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| 02:04.19 | saint_ | beside voicemail email notification, can asterisk send emails from a context directly ? |
| 02:09.25 | pabelanger | sure, why not |
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| 02:41.53 | saint_ | is this correct: Set(CALLERID(name)=${DB(whitelist/${CALLERID(num)})}) |
| 02:51.17 | saint_ | so when I try to call my extension, I have Everyone is busy/congested at this time in the logs. and it does not ring |
| 02:51.25 | saint_ | The set is in front of me, and is hung up. |
| 02:51.28 | saint_ | How can I trace that ? |
| 02:51.31 | saint_ | or debug it ? |
| 02:51.35 | saint_ | or is it a known bug ? |
| 02:51.47 | saint_ | it went into this state after I reached 10 non answered calls |
| 02:54.24 | saint_ | oh great. now when i call my google number, it's not even showing up in my asterisk |
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| 03:00.46 | tmrhmd | What's the most mature Python library to interact with Asterisk? |
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| 03:14.09 | saint_ | how can i send 2 emails in the voicemail.conf , for a specific voicemail ? |
| 03:14.15 | saint_ | do I put a | as a separator , or a space ? |
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| 03:23.03 | ChannelZ | I think you could do just two, using the 'pager' email as a second (though I don't actually know if the content of what it sends is different..) |
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| 03:26.14 | morfin | hello |
| 03:26.52 | morfin | i need a help: i have asterisk 11 with realtime and do not know why but it does not show softphones in sip show peers |
| 03:30.50 | [TK]D-Fender | it won't unless you set rtcache=yes |
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| 03:49.38 | nc11235 | i am migrating a Glenayre (now Movius, I guess) voicemail system to asterisk and i would like to preserve the existing Glenayre subscriber messages and greetings. I have made a backup tape and can read the tape on my linux box. but i have not been able to transform the Glenayre audio into wav, or anything for that matter. has anyone here had any experience with converting Glenayre audio? |
| 03:50.22 | nc11235 | for starters i don't even know what format it is in. |
| 03:55.43 | [TK]D-Fender | nc11235: it's been mentioned here a grand total of 6 times.. ever. And from people moving away from and no mention of getting to keep any of it. |
| 03:56.05 | [TK]D-Fender | nc11235: As they are supposed to support their own product, perhaps you should ask them directly |
| 03:56.25 | nc11235 | we don't have support…it is expensive and we are cheap :-) |
| 03:56.51 | nc11235 | but thank you [TK]D-Fender |
| 03:57.47 | [TK]D-Fender | You could probably cheat by recording the audio as played back through it.... |
| 03:58.22 | [TK]D-Fender | Of course there will be some loss for doing it, but maybe it'll be enough for you |
| 03:58.29 | nc11235 | hmmm that's an idea. i don't know how bad the honchos want to keep this audio for the customers |
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| 04:06.36 | [TK]D-Fender | I'd say about ...."it is expensive and we are cheap" ... much :) |
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| 08:03.39 | madhatt3r | helloAll |
| 08:08.09 | ChannelZ | ahoyhoy |
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| 08:26.23 | apb1963_ | Hello Mr. Burns |
| 08:27.31 | apb1963_ | ChannelZ: That's you |
| 08:32.27 | ChannelZ | I seem to be having difficulty operating my telephone machine. |
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| 08:47.49 | apb1963_ | You must crank the handle |
| 08:52.38 | *** join/#asterisk dago100 (~dago100@cust.static.46-14-221-197.swisscomdata.ch) |
| 08:54.21 | dago100 | Hi all, I bought a digium B410P Card. Phone calls in and out function. But when I do a IVR I get no dtmf inbound. |
| 08:54.59 | ChannelZ | wtf no dtmf? gd! |
