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| 01:40.39 | scoobysnack | Are the headings in sip.conf functional in any way. For instance if I group sip devices in one section called [students] and another section called [faculty] ? |
| 01:41.48 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 01:41.56 | Penguin | Yep, they are significant. |
| 01:42.21 | Penguin | That can be used for either a sip device or as a template if you mark it as such. |
| 01:42.58 | Penguin | If you don't mark it as a template, it will be a device named 'students'. |
| 02:02.30 | scoobysnack | Is there a way to group a lot of sip devices together--a sort of label for a lot of them? |
| 02:02.41 | *** join/#asterisk Katty (~Angela@68-184-14-250.dhcp.stls.mo.charter.com) |
| 02:02.48 | Penguin | What do you mean by "group" them? |
| 02:05.19 | scoobysnack | I guess I could just comment the groups with semi-colons |
| 02:05.32 | scoobysnack | ;;;;;;;;;;;;;;;;;;;;;;;Students;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; |
| 02:05.38 | scoobysnack | ;;;;;;;;;;;;;;;;;;;;;;;Faculty;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; |
| 02:05.41 | scoobysnack | that kinda thing |
| 02:07.12 | scoobysnack | I was just wanting an easy way to identify what group a device belonged to in my organization. |
| 02:07.35 | scoobysnack | This is done fucntionally with contexts in extensions.conf |
| 02:09.49 | Katty | eyes scoobysnack with a fork |
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| 02:14.30 | Penguin | It would be purely a human concept. Although you can define a group by way of variable, and call it in dial plan if that's the sort of thing you need to do. |
| 02:14.54 | file | yawns |
| 02:17.30 | Katty | hi file |
| 02:17.34 | Katty | how's your room |
| 02:20.40 | file | acceptable! |
| 02:20.44 | file | the wifi isn't annoying me |
| 02:20.51 | Katty | that's a miracle. |
| 02:22.30 | file | what's up in here this evening? |
| 02:25.29 | Katty | no idea. |
| 02:25.39 | Katty | i'm cooking sausage and eggs, tho. how very exciting? |
| 02:29.08 | *** join/#asterisk mae_tae (7960eecf@gateway/web/freenode/ip.121.96.238.207) |
| 02:29.14 | mae_tae | hello people... |
| 02:30.43 | mae_tae | hello people, i am getting "service unavailable 503" error when i softphone to softphone or grandstream phone to softphone but other way works fine, how do i fix this |
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| 03:12.16 | mae_tae | <PROTECTED> |
| 03:12.30 | mae_tae | i dont have idea on how to fix this |
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| 03:20.28 | Penguin | Pastebin the entire sip debug of that call. |
| 03:20.42 | Penguin | mae_tae: ^ |
| 03:20.56 | jploh | hi tae :D, can you pastebin the other messages near that line? |
| 03:21.17 | mae_tae | ok |
| 03:23.22 | mae_tae | Penguin: do i need to paste the 500 lines |
| 03:23.35 | Penguin | Pastebin it. All of it. |
| 03:26.19 | mae_tae | http://pastebin.com/n0XrU9ew <--- here's the link |
| 03:27.05 | Penguin | That is not a sip debug. |
| 03:28.03 | Penguin | I want a full sip debug of the problem you described, where one phone makes a call to the other phone and it fails. |
| 03:28.22 | jploh | mae_tae: type sip set debug on at the console |
| 03:30.33 | mae_tae | ok, ill try to remote |
| 03:41.25 | mae_tae | http://pastebin.com/hVb1wxxa |
| 03:41.45 | mae_tae | Penguin: that's the result of sip set debug peer 108 |
| 03:42.30 | Penguin | That's 13 lines. This isn't what I asked for. |
| 03:42.59 | mae_tae | sorry im really new to asterisk, im the one that install this |
| 03:43.53 | Penguin | I'll tell you specifically one more time. Pastebin the ENTIRE SIP DEBUG of the failed call. |
| 03:43.57 | Penguin | Nothing less. |
| 03:44.07 | mae_tae | ok |
| 03:44.46 | Penguin | sip set debug on |
| 03:45.04 | Penguin | Make the call, which you anticipate will end in failure. |
| 03:45.07 | Penguin | sip set debug off |
| 03:45.16 | Penguin | Copy all of it, paste to the pastebin. |
| 03:45.22 | mae_tae | ok |
| 03:57.54 | mae_tae | Penguin: http://pastebin.com/9JmBmCcv |
| 04:00.32 | Penguin | I don't see anything about 108 in here at all. Seems like your phone didn't make call to asterisk. |
| 04:01.41 | mae_tae | actually if call from our grandstream to other grandstream phone, it works very fine but im going to call to softphone... it will not work |
| 04:04.13 | mae_tae | its only a problem with softphone to softphone or grandstream to softphone |
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| 05:11.56 | mae_tae | how do i fix this "service unavailable - 503" |
| 05:12.17 | mae_tae | its the result of calling a phone device to softphone |
| 05:13.07 | ChannelZ | dunno. Fix why the device is unavailable. |
| 05:13.14 | ChannelZ | It could be a dozen different things. |
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| 05:47.25 | ChannelZ | FFffffuuuuuu I almost forgot about Homeland tonight |
| 06:01.35 | *** join/#asterisk crumb (crumb@gateway/shell/cadoth.co/x-ukygoymstbovpyhh) |
| 06:01.47 | crumb | hey, quick questions about phones |
| 06:02.34 | crumb | let's say you have a phone set with a landline, and you want to add a voip line on the same line, would that work? |
| 06:03.54 | crumb | 1 phone with 1 landline + 1 line from ATA |
| 06:04.31 | crumb | or would you need some kind of switch |
| 06:05.09 | crumb | man those look expensive |
| 06:05.36 | crumb | is there an ATA with a switch built in |
| 06:05.48 | ChannelZ | I presume you're asking how to use the landline in a voip system (like asterisk).. in which case you need an ATA |
| 06:05.53 | crumb | with something like a landline pass-thru |
| 06:06.12 | crumb | i'm talking about two independent lines |
| 06:06.32 | ChannelZ | Well voip is over the internet.. so that "line" is your internet connection, whatever that is. |
| 06:06.33 | crumb | one from over copper landline |
| 06:06.51 | crumb | another through the ATA |
| 06:08.25 | ChannelZ | An ATA like the SPA3102 would let you connect an analog phone, and/or an analog line to your system. |
| 06:08.35 | crumb | cool! |
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| 06:08.54 | crumb | still manufactured? |
| 06:08.58 | ChannelZ | It shows up as 2 SIP devices |
| 06:09.18 | ChannelZ | As far as I know. It's Cisco now (was Linksys) |
| 06:09.35 | crumb | $64 bucks.. pricey :/ |
| 06:09.52 | crumb | is the PAP2 also capable of passing through analog? |
| 06:10.45 | ChannelZ | the PAP2 is only for handsets IIRC |
| 06:11.06 | crumb | pap2 looks just like the spa3102 |
| 06:11.34 | ChannelZ | So? They have like 4 different boxes that all look alike |
| 06:11.54 | crumb | what do you mean by handsets |
| 06:12.01 | crumb | they have 2 fxs ports |
| 06:12.46 | ChannelZ | Yes. FXS = Connects to a handset (an analog phone) FXO = Connects to a telephone line |
| 06:13.00 | ChannelZ | The 3102 has one of each. I'm still unclear what you're actually trying to accomplish |
| 06:13.31 | crumb | i want both sip and analog voice capability from a single phone receiver |
| 06:14.58 | crumb | ok, seems the grandstream ht-386 has a two fxs ports as well with pstn pass-through capability |
| 06:15.30 | ChannelZ | So the 3102 would do that. It can pass your analog telephone line to your analog handset, and do VoIP. Although if you want to do it direct through an ITSP I'm not sure off-hand how you'd choose how to dial out.. |
| 06:16.41 | crumb | i'll go with the cheaper ata :/ |
| 06:16.42 | ChannelZ | It looks to me like the HT-386 is 2 FXS only. |
| 06:16.50 | crumb | oh |
| 06:16.54 | crumb | http://www.asteriskguru.com/tutorials/grandstream_handytone_386_configuration.html |
| 06:17.44 | ChannelZ | "Grandstream HandyTone-386 is a multi-port, all-in-one, Dual FXS Analog Adapter"... right on the first line |
| 06:17.55 | ChannelZ | The 488 has one of each apparently |
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| 06:19.29 | ChannelZ | That might be old/discontinued models. From Grandstream's appalling website, there's this: http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht503 |
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| 06:29.34 | RudyValencia | Hi, I'm trying to set up Asterisk with a SIP proxy. Whenever we try to call an extension defined in the Asterisk dialplan we either get OK or silence. What could be happening? |
| 06:32.42 | ppc | RudyValencia: i'd try using a softphone first to see if it's the phone or asterisk |
| 06:33.12 | RudyValencia | ppc: We are using Blink as our preferred softphone. |
| 06:33.39 | ppc | does it work at all? |
| 06:34.01 | robert_ | nope. |
| 06:34.44 | RudyValencia | robert_ is working on it with me. |
| 06:34.50 | robert_ | indeed. |
| 06:35.02 | ppc | fyi im not expert by any means so don't expect me to get it working |
| 06:35.14 | robert_ | yeah, I know. |
| 06:35.31 | ppc | so do any of the extensions work? |
| 06:35.38 | robert_ | no |
| 06:35.43 | ppc | one server? |
| 06:35.46 | robert_ | we have a mock hold line set up |
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| 06:35.47 | robert_ | yes. |
| 06:36.03 | robert_ | even calling amongst ourselves doesn't work. |
| 06:36.07 | ppc | did you look at the asterisk console? |
| 06:36.14 | ppc | w/ verbosity turned up |
| 06:37.17 | robert_ | yes |
| 06:37.27 | ppc | not throwing any errors or anything odd? |
| 06:37.34 | robert_ | it answers with the macro and then simply starts and stops. |
| 06:38.52 | ppc | robert_: copy/paste whats in the log to pastebin or something |
| 06:39.06 | robert_ | the log? |
| 06:39.12 | robert_ | as in syslog or the console? |
| 06:39.17 | ppc | the asterisk console |
| 06:40.02 | ppc | I think you do asterisk -rvvvvv |
| 06:40.13 | robert_ | http://codepad.org/fniel78g |
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| 06:41.37 | ppc | robert_: how about the dialplan |
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| 06:42.35 | robert_ | ppc: http://codepad.org/85h2Fvyt |
| 06:43.47 | ppc | robert_: you have the extensions set to be 2000 and 3000? |
