-18 14:58:41] VERBOSE[4901] pbx.c: -- Executing [s@macro-confirm:5] Read("SIP/Traceroutes-000027f6", "INPUT,,1,,,4") in new stack |
| 19:34.45 | [TK]D-Fender | [2013-07-18 14:58:45] VERBOSE[4901] pbx.c: -- Executing [s@macro-confirm:6] GotoIf("SIP/Traceroutes-000027f6", "0?,1:t,1") in new stack |
| 19:34.54 | [TK]D-Fender | 4 seconds of no confirmation from the look of it |
| 19:35.13 | [TK]D-Fender | [2013-07-18 14:58:45] VERBOSE[4901] app_read.c: -- User entered nothing. |
| 19:35.48 | [TK]D-Fender | So if you're saying you entered something... perhaps your DTMF settings are bad.... |
| 19:37.22 | [TK]D-Fender | [2013-07-18 14:58:04] VERBOSE[4901] pbx.c: -- Executing [3303049337@from-queue:1] Set("Local/3303049337@from-queue-0a13;2", "QAGENT=3303049337") in new stack |
| 19:37.30 | *** join/#asterisk micols (~t@81.161.188.225) |
| 19:37.31 | [TK]D-Fender | Curiously right about the 30s mark |
| 19:38.04 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 19:38.35 | *** join/#asterisk serafie (~erin@206.255.84.113) |
| 19:39.11 | wolrah | I have "agent timeout" set to 30 seconds, if the cell network was to be slow for whatever reason and it's 31 seconds after Asterisk thinks it's sent the call would it make sense for this to happen? |
| 19:39.28 | igcewieling | cjm_: You have an odd problem. What is 10.1.1.44? |
| 19:40.10 | igcewieling | [TK]D-Fender: Pastebin! 8-) |
| 19:40.43 | [TK]D-Fender | igcewieling: I cherry picked and did those individually :) |
| 19:40.46 | *** join/#asterisk random_cat (~random_ca@gateway/tor-sasl/randomcat/x-49498005) |
| 19:41.08 | [TK]D-Fender | [15:39]wolrahI have "agent timeout" set to 30 seconds, if the cell network was to be slow for whatever reason and it's 31 seconds after Asterisk thinks it's sent the call would it make sense for this to happen? <- time's up = time's up |
| 19:41.29 | igcewieling | [TK]D-Fender: you have enough karma you could never use pastebin |
| 19:41.50 | [TK]D-Fender | :p |
| 19:42.16 | wolrah | ok, will try bumping that up to 60 seconds and see if that helps. The goal on 30 seconds was to ring the remote phones more frequently if they're not answered, maybe that was ill advised |
| 19:43.55 | wolrah | cell network is certainly full of odd delays, often I'll be driving with another tech when a call comes through and my phone can start ringing 10-20 seconds earlier than theirs, even though we're on the same network |
| 19:44.54 | wolrah | wishes for a day where cellular voice as a standalone thing is in the past and we're just running SIP+RTP over LTE |
| 19:45.03 | [TK]D-Fender | So what you;re saying is ... you should have known this already :) |
| 19:45.29 | wolrah | I didn't put two and two together |
| 19:45.29 | wolrah | so yea, derp on my part |
| 19:45.34 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 19:45.34 | wolrah | forest, trees, and such |
| 19:47.02 | wolrah | i thought that was solely the ring timer, and that as long as the physical pickup of the line happened within that time one wouldn't have to rapid-fire the 1 button |
| 19:56.58 | *** join/#asterisk dtascom (~dtascom@cpe-142-129-156-205.socal.res.rr.com) |
| 19:59.56 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.15) |
| 20:02.11 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 20:02.34 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 20:03.21 | *** join/#asterisk nix8n82 (~AndChat27@161.sub-97-36-157.myvzw.com) |
| 20:08.11 | *** join/#asterisk peter__ (~peter@x2f00811.dyn.telefonica.de) |
| 20:11.02 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:11.39 | *** join/#asterisk module000 (~module000@CEDAR-CREEK.edge9.Dallas1.Level3.net) |
| 20:12.35 | module000 | Is Flite the standard text-2-speech module for asterisk? Googling shows a few available, but flite had the most page hits. Anything other than flite good to take a look at? (this is for an IVR) |
| 20:15.55 | _Corey_ | module000: We usually use Cepstral |
| 20:16.28 | module000 | _Corey_: thanks, going to check that one out now |
| 20:20.39 | [TK]D-Fender | Google's TTS API is pretty good if you don't mind the online requirement |