| 08:55.35 | apb1963_ | I had that problem once |
| 08:55.49 | apb1963_ | ChannelZ told me how to fix it. |
| 08:55.56 | ChannelZ | I did? |
| 08:56.13 | apb1963_ | Well I didn't read it on the back of a cracker jack box |
| 08:56.34 | ChannelZ | Hmm. |
| 08:56.42 | apb1963_ | well I might have |
| 08:56.54 | apb1963_ | But it's fairly unlikely. |
| 08:57.17 | apb1963_ | After all.. I don't eat cracker jacks |
| 08:58.06 | ChannelZ | Well I don't really know anything about BRI so I'm not sure how I helped |
| 08:58.29 | apb1963_ | It might have been in a fortune cookie... I don't generally eat them either, but they come with the meal. |
| 08:58.36 | ChannelZ | Either some DAHDI setting or perhaps how your IVR is designed is not actually working, I dunno |
| 08:59.27 | ChannelZ | You could turn on DTMF debug and see if the thing is even aware of what is going on |
| 08:59.44 | apb1963_ | I have a dim recollection of something about "signal ringing" and a timeout of something somewhere... but, you can't always believe fortune cookies. |
| 09:00.56 | apb1963_ | I've just recently started a fortune cookie collection for exactly this reason... I can't remember what they all say. I'm up to two now. |
| 09:01.18 | apb1963_ | neither says anything about dtmf though |
| 09:04.21 | apb1963_ | does he know how to turn on dtmf? |
| 09:04.33 | apb1963_ | s/dtmf/dtmf debugging/ |
| 09:06.58 | din3sh | apb1963_: which asterisk version? |
| 09:07.22 | apb1963_ | dago100 which asterisk version? |
| 09:08.10 | din3sh | yeah sorry :p |
| 09:08.16 | apb1963 | no worries :) |
| 09:08.32 | din3sh | i had similar dtmf issues inbound |
| 09:08.40 | apb1963 | I however, am running latest & greatest.. for the record. |
| 09:08.44 | din3sh | had or stll have |
| 09:09.01 | din3sh | asterisk 12? |
| 09:09.02 | apb1963 | so you didn't resolve it? |
| 09:09.08 | apb1963 | 12? there's a 12? |
| 09:09.13 | din3sh | haha |
| 09:09.20 | apb1963 | hurries to upgrade |
| 09:09.33 | din3sh | gd luck :P |
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| 09:09.45 | apb1963 | No it's 1.2 |
| 09:09.49 | apb1963 | ;P |
| 09:10.15 | din3sh | https://issues.asterisk.org/jira/browse/ASTERISK-19610 |
| 09:10.28 | apb1963 | Did you know vicidial uses something like that? 1.2 or 1.4 or something |
| 09:10.40 | din3sh | Affects Version/s: 1.8.10.1, 1.8.11.0, 10.2.1, 10.3.0 |
| 09:10.44 | dago100 | 11.2.1 |
| 09:10.56 | dago100 | asterisk version is 11.2.1 |
| 09:11.10 | eirirs | <-- runs 1.6 |
| 09:11.15 | eirirs | it isn't affected, no? :P |
| 09:12.18 | apb1963 | I refuse to debug anything that requires millisecond timing |
| 09:12.25 | din3sh | lol |
| 09:12.45 | din3sh | i am having serious hdlc errors on a dahdi line |
| 09:12.53 | apb1963 | I'm gettin' too for dat |
| 09:12.58 | din3sh | smtimes inbound calls dont even reach asterisk |
| 09:13.03 | apb1963 | ^old^ |
| 09:13.09 | din3sh | too old you mean? |
| 09:13.10 | din3sh | ;) |
| 09:13.23 | apb1963 | don't make fun of me |
| 09:13.40 | apb1963 | it'll happen to you |
| 09:13.59 | apb1963 | what were we talking about? |
| 09:14.06 | din3sh | dtmf |
| 09:14.07 | din3sh | :D |
| 09:14.16 | apb1963 | he's running 11.2.1 |
| 09:14.45 | apb1963 | hey... so am I |
| 09:14.47 | din3sh | dago100: the incoming call might be coming from a pulse dialing phone |
| 09:14.50 | din3sh | :D |
| 09:16.23 | dago100 | I use libbri -1.4.14, and I calling from mobile. |
| 09:17.19 | apb1963 | dago100 try dtmfmode=inband or dtmfmode=auto |
| 09:18.24 | din3sh | inband messes up dtmf for internal sip phone users |
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| 09:19.08 | din3sh | you have to turn on dtmf debug |
| 09:19.18 | din3sh | and see if the dtmf at least reaches asterisk |
| 09:19.34 | dago100 | Ok I can try, I tunrned on dtmf debuggin in logger.conf, but I do not sea any log entry. |
| 09:19.37 | din3sh | if rxgain is too high, smtimes dtmf is read twice |