| 06:44.08 | robert_ | yeah. |
| 06:44.40 | ppc | robert_: i dont think you can set extensions to be 2000/3000/4000/5000 |
| 06:44.53 | ppc | I think it needs to be like 2001/3001/4001 |
| 06:44.55 | robert_ | we had it working on ast before. |
| 06:45.05 | robert_ | (standalone) |
| 06:45.54 | RudyValencia | we decided to add a SIP proxy because sometimes inbound calls would route directly to voicemail |
| 06:46.47 | RudyValencia | and occasionally ast crashed, taking down our softphone connections temporarily (until we restarted ast) |
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| 07:01.47 | StaRetji | hello everyone, hope you had nice weekend |
| 07:02.41 | StaRetji | I need asterisk guru to hire as freelancer, need to resolve issue with hangupcause |
| 07:03.28 | StaRetji | i am kind of in trouble, so i will pay for help |
| 07:04.09 | StaRetji | i've read wiki, but couldn't find solution |
| 07:06.14 | StaRetji | i am playing ivr to client, before i send them to another context, but Master.csv shows it as answered, even though i send Playback(enterthepin,noanswer) |
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| 07:16.54 | RudyValencia | Can anyone suggest a sample dialplan that provides a basic PBX with simple IVR routing, voicemail and access to flowroute? |
| 07:23.53 | StaRetji | is there way to force hangupcause in a context |
| 07:24.25 | StaRetji | if line is lost to send Hangup(34) |
| 07:24.29 | StaRetji | ? |
| 07:29.40 | StaRetji | it looks like i can control hangupcause when i hangup, but if user hangsup i cant control it |
| 07:31.48 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 07:32.43 | PLMg | hey, anyone willing to give a hand in forwarding calls from asterisk to freeswitch? |
| 07:33.13 | RudyValencia | would http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb still apply to Asterisk 11.5.1 and Kamailio 4.0.2 ? |
| 07:34.42 | StaRetji | ok, i need to know if ivr and dtmf works in earlymedia? please, anyone? |
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| 07:54.38 | PLMg | can anyone help with finding the location inside extensions.conf where calls are being handeld? |
| 07:55.04 | mae_tae | whois jplog |
| 08:03.10 | ChannelZ | PLMg: that depends on the peer and what context the calls are occuring in. |
| 08:04.56 | RudyValencia | Any suggestions for a sample dial plan that provides PBX services, a simple IVR and calling to/from an online PSTN gateway? |
| 08:05.57 | ChannelZ | That's like asking "how long is a piece of string" |
| 08:06.39 | ChannelZ | "a simple IVR" that does what? What PSTN gateway? What "PBX services?" |
| 08:07.12 | ChannelZ | It's not really a copy/paste thing. |
| 08:07.13 | PLMg | ChannelZ I have this: Route the calls to the gateway. In /etc/asterisk/extensions.conf context where your calls are being handled, forward the calls to the gateway. Here, when someone dials 85001, the call is sent to the fs-gw defined above. |
| 08:07.30 | RudyValencia | We just want something like, "for sales press 1, for support press 2, ..." |
| 08:07.47 | PLMg | now, where are my calls being handeld :) |
| 08:07.47 | ChannelZ | ~primer |
| 08:07.48 | infobot | hmm... primer is http://burner.com/asterisk-primer |
| 08:07.50 | RudyValencia | as for the services, just stuff that you take for granted on traditional phones |
| 08:08.00 | RudyValencia | and we use flowroute for a gateway |
| 08:08.43 | PLMg | ia have to enter exten => 85001,1,Dial(SIP/fs-gw/${EXTEN}) |
| 08:08.44 | PLMg | exten => 85001,2,Hangup but I do not know exactly where |
| 08:09.45 | ChannelZ | They will be under a [context] |
| 08:11.24 | ChannelZ | If you don't understand dialplan contexts and how devices enter the dialplan, you've got to start with the basics.. |
| 08:12.08 | PLMg | I do have a grasp on them but I am confused in regards to [context] I am guessing I have to enter them somewhere in outbound |
| 08:12.25 | PLMg | as in dial out |
| 08:13.01 | kaldemar | PLMg: who/what wrote your dialplan? |
| 08:13.28 | PLMg | freepbx |
| 08:13.45 | PLMg | but I will manualy enter this and it will not get overwriten by freepbx |
| 08:13.52 | ChannelZ | A device -- the phone you pick up and dial -- has a set context, where it enters the dialplan when you dial things. |
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| 08:14.12 | kaldemar | you better ask in #freepbx for the modification without it writing your changes over. |
| 08:14.45 | PLMg | its fine, freepbx will not change my modifications |
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| 08:15.43 | PLMg | so do I just put my 2 lines between [macro-outisbusy] and [macro-dialout]? It is just an example |
| 08:16.31 | kaldemar | ask in #freepbx. they will know better. |
| 08:17.03 | PLMg | ok, will do |
| 08:17.08 | PLMg | thx for the help anyway :) |
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| 09:42.38 | pzn | There is an "operator panel" that works with a proprietary PBX. I need to integrate it with asterisk and making the operator panel manage proprietary+asterisk in the migration window (a few weeks). how can I get real-time call status (ring, answer, ...) from asterisk? |
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| 10:10.18 | StaRetji | lost internet connection, so I will have to ask again, sorry folks |
| 10:13.54 | StaRetji | I need help, I have ivr with exten => _X.,n(ivrauth1),Playback(eng1,noanswer) |
| 10:14.50 | StaRetji | http://pastebin.com/fgKTyUzC |
| 10:15.11 | StaRetji | however, calls show up in Master.csv as answered |
| 10:15.43 | StaRetji | I am trying to produce earlymedia Playback and read |
| 10:16.04 | StaRetji | is it possible? not to bill asterisk ivr? |
| 10:21.16 | StaRetji | Could this line be the problem? exten => _X.,n,Read(digits,,1,30) |
| 10:23.04 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 10:23.07 | StaRetji | hm maybe should be done like this: exten => _X.,n,Read(digits,,1,noanswer) |
| 10:24.23 | kaldemar | you can't just come up with options and expect them to work. app Read does not have a noanswer option. Read(digits,,1,30) <-- and your timeout is in the wrong place there. |
| 10:24.41 | StaRetji | yes, I understand |
| 10:24.46 | StaRetji | 30 wqs on wrong place |
| 10:25.11 | StaRetji | so, exten => _X.,n,Read(digits,,1,noanswer) this wont work? |
| 10:25.32 | kaldemar | core show application Read |
| 10:26.50 | StaRetji | thx, kaldemar, howerver, I was desperate and changed and reloaded |
| 10:27.17 | StaRetji | it seems ivr calls are now "NO ANSWER" |
| 10:28.44 | StaRetji | of course, core show application Read says oposite |
| 10:29.14 | kaldemar | maybe n does that. you're now feeding options n, o, a, s, w, e and r to it. |
| 10:29.29 | kaldemar | s, i and n being valid options. |
| 10:30.21 | StaRetji | yes |
| 10:30.22 | StaRetji | hm |
| 10:30.39 | StaRetji | mate, kaldemar, thx so much |
| 10:30.48 | StaRetji | n: to read digits even if the line is not up. |
| 10:32.06 | kaldemar | a peek at the source tells that without n the application answers the channel. |
| 10:32.17 | StaRetji | my GOD |
| 10:32.31 | StaRetji | i am stupid, but luckili noanswer starts with n |
| 10:32.32 | StaRetji | hahahahha |
| 10:32.35 | StaRetji | it seems it work :) |
| 10:32.42 | kaldemar | too bad the app documentation does not say it. |
| 10:32.44 | StaRetji | I will change noanswer to n and test |
| 10:33.32 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:1c62:94ec:e289:8b53) |
| 10:34.10 | *** join/#asterisk jploh (~textual@49.144.39.51) |
| 10:34.30 | StaRetji | exten => _X.,n,Read(digits,,1,n) |
| 10:34.33 | StaRetji | yep, works ;) |
| 10:34.53 | StaRetji | kaldemar: your like lucky medallion :) |
| 10:34.54 | StaRetji | thx |
| 10:36.14 | kaldemar | np |
| 10:44.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.70) |
| 10:51.04 | *** part/#asterisk StaRetji (~LittleAll@178.79.6.17) |
| 10:57.13 | *** join/#asterisk kleszcz (~tick@linuxmafia.pl) |
| 11:04.05 | *** join/#asterisk StaRetji (~LittleAll@178.79.6.17) |
| 11:04.30 | StaRetji | kaldemar: I have another problem, it seems clients have problem with dtmf tones not being recognized |
| 11:04.52 | StaRetji | I've set dtmfmode=auto |
| 11:05.23 | StaRetji | cause it seems it wont work with dtmfmode=rfc2833 |
| 11:05.30 | StaRetji | any tip? thanks |
| 11:06.03 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 11:09.44 | kaldemar | StaRetji: change to mode. |
| 11:13.24 | StaRetji | change to mode, what that means mate? |
| 11:13.26 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-opsmipabmwfuseuk) |
| 11:14.16 | kaldemar | StaRetji: i meant change THE mode. typo. |
| 11:14.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.70) |
| 11:14.59 | StaRetji | well, I tried auto |
| 11:15.24 | StaRetji | but thy said pressing digits does nothing, indeed, cli says User entered nothing |
| 11:15.35 | StaRetji | I've set to rfc2833 to test again |
| 11:15.44 | kaldemar | there are others too. |
| 11:16.59 | StaRetji | inband says it works only on ulaw alaw, there is info that remains |
| 11:17.32 | StaRetji | I have sip to sip clients, but some of them have handset clients :/ |
| 11:27.35 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xylpyvrbhnkwdnbs) |
| 11:35.52 | StaRetji | it seems that for some works, for some no, so I have no idea, I thought auto will be best choice |
| 11:41.55 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 11:56.31 | *** join/#asterisk din3sh (2988feba@gateway/web/freenode/ip.41.136.254.186) |
| 11:56.57 | din3sh | hey all |
| 11:57.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:57.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:59.20 | din3sh | hey all |
| 11:59.51 | din3sh | is it possible to trigger a 3-way conference from a SIP phone and transfer that conference call to another phone? |
| 12:00.03 | din3sh | What is needed for such a scenario to work? |
| 12:03.10 | *** part/#asterisk StaRetji (~LittleAll@178.79.6.17) |
| 12:04.15 | *** join/#asterisk viasanctus (~viasanctu@unaffiliated/viasanctus) |