| 20:23.31 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 20:25.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:25.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:33.45 | *** join/#asterisk peetaur2 (~peter@x2f00811.dyn.telefonica.de) |
| 20:34.47 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:34.48 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:36.31 | *** join/#asterisk mwally (~mwally@mwally.hbx.us) |
| 20:36.53 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:37.13 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 20:37.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.234) |
| 20:41.31 | *** join/#asterisk Rumbles (~Rumbles@cpc25-shef11-2-0-cust142.17-1.cable.virginmedia.com) |
| 20:55.59 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mcqrayziwwmxjgls) |
| 21:02.48 | *** join/#asterisk russum (~russum@92-62-22-26.customer.bnet.at) |
| 21:03.32 | *** part/#asterisk russum (~russum@92-62-22-26.customer.bnet.at) |
| 21:07.00 | *** join/#asterisk navaismo (~navaismo@189.191.233.140) |
| 21:11.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:15.56 | *** join/#asterisk lvlolvlo_ (~lvlolvlo@unaffiliated/lvlolvlo) |
| 21:32.05 | *** join/#asterisk guitarHester (~hester@nat/digium/x-dppuaacsiafcrrsu) |
| 21:37.53 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 21:48.19 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:53.01 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-30.usfamily.net) |
| 22:00.34 | navaismo | Hello guys. Is there a way to count the number o persons in a call? Without Meetme or Confbridge. |
| 22:02.42 | *** part/#asterisk dtx (~dtx@cdf-imaging.com) |
| 22:02.46 | lorsungcu | anyone know offhand if dtmf is logged in CEL, or if it's possible to have it logged? |
| 22:12.39 | WIMPy | navaismo: 2 if in a successfull Dial(), otherwise 1. |
| 22:12.50 | navaismo | ¬¬ |
| 22:13.55 | navaismo | how about a conference phone in one of the sides? Or a 3-way call generate by remote end? Just saying... |
| 22:14.37 | WIMPy | That's not a 3-way call. It's two normal calls mixed together at one end. |
| 22:15.10 | WIMPy | Not different from someone using two phones on speakerphone side-by-side. |
| 22:15.28 | navaismo | guess can't explain/communicate better |
| 22:15.44 | WIMPy | Back in the days of the PSTN, there was a notification for it. |
| 22:20.52 | pigpen | this is stupid, but how to you get in.tftp to log to syslog? |
| 22:23.14 | _Corey_ | pigpen: Add -v to the command line |
| 22:23.25 | _Corey_ | If you're using init.d you'd put it in there |
| 22:23.25 | pigpen | I have 10 -v |
| 22:23.30 | _Corey_ | hmmm |
| 22:23.34 | pigpen | yeh. |
| 22:23.37 | pigpen | yeah. weird. |
| 22:23.45 | WIMPy | Did you restart inetd? |
| 22:24.09 | *** join/#asterisk Imposs (d11f36e2@gateway/web/freenode/ip.209.31.54.226) |
| 22:24.13 | _Corey_ | when I said init.d, I obviously meant xinetd.d :) |
| 22:24.27 | _Corey_ | here's one I have that logs: |
| 22:24.28 | _Corey_ | server_args = -s /tftpboot -v -v |
| 22:24.43 | Imposs | has anyone ever installed app_cbmysql |
| 22:24.57 | Imposs | I have built asterisk from src ver 11.5 |
| 22:24.57 | pigpen | yeah, I don't have any ftp, tftp or such |
| 22:25.06 | pigpen | using in.tftp and syslog-ng |
| 22:25.23 | _Corey_ | well, this is: |
| 22:25.25 | _Corey_ | server = /usr/sbin/in.tftpd |
| 22:25.28 | pigpen | vsftpd logs to it's own. |
| 22:25.45 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 22:26.01 | Imposs | when I try to make cbmysql I get make: *** [app_cbmysql.o] Error 1 |
| 22:26.13 | pigpen | yeah, I need to google syslog-ng |
| 22:26.48 | navaismo | Imposs, answere in freepbx channel |
| 22:27.00 | navaismo | the real error is above that line |
| 22:30.25 | pigpen | I have a cyberdata speaker that doesn't look like it is grabbing it's config |
| 22:30.28 | pigpen | bastard. |
| 22:31.41 | pigpen | tcpdump seems to be in my future |
| 22:48.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.181) |
| 23:02.36 | *** join/#asterisk serafie (~erin@206.255.84.113) |
| 23:12.32 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 23:21.36 | *** join/#asterisk andrewyager (~andrewyag@8-104-141-114.static-dsl.realworld.net.au) |
