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| 04:09.51 | Samot | Anyone use agiphp? I'm running into an issue with the say_phonetic and say_alpha functions. They cause the call to die. |
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| 07:17.27 | pic0frame | Hello friends, Is it possible to redirect a call if the que has no members logged in? |
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| 07:21.17 | AviiNL | There should be a setting to not go into the queue if there are no members/agents, then it'll continue with the next line in the dialplan |
| 07:21.36 | AviiNL | pic0frame: ^ |
| 07:22.13 | pic0frame | AviiNL: I taought so... but I can find the function |
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| 07:26.42 | AviiNL | Oh, we are using the ExecIf for that |
| 07:27.06 | AviiNL | lookup QUEUE_MEMBER |
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| 07:33.35 | pic0frame | AviiNL: ohh thx! Ill look it up |
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| 08:08.54 | madduck | what should I use as first argument to Dial() when all I really want is just Voicemail? |
| 08:09.12 | madduck | confused |
| 08:12.10 | AviiNL | madduck, Don't use the Dial application but instead use Voicemail ? |
| 08:35.05 | madduck | AviiNL: doh ;) |
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| 09:03.17 | BarthezZ | I have two asteriskes, one (ast-A) connected to an pstn uplink (SIP) and sending calls to the other one (ast-B) which has the phones connected... Is there a way to prevent ast-A from propagating ringing events from ast-B to the PSTN? |
| 09:04.45 | BarthezZ | This is due to an issue with an 180 ringing after the 183 session progress, where the pstn uplink decides that this is invalid and stops sending out any audio |
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| 10:57.49 | linjan | hello everybody. im trying to find the best way to provide web access for call listening. |
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| 10:58.16 | linjan | simple, like sip-phone. it may be as a anonymous sip client as a member of conference. |
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| 11:20.29 | Syco | Hello, I'm having some problems with setting up dahdi 2.11.1, libpri 1.5.0, wanpipe 7.0.19 and asterisk 13.9.1 (I don't think versions matter, it never worked). |
| 11:20.29 | Syco | Calls come in from the telephony switch with CPC=10, in my dialplan I only have 1 line that does Dial(DAHDI/g0/xxxxxxxxxx), but calls go out with CPC=0. I can see the CPC in the traces I run on the telephony switch. |
| 11:20.29 | Syco | Where is the CPC setting? Can it be changed? |
| 11:20.29 | Syco | Thanks |
| 11:21.48 | WIMPy | What is CPC? |
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| 11:24.29 | Syco | it's the "Calling Party Category", 10 is Ordinary calling subscriber, 0 is unknown. I was doing some reading here http://www.dialogic.com/webhelp/IMG1010/10.5.0_ER4/WebHelp/Description/Interworking/sipss7interworking.htm |
| 11:24.29 | Syco | I'm starting to think this flag is only related to ss7 and that I lose it in the ss7/euroisdn translation. But I would need some confirmation on that. |
| 11:25.34 | WIMPy | Ah. SS7. You didn't mention that. |
| 11:25.55 | WIMPy | So what ARE you using? |
| 11:29.25 | Syco | that's because I consider ss7 out of my setup, here's what I have: provider --> ss7 --> telsis --> euroisdn --> pri/sangoma --> asterisk. |
| 11:29.25 | Syco | The telsis does the ss7/auroisdn conversion, So I can see telsis sending the call to asterisk in euroisdn with cpc=10 and receiving it back with cpc=0 |
| 11:30.12 | WIMPy | Where do you see that cpc on the euroisdn link? |
| 11:33.34 | Syco | I use the telsis call logger utility, it shows 4 entries per call: |
| 11:33.34 | Syco | call in, ss7, (provider to telsis), cpc=10 |
| 11:33.34 | Syco | call out, euroisdn, (telsis to asterisk), cpc=10 |
| 11:33.34 | Syco | call in, euroisdn, (asterisk to telsis), cpc=0 |
| 11:33.34 | Syco | call out, ss7, (telsis to provider), cpc=0 |
| 11:34.24 | WIMPy | So you don't actually see it. |
| 11:35.15 | WIMPy | It doesn't exist. |
| 11:37.56 | Syco | I can't get a trace, but what you mean it doesn't exists? |
| 11:37.56 | Syco | you think the telsis is just showing me the last value for that call (10), and when the call goes out again euroisdn doesn't contain a CPC field, so it gets set to 0 automatically? |
| 11:39.36 | WIMPy | yes |
| 11:40.27 | WIMPy | Maybe you can configure the telsis to set somethign else for calls comming from the euroisdn interface. But it's not transmitted there. |
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| 11:43.08 | Syco | Or I could use ss7 everywhere and avoid the translation, well, thanks very much |
| 11:43.18 | Syco | I'll give it a go later today |
| 11:44.10 | WIMPy | That would be another possibility, yes. |
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| 11:58.25 | zeorin | Hello, I'm trying to help out a small non-profit move away from their current telecoms solution. They've got an after hours call forwarding requirement. It's really important that those calls get answered (think suicide hotlineâalthough it's not that). I don't know what hardware I'll need for an Asterisk computer to run it on. I've drawn up the scenario here: http://pastebin.com/PFnhN8md (1 min read). |
| 11:58.31 | zeorin | Please could someone take a look and give me an estimate on the the hardware? I'm a Linux veteran but way out of my depth with telephony. |
| 11:58.45 | zeorin | I appreciate any help ⺠|
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| 12:01.45 | dadrc | if you have just the one phone line, basically anything will da |
| 12:01.48 | dadrc | *do |
| 12:02.28 | zeorin | dadrc: currently there's just the one phone line. Surely I'd need a second line to forward a call to an external number? |
| 12:02.36 | WIMPy | Using POTS lines is a PITA. |
| 12:02.58 | WIMPy | You'd usually tell the exchange to do so. |
| 12:03.18 | dadrc | If you want to do the forwarding yourself, yeah |
| 12:03.34 | zeorin | WIMPy: not sure what you mean? Tell the exchange to set up the forwarding as I've described? |
| 12:04.05 | WIMPy | Yes. That's the nature of forwarding. |
| 12:04.32 | zeorin | We're in South Africa. When it comes to fixed lines, there's only one provider: Telkom, a state-owned, corrupt, inefficient, expensive, and incompentent monopoly. |
| 12:05.11 | zeorin | So currently the best they've been able to do for us is this: caller calls, auto-detect region, play a recorded message telling them what number to call. That's it. |
| 12:05.37 | zeorin | When we've needed to change the numbers, sometimes they get it wrong, sometimes it takes 1 month to effect the change, etc. |
| 12:06.03 | WIMPy | Shouldn't be a problem to forward them to the correct number, either. |
| 12:06.19 | zeorin | Also, the numbers are volunteer's. Unfortunately that means a lot of calls don't get answered because they're not as vigilant as employees might be. We're trying to increase the number of answered calls. |
| 12:06.45 | zeorin | Anyway, the telco is not able to give us anything better that we can actually afford. |
| 12:06.53 | WIMPy | Sounds like you'd need a few more lines. |
| 12:07.09 | zeorin | We'll settle for one just now. The calls aren't very frequent. |
| 12:07.26 | zeorin | I.E. just one extra line, if that's enough to forward one call at a time. |
| 12:07.43 | WIMPy | But if you wann to pass the calls on yourself, that means at least a 2nd channel. |
| 12:07.52 | zeorin | We're OK with that. |
| 12:08.27 | WIMPy | But you really shouldn't try that with POTS lines. You're likely unable to find out if the call was answered. |
| 12:08.46 | WIMPy | Either get an ISDN line or use an ITSP instead. |
| 12:09.19 | zeorin | Would that serve the needs of the scenario I described better? |
| 12:10.07 | WIMPy | yes |
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| 12:12.36 | zeorin | WIMPy: thanks for your feedback, it's been quite valuable. I think this is a little more involved that I anticipated. I still think we'll go for an in-house PBX rather than using the telco (they really suck, and there aren't any alternatives), but it's going to take more time than we thought. |
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| 13:51.24 | DanQuinney | I'm trying to get ODBC MySQL storage setup for voicemail messages but keep seeing the warning "app_voicemail.c:3502 retrieve_file: SQL Get Data error! coltitle=category" http://pastebin.com/EPgxxT7m Does anyone have any suggestions? |
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| 15:10.26 | seik0 | your odbc was unable to retrieve category column |
| 15:10.41 | seik0 | DanQuinney: |
| 15:11.40 | seik0 | more exact: odbc was unable to retrieve data from column with name "category" |
| 15:12.36 | DanQuinney | The issue goes away if I set a variable called 'VM_CATEGORY' |
| 15:13.18 | DanQuinney | Although I'm unsure why |
| 15:13.23 | seik0 | maybe it is reverse: odbc was unable to set null to category? |
| 15:13.36 | seik0 | what type is category column? |
| 15:14.25 | DanQuinney | http://pastebin.com/KCSNt3XG |
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| 15:22.06 | seik0 | quite simple though |
| 15:22.10 | seik0 | nothing special |
| 15:23.22 | DanQuinney | Indeed, I'll have one of our guys look over app_voicemail.c |
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| 15:30.07 | seik0 | you will soon realize, that it is not in app_voicemail +) |
| 15:30.40 | seik0 | odbc to oracle works ok with empty category |
| 15:36.14 | DanQuinney | http://forums.asterisk.org/viewtopic.php?f=1&t=81495 |
| 15:36.26 | DanQuinney | Seems to be very similar to the issue I'm seeing |
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| 17:20.20 | jrun | i see documentations at the beginning of the source code files. deoxygen i guess... where are they stored after being compiled? |
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| 17:21.20 | Gugge | jrun: the same place they were before a compile. In the source code :) |
| 17:21.35 | file | some files have embedded XML for documentation |
| 17:21.47 | file | it's extracted as part of the install process into a master XML document |
| 17:22.07 | jrun | Gugge: hahaha |
| 17:22.18 | file | and is loaded on startup, it is what allows things like "core show application Dial' to work |
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| 17:22.41 | file | by default it's stored in /var/lib/asterisk/documentation |
| 17:23.06 | file | it's also used to populate the wiki with documentation in an automated fashion so everything is in sync |
| 17:23.07 | jrun | i've compiled asterisk fine but i don't plan on running it now. i was wondering if there was a build direcotry where these docs get stored. |
| 17:23.37 | file | they are also in the "doc" directory of the build directory |
| 17:23.42 | jrun | the wiki is fine too but i rather not go back and forth between console and X for web |
| 17:24.06 | jrun | uh, thanks. |
| 17:24.37 | file | you may have to manually do "make doc" to have the document be created |
| 17:24.51 | jrun | make progdoc? |
| 17:25.04 | file | progdoc is doxygen developer documentation |
| 17:25.10 | file | doc is end-user documentation |
| 17:25.18 | jrun | i need the dev docs |
| 17:25.30 | jrun | well, i guess user doc too. |
| 17:26.08 | file | progdoc will generate HTML for the developer stuff using doxygen, doc will just generate the XML for end-user |
| 17:28.24 | jrun | i've got started on doc/api/index.html. i should get my links going to stay on console, i guess. |
| 17:30.32 | jrun | hmm, i've what's up on wiki... i've been looking for the embedded docs in a print-pretty incarnation :) |
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| 17:49.29 | jrun | in app/app_directed_pickup.c: right above definition of find_by_channel(); comments state that this is a helper funtion to walk through ALL channels checking NAME and STATE. |
| 17:50.03 | jrun | the STATE part is what i'm interested in but i don't see how, say, i can search for a channels on hold. any help? |
| 17:51.53 | WIMPy | Not sure what the question is. You search for channels in the HOLD sate. |
| 17:52.12 | jrun | althogh, i'm not even sure how to look for hold state. would that be done by examining ast_channel.music_state ? |
| 17:53.30 | jrun | hmm, there is hold_state too :) |
| 17:53.56 | WIMPy | Just .state IIRC. |
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| 19:40.19 | cyberfab007 | Hey anyone have any resources on add local presence to asterisk dial plan for vicidial , I am having some issues finding anything |
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| 23:39.54 | znf | >xau 30 usd to ron |
| 23:40.11 | znf | oops, wrong channel |
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