| 12:10.40 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:10.40 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:17.46 | *** join/#asterisk Weezey (~ohno@199.115.91-75.ip.mltn.standardbroadband.ca) |
| 12:18.57 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 12:20.00 | WIMPy | din3sh: On, not from. |
| 12:20.30 | *** join/#asterisk Rico (~rico@unaffiliated/rico29) |
| 12:20.31 | din3sh | yeah on not from :P |
| 12:20.33 | Rico | hi ther |
| 12:20.36 | Rico | e |
| 12:21.10 | Weezey | where? |
| 12:21.15 | Rico | here |
| 12:21.31 | Weezey | oh |
| 12:23.27 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:24.37 | Rico | I have a problem with cdr_odbc, I can't find why it does not work : http://pastebin.com/K23kb6v1 |
| 12:24.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 12:24.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 12:26.12 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 12:29.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:34.27 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 12:38.35 | Rico | for persons who just arrived : |
| 12:38.36 | Rico | I have a problem with cdr_odbc, I can't find why it does not work : http://pastebin.com/K23kb6v1 |
| 12:38.47 | Rico | if somebody can take a look... thanks |
| 12:39.30 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:40.19 | *** join/#asterisk rolandow (~roland@s559700f1.adsl.online.nl) |
| 12:41.15 | rolandow | hi, good afternoon! |
| 12:41.28 | rolandow | i have this in my debug log: [2013-09-30 13:37:38] DEBUG[843][C-000018d9] chan_sip.c: Got redirecting from number +31630597085 |
| 12:41.40 | rolandow | though the ${CALLERID(num)} gives me Anonymous |
| 12:42.00 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-ldvwuorzftqyrjnu) |
| 12:42.11 | rolandow | isn't that weird? |
| 12:46.29 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
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| 13:07.03 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 13:12.07 | *** join/#asterisk Katty (~Angela@68-184-14-250.dhcp.stls.mo.charter.com) |
| 13:12.21 | Katty | morning |
| 13:14.03 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:14.17 | Dovid | hi. does anyone know if I can set outboundproxy via the dial plan ? |
| 13:19.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.70) |
| 13:22.19 | [TK]D-Fender | Nope |
| 13:22.37 | *** join/#asterisk serafie (~erin@nat/digium/x-fqdkshvpwndbcpxc) |
| 13:26.57 | *** join/#asterisk ghost75_ (~quassel@dslb-088-066-187-096.pools.arcor-ip.net) |
| 13:27.28 | ghost75_ | is there any way to see what is killing my * process? |
| 13:28.56 | wdoekes | that depends on the type of killing |
| 13:29.29 | wdoekes | is it the kernel (out-of-memory, segfault, sigbus)? |
| 13:29.43 | ghost75_ | hmmm |
| 13:29.59 | wdoekes | is it overloaded, but not killed? |
| 13:30.22 | Penguin | Are we talking cold-blooded murder, or a mercy killing? |
| 13:30.36 | wdoekes | does someone have AMI or other access and is he explicitly shutting asterisk down? |
| 13:31.24 | ghost75_ | dont think so |
| 13:31.30 | wdoekes | ghost75_: the kernel will list the reason in dmesg and/or kern.log |
| 13:32.12 | ghost75_ | [1108047.974460] asterisk[3220]: segfault at db80ce3c ip b665819e sp b56ce8b0 error 5 in libspandsp.so.2.0.0 [b6637000+b4000] |
| 13:32.51 | wdoekes | there you have it.. it's self-defence |
| 13:33.29 | wdoekes | *se |
| 13:33.51 | bacobart | no |
| 13:33.53 | Qwell | wdoekes: I like it. "My code didn't crash, it happened in self defense" |
| 13:33.53 | bacobart | defence is the correct way |
| 13:34.11 | bacobart | defense is burger king english |
| 13:34.17 | ghost75_ | isnt spandsp for fax? |
| 13:35.11 | eirirs_ | hehe |
| 13:35.17 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 13:37.55 | wdoekes | ok bacobart, I stand corrected |
| 13:39.07 | wdoekes | but yes, ghost75_, spandsp is fax indeed |
| 13:39.37 | ghost75_ | libspandsp resides in /usr/lib |
| 13:39.43 | wdoekes | recompile spandsp with debug symbols if you want to find out why it crashed |
| 13:39.45 | ghost75_ | shouldnt be in /usr/lib/asterisk ? |
| 13:40.00 | wdoekes | no.. it's not a part of asterisk, it is used by asterisk |
| 13:42.12 | ghost75_ | can i exclude it in modules.conf? |
| 13:42.34 | wdoekes | you surely can |
| 13:42.43 | wdoekes | res_fax_spandsp.so |
| 13:45.57 | rolandow | if i see this in my logs: Got redirecting from number +31630597085 .. would this be the calier id? |
| 13:50.56 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:50.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.41 | *** join/#asterisk scoobysnack (~aaffcc002@gateway/tor-sasl/aaffcc0022ee) |
| 13:53.43 | scoobysnack | I'm using a Snom 720 and when I miss a call and try to dial it from the menu, I get an error. |
| 13:55.17 | scoobysnack | sip set debug shows it doing this To: <sip:+12018248357@199.132.130.182;user=phone>;tag=as187653ee3 |
| 13:55.52 | scoobysnack | I think the problem may be my dialplan doesn't have anything resembling stuff after the @ |
| 13:55.56 | [TK]D-Fender | You should consider showing the complete actual debug and not just a single line like that,. |
| 13:55.56 | scoobysnack | I'm unsure |
| 13:55.59 | scoobysnack | Any ideas? |
| 13:56.06 | [TK]D-Fender | ~pb |
| 13:56.06 | 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. |
| 13:56.15 | [TK]D-Fender | ^^^ |
| 13:56.49 | scoobysnack | Short of that, do you have any thoughts as to why it wouldn't dial? |
| 13:57.23 | [TK]D-Fender | No. |
| 13:57.23 | scoobysnack | Do you have to have anything special in your dialplan to handle the @ etc, etc. |
| 13:57.28 | [TK]D-Fender | We have to see the actual evidence |
| 13:57.33 | scoobysnack | Should it be trimmed first? |
| 13:57.35 | [TK]D-Fender | there is no "@" |
| 13:57.37 | [TK]D-Fender | No. |
| 13:57.40 | scoobysnack | Right that's not going to happen. |
| 13:57.50 | [TK]D-Fender | You do not have a proper understanding about how SIP headers work |
| 13:57.54 | scoobysnack | I did that before here and had hack attempts on my system. |
| 13:57.57 | scoobysnack | so |
| 13:58.01 | scoobysnack | nope the hell out of that |
| 13:58.18 | [TK]D-Fender | The attacks you tend to get are from random scanners |
| 13:58.20 | scoobysnack | You may be one of the good guys but there are a LOT of other people in this room that may not be |
| 13:58.35 | [TK]D-Fender | That's why firewalls were invented |
| 13:58.47 | scoobysnack | so you have no other ideas |
| 13:58.56 | scoobysnack | as to possibilities |
| 13:58.59 | [TK]D-Fender | You've told us precisely nothing |
| 13:59.07 | scoobysnack | lol |
| 13:59.08 | scoobysnack | ok |
| 13:59.11 | scoobysnack | thanks |
| 13:59.17 | scoobysnack | anyone else |
| 13:59.18 | [TK]D-Fender | Do you ask your doctor what's wrong and NOT let them examine you? |
| 13:59.25 | [TK]D-Fender | "Hi, it doesn't work, WHY!?!?!" |
| 13:59.25 | scoobysnack | yes |
| 13:59.28 | scoobysnack | webmd |
| 13:59.35 | scoobysnack | I just told you the symptoms |
| 13:59.43 | scoobysnack | the doctor could give me possibiliites |
| 13:59.57 | [TK]D-Fender | Using the internet for self-diagnosis is like representing yourself in court. You hav a fool for a client |
| 14:00.00 | scoobysnack | you're asking me to display personal info here and have hack attempts from experts |
| 14:00.02 | scoobysnack | no thanks |
| 14:00.11 | scoobysnack | anyone else? |
| 14:00.47 | [TK]D-Fender | mask the IP's then. |
| 14:00.53 | [TK]D-Fender | This isn't Raw-Cat Sigh Hence |
| 14:01.09 | [TK]D-Fender | And nobody here cares, and the only people out there hacking are using bulk scanners just to find you |
| 14:01.21 | scoobysnack | untrue |
| 14:01.22 | [TK]D-Fender | Has nothing to do with people sitting in IRC waiting to find even a trace by hand |
| 14:01.28 | scoobysnack | nice opinion of the room |
| 14:01.29 | scoobysnack | but untrue |
| 14:01.54 | scoobysnack | ok thanks. I'll figure it out or pay for support |
| 14:01.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 14:03.49 | jmetro | or you know, edit the personal stuff out. |
| 14:04.19 | rolandow | [TK]D-Fender: can you see why this call is being dropped? http://pastebin.com/er7vPWWb |
| 14:04.52 | [TK]D-Fender | jmetro: We've already heard his verdict on this. Let him search and hope he learn enough to understand the failure. Or that he doesn't waste a huge amount of time doing that just to pay for the answer in the end. |
| 14:05.30 | jmetro | [TK]D-Fender: i dunno about you, but i always find-replace all my stuff if i post it on here |
| 14:06.07 | [TK]D-Fender | jmetro: He's resistant to showing anything. Not worth arguing through that kind of paranoia. |
| 14:06.26 | scoobysnack | lol |
| 14:06.28 | [TK]D-Fender | rolandow: Pastebin another call without verbose and channel debug enabled, not core |
| 14:06.35 | scoobysnack | some people |
| 14:06.54 | scoobysnack | I'm thinking it may be because it puts a + at the front of the number |
| 14:07.07 | rolandow | [TK]D-Fender: ok .. i haven't been able to reproduce the problem yet.. |
| 14:07.21 | file | O.o |
| 14:09.29 | *** join/#asterisk nny (~Scott@cpe-075-182-017-074.sc.res.rr.com) |
| 14:11.04 | nny | Dealing with some Vega50 stuff. (Horrible gateway btw). I notice 1.8 has the function "volume" now. Is it pretty straight forward to use? I am probably just going to argue with the gateway for now to adjust but it caught my eye |
| 14:11.30 | [TK]D-Fender | nny: It's a function. It works just like any other. |
| 14:12.46 | scoobysnack | Yep...that was it |
| 14:12.51 | scoobysnack | No + in my dialplan |
| 14:13.02 | Katty | eyes scoobysnack with a fork again |
| 14:13.19 | nny | yeah was kind of pinging for reliability's sake, being a new one. I'll use the gateway for now, I don't feel like setting it independently in the dialplan |
| 14:13.37 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:14.25 | scoobysnack | I guess there aren't enough experts in here today to offer that as a suggestion. |
| 14:14.36 | scoobysnack | I wonder if that's a common rookie mistake. |
| 14:14.53 | [TK]D-Fender | nny: "New"? func_volume has been around for a LONG time.... |
| 14:15.27 | [TK]D-Fender | scoobysnack: We could have. But I'm not going to waste my time throwing random guesses at problems with no backup. |
| 14:16.09 | scoobysnack | It sounds like anything out of your perceived notions of how things should work is a waste of your time. |
| 14:16.45 | scoobysnack | Anyhow, I fixed it. So Yay for self-sufficiency. |
| 14:16.53 | [TK]D-Fender | \o/ |
| 14:17.51 | scoobysnack | lol |
| 14:18.00 | nny | [TK]D-Fender: hmmph. Didn't know, guess it's new to me ;) |
| 14:18.24 | *** part/#asterisk scoobysnack (~aaffcc002@gateway/tor-sasl/aaffcc0022ee) |
| 14:18.26 | [TK]D-Fender | nny: Don't let the mould accumulate ;) |
| 14:18.52 | nny | [TK]D-Fender: heh |
| 14:23.43 | nny | sigh, sangoma support has gone seriously down hill this year |
| 14:23.59 | nny | too many pointy hat issues I imagine |
| 14:24.41 | nny | oh, you just paid for our rebranded inferior product? Send us an email telling us how miserbale the documentation is! |
| 14:25.26 | [TK]D-Fender | What has Sangoma rebranded? |
| 14:25.52 | nny | i assumed the vega 50 was an aquisition or other type of dealing, it sucks |
| 14:26.02 | nny | wwell the vega line in general, next time mediatrix for me |
| 14:26.08 | Katty | jmetro: https://scontent-b-ord.xx.fbcdn.net/hphotos-ash4/1383299_10100741101374077_2134845130_n.jpg <- kitty i might be adopting, pending vet visit :> |
| 14:27.06 | *** join/#asterisk Jacke (~jacke@iam.ss7.pl) |
| 14:27.22 | nny | wouldn't be so bad but Sangoma support went from "Hi, this is XXX, a highly trained engineer willing to assist with the less documented features of product XXX" to "Send us an email using this generic form or search our limited wiki, k thanks, bye" |
| 14:29.28 | [TK]D-Fender | nny: Oh yeah, those.. well technically Sangoma bought out Vegastream... |
| 14:29.46 | [TK]D-Fender | nny: So not really "rebranded", just not us up to speed on the support I guess |
| 14:31.02 | nny | And just so I don't sound bitter, here is the wiki page for "Configuration" http://wiki.sangoma.com/Vega-50-Configuration and I have yet to find documentation on echo canceller aggressiveness or any other "CLI" feature. I had to hunt to find the way to adjust the gain, and only am guessing at this point on the syntax based on a statement on the FAQ here... http://wiki.sangoma.com/Vega-FAQ |
| 14:31.48 | nny | so I have to assume "set ._advanced.pots.fxs.1.digital_rx_gain=3" for FXO adjustment (tx/rx) is "set ._advanced.pots.fxo.1.digital_rx_gain=3" |
| 14:34.42 | nny | also wondering if they are referring to FXS from the far end or not... gah. Time to "contact" support. Sorry, ranting a bit |
| 14:37.51 | *** join/#asterisk lorsungcu (~anonymous@mo1.marathonortho.com) |
| 14:38.18 | Penguin | I guess someone is trying out a new technique for SIP toll fraud... |
| 14:38.57 | Penguin | There's this one host that has been attempting calls lately, but subsequent attempts take place after one to two hours. |
| 14:41.36 | Penguin | They must be trying to get around rate-limited blocking mechanisms. |
| 14:43.18 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 14:44.03 | qakhan | hi all. is there any function for charindex in asterisk 1.8 dialplan? |
| 14:45.11 | *** join/#asterisk bombev (~User@73.109.Globcom.Net) |
| 14:45.39 | [TK]D-Fender | qakhan: As in? |
| 14:46.56 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-111-87.user.veloxzone.com.br) |
| 14:47.38 | bombev | I am experinece strange problem |
| 14:47.38 | wdoekes | qakhan: CUT ;) |
| 14:47.39 | bombev | http://pastebin.com/crtXJqpb |
| 14:48.03 | bombev | I am trying to send two arguments within macro context but somehow it doesnt work properly |
| 14:49.28 | Penguin | Did you read core show application Dial and look how to format the M() option? |
| 14:49.29 | *** join/#asterisk Prosouth__ (~sabayonus@62-2-198-100.static.cablecom.ch) |
| 14:49.53 | qakhan | i am getting records from SQL through ODBC. there is numbers in the start of the record. i need to get these numbers in separate veriable and remaining data in separate veriable |
| 14:50.00 | Prosouth__ | Hi, is there a way to call our asterisk server from outside in order to call a customer with our asterisk SIP number? |
| 14:50.30 | Penguin | Do it the same way you do from the "inside." |
| 14:50.35 | bombev | Penguin I have to use ^ as delimiter |
| 14:50.35 | Penguin | (with a phone) |
| 14:50.36 | bombev | right |
| 14:50.40 | [TK]D-Fender | Prosouth__: There is no such thing as a "SIP number". And you can setup whatever acees to your server that you wish |
| 14:50.43 | Penguin | bombev: Try it. |
| 14:50.58 | [TK]D-Fender | access* |
| 14:51.14 | *** join/#asterisk fenja (~fenja@37.77.178.162) |
| 14:51.27 | dongola7 | Prosouth__: think you need to look at the DISA application. |
| 14:51.43 | [TK]D-Fender | No |
| 14:51.45 | Penguin | Or use your SIP phone from another location. |
| 14:51.56 | [TK]D-Fender | Not required. |
| 14:52.01 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-lvqzfnywyrykhfsd) |
| 14:52.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:52.01 | [TK]D-Fender | A better description would help |
| 14:52.46 | [TK]D-Fender | Prosouth__: Please reprasre yoru question so we can be sure what your options are |
| 14:52.47 | Prosouth__ | [TK]D-Fender: I meant : I already have an Asterisk server working properly but I was looking for a way to call my asterisk server from the outside with my cellphone in order to call a customer with my "asterisk configured number" |
| 14:53.20 | [TK]D-Fender | Prosouth__: So you do want to call in... and then have it dial OUT again just to use that connection instead of dialing direct from your cell? |
| 14:53.32 | Prosouth__ | exactly |
| 14:53.53 | [TK]D-Fender | Prosouth__: Well "DISA" is certainly an options then |
| 14:54.02 | [TK]D-Fender | Prosouth__: "core show application disa" |
| 14:54.28 | bombev | Penguin same thing happens |
| 14:54.45 | [TK]D-Fender | Prosouth__: Or just set up a pattern to match what you want it to dial. this is dialplan basics and you have to determin how to choose to hand out access, maybe bury the menu option for it,m etc |
| 14:55.00 | Prosouth__ | Thank you very much, I'm going to investigate in this way |
| 14:55.02 | [TK]D-Fender | bombev: PASTEBIN |
| 14:55.22 | Penguin | ... the new call. |
| 14:56.51 | bombev | [TK]D-Fender http://pastebin.com/Dy4F7Tke |
| 14:56.59 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:57.57 | [TK]D-Fender | bombev: The macro is called jsut fine |
| 14:58.21 | bombev | ok but then why I cant |
| 14:58.22 | [TK]D-Fender | bombev: And you have forgetten the most basic aspects of how they work. |
| 14:58.23 | Penguin | Your MixMonitor() syntax is wrong. |
| 14:58.36 | Penguin | What is the new problem? |
| 14:58.55 | [TK]D-Fender | Penguin: Perhaps, but he no concept of the proper VARIABLE names when passing parameters. |
| 14:59.04 | Penguin | I see that. |
| 14:59.12 | [TK]D-Fender | bombev: arg1, arg2, arg3, etc |
| 14:59.22 | qakhan | wdoekes need to get charidex in a string |
| 14:59.33 | [TK]D-Fender | bombev: read THE BOOK. You have skipped the most important part of gosubs/macros |
| 15:00.07 | bombev | well I did it with , |
| 15:00.24 | bombev | but still cant pass the variables |
| 15:00.57 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:01.11 | bombev | I will try tomorrow again |
| 15:01.19 | bombev | but I am missing something |
| 15:02.45 | wdoekes | qakhan: ${LENGTH(${CUT(mystring,charImLookingFor,1)})} |
| 15:02.58 | wdoekes | or LEN, I don't remember |
| 15:05.50 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 15:06.58 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:08.53 | qakhan | wdoekes e.g "198273$ This is new record in DB" |
| 15:09.10 | qakhan | i want to get all numbers before $ sign |
| 15:09.54 | Penguin | Does a variable contain that string that you're showing as an example? |
| 15:10.06 | qakhan | yes |
| 15:10.10 | wdoekes | Set(var=198273$ This is new record in DB); NoOp(${CUT(var,$,1)}) |
| 15:10.11 | [TK]D-Fender | qakhan: cor show function CUT |
| 15:10.13 | [TK]D-Fender | ^ |
| 15:10.27 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-111-87.user.veloxzone.com.br) |
| 15:11.06 | qakhan | let me give u example: 12345$$This is variable string |
| 15:11.13 | qakhan | splitter is $$ |
| 15:11.25 | qakhan | no. on left and string on right are variable |
| 15:11.34 | qakhan | but $$ is constant |
| 15:11.50 | Penguin | It's all the same string, and CUT will take care of it. |
| 15:12.32 | Penguin | wdoekes even told you EXACTLY what you need to use to make it work. He didn't even make you read the instructions. |
| 15:12.54 | qakhan | ok let me check |
| 15:14.35 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 15:15.25 | wdoekes | perhaps he opened IRC in write-only mode? |
| 15:16.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 15:19.01 | *** part/#asterisk russum (~russum@ool-4a599e57.dyn.optonline.net) |
| 15:23.29 | *** part/#asterisk asghar144 (~asghar144@host118-24-dynamic.8-87-r.retail.telecomitalia.it) |
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| 15:42.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:43.58 | *** join/#asterisk slidesinger (~slidesing@c-68-44-123-117.hsd1.nj.comcast.net) |
| 15:47.06 | cian1500ww | Hi, I've a konftel 300ip that is dropping all calls exactly 30 mintues after they begin with the following error on the phone: (No session refresh received after 900s (expiration period=1800s), stopping session now!) Asterisk reports it as being "hungup" in the logs. |
| 15:47.12 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 15:50.05 | Dovid | TK: Are you going to be at Astricon? |
| 15:57.30 | [TK]D-Fender | Dovid: Nope. |
| 16:01.18 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 16:04.33 | qakhan | Function Cut not registered |
| 16:04.39 | Penguin | CUT, not Cut. |
| 16:04.47 | qakhan | :( |
| 16:04.53 | qakhan | Sorry ..... |
| 16:05.30 | Penguin | By not following instructions, you only hurt yourself. |
| 16:08.16 | qakhan | yes you are right |
| 16:16.22 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 16:16.22 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:19.38 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 16:21.13 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
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| 16:25.53 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:30.08 | *** join/#asterisk undecided (~kvirc@49.156.43.10) |
| 16:31.03 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5E62E.dip0.t-ipconnect.de) |
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| 16:32.40 | undecided | is it possible to programatically send an event to an Asterisk live channel? |
| 16:33.16 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 16:34.06 | [TK]D-Fender | undecided: What is an "event"? |
| 16:34.39 | Katty | dances with [TK]D-Fender |
| 16:34.54 | undecided | for example, Read() application is waiting for some digits |
| 16:34.56 | [TK]D-Fender | Katty: If we don't, nobody will...\ |
| 16:35.19 | undecided | so I want to send them from linux cli |
| 16:35.23 | [TK]D-Fender | undecided: Describe a complete scenario so we can determin what approach is appropriate... |
| 16:35.36 | vlad_starkov | Question: Can anyone finally point me, what does this mean? |
| 16:35.41 | vlad_starkov | [2013-09-30 20:28:07] WARNING[23984]: chan_sip.c:3905 __sip_xmit: sip_xmit of 0xb468bf18 (len 788) to <ip_addr_removed>:60550 returned -2: No such file or directory |
| 16:35.41 | vlad_starkov | [2013-09-30 20:28:17] WARNING[10940]: chan_sip.c:3905 __sip_xmit: sip_xmit of 0xb4805a08 (len 784) to <ip_addr_removed>:38464 returned -2: Success |
| 16:35.59 | undecided | [TK]D-Fender: I just did |
| 16:36.51 | undecided | passing digits for Read() application from outside is a good example |
| 16:38.35 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:38.46 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 16:39.56 | [TK]D-Fender | undecided: What state is the call in prior? What do you expect to do with the pout? |
| 16:40.04 | [TK]D-Fender | unecAny other steps? |
| 16:40.11 | [TK]D-Fender | undecided: Triggered when/how? |
| 16:40.52 | [TK]D-Fender | output* |
| 16:41.25 | undecided | the call state is 'answered' |
| 16:41.47 | undecided | so caller requested to enter digits |
| 16:42.16 | undecided | I want a script to enter these digits |
| 16:42.40 | [TK]D-Fender | Read has a user enter digits. What is this "script" getting digits? |
| 16:42.47 | [TK]D-Fender | (entering rather) |
| 16:43.14 | [TK]D-Fender | What happens to the other leg of the call? |
| 16:43.38 | undecided | the script will enter these digits instead of caller |
| 16:43.54 | [TK]D-Fender | What script? |
| 16:44.04 | undecided | what do you mean other leg? |
| 16:44.07 | [TK]D-Fender | Nothing is entering digits yet execpt for the actuall call channel |
| 16:44.08 | undecided | that's an ivr |
| 16:44.45 | WIMPy | What kind of sense does it make to control an IVR from a script? |
| 16:44.46 | undecided | the script is what I'm after |
| 16:44.55 | [TK]D-Fender | Stop using the word script here |
| 16:45.04 | [TK]D-Fender | You are confusing things |
| 16:45.05 | undecided | I still don't knwo how to do this |
| 16:45.12 | undecided | maybe in bash |
| 16:45.19 | WIMPy | Do what EXACTELY? |
| 16:45.22 | [TK]D-Fender | You need to be a LOT clearer about what the actual asterisk channels are doing and who is expected to enter what. |
| 16:47.09 | undecided | a caller calls into asterisk and hear an IVR requesting to enter digits |
| 16:47.32 | paulc | undecided: You want a way for a script to generate those digits, so the IVR can capture them, right? |
| 16:47.32 | undecided | I see that call at the log for the example |
| 16:47.53 | undecided | and use a script to enter these digits into his channel |
| 16:48.19 | WIMPy | Are you talking about an IVR at your end or do you want to autiomate a call to a remote IVR or what? |
| 16:48.26 | [TK]D-Fender | entering digits for them is not the READ() you mentioned at the start |
| 16:48.41 | [TK]D-Fender | undecided: READ does not ENTER digits... Read() GETS digits |
| 16:48.51 | [TK]D-Fender | undecided: Your concept is completely backwards |
| 16:48.55 | undecided | WIMPy: ivr at my end |
| 16:49.23 | [TK]D-Fender | undecided: So what is the trigger for INSERTING DTMF (which is what you seem to be wnating to do now) |
| 16:49.26 | undecided | paulc: yes, but i need it send to that channel |
| 16:49.38 | WIMPy | So what's the point of entering digits other than by the caller? |
| 16:50.12 | WIMPy | There are tons of other ways to do call flow on the server. |
| 16:50.54 | undecided | [TK]D-Fender: lets say the trigger is just a System() call just before the Read app |
| 16:51.11 | undecided | system() will kick out a script |
| 16:51.36 | [TK]D-Fender | undecided: You are still being extremely vague |
| 16:51.54 | [TK]D-Fender | uneAnd describing the "how" that you envision and not properly descirbing the circumstances |
| 16:53.16 | undecided | I know it doesn't make much sense, and you're looking for some logic here |
| 16:53.29 | undecided | but that is a part of something bigger |
| 16:53.30 | [TK]D-Fender | [12:51]undecidedsystem() will kick out a script <- when does this happen? How long does it wait? Does it even wait? Why are you automating the response to your won Read() command? |
| 16:53.41 | undecided | so to simplify things I just gave that exampl |
| 16:53.42 | [TK]D-Fender | Your description is still pretty bad. |
| 16:53.57 | [TK]D-Fender | You are not giving a proper set of circumsatnces we can suggest an approach to |
| 16:54.10 | undecided | my english is not perfect |
| 16:54.17 | [TK]D-Fender | You are in fact trying to tell us what pieces you expec to use to do this action you have not clearly defined |
| 16:54.18 | undecided | but I;m trying |
| 16:54.31 | WIMPy | That did not siplyfy things. |
| 16:55.28 | [TK]D-Fender | undecided: Do not describe the actions in the solution, describe the circumstances of the call PRIOR to this action you would liek to take. Then describe the TRIGGER for it. Then describe the actual actions you want it to take upon being triggered |
| 16:55.55 | undecided | lets say I'm sitting in front of my asterisk cli logger |
| 16:56.13 | undecided | and I see you calling into my IVR |
| 16:57.07 | undecided | then I see the READ app is waiting for some digits in your channel |
| 16:57.10 | undecided | ok? |
| 16:57.25 | [TK]D-Fender | so far... |
| 16:57.34 | undecided | now I want to use a bash script to send tyhese digits instead of you |
| 16:57.37 | [TK]D-Fender | so why is this person SITTING on a read and not ansswering it? |
| 16:58.15 | [TK]D-Fender | What triggers this automated response you are looking for? |
| 16:58.21 | undecided | let's imagine I told him to ignore it for the test |
| 16:58.28 | [TK]D-Fender | How is that decision made? |
| 16:59.27 | undecided | the trigger can be programmed in various ways |
| 16:59.31 | undecided | but for now |
| 16:59.33 | paulc | undecided: What you need is a way to MAKE calls.. then SEND the DTMF tones, right? The question is.. do you want Asterisk to do that? If so, it can be done.. with a "call file" that points to smoe dialplan that will send the right DTMFs.. which maybe it gets from a variable, that you populate in the call file. |
| 16:59.42 | [TK]D-Fender | paulc: We're almost there, hold on a sec |
| 16:59.46 | undecided | lets say I want to kick out the script manually |
| 16:59.50 | [TK]D-Fender | paulc: there is a HUMAN involved here |
| 17:00.12 | WIMPy | undecided: Unless you ask a real question, I can only recommend you read about AMI. |
| 17:00.18 | paulc | goes to grab fresh water... |
| 17:00.49 | [TK]D-Fender | undecided: then there is an AMI command to execute a dialplan app against a channel |
| 17:01.04 | [TK]D-Fender | undecided: And you'd have it call SendDTMF |
| 17:01.54 | undecided | so with that AMI I can inject something into a channel, right? |
| 17:02.18 | [TK]D-Fender | undecided: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_PlayDTMF |
| 17:02.59 | undecided | thanks. reading |
| 17:04.13 | boom^time | Having a weird issue where calling via AMI with the exact same configuration/commands that worked perfectly find last Friday, now give me a failure and a SIP/2.0 487 Request Cancelled |
| 17:04.34 | boom^time | while the call file which does the exact same thing works perfectly fine. |
| 17:04.36 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 17:05.04 | [TK]D-Fender | boom^time: What does AMI have to do with that SIP RESPONJSE you are showing? |
| 17:05.06 | boom^time | The phone will ring once and then instantly hangup. Restarted the phone, asterisk |
| 17:05.07 | [TK]D-Fender | boom^time: Apples & oranges |
| 17:05.15 | boom^time | Nothing at all. |
| 17:05.35 | [TK]D-Fender | boom^time: You'd also have to show us the complete call |
| 17:05.51 | boom^time | Correct, but the calls originate fine with a call file and not with AMI. Sure, one moment. |
| 17:06.30 | boom^time | http://pastebin.com/xAL6L41s |
| 17:07.41 | [TK]D-Fender | boom^time: Som details and backup would help.... |
| 17:08.21 | vlad_starkov | Question: what is better in UDP packets loosing environment: G.711 or G.722? |
| 17:08.24 | boom^time | [TK]D-Fender, sorry but what details? |
| 17:08.30 | boom^time | I mean, what more details do you want specifically |
| 17:08.42 | [TK]D-Fender | vlad_starkov: Should be the same really. |
| 17:09.04 | [TK]D-Fender | vlad_starkov: iLBC and G.729 compensate better, but obviously at a loss of initial quality |
| 17:09.05 | boom^time | here is the ami command i'm ussing http://pastebin.com/DiQbEMvQ |
| 17:09.07 | Penguin | I'm still trying to figure out why asterisk needs to answer its own ivr. |
| 17:09.43 | [TK]D-Fender | Penguin: Technically his description now has HIM sitting at watching it execute... and then manually triggering it. |
| 17:10.02 | Penguin | But what circumstances require this? |
| 17:10.06 | [TK]D-Fender | boom^time: And how long is it ringing for? |
| 17:10.25 | boom^time | [TK]D-Fender, maybe half a ring |
| 17:10.26 | Penguin | If a caller is asked to enter some digits, why would the PBX need to generate the answer? |
| 17:10.28 | [TK]D-Fender | Penguin: Noone said his purpose had to make sense... just how it gets accomplished :) |
| 17:10.48 | [TK]D-Fender | Penguin: And it's really him the "administrator", forcing a response for some "testing" reason. |
| 17:10.56 | [TK]D-Fender | Penguin: Which yes... does sound pointless. |
| 17:11.09 | [TK]D-Fender | Penguin: But that's besides the point right now |
| 17:12.24 | vlad_starkov | [TK]D-Fender: Can I use OPUS codec in Asterisk? |
| 17:12.39 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:13.10 | [TK]D-Fender | vlad_starkov: https://www.google.ca/#q=asterisk+opus+codec |
| 17:14.16 | vlad_starkov | [TK]D-Fender: Is it really ready for production system? |
| 17:14.49 | [TK]D-Fender | vlad_starkov: It can't even be officially distributed with Asterisk.... guess why my usage would be? NONE. |
| 17:15.12 | [TK]D-Fender | vlad_starkov: no ifficial support = take what you can get, or don't do it at all. |
| 17:15.25 | vlad_starkov | [TK]D-Fender: ) |
| 17:15.28 | vlad_starkov | ok |
| 17:15.38 | [TK]D-Fender | vlad_starkov: I can't speak for any issues it may have with the implementations out there. I could imagine it works fine... but would not personally touch |
| 17:16.46 | vlad_starkov | [TK]D-Fender: Ok. So am I right thinking that better to use G.729 in packets loosing networks? |
| 17:17.04 | anonymouz666 | vlad_starkov: I don't think so |
| 17:17.09 | [TK]D-Fender | [13:09][TK]D-Fendervlad_starkov: iLBC and G.729 compensate better, but obviously at a loss of initial quality |
| 17:18.12 | vlad_starkov | I mean what everyone uses in production systems with Cisco SPA and Gigaset IP DECT phones? |
| 17:18.29 | vlad_starkov | anonymouz666: what is your opinion? |
| 17:18.37 | Penguin | When a telephone repair person goes out and determines where a wire is broken between the NID and the CO, and says something such as it is 237 feet from the box, are they using a TDR to measure that distance? |
| 17:18.58 | [TK]D-Fender | vlad_starkov: I've never heard of any of those supporting anything but the standard codecs.... G.711/729/723/722 |
| 17:19.06 | [TK]D-Fender | vldMAYBE iLBC depending |
| 17:20.59 | anonymouz666 | vlad_starkov: My opinion is that G711 will have a better MOS if you compare with g729 with 2%, 6%, 10% even 20%. |
| 17:21.33 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 17:22.23 | vlad_starkov | anonymouz666: what does MOS stands for? |
| 17:23.29 | anonymouz666 | Mean Opinion Score |
| 17:24.40 | vlad_starkov | anonymouz666: I have an issue with one of my customers, which has too many UDP packets drops |
| 17:26.49 | vlad_starkov | He is using G.711 and when the network drops RTP packets it sounds like it cuts sounds in the word |
| 17:28.05 | vlad_starkov | I just looking for some method to smooth this. |
| 17:30.34 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 17:34.17 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 17:35.25 | boom^time | [TK]D-Fender, this is a full sip debug/asterisk debug/verbose of just one ami origination. http://pastebin.com/VmYrYAba |
| 17:35.45 | boom^time | The phone will ring for about half a ring and then just end. It's the strangest thing, nothing has changed since I last touched this and it was working fine. |
| 17:38.45 | anonymouz666 | vlad_starkov: why your network drops RTP packets? |
| 17:40.13 | ChannelZ-Wk | boom^time: These debugs are a mess to look at, but it seems to me like a Hangup is being sent.. see line 159 etc |
| 17:40.29 | vlad_starkov | anonymouz666: its our customer's uplink |
| 17:40.43 | [TK]D-Fender | boom^time: paste again with the complete details including the original originate |
| 17:41.10 | boom^time | Those details are from the instant the call starts until it fails, debug+sipdebug+verbose |
| 17:41.46 | boom^time | here is the AMI originate command http://pastebin.com/s1vejE1Y |
| 17:41.47 | [TK]D-Fender | ChannelZ-Wk: [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7078 sip_hangup: Hanging up channel in state Down (not UP) |
| 17:41.55 | [TK]D-Fender | ChannelZ-Wk: * is killing it... |
| 17:42.21 | boom^time | But why I wonder? yeah it follows the sip 100 with a cancel |
| 17:42.41 | [TK]D-Fender | boom^time: Show again witht he debug that also shows the complete originate |
| 17:42.43 | boom^time | It's not complaining about the context/exten/priority not existing, as it does. I would understand a cancel in that situation |
| 17:43.18 | [TK]D-Fender | boom^time: You shouldn't get a dialplan error until it answers anyway.... |
| 17:43.21 | boom^time | I'm not sure what you mean by the complete originate. With AMI what I copied for you is all you get, unless you want the AMI command which I just pasted |
| 17:43.39 | boom^time | There is nothing else that I skipped |
| 17:43.41 | [TK]D-Fender | boom^time: we see the OTHER events being sent but not the inbound originate itself thart starts all of this |
| 17:43.44 | boom^time | But i'll grab it for you without debug |
| 17:43.52 | [TK]D-Fender | And yes, we wan't debug. |
| 17:43.56 | [TK]D-Fender | want* |
| 17:44.28 | boom^time | There is no inbound originate [TK]D-Fender , just the local AMI command that causes it to start. |
| 17:44.36 | boom^time | I clear my terminal before I hit return |
| 17:44.43 | boom^time | and copy/paste everything from that until it's done. |
| 17:44.58 | boom^time | (hit return as in sending the AMI command manually) |
| 17:46.08 | boom^time | <PROTECTED> |
| 17:46.18 | [TK]D-Fender | boom^time: dump AMI at the same time. Wee see the OTHER events that follow the originate, but not the originate itself |
| 17:46.26 | boom^time | Okay |
| 17:47.16 | boom^time | http://pastebin.com/YP7e5jwH |
| 17:49.46 | [TK]D-Fender | [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:3874 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 10.1.10.70:5060 |
| 17:49.48 | [TK]D-Fender | [Sep 30 13:31:27] DEBUG[26278][C-00000000]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/501-00000000' |
| 17:50.54 | [TK]D-Fender | not seeing much else, I see it saying that's it's trying to do something, then stating a hangup which sounds like it thinks it failed to actually push the packet out. |
| 17:51.15 | [TK]D-Fender | Even thoguh we see a SIP response following |
| 17:51.19 | [TK]D-Fender | This i odd... |
| 17:51.21 | boom^time | hmm, yeah it starts up and then instantly cancels |
| 17:51.24 | [TK]D-Fender | I would certain upgrade first... |
| 17:51.29 | [TK]D-Fender | And retest |
| 17:51.40 | *** join/#asterisk Defraz (~Defraz@gump.fuzecore.com) |
| 17:53.42 | boom^time | What I think is really odd is that if I use a call file with the exact same channel/context it works fine. |
| 17:53.49 | boom^time | as opposed to an AMI originate. |
| 17:54.27 | [TK]D-Fender | boom^time: I see why :) |
| 17:54.40 | [TK]D-Fender | boom^time: Someone didn't read the instructions... |
| 17:54.48 | [TK]D-Fender | boom^time: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate |
| 17:55.16 | [TK]D-Fender | boom^time: http://pastebin.com/YP7e5jwH <- #7 |
| 17:55.16 | *** join/#asterisk crankyadmin (~crankyadm@its.hackerti.me) |
| 17:55.48 | boom^time | wow. |
| 17:56.27 | boom^time | haha thank you [TK]D-Fender. And here I thought I saw you stumped on something for the first time. |
| 17:57.45 | boom^time | My fault for assuming WaitTime: from a call file and Timeout: in AMI would take the same unit of time. |
| 17:58.11 | [TK]D-Fender | ~assume |
| 17:58.11 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 17:58.14 | [TK]D-Fender | yum... |
| 18:00.52 | [TK]D-Fender | yup* |
| 18:03.02 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 18:04.54 | *** join/#asterisk rsaffi (~rsaffi_@187.109.36.11) |
| 18:05.03 | crankyadmin | Hi. I have a Asterisk (1.8.13.1~dfsg-1ubuntu2) server running a dial plan that is triggered from a call file. But neither the 'h' or 'failed' extensions are every hit. Been digging for about a day on this with no joy. Any immediates thoughts jump out? |
| 18:06.07 | [TK]D-Fender | crankyadmin: Your dialplan has syntax errors. The extens are not in the proper place. Sun-spots. |
| 18:06.23 | [TK]D-Fender | #anything |
| 18:06.58 | crankyadmin | Sorry I don't understand Sun-spots? |
| 18:07.42 | [TK]D-Fender | You know when the Sun (that red-giant star of ours) puts our flares? |
| 18:08.22 | [TK]D-Fender | Also... fluctuations in the planet's magnetic axis. |
| 18:08.32 | *** join/#asterisk echo083 (~adam@gateway/tor-sasl/echo083) |
| 18:08.36 | echo083 | hello |
| 18:08.45 | echo083 | can you tell me which codec is the best ? |
| 18:08.52 | echo083 | best quality |
| 18:09.30 | navaismo | g722 ?¿?¿ |
| 18:09.50 | Tim_Toady | echo083: g711 usually, unless you can use wideband codecs like g722 |
| 18:10.14 | echo083 | Tim_Toady, g711 is not in the list :( |
| 18:10.26 | Tim_Toady | what is in the list then? |
| 18:10.36 | boom^time | ulaw |
| 18:10.49 | echo083 | Tim_Toady, i'm making a copy paste :) |
| 18:10.58 | Penguin | or alaw, depending on region/country. |
| 18:11.23 | echo083 | gsm alaw ulaw lpc10 speex adpcm siren14 siren7 g722 g723 slin g726 g729 ilbc g726aal2 |
| 18:11.41 | echo083 | Penguin, i'm in great britain |
| 18:12.22 | Tim_Toady | echo083: ulaw/alaw is g711 |
| 18:12.33 | echo083 | i'm usually using gsm the person i call here my voice perfectly but me i'm hearing sizzlings |
| 18:12.38 | echo083 | Tim_Toady, ahhhh ok ! |
| 18:12.53 | echo083 | hear* |
| 18:13.32 | Penguin | G.711u or G.711a = ulaw or alaw |
| 18:13.46 | echo083 | i'll alaw and ulaw so :) |
| 18:13.50 | echo083 | i'll try* |
| 18:24.24 | *** join/#asterisk russum (~russum@86.104.14.194) |
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| 19:40.27 | *** join/#asterisk cobolfoo (~cobolfoo@modemcable206.108-200-24.mc.videotron.ca) |
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| 19:46.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:54.42 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 19:58.20 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 19:59.18 | MLNoah | I have 2 Asterisk 11.5 boxes clustered together using res_corosync to share device state information. Is there a supported way to make it so that if box A changes the device state to NOT_INUSE for a specific Custom: device, box B also changes its device state for that Custom: device? |
| 20:00.23 | *** part/#asterisk nny (~Scott@cpe-075-182-017-074.sc.res.rr.com) |
| 20:00.26 | MLNoah | I have phones that are occasionally changing registrations between the two boxes (via DNS SRV), so sometimes a phone might set a forward on when registered to A, then clear the forward when registered to B, so simply setting Custom:${EXTEN}@forwarding to NOT_INUSE on B wouldn't clear the shared hint, since Custom:${EXTEN}@forwarding on A would still be showing INUSE |
| 20:02.32 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 20:10.34 | wltjr | anyone know why polycom vvx300 phones refuse to register I keep getting username mismatch, have <11>, digest has <> |
| 20:11.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.46) |
| 20:11.14 | wltjr | I updated firmware, as other reported that issue being resolved wih newer firmware, running 5.0.0.6874, might try to revert to the lastest 4.x version |
| 20:12.04 | Chainsaw | Only have IP670 & IP7000 at work, sorry. That's a different firmware train entirely. |
| 20:17.17 | *** join/#asterisk datacrusher (~datacrush@unaffiliated/datacrusher) |
| 20:18.00 | wltjr | might call polycom there is no reason they should not register must be a firmware thing, I might try another version since its easy to update with their web interface |
| 20:26.06 | cobolfoo | How I can program Asterisk to transfer to external phone number if a extension is not answering after 4 rings? |
| 20:26.31 | ghost75_ | with a dialplan :> |
| 20:27.19 | cobolfoo | I do a Dial() command but it hangup right after. |
| 20:28.06 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 20:30.08 | *** join/#asterisk felipealmeida (~user@187-15-226-96.user.veloxzone.com.br) |
| 20:31.01 | ghost75_ | show code |
| 20:32.04 | cobolfoo | I use Asterisk GUI, created a voice menu that only do a Dial command. |
| 20:32.18 | cobolfoo | Do FollowMe feature can be used to do the same thing ? |
| 20:32.40 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:33.02 | Penguin | Use another Dial() |
| 20:33.48 | ghost75_ | asterisk gui still exists? |
| 20:34.12 | *** join/#asterisk felipealmeida (~user@187-15-226-96.user.veloxzone.com.br) |
| 20:34.16 | Penguin | Exist and in development are two different things. |
| 20:34.31 | Penguin | Just because it isn't in dev does not mean it ceases to exist. |
| 20:37.12 | ghost75_ | better to use freebpx when gui is needed |
| 20:40.35 | wltjr | cobolfoo: http://pastebin.com/Bq4MTVSs |
| 20:42.32 | cobolfoo | Why are you calling an extern first ? |
| 20:42.40 | cobolfoo | erm extension |
| 20:43.21 | Penguin | The whole thing is an extension. |
| 20:43.25 | Penguin | This is extension 0. |
| 20:43.32 | Penguin | It calls some SIP phones first. |
| 20:43.44 | Penguin | Then there's some useless thing about callerid. |
| 20:44.04 | Penguin | Then it calls a phone number via a SIP peer named trunk. |
| 20:44.24 | Penguin | After than the extension goes to voicemail. |
| 20:44.39 | Penguin | Those four lines are one extension. |
| 20:44.52 | wltjr | I set caller id to who called me when I have it ring my cell phone, otherwise it shows caller id from * |
| 20:45.11 | Penguin | It's already that. |
| 20:45.12 | cobolfoo | ok. |
| 20:45.17 | wltjr | but bad idea to have 0 call you, clients drive me nutz... |
| 20:45.18 | Penguin | You don't need to set it again when it is already that. |
| 20:45.41 | wltjr | Penguin: I had to for some reason it wasn't what I wanted idk maybe I can lose |
| 20:45.59 | Penguin | It may depend on what else you have going on in your dial plan. |
| 20:46.20 | wltjr | Penguin: idk might be uncessary have to test, but not sure I care ;) |
| 20:49.55 | wltjr | what is the deal with polycom you can't contact them for support but have to go through authorized partner or something... trying to move from pice of crap cisco spa phones to polycom and can't get the polycoms to register with *... h8 voip |
| 20:50.39 | Penguin | I would guess that you didn't configure the pertinent phone settings correctly. |
| 20:51.44 | ghost75_ | normally isp doesnt allow to change caller id |
| 20:53.29 | wltjr | Penguin: cleary, but there is very little to it really, their config is much better, let me clean it up and pastbin |
| 20:54.37 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 20:55.20 | wltjr | http://pastebin.com/eiSAAMNf |
| 21:01.23 | wltjr | its weird line 3 was able to register but not others earlier it was other lines registering |
| 21:01.42 | wltjr | but when line 3 registers its on port 5060, and I am telling it to register on port 5062, like the old SPA-501g was |
| 21:02.18 | Penguin | That is the server port. Asterisk isn't listening on those other ports; it's listening on 5060. |
| 21:03.11 | Penguin | For the client port, you'd have to specify the client port in each peer definition as well as the phone. |
| 21:04.25 | wltjr | Penguin: works fine now with the Cisco phones |
| 21:04.28 | wltjr | Penguin: one sec |
| 21:04.41 | Penguin | In the Cisco phone, you change the CLIENT port. |
| 21:04.52 | Penguin | And then you specify the client port for the asterisk devices. |
| 21:05.10 | wltjr | Penguin: http://pastebin.com/UM6Cb4iy |
| 21:05.22 | wltjr | Penguin: orginally with polycoms I did no set the port |
| 21:05.40 | wltjr | right now I have just 1 extension configured on port 5060 and its failing registration |
| 21:05.56 | wltjr | this crap username mismatch, have <11>, digest has <> |
| 21:06.04 | Penguin | Extensions don't have ports. |
| 21:06.07 | Penguin | What's with all the different port numbers? Is something wrong with the normal 5060? |
| 21:06.46 | wltjr | Penguin: no clue, not sure why I did that, might have been a quirk with the spa-501g, I think they can't listen all on port 5060 or something |
| 21:07.17 | wltjr | I never realized those phones were such junk I have an old linksys one that has all the same issues, can't disable call waiting without making phone 1 line and so may other stupid things |
| 21:07.40 | Penguin | If the phone was on a different network segment with a NAT in between or something like that, I could understand all the different ports. |
| 21:08.30 | Penguin | I'm not sure where the client port setting would be in the polycom. I usually use the Linksys phones, and it's pretty clear in the web UI. |
| 21:08.33 | wltjr | Penguin: no its on its own network, vlan and everything, some of that is on a different network for now, putting in smart switches so that will be corrected |
| 21:08.56 | wltjr | just makes no sense why the polycomes keep failing authorization and not sending over the right stuff |
| 21:09.05 | wltjr | here is a single line config http://pastebin.com/qHh49EZQ |
| 21:10.40 | Penguin | address, auth.userId, and auth.password ... look fine, so I would think it would register. |
| 21:10.57 | wltjr | Penguin: me to, and it does sometimes... its pretty crazy |
| 21:12.25 | *** join/#asterisk brut- (~brut-@h184-61-140-187.pqlkmn.dedicated.static.tds.net) |
| 21:12.30 | Penguin | Did you check the sip debug to see if the phone even sends a REGISTER at all? |
| 21:12.54 | wltjr | Penguin: I am seeing the registration fail |
| 21:13.01 | wltjr | Registration from '<sip:11@10.4.2.2>' failed for '10.4.2.4:5060' - Username/auth name mismatch |
| 21:13.13 | Penguin | Oh. That's pretty clear cut. |
| 21:13.17 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 21:13.31 | wltjr | yes, and I have make sure the username and password are correct, but the problem is before that |
| 21:13.35 | wltjr | username mismatch, have <11>, digest has <> |
| 21:13.47 | wltjr | I googled that and others reported same issue but firmware update worked for them does not for me |
| 21:14.40 | Penguin | I have trouble remembering if digest is what the phone sends or if it's what is in asterisk. I feel like the digest is the phone. |
| 21:15.03 | brut- | hi folks - quick question: does anyone know if asterisk has the option of sending an in-dialog OPTIONS ping during call setup? I haven't found anything in the docs or in any of my configs for it. |
| 21:16.50 | wltjr | Penguin: I think it is, I am pretty sure the problem is on the phoens end, * hasn't changed, I am just swapping out phones, should be simple I cannot believe I am stuck at registration |
| 21:19.56 | wltjr | Penguin: same problem firmware update resolved it -> http://forums.asterisk.org/viewtopic.php?f=1&t=79707 |
| 21:25.52 | wltjr | thinking polycom is worse than cisco on the phone with them but they want me to go through the company I got them from when this could be some software bug, really stupid... |
| 21:26.52 | wltjr | just tried on another have 7 of them here, just configured server, 1 extension, and same thing, fails to register, seem all entrly level phones are crap, might just go with grandstream... |
| 21:27.09 | wltjr | or buy used 7960s off ebay.. |
| 21:34.13 | wltjr | polycom is making me call the vendor, so the vendor can call Polycom with me on the phone, give them their polycom certified # so polycom will speak to me, completely stupd |
| 21:34.29 | wltjr | here is same problem on their support forum http://community.polycom.com/t5/VoIP/reprogramming-Polycom-301-via-Web-GUI/td-p/12304 |
| 21:34.48 | Penguin | Is the solution listed? |
| 21:38.15 | *** join/#asterisk pigeonflight (~macuser@72.252.224.80) |
| 21:38.38 | mic_ | hello |
| 21:38.48 | mic_ | just before I order a straight jacket for myself |
| 21:38.57 | mic_ | type = friend vs peer |
| 21:39.14 | mic_ | friend just makes sure it's matched by the username as well? |
| 21:39.23 | Penguin | Not entirely. |
| 21:39.56 | mic_ | Or only by the username? |
| 21:39.59 | pigeonflight | are these services like tollfreeforwarding and ringcentral needed in order to get a tollfree number? |
| 21:40.47 | Penguin | Using type=peer causes matching by IP/port. user matches by username. friend creates both a peer and a user and attempts to match incoming calls by username first; outbound calls will be treated like type=peer. |
| 21:41.42 | mic_ | Penguin: well, that would mean outbound calls should be matched by IP/port |
| 21:41.55 | Penguin | Outbound calls are not matched. |
| 21:41.58 | Penguin | so no. |
| 21:42.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.174) |
| 21:42.02 | mic_ | aa. |
| 21:42.05 | mic_ | crap, right. |
| 21:42.13 | mic_ | treated != matched. |
| 21:43.24 | Penguin | If you make a call to a device that is type=friend, the call will use the IP just the same as if it was type=peer. |
| 21:43.30 | Penguin | That's what I meant by treated. |
| 21:44.02 | mic_ | Yes. |
| 21:44.14 | mic_ | Ok, I am clear. |
| 21:44.20 | mic_ | Thanks a lot, I can go to bed now ;) |
| 21:48.52 | Penguin | pigeonflight: Most ITSPs have toll free numbers. |
| 21:49.49 | pigeonflight | Penguin: who are the trusted names among ITSPs? |
| 21:49.56 | Penguin | Which country? |
| 21:49.59 | pigeonflight | this is all new to me |
| 21:50.05 | pigeonflight | US/CAN toll free |
| 21:50.09 | Penguin | VoIP.ms |
| 21:51.49 | pigeonflight | Penguin: any others? for comparison? |
| 21:51.49 | Penguin | flowroute |
| 21:52.01 | brut- | I use voicepulse myself |
| 21:52.12 | brut- | they work pretty decent, never used any of their 800 stuff though, but they do have it |
| 21:52.19 | pigeonflight | and these guys have minimum monthly fees? |
| 21:52.55 | brut- | my voicepulse is like $12/month minimum |
| 21:53.01 | pigeonflight | goes off to do more investigation |
| 21:53.03 | brut- | pstn termination fees, etc |
| 21:53.16 | pigeonflight | I need sip termination only |
| 21:53.31 | Penguin | What do you think you're going to terminate to? |
| 21:53.38 | brut- | yeah, the PSTN charge is to receive calls from the PSTN world on your SIP side |
| 21:53.44 | Penguin | They don't terminate to SIP. |
| 21:53.55 | brut- | calls still have to get onto the SIP network from the PSTN somehow |
| 21:53.56 | pigeonflight | :) learning the terminology |
| 21:54.02 | Penguin | They terminate to the PSTN like every other VoIP telephone company. |
| 21:54.06 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:54.11 | Penguin | ~termination |
| 21:54.31 | pigeonflight | I was thinking from tollfree number --> SIP = sip termination |
| 21:54.39 | pigeonflight | still have a bit of reading to do |
| 21:54.43 | Penguin | ~origination |
| 21:54.45 | Penguin | wtf |
| 21:54.50 | Penguin | ~itsp |
| 21:54.51 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:55.10 | Penguin | From a toll-free number to you by SIP is origination. |
| 21:55.20 | drmessano | SIP <-----> Ball of wire known as PSTN <-----> SIP |
| 21:55.28 | pigeonflight | I pretty much want to do 99% RECEIVE calls |
| 21:55.42 | Penguin | You'll want an ITSP. |
| 21:55.56 | Penguin | If you don't need to get calls from the PSTN, you don't need the ITSP. |
| 21:56.33 | pigeonflight | Penguin: I would need to get calls from the PSTN... since most toll free callers would be PSTN right? |
| 21:56.42 | Penguin | Yes. |
| 21:57.01 | Penguin | If I wanted to call you by dialing your toll-free number, I'd use the PSTN to do it. |
| 21:57.05 | pigeonflight | but all my termination would pretty much be SIP |
| 21:57.20 | Penguin | Where will you be terminated these calls? |
| 21:57.33 | Penguin | s/terminated/terminating/ |
| 21:58.02 | Penguin | Will you be calling people's regular phone numbers and expecting them to pick up their phones? |
| 21:58.10 | pigeonflight | Plivo based endpoints |
| 21:58.36 | pigeonflight | all soft phones |
| 21:59.09 | pigeonflight | where do I go to get good prices on SIP phones btw? |
| 21:59.10 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 21:59.25 | Penguin | I'll rephrase: from your phones that are peered to asterisk, will you ever be calling regular phone numbers of people who have phones in their houses? |
| 21:59.47 | pigeonflight | not for this project |
| 21:59.56 | pigeonflight | the use case will arise for other projects |
| 21:59.58 | Penguin | Then there is no termination service needed. |
| 22:00.17 | pigeonflight | but ITSP is needed for origination, correct? |
| 22:00.19 | Penguin | If you aren't sending calls out to the PSTN, you don't need the termination service from an ITSP. |
| 22:00.37 | Penguin | To get calls FROM the PSTN, you'll use the ITSP's origination service. |
| 22:00.48 | pigeonflight | right |
| 22:01.05 | Penguin | They'll offer you both; you just don't have to use outgoing calling. |
| 22:01.09 | pigeonflight | Penguin: Is there such as thing as signing up only for origination services? |
| 22:01.21 | Penguin | See previous statement. |
| 22:02.01 | Penguin | Usually, they just provide "service." You either use what they provide or you don't. Termination service will be available, even if you never send a call outbound through them. |
| 22:02.25 | Penguin | If you sign up and have credit available, you CAN make calls through them, but you don't have to. |
| 22:02.52 | Penguin | Buy a phone number or port your existing number in, and that's how you use their origination services. |
| 22:03.49 | pigeonflight | Penguin: so I went to this place called tollfreenumbers.com and saw a vanity number that I liked, but someone told me not to buy from them |
| 22:03.55 | *** join/#asterisk xzarth (~krikkit@dh207-37-146.xnet.hr) |
| 22:04.00 | pigeonflight | Penguin: does it matter? |
| 22:04.07 | Penguin | Check for it from another provider. |
| 22:04.26 | pigeonflight | like voip.ms I guess? |
| 22:04.33 | Penguin | With local numbers, most providers have the same pool available. I don't know how it works with toll-free numbers. |
| 22:05.50 | wltjr | finally got polycom on phone after 3 way call with resller... now polycom is realizing I am dealing with some bug in phone... nice first experience... |
| 22:06.48 | wltjr | first time I have ever had registration issues... the most basic voip stuff... vvx 300 very nice phone, nice web gui etc, but dont work :( |
| 22:07.19 | wltjr | also pretty crazy their config defaults to 24 calls per line, wow... |
| 22:07.20 | Penguin | pigeonflight: If you'll PM me the number you're looking for, I'll check in voip.ms SMS800 database to see if they can get it. |
| 22:08.25 | pigeonflight | Penguin: I'm looking at 400 minutes per day, I know that's modest compared to carriers but would these ITSPs still be the way to go? |
| 22:08.53 | pigeonflight | we're currently doing 150 minutes and looking for better pricing |
| 22:09.03 | drmessano | I do 100 minutes a month on my flowroute account for home. Not a problem |
| 22:09.12 | Penguin | Per minute pricing for toll-free is going to add up fast. |
| 22:09.26 | pigeonflight | Penguin: that is also a problem at the moment |
| 22:09.39 | pigeonflight | which is why we're looking for smaller segments |
| 22:10.16 | Penguin | It won't be a problem to do 400 minutes a day, but I was just thinking of the $10 per day it will cost. |
| 22:12.07 | pigeonflight | Penguin: it's for a company that will have a call center with around 10 agents behind sip phones |
| 22:14.04 | pigeonflight | so $10 for 400 minutes is a competitive price then? |
| 22:14.44 | Penguin | I think voip.ms price is 2.5c per minute for toll-free. |
| 22:15.20 | pigeonflight | I guess I'll shop around |
| 22:16.02 | Katty | evening |
| 22:16.43 | Katty | file: are you surviving alabama ok? |
| 22:16.50 | file | yes |
| 22:17.19 | Katty | good. |
| 22:18.06 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 22:18.08 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 22:19.04 | *** join/#asterisk pigeonflight (~macuser@davidbain.xen.prgmr.com) |
| 22:19.51 | carrar | Got a emergency underground shelter in Alabama? :) |
| 22:19.56 | *** part/#asterisk russum (~russum@86.104.14.194) |
| 22:20.27 | file | I know to go to the stairs if something weather-like were to occur. |
| 22:20.38 | file | stairwell, that is |
| 22:20.58 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:21.49 | Katty | good. |
| 22:27.49 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:28.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:29.38 | Katty | fender bender. |
| 22:29.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zznafmsrztfycqkl) |
| 22:30.58 | [TK]D-Fender | Kit Kat kitty katty |
| 22:33.02 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:39.14 | pigeonflight | anyone done business with H3.net? |
| 22:47.12 | wltjr | think I tracked down the problem http://pastebin.com/3KJ7PQij looks like the phone is sending an empty user name for some reason... |
| 23:25.28 | crumb | where the instructions for installing g729 codec |
| 23:27.53 | crumb | with licenses |
| 23:29.55 | Penguin | http://downloads.digium.com/pub/telephony/codec_g729/README |
| 23:30.19 | crumb | yeah, that's what i've been reading :/ |
| 23:30.34 | Penguin | Did you not just request the instructions? |
| 23:30.45 | crumb | i don't know -_- |
| 23:31.17 | Penguin | Albeit a poorly constructed question, I believe that you just asked where the instructions are to install the codec. |
| 23:31.20 | Penguin | I provided it to you. |
| 23:31.55 | crumb | well, somebody made their own g729 codec implementation for arm and i finally got the license key |
| 23:32.05 | crumb | he said the installation is similar to digium's |
| 23:32.36 | Penguin | If your license is not from Digium, ask your license vendor for his instructions. |
| 23:33.18 | crumb | d'oh.. supposed to be using asterisk 11 |
| 23:33.19 | crumb | sorry |