| 09:19.59 | dago100 | rxgain and txgain is 0.0 |
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| 09:20.14 | din3sh | wht numbers do you require to key in in the IVR? |
| 09:20.21 | apb1963 | did you reload logger? |
| 09:20.39 | dago100 | yes I did. |
| 09:21.00 | dago100 | By the way I use asterisk with freepbx |
| 09:21.12 | dago100 | 1,2,3 and 9 |
| 09:21.17 | dago100 | IVR keys. |
| 09:21.45 | apb1963 | debug level? |
| 09:22.27 | ChannelZ | I think you need to 'core set debug channel DAHDI/1-1' or whatever channel you've got a call going on |
| 09:23.44 | dago100 | debug 5 |
| 09:23.58 | apb1963 | try 10 just for giggles |
| 09:24.19 | ChannelZ | hmm no that's just for the console. It should log to a log if you have dtmf enabled for one |
| 09:24.38 | ChannelZ | (I just tried) |
| 09:25.44 | din3sh | apb1963: did you get old with age or debugging? |
| 09:25.46 | apb1963 | logger.conf defaults to everything to log |
| 09:26.09 | apb1963 | of couse if you've modified it like I did, then everything goes to the console too |
| 09:26.34 | apb1963 | din3sh: debugging made me old |
| 09:27.05 | dago100 | I activated dtmf, I see no logging entry. |
| 09:27.20 | dago100 | dtmf => notice,warning,error,debug,verbose,dtmf |
| 09:27.32 | apb1963 | I didn't active dtmf, but I get dtmf messages |
| 09:27.39 | apb1963 | s/active/activate/ |
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| 09:28.11 | apb1963 | errrm... is dtmf => a legal construct? |
| 09:28.52 | kaldemar | you can put there what ever you want. it's just a name for a file. |
| 09:29.57 | dago100 | apd1963: you sad to set dtmfmode=auto, where do I set this? |
| 09:30.10 | kaldemar | i prefer using console. then you can enable signaling debug on the BRI and see it inline with the possible dtmf prints. |
| 09:30.27 | apb1963 | dago100: Pay attention to kaldemar. He da man |
| 09:30.38 | kaldemar | pfft |
| 09:30.42 | apb1963 | no seriously |
| 09:30.57 | apb1963 | bows in reverence |
| 09:31.34 | dago100 | kaldmar: Ho I can I set this debug on signaling on? |
| 09:31.49 | apb1963 | I just say random things that sound good... kaldemar actually knows what he's talking about. |
| 09:32.17 | apb1963 | like salmon on a Wednesday. Yum. Good. |
| 09:32.53 | apb1963 | You can tune a piano but you can't debug a fish. |
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| 09:35.27 | kaldemar | dago100: pri set debug 2 span <span> |
| 09:39.08 | dago100 | kaldemar: I see still no dtmf signal. Even when i do a dahdi_monitor -v on the channel I see no dtmf. |
| 09:41.15 | kaldemar | unfortunate. i guess you don't hear them either? |
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| 09:42.57 | dago100 | When i do call from my extension I press the numbers I here the dtmf signal, but when I do on the outside phone, I don't hear anything. |
| 09:43.45 | dago100 | relaxdtmf is set to yes. |
| 09:46.07 | din3sh | i had an issue with incoming dtmf on meetme room |
| 09:46.48 | din3sh | i inadvertently found that if you key in dtmf rapidly, asterisk doesnt understand |
| 09:47.10 | din3sh | if you pause 1 sec between the pin code numbers, the dtmf is then read properly |
| 09:48.08 | din3sh | but dago100 says he inputs only 1 number, it doesnt work |
| 09:48.09 | dago100 | also when I press only one key, it won't understand. |
| 09:48.33 | din3sh | have you tried calling from another number from another provider? |
| 09:49.31 | dago100 | Yes, I tried to call from a other ISDN Phone to asterisk over landline and is the same. |
| 09:51.03 | din3sh | is it a production * box? |
| 09:51.41 | din3sh | try asterisk version 1.8.20 |
| 09:52.26 | dago100 | Why asterisk 1.8? |
| 09:53.39 | dago100 | 11.2 is the next LTS |
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| 10:12.05 | eirirs | tvilsomt |
| 10:12.23 | TSM | anyone have any idea why RPID that is being received via a SIP trunk is not being passed back to the callers phone? it works localy between extensions but not via the trunk, ive run sip debug and can see the data on the trunk but not to the extension |
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| 10:53.35 | WIMPy | din3sh: Just to be sure: Where exactely do you see those DTMF issues? It has nothing to do with entering extensions, has it? |
| 10:56.02 | Rico29 | where can I find asterisk compatible XML scripts for aasta phones (acd, etc...) ? |
| 10:56.49 | din3sh | in my case, i tested from inbound dahdi numbers |
| 10:58.51 | WIMPy | But what kind of function in the dialplan? |
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| 11:04.47 | din3sh | meetme conf room with pin access |
| 11:10.16 | WIMPy | Ok, that's what I thought. |
| 11:10.36 | din3sh | what is the problem according to you~? |
| 11:10.57 | WIMPy | Bad DTMF detection :-) |
| 11:11.45 | din3sh | lol |
| 11:11.47 | WIMPy | No, the thing is that there is an issue with entering extensions too fast if you use dundi. But that's obviousely not your issue. |
| 11:11.52 | din3sh | i figured that much also |
| 11:12.01 | din3sh | naa |
| 11:12.09 | din3sh | interstingly |
| 11:12.17 | din3sh | i tried from 2 numbers |
| 11:12.46 | din3sh | 1 ISDN number from an * box, failed each time the user entered the keys rapidly |
| 11:13.09 | din3sh | i used a regular analogue pstn line, li could get into the meetme room no matter |
| 11:13.14 | din3sh | i keyed in fast or slow |
| 11:13.26 | din3sh | *it could |
| 11:13.55 | WIMPy | Unfortunatly telephones generate DTMF in vvarious ways. |
| 11:14.01 | din3sh | the 2 difference might be line quality, rx or txgain |
| 11:14.16 | din3sh | the analogue line , i could hear the annoucement louder |
| 11:14.26 | WIMPy | Some will even buffer your input and send it at a fixed maximum speed. |
| 11:14.42 | din3sh | but note that both isdn and pstn line are from same provider |
| 11:15.19 | din3sh | the issue further confused me because some countries could dial in and key in the dtmf fast |
| 11:15.51 | din3sh | while for south african callers to work, they had to input the dtmf keys with a pause of 1sec between them |
| 11:18.09 | WIMPy | Well, gain should never exist (neither positive nor negative) als long as you stay digital. Otherwise you've got a serious issue. |
| 11:18.55 | WIMPy | And noone knows what kind of shitty CODEC and connection gets used on international calls these days. |
| 11:19.04 | din3sh | lol |
| 11:19.19 | WIMPy | It might even be due to echo. |
| 11:19.19 | din3sh | the client wouldnt hear it anyways, he said my system is messed up |
| 11:19.20 | din3sh | :D |
| 11:19.39 | din3sh | there's a hardware echo cancel module on the card |
| 11:20.31 | WIMPy | They fail regularly when the delay gets too long. |
| 11:21.05 | WIMPy | Which really isn't taht bad because it makes normal conversation a little hard anyway. |
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| 11:29.57 | c0rnoTa | is there any one alive? |
| 11:31.01 | din3sh | meanwhile |
| 11:31.11 | din3sh | i havent been able to track down the HDLC errors |
| 11:31.45 | ectospasm | din3sh: Digium card? |
| 11:32.35 | c0rnoTa | <PROTECTED> |
| 11:33.00 | ectospasm | c0rnoTa: what version of Asterisk? |
| 11:33.23 | c0rnoTa | ectospasm: 1.8.20.1. But was the same issue on 1.8.12.0 |
| 11:33.53 | din3sh | ecto: yup, dual span TE220B |
| 11:33.55 | ectospasm | c0rnoTa: have you filed a bug report on http://issues.asterisk.org/jira? Follow this guide: 1. CPU utilisation is fine certainly not stressed. The CPU load doesn't ight, the total number of channels is the same, so it doesn't look like it. |
| 11:34.15 | ectospasm | oops, paste fail |
| 11:34.22 | ectospasm | c0rnoTa: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
| 11:34.36 | ectospasm | din3sh: have you registered the TE220 with Digium? |
| 11:34.44 | c0rnoTa | ectospasm: ok, thanks |
| 11:34.47 | ectospasm | din3sh: (http://digium.com/register) |
| 11:35.15 | WIMPy | din3sh: Yes