| 23:24.30 | *** join/#asterisk stevetodd (~blurryrun@199.87.121.1) |
| 23:27.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 23:28.08 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 23:28.34 | stevetodd | is there an easy way to mark a channel in one call and find the marked channel in another call? |
| 23:29.11 | WIMPy | Maybe the GROUP* functions can help you. |
| 23:29.25 | WIMPy | But maybe you should tell us what exactely you want to do. |
| 23:29.54 | igcewieling | stevetodd: I think Asterisk 11 has something like that, but I can't remember what it would be called. Try https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation |
| 23:30.50 | stevetodd | WIMPy: thanks for your response. Basically I'm doing a cheap call park. A calls B. B call park extension and transfers A there. C calls call park pickup extension and gets A. |
| 23:31.39 | WIMPy | But you set (or get) an index there already. |
| 23:31.55 | stevetodd | while I can store the channel name when B calls call park, because of the transfter the channel name can be either A's or B's. |
| 23:31.55 | WIMPy | I guess I don;t understand the question. |
| 23:33.53 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 23:35.20 | stevetodd | sorry, that was confusing. A calls B. B calls an extension that just plays hold music and stores the channel name in something like astdb. B transfers his call with A to his call with the hold music. C calls a retrieval extension that looks in the db for the channel name and use bridge to bridge the call. |
| 23:36.09 | stevetodd | this works about half the time |
| 23:36.29 | stevetodd | because of the transfer, it is a 50/50 chance that i stored the wrong channel name |
| 23:36.30 | WIMPy | Whay don;t you just use the parking stuff? |
| 23:36.44 | stevetodd | multiple customers on the same pbx |
| 23:37.25 | WIMPy | That should work, IIRC? |
| 23:37.37 | WIMPy | I never understood the concept of parking anyway. |
| 23:37.46 | WIMPy | Transfers have been invented. |
| 23:38.32 | stevetodd | transfers don't fit all use cases |
| 23:38.53 | stevetodd | for example, you don't know where to transfer the call to |
| 23:39.05 | stevetodd | but you have a PA system |
| 23:42.10 | navaismo | stevetodd, IIR the features.conf have a method to create another parking lot different context |
| 23:42.11 | robl^ | you can still used the built-in parking, if you handle it correctly. You don't have to to use the autoparking -- i.e. transfering to 700 and then wait for Allison to tell you where it is parked. You can park to a specific extension. UIf you are careful in dial-plan construction, you can keep one customer from picking up calls for other customers |
| 23:46.39 | robl^ | navaismo: correct. I think multiple parking lots were added in Asterisk 1.6 era. |
| 23:46.43 | stevetodd | navaismo, robl^: Thanks, I'll take another look. I'm actually doing this all in AGI and am looking to go beyond the standard asterisk feature set for this if I can. Really the only thing that is stopping me is finding the channel. |
| 23:46.51 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 23:47.57 | stevetodd | I may just end up using a meetme conference for it, though I'd rather not go down that route. |
| 23:48.55 | WIMPy | If you were using AMI you could just redirect the channel. |
| 23:49.07 | robl^ | stevetodd: just saying there are multiple solutions for your functional need. Asterisk has some built-in, which is worth taking a look at before re-inventing the wheel |
| 23:49.30 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 23:51.01 | stevetodd | robl^: hey thanks. I've got some requirements I can't bring up here that make that a little less feasible, but I appreciate your responses here. I've gtg, thanks all! |
| 23:52.29 | eirirs_ | re-invents the sleep |
| 23:53.08 | navaismo | stevetodd, my idea is try saving the Company+CID+UNIQUEID(or something like that) to astdb then when you want to pickup dial a code(previously defined in your dialplan) for the company and list the company-* whn you get the desired press 1 save the lastname and redirect |
| 23:53.14 | navaismo | LOL too much work |
| 23:53.47 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |