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| 05:32.53 | wrouesnel | quick question: running 1.8.10, I can't seem to add a string with spaces in it to the asterisk database |
| 05:33.37 | wrouesnel | ok never mind, got my syntax out of order. |
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| 06:44.54 | gavimobile | folks I added security => security to my logger.conf as mentioned in http://ofps.oreilly.com/titles/9781449332426/asterisk-Security.html#example-accountscan |
| 06:45.16 | gavimobile | if I understand correctly, now with the output I can use fail2ban to block only the ones I don't want? |
| 06:45.33 | gavimobile | cause some of the output it might be not bad |
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| 07:14.07 | fling | How do dial both legs and bridge them together in a simple way? |
| 07:14.19 | fling | Without writing a huge dialplan. |
| 07:16.55 | kleszcz | Dial(SIP/john@foo.com) ? |
| 07:25.41 | kaldemar | fling: both? do you mean just any two? |
| 07:26.14 | fling | kaldemar: yes |
| 07:26.33 | fling | kaldemar: right, and how to dial the second one? |
| 07:26.37 | kaldemar | where do you want to trigger it from? |
| 07:26.46 | fling | s/kaldemar/kleszcz |
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| 07:26.57 | fling | kaldemar: idk :P |
| 07:27.18 | fling | I have a sip softphone |
| 07:27.36 | kaldemar | fling: with a phone (=from dialplan)? with a script (=AMI, CLI)? |
| 07:27.45 | fling | from dialplan! |
| 07:27.49 | kaldemar | "core show application Originate" |
| 07:27.53 | fling | thanks |
| 07:31.57 | fling | http://dpaste.com/1270131/ |
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| 07:32.30 | fling | kaldemar: ^ Zap/1/123456 is what dialed first, right? And then it goes into extension 1@greeting? |
| 07:32.37 | fling | I like originate |
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| 07:43.46 | fling | kaldemar: I don't have originate cli command |
| 07:44.49 | fling | found it! it is channel originate |
| 07:49.22 | fling | kaldemar: it is working! thanks :P > pbx*CLI> channel originate sip/some-peer/first-leg-did extension second-leg-did@some-context |
| 07:55.58 | fling | How to monitor ongoing calls/bridges from cli? |
| 07:59.41 | Kalidarn | wasn't it renamed from Zap? |
| 08:00.22 | fling | kaldemar: what renamed to Zap? |
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| 08:05.03 | Kalidarn | the channel name |
| 08:05.09 | Kalidarn | fling: i'm not kaldemar |
| 08:07.26 | fling | Kalidarn: I'm using tabs bad :D |
| 08:07.41 | fling | sip renamed to zap? woot? |
| 08:08.19 | Kalidarn | no no |
| 08:08.26 | Kalidarn | oh i thought i read something about zaptel |
| 08:09.10 | fling | oh! |
| 08:09.30 | fling | I'm now originating calls! |
| 08:12.42 | kaldemar | fling: nothing has been renamed to zap. zaptel was renamed to DAHDI years ago. |
| 08:13.18 | kaldemar | fling: enable verbosity with "core set verbose 10" and you'll see dialplan flow on CLI has it happens. |
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| 08:29.21 | giorgiodinapoli | good morning guys |
| 08:49.50 | wrouesnel | is there some way to determine what queue a device is a member of from the dialplan? |
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| 08:54.44 | Gugge | wrouesnel: yes |
| 08:54.50 | Gugge | but not an easy way :) |
| 08:55.39 | Gugge | but QUEUE_MEMBER_LIST on each queue would give the right info |
| 08:57.00 | gavimobile | can I take this out if im using virtual directories? DocumentRoot "/var/www/html" |
| 08:58.51 | Gugge | gavimobile: asterisk has a documentroot setting? :) |
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| 09:02.34 | gavimobile | Gugge: im lost :-p |
| 09:02.37 | gavimobile | thought I was in httpd |
| 09:02.41 | gavimobile | wrong post |
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| 09:09.33 | CommaCrazy | hi all |
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| 09:21.52 | MariusKarthaus | So can anyone point me to / tell me what the difference between the 1.8.* and the 11.* version trees are. They both seem to be maintained and the ubuntu folks seem to stick with the 1.8.* version. I'm wondering if a change to 11.* is usefull in terms of stability, functionality etc ? |
| 09:32.05 | kaldemar | MariusKarthaus: http://svn.digium.com/svn/asterisk/tags/11.4.0/CHANGES |
| 09:34.20 | kaldemar | MariusKarthaus: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 09:39.19 | MariusKarthaus | So if I understand this correctly there is no funcdamential difference between the 1.* and 10.*/11.* series, they just decides to not go for 1.9 and change the versioning sceme |
| 09:41.05 | MariusKarthaus | I was under the impression that with the mark of 10.* it would have been a reworked codebase that would have significant impact. |
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| 09:42.07 | kaldemar | the version number change was just a change of numbering scheme. there are quite significant changes netween the branches though. |
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| 10:00.58 | MariusKarthaus | kaldemar: ok, thank you |
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| 10:03.47 | BorjaGVO | Hello. I'm trying to use xmpp to distribute states. Both xmpp clients are connected to openfire server but state information won't go across. I folowed steps in http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265044.html#DeviceStates_id36025342 in order to test it. Full log: http://pastebin.com/i7s0mTvb. Any ideas on what can be happening? |
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| 10:07.40 | ipalmer | Morning all, I'm running * 1.8.5 and have a couple of SIP channels get stuck after using chanspy. I want to manually end these channels, I have tied channel hangup request but this hasn't worked. I have also been advised to use the ami to redirect the channel 'off the cliff' not sure what this means but I have tried to redirect the stuck channel into a non existent context but it still appears stuck. Any ideas how I can ki |
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| 10:14.56 | Greenlight | ipalmer: Restart Asterisk |
| 10:15.22 | ipalmer | Greenlight: Thanks want to avoid this if possible, it's a live system |
| 10:16.31 | Greenlight | If a hangup request failed, and also a redirect, then you've little choice |
| 10:17.05 | Greenlight | I've seen that sort of behaviour with ChanSpy channels, although not since upgrading to 11 |
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| 10:17.17 | ipalmer | Greenlight: Thought as much but it seems a bit of a big hammer for one extension |
| 10:17.49 | Greenlight | It seems to be happening quite regularly for you though? |
| 10:18.30 | ipalmer | Greenlight: It's the same extension as last week, it seemed to resolve itself in the end but the same extension popped up stuck again this morning |
| 10:19.05 | ipalmer | Greenlight: Maybe it's the same agent not ending the chanspy properly |
| 10:19.06 | Greenlight | Do you know what it's trying to spy on ? |
| 10:19.33 | Greenlight | I don't see how they can end it "improperly" -- don't they just hangup ? |
| 10:19.52 | ipalmer | Another sip endpoint which is taking a call from the isdn line |
| 10:20.46 | ipalmer | I assume they do just hangup but as I'm not there can't be sure, maybe the person they were spying on ended their call before they ended the spy |
| 10:21.00 | ipalmer | and this caused an issue |
| 10:21.11 | Greenlight | Current version is 1.8.22, so it's worth updating to that at the very least |
| 10:21.46 | ipalmer | I'm currently on 1.8.5 |
| 10:23.05 | ipalmer | maybe I'll update them to 11, I'm just looking at the docs. Do you know if much work has been done with chanspy in this version? |
| 10:23.54 | Greenlight | I know my agents use it extensively and I've not had any of those sorts of issues with 11 ever. |
| 10:24.17 | ipalmer | Cool, did you have problems before then? |
| 10:24.27 | Greenlight | Yes, similar to what you describe |
| 10:25.01 | Greenlight | I just had a cron job restart asterisk nightly to clear them back in the day :) |
| 10:25.20 | ipalmer | Hmmm, looks as though this is the way to go then, looks like a weeks worth of testing in store for me then |
| 10:25.38 | Greenlight | 1.8 -> 11 is quite painless to be honest |
| 10:25.52 | ipalmer | I like your style re the cron job, trouble is they're 24*7*365 |
| 10:27.50 | ipalmer | doh, first issue i can see is Macros no longer work but not a big problem |
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| 12:43.49 | alexhld | hey all, curious issue with dahdi, in asterisk cli "dahdi show status" shows all spans OK, on bash prompt "dahdi_cfg -tvv" doesnt show any errors, however on the asterisk cli "pri show spans" is blank, no output at all |
| 12:43.54 | alexhld | any ideas? |
| 12:45.43 | WIMPy | No pri support enabled? |
| 12:45.55 | alexhld | hang on, im being dumb, dahdi-channels isnt included in chan_dahdi.conf |
| 12:45.58 | alexhld | obvious |
| 12:46.00 | alexhld | doh |
| 12:47.04 | alexhld | nevermind |
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| 14:00.14 | Katty | o/ |
| 14:00.29 | [TK]D-Fender | <PROTECTED> |
| 14:03.45 | leifmadsen | \o |
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| 14:20.17 | Kobaz | anyone have any problems with rtptimeout dropping lan to lan calls randomly? |
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| 14:26.02 | workingcats | hello, i was wondering if it was possible to turn off the meetme notification sounds dynamically? |
| 14:26.29 | workingcats | wait sorry |
| 14:27.14 | workingcats | that wouldn't actually work, let me rephrase: is there a way to have the meetme sounds only for one of the parties of the conference? it is easy to distinguish the party that should have the noise from those who shouldnt |
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| 14:34.29 | workingcats | oh i think i may have found something after all |
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| 14:36.12 | workingcats | no i didnt actually |
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| 15:34.09 | Kobaz | [2013-06-25 11:20:27] NOTICE[14163]: chan_sip.c:25939 check_rtp_timeout: Disconnecting call 'SIP/420-00000c0d' for lack of RTP activity in 61 seconds |
| 15:34.34 | Kobaz | rtptimeout=0 |
| 15:34.51 | Kobaz | [2013-06-25 11:20:27] NOTICE[14163]: chan_sip.c:25939 check_rtp_timeout: Disconnecting call 'SIP/420-00000c0d' for lack of RTP activity in 61 seconds |
| 15:34.55 | Kobaz | er wrong paste |
| 15:34.58 | Kobaz | rtpholdtimeout=0 |
| 15:35.20 | igcewieling | Kobaz: looks like you didn't sip reload |
| 15:35.24 | Kobaz | nope |
| 15:35.27 | Kobaz | sip show settings...: |
| 15:35.31 | igcewieling | just comment out the settings and see what happens |
| 15:35.44 | Kobaz | <PROTECTED> |
| 15:35.44 | Kobaz | <PROTECTED> |
| 15:35.52 | Kobaz | yeah, commented out does the same thing |
| 15:36.05 | igcewieling | Kobaz: none of those settings on any of your peers? |
| 15:36.09 | Kobaz | nope |
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| 15:36.16 | igcewieling | then file a bug report on jira |
| 15:37.05 | Kobaz | well, yeah i know it's a bug |
| 15:37.27 | Kobaz | just wondering if there's some super-secret way to disable rtptimeout |
| 15:37.28 | igcewieling | gads, I love clustered Asterisk (most of the time) |
| 15:37.57 | igcewieling | Kobaz: I think the workaround is for clients to send RTP. |
| 15:38.32 | Kobaz | haha |
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| 15:40.01 | cdavis | Did anything replace chan_mobile? I have a bluetooth adapther that I want to try to get my cell phone working with |
| 15:40.04 | igcewieling | What is so funny. Clients should be sending RTP. If they are not, then you have a problem. |
| 15:40.55 | igcewieling | cdavis: unless you just want to use it for fun, don't bother with chan_mobile / chan_dongle. you should read the upgrade*.txt files, the change should be mentioned there. |
| 15:41.15 | Greenlight | Some clients don't sent RTP during "silent" periods, or during "hold". |
| 15:41.18 | Greenlight | *send |
| 15:41.34 | Greenlight | That's usually configurable on the client though |
| 15:41.35 | igcewieling | Greenlight: As asterisk does not support VAD, the first one is not a valid |
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| 15:42.12 | Greenlight | Asterisk may not support it, but alas some clients try it anyway |
| 15:43.04 | cdavis | igcewieling: thanks |
| 15:43.47 | igcewieling | Greenlight: those clients have had issues since the beginning of (asterisk) time. |
| 15:46.37 | Greenlight | Or since an "user" decided to try a new setting :) |
| 15:47.35 | Kobaz | Greenlight: cisco spa phones |
| 15:47.46 | Greenlight | Cisco eww |
| 15:47.50 | Kobaz | heh yeah |
| 15:48.10 | igcewieling | Greenlight: "Cisco" SPA phones are good for the price |
| 15:48.20 | Kobaz | heh true, linksys |
| 15:48.22 | Katty | ICE WEASEL |
| 15:48.28 | Katty | hugs Kobaz |
| 15:48.31 | Kobaz | yay |
| 15:48.31 | Katty | returns to lurking |
| 15:48.36 | Kobaz | aw |
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| 15:50.02 | igcewieling | squirrel grrl! |
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| 16:12.00 | Weezey | Is it possible to set up a Queue on one asterisk box that has agents on a different box? |
| 16:12.15 | Greenlight | It sure is |
| 16:12.51 | Greenlight | Asterisk just sees a sip endpoint |
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| 16:13.07 | Greenlight | It doesn't matter if that endpoint is another box, or an actual handset |
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| 16:23.19 | Weezey | Greenlight: awesome! Thanks! |
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| 16:35.30 | jagster` | anyone know where i can find the man page for the command "reload" on asterisk 1.4 |
| 16:35.47 | jagster` | i see logger and module reload but not plain "reload" |
| 16:38.05 | Greenlight | 1.4 eeek |
| 16:40.27 | jagster` | looks like 5) Reload All : asterisk -rx reload |
| 16:41.21 | igcewieling | jagster`: "help" in the CLI |
| 16:41.29 | jagster` | sweet thanks igcewieling |
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| 16:57.48 | rrittgarn | anybody in the chicagoland area work with Kamalio? |
| 16:58.57 | igcewieling | I doubt anyone would admit something like that. 8-| |
| 16:59.58 | rrittgarn | lol... guess I am in the wrong channel for that... |
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| 17:24.55 | beanie | hello, although this is potentially a bit of a crossover issue between free pbx and asterisk, i'd like to rule out whether it is an issue: i cab't get access to the 7777 simulate inbound call option on my freepbx, I have set up a blank inbound route for the purpose. The reason why I wonder if it is not working is because I did an update of the Asterisk modules and did not realise i actually |
| 17:24.55 | beanie | manually needed to delete the old ones. Does anybody have any idea of whether this would actually cause an issue? |
| 17:25.35 | igcewieling | beanie: This is a FreePBX issue, as asterisk has neither 7777 nor call simulation built in. |
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| 17:27.43 | igcewieling | beanie: what modules needed to be deleted. |
| 17:27.52 | igcewieling | You must have upgraded to a new major Asterisk version |
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| 17:33.45 | beanie | igcewieling, hello and thanks for your message :-) I just ran a yum update |
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| 17:34.23 | beanie | bit of a linux newbie - i thought linux was like windows, you download an update it gets rid of unneccessary files as part of the process, you back up any custom stuff if you want to keep it manually |
| 17:35.08 | igcewieling | beanie: if you upgrade your kernel and are using DAHDI, you MUST rebuild and reinstall DAHDI |
| 17:36.16 | beanie | ok, so is it unlikely that, save for what your advising me, that two sets of modules are installed and conflicting with each other? |
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| 17:40.52 | igcewieling | beanie: correct. |
| 17:42.32 | igcewieling | beanie: the kind of conflict you describe is very very uncommon on Liunx |
| 17:42.46 | jmetro | whatd i miss? |
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| 17:53.26 | igcewieling | jmetro: tomorrow's lottery numbers |
| 17:53.55 | jmetro | :< |
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| 18:17.54 | *** join/#asterisk wrouesnel (~wrouesnel@203-206-134-36.perm.iinet.net.au) |
| 18:18.39 | wrouesnel | why would an extension of _X not be matching a simple digit like 2 or something? |
| 18:18.58 | [TK]D-Fender | wrouesnel: Show us your exact code and your call debug |
| 18:18.59 | igcewieling | _X would match "2" |
| 18:19.01 | [TK]D-Fender | ~pb |
| 18:19.01 | 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. |
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| 18:19.09 | [TK]D-Fender | indded it should |
| 18:19.19 | [TK]D-Fender | Which means either your change is not applied.. or you did something else wrong around it |
| 18:19.30 | igcewieling | my guess is the phone dialplan |
| 18:19.47 | wrouesnel | I can't see how. I have some literal single-digit extensions in the same dialplan, which work |
| 18:20.11 | jmetro | put verboses here and there |
| 18:20.16 | jmetro | say whats happening |
| 18:20.17 | [TK]D-Fender | wrouesnel: Something is wrong around it. Show us the call and your dialplan. |
| 18:20.22 | jmetro | and paste the cli output in a pb |
| 18:21.53 | wrouesnel | this is the dialplan fragment its in: http://pastebin.com/cPrA6uwu |
| 18:22.41 | igcewieling | wrouesnel: 1) stop reading docs which became obsolete in 2005 |
| 18:23.38 | [TK]D-Fender | wrouesnel: Show us the call... |
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| 18:23.43 | jmetro | same => n it up |
| 18:24.15 | igcewieling | wrouesnel: you should read the Asterisk book. |
| 18:24.18 | igcewieling | ~book |
| 18:24.18 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:25.00 | ChannelZ | of course none of this addresses his issue. Need to see console output. |
| 18:25.25 | wrouesnel | this is the call: http://pastebin.com/GNLPWh9q |
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| 18:28.23 | [TK]D-Fender | wrouesnel: I'd be guessing you didn't apply your changes. |
| 18:28.40 | igcewieling | wrouesnel: "reload" in the Asterisk CLI and try again |
| 18:28.42 | [TK]D-Fender | wrouesnel: "dialplan show" in CLI should show you what's active... perhaps you forgot to "reload" |
| 18:29.01 | ChannelZ | yeah or something barfed in a parse above |
| 18:29.03 | jmetro | dialtab reltab |
| 18:29.19 | ChannelZ | eh? |
| 18:30.05 | igcewieling | Oddly good dialplan for someone using docs out of a museum though. |
| 18:30.11 | ChannelZ | Oh. Tab key you are trying to say. |
| 18:30.43 | wrouesnel | igcewieling: this was all set up years ago and very rarely needs altering |
| 18:30.45 | ChannelZ | Inflexible, but still probably functonal. |
| 18:31.03 | wrouesnel | but i think i spotted it - a DB lookup is returning null and kicking it off to the i context. |
| 18:31.14 | wrouesnel | what's the new hotness in dialplans anyway? |
| 18:31.35 | igcewieling | wrouesnel: "n" priority, priority labels, the "same" extension |
| 18:31.43 | ChannelZ | extensions made by whistling sounds rather than dialing numbers. |
| 18:34.02 | wrouesnel | ok that was weird. _X didn't work, _X! did, and then i found my other bug. strange. |
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| 18:34.26 | igcewieling | wrouesnel: indeed. ! isn't very useful |
| 18:34.57 | [TK]D-Fender | _X should work |
| 18:35.08 | [TK]D-Fender | And there was no time it wouldn't |
| 18:35.19 | wrouesnel | ok no looks like _X is working, and it was just the other bug in a DB evaluation |
| 18:35.22 | [TK]D-Fender | There was never a stated bug around this... ever in my time in * |
| 18:35.25 | igcewieling | of course, doing what was asked (pastebinin the output of dialplan show") would tell us if there was a reload issue or not |
| 18:35.46 | jmetro | whatever you do, dont do a "dialtab savtab" |
| 18:36.02 | wrouesnel | the specific line that was going wrong was _X,1,GotoIf($[${DB(DoctorExtensions/${EXTEN})}]?:i,1) which should've been _X,1,GotoIf($[${DB_EXISTS(DoctorExtensions/${EXTEN})}]?:i,1) |
| 18:36.33 | igcewieling | and yet, the ERROR message you pasted indicated that extension 2 was not found. |
| 18:36.39 | [TK]D-Fender | wrouesnel: No, it still should have matched the first line... |
| 18:36.58 | igcewieling | wrouesnel: in any case glad you got it working. |
| 18:37.00 | wrouesnel | good point - i don't know. I was reload 'ing constantly while testing it. |
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| 18:39.00 | blizzow | Anyone here have recommendations on getting call data automatically pushed into salesforce? I see a couple commercial "CTI" connectors (Ingenius, Camrivox) out there but was wondering what else might be out there. |
| 18:40.59 | ChannelZ | not specifically, but via AGI et al you can barf pretty much anything you want anywhere you want, so it's probably more of a Salesforce question - what sort of external API do _they_ have |
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| 18:43.18 | blizzow | ChannelZ: Yeah, I was hoping that someone made a plugin or had something written already. |
| 18:45.05 | igcewieling | blizzow: thats funny |
| 18:45.59 | igcewieling | blizzow: most things which directly generate revenue is stuff people don't release as open source. Examples include queue management, billing, telemarketing, etc |
| 18:48.47 | blizzow | igcewieling: I get requests on a quarterly basis to automatically push call data into salesforce. The latest requests also include automated transcription of calls to text and also push that into salesforce. |
| 18:49.40 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 18:49.49 | [TK]D-Fender | blizzow: Go for it.. this is up to you.... |
| 18:50.11 | igcewieling | blizzow: you have multiple things then 1) what data 2) how to extract from asterisk (only thing we can help with) and 3) how to push that data into sales force |
| 18:50.31 | igcewieling | once you get to step 2 and have specific questions, you'll start getting specific answers. |
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| 18:58.12 | wrouesnel | thanks for the help guys! time for me to go to bed |
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| 19:06.11 | beanie | igcewieling - thanks for those comments! I don't have any hardware set up even those i'm aware that DAHDI is the Digium Asterisk Hardware Device Interface - i'm simply going to be using trunks through another company who will provide a uk landline number :-) Is the point made earlier still relevant? |
| 19:06.23 | beanie | oh and softphones! |
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| 19:10.24 | beanie | Blizzow - that's quite fascinating by the way, how goods Asterisk with transcribing call recordings - I would imagine for it to be reliable it would pretty much have to be state of the art! |
| 19:11.14 | blizzow | beanie: I have no clue, it's just a feature request. |
| 19:11.44 | beanie | ah fair enough - could be quite interesting, most telephone banking services in the uk recognise the caller's voice so i guess asterisk must be going in the same direction |
| 19:11.58 | blizzow | They also want automated QA grading based on call transcription. |
| 19:12.54 | beanie | well how would that work in practice!! It would require the pickup of certain key words being used that an Adviser was not aware of to manipulate the system...there's nothing better than a human QA checking the recordings after! |
| 19:14.35 | beanie | if i'm getting your drift correctly, there'd be all sorts of issues with recognising different dialects, frequencies of voice, quality of the call etc |
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| 19:18.45 | blizzow | beanie: I'm more interested in getting call data (caller(agent name), callee (number), call duration, and a link to the call recording) pushed into a salesforce lead record after a call is completed. |
| 19:18.55 | blizzow | I don't even care about transcription at this point. |
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| 19:19.48 | igcewieling | blizzow: check the CDRs for that information, you'll have to do your own call recording linking of course |
| 19:20.07 | beanie | my knowledge is exceptionally basic to the extent i'm not even a competetent linux user to be quite honest! I'm learning slowly but it's very interesting what your client/employer is asking about. They are clearly looking at the business model and the challenge i guess is interpreting business need into IT Solution |
| 19:20.47 | [TK]D-Fender | Asterisk does not include any voice recognition. That's all 3rd party, and they all suck by-and-large |
| 19:21.49 | beanie | aha - so D-Fender how are other voice recognition services more affective? my bank, LLoyds in England they have a really smart voice recognition which is very accurate! |
| 19:21.59 | beanie | effective* |
| 19:23.07 | igcewieling | beanie: they spend the GDP of a small country to purchase the high end software |
| 19:23.07 | rojozx | hello.. I am trying to dial to an ivr and supress the ivr greeting sending my call directly to the extension i want to reach. I use SENDDTMF to dil the extension but there is still another message before the extension starts to ring. I tried to use the G option in the DIAL command which sends both calls to another context then bridge them again but I still get the second message. I use the G option as documented here http://wiki.projectdiastar.org/index |
| 19:23.35 | igcewieling | beanie: keep in mind, if you have a small vocabulary voice rec is reasonably good. For example, in an IVR |
| 19:23.42 | jmetro | rojozx: huh |
| 19:23.56 | jmetro | youre trying to allow people to dial extensions before the greeting is finished? |
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| 19:24.01 | igcewieling | rojozx: that won't work. See the D() option to Dial |
| 19:24.27 | igcewieling | The "w" digit in D() will wait .5 second |
| 19:25.26 | rojozx | I looked at all the options.. I dont want the caller to hear any greeting |
| 19:26.10 | rojozx | The first greeting is dealt with by sending the dtmf of the extension |
| 19:26.34 | jmetro | why not just put them the way you want them to go |
| 19:26.45 | jmetro | and avoid the greeting in the first place |
| 19:26.58 | rojozx | i have no control over the ivr |
| 19:27.10 | rojozx | I want to bypass the messages |
| 19:27.22 | igcewieling | jmetro: he is "hacking" the IVR |
| 19:27.26 | rojozx | and dial directly to the extension |
| 19:27.28 | jmetro | ... lol |
| 19:27.33 | rojozx | no hacking.. |
| 19:27.42 | igcewieling | rojozx: The D() option to dial should not send audio back to the caller |
| 19:27.48 | rojozx | i just dont want to waste time listening to ads or promotions |
| 19:28.07 | igcewieling | rojozx: you are trying to do something it was never designed for. that is hacking. |
| 19:28.20 | jmetro | igcewieling: i suggest you read the book.. |
| 19:28.24 | rojozx | The D option working fine |
| 19:28.27 | igcewieling | jmetro: REALLY? |
| 19:28.33 | rojozx | as it bypasses the first greeting |
| 19:29.05 | igcewieling | rojozx: why can't you extend the data passed to D() to handle the following greeting too? |
| 19:29.29 | jmetro | igcewieling: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-Revolution-SECT-3.html |
| 19:29.29 | jmetro | L3 |
| 19:29.30 | jmetro | :3* |
| 19:29.37 | rojozx | Thats why I am here... |
| 19:30.14 | igcewieling | jmetro: that is pretty much what I was saying |
| 19:33.22 | beanie | so must i rebuild and reinstall dahdi even though i don't use any hardware based phones, i just use softphones |
| 19:33.48 | igcewieling | beanie: or stop loading dahdi and stop loading chan_dahdi |
| 19:34.37 | jmetro | dont need dahdi for sip phones anyway |
| 19:35.38 | igcewieling | you need it for meetme if you are using meetme |
| 19:35.44 | beanie | ahhhh |
| 19:36.26 | beanie | oh crumbs, im now connecting to my server as an external rather than as an internal user and i'm getting no sound inbound, not sure about outbound |
| 19:36.56 | beanie | just was going to ask about the specific effor message fro 7777 which is different from dialing an unrecognised number |
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| 19:53.18 | eirirs_ | teh stuph |
| 19:54.13 | [TK]D-Fender | "7777" means nothing to us, and your meaning by "internal" vs "external" user is also vague |
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| 19:57.19 | beanie | ooops sorry! 7777 is the simulate inbound call function in free pbs - internal in my context means people on my internal home network and external, people connecting from outside of where my local network is located |
| 19:57.25 | beanie | pbx |
| 19:58.30 | [TK]D-Fender | Go look at your calls. |
| 19:58.50 | beanie | as in the error msgs? |
| 19:59.03 | igcewieling | beanie: You have not said a non-freepbx thing in ages. Maybe you should start asking on............#FreePBX |
| 19:59.34 | [TK]D-Fender | Ans in the ENTIRE call with SIP DEBUG enabled. |
| 19:59.45 | [TK]D-Fender | So you can see if it's passing wrong information somewhere. |
| 19:59.56 | beanie | http://bin.cakephp.org/view/1550046809 |
| 20:00.06 | [TK]D-Fender | SIP DEBUG <- |
| 20:00.09 | beanie | i did look through this but couldnt see anything obvious |
| 20:00.10 | [TK]D-Fender | "sip set debug on" <----------- |
| 20:00.10 | beanie | yeah sure |
| 20:00.14 | beanie | just about to do that again |
| 20:00.25 | [TK]D-Fender | <PROTECTED> |
| 20:00.30 | [TK]D-Fender | that is 777 not 7777 |
| 20:00.39 | [TK]D-Fender | pay attention to the number being processed |
| 20:00.52 | beanie | yeah but if you look further down i did dial it correctly and it still did not work :-) |
| 20:01.16 | [TK]D-Fender | And do you have an ANY/ANY route set up for this? |
| 20:01.21 | [TK]D-Fender | I imagine not.... |
| 20:01.27 | beanie | certainly do :-) |
| 20:01.40 | [TK]D-Fender | This is a crappy test and you should be using a Custom Destination for this |
| 20:01.49 | [TK]D-Fender | not properly... |
| 20:02.23 | beanie | well, is it not a case of simply creating an inbound destination and leaving everything blank except for the ivr option at the bottom |
| 20:03.43 | [TK]D-Fender | beanie: Do you also see that feature code enabled under that value in the first place? |
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| 20:10.58 | beanie | well i was trying to look but frustratingly i can't get access to the gui atm because of some firewall configs that have been changed by the kindness of someone very helpful :-) |
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| 20:18.38 | [TK]D-Fender | starts packing up... |
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| 20:24.31 | beanie | yeah 7777 is on my extension list which i printed so presumably it wouldn't print it unless it was enabled |
| 20:24.33 | beanie | :) |
| 20:25.26 | beanie | this is is the latest sip errors http://bin.cakephp.org/view/1156180094 |
| 20:27.45 | beanie | "Authorization: Digest username="048665",realm="asterisk",nonce="3a308d48",uri="sip:7777@2.122.1.225:5060",response="023214fc3119cf6fdf2afef9fc66b0b2",algorithm=MD5 |
| 20:27.45 | beanie | Reason: SIP;description="RTP stream closed due to inactivity; mediaType=AUDIO"" is that the issue |
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| 21:34.03 | saint_ | hi all - can someone explain to me how to setup an outbound caller ID for a sip provider, in asterisk 11.4 ? |
| 21:35.37 | [TK]D-Fender | saint_: Change the callerid. |
| 21:35.40 | [TK]D-Fender | saint_: Dial. |
| 21:35.47 | [TK]D-Fender | saint_: The End. |
| 21:36.34 | saint_ | like Set(CALLERID(xxx)) ..? |
| 21:36.54 | [TK]D-Fender | yes |
| 21:38.38 | saint_ | Gee that was an easy one. |
| 21:38.49 | saint_ | thanks for the info. Is there a way to pass a CNAME too ? |
| 21:39.08 | saint_ | CNAM I meant |
| 21:40.08 | saint_ | [TK]D-Fender: how do you manage the option for users who want to hide their caller ID ..? |
| 21:40.50 | [TK]D-Fender | look for other functions with "CALLER" in them.... |
| 21:42.48 | saint_ | and is there a way to assign a variable to "fist last" <number> so I can simply do Set(CALLERID(all)=${VARIABLE_NAME}) ..? or is it illegal ? |
| 21:44.16 | [TK]D-Fender | sure |
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| 21:53.20 | *** join/#asterisk Mon|A|rch (~sbean@72.29.180.35) |
| 21:55.25 | Mon|A|rch | hey, I'm testing out auto-dialout using .call files. I've got the spool module loaded (i believe), and have a decently formatted file format. When i move files to the spool though, after a while I get an error: call could not go through reason (1) hangup |
| 21:55.32 | Mon|A|rch | I'll pastebin the cli log and the call file |
| 21:56.22 | Mon|A|rch | asterisk 1.8 btw |
| 21:58.51 | Mon|A|rch | also, the files are being deleted, but not moved to outgoing_done |
| 22:01.25 | Mon|A|rch | call file: http://pastebin.com/M1eCfYax |
| 22:01.39 | Mon|A|rch | some personal info edited out |
| 22:03.01 | Mon|A|rch | even the most general advice on why call files would fail would be helpful |
| 22:05.29 | igcewieling | don't use quotes or the leading 1 in the callerid |
| 22:06.54 | Mon|A|rch | good to know |
| 22:07.51 | Mon|A|rch | so just: CallerID: some id <1234567890> |
| 22:07.52 | Mon|A|rch | ? |
| 22:07.57 | Mon|A|rch | also, thanks igc |
| 22:08.05 | igcewieling | CallerID: some id <234567890> |
| 22:08.12 | igcewieling | no leadig 1 |
| 22:08.21 | Mon|A|rch | er |
| 22:08.23 | Mon|A|rch | yeah |
| 22:08.28 | Mon|A|rch | that was just me being dumb |
| 22:08.38 | Mon|A|rch | didn't actually mean to include the 1 |
| 22:08.51 | igcewieling | see if that makes any difference. MOST carriers don't care about the CallerID, but some do. |
| 22:09.28 | Mon|A|rch | it seemed like the call never was originated |
| 22:09.29 | Zopieux | anyone else has this annoying "no reply to our critical packet" with SIP? |
| 22:09.34 | Mon|A|rch | I'm trying to find the error in the logs |
| 22:09.47 | igcewieling | You can also use Channel: Local/915555555555@somecontext and dial out from somecontext, then you can see everything in the CLI |
| 22:10.01 | Mon|A|rch | actually, the error I'm having now gave me that iirc |
| 22:10.09 | Mon|A|rch | igcewieling, good idea |
| 22:20.16 | Mon|A|rch | igcewieling, neither of those fixed it unfortunately |
| 22:20.16 | Mon|A|rch | however |
| 22:20.26 | Mon|A|rch | i found the logs |
| 22:20.26 | Mon|A|rch | http://pastebin.com/ica9232s |
| 22:21.00 | Mon|A|rch | i think the Loclal/exten@context might work though |
| 22:21.21 | Mon|A|rch | it actually hit my dialplan |
| 22:24.49 | Mon|A|rch | still, I'd like to know what the deal with that is, not big into hacking my way around |
| 22:25.17 | Mon|A|rch | also, if i use Local/exten@context, do i need to specify Context, Extension and Priority in the file? |
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| 22:37.38 | igcewieling | Mon|A|rch: yes |
| 22:38.13 | igcewieling | Mon|A|rch: unless you have MANY calls, most people use Local/ |
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| 22:38.22 | igcewieling | it makes everything easier |
| 22:38.33 | Mon|A|rch | I can see that |
| 22:38.55 | Mon|A|rch | so, the internet tells me that the call file system chockes with high volume |
| 22:39.09 | Mon|A|rch | how serious is that? what sort of volume does that happen at? |
| 22:39.33 | Mon|A|rch | my use would only be at most 3000 calls a day |
| 22:39.42 | Mon|A|rch | possibly 4000 |
| 22:40.00 | Mon|A|rch | in batches of maybe 100 at a time |
| 22:41.29 | Mon|A|rch | the cycle sort of looks like: server makes file, server sftp's file to asterisk server, mv's file to spool |
| 22:41.31 | Mon|A|rch | rinse, repeat |
| 22:41.43 | Mon|A|rch | so pretty rapid-fire |
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| 23:27.02 | pizlam_ | hello all |
| 23:28.39 | pizlam_ | can anyone help me (newb) with a vicidial express install using asterisk 1.4.44 with dahdi 2.6.1 and an analog TDM800P (digium) with telco lines |
| 23:29.00 | pizlam_ | getting "that is an invalid ext" evertime I dial in or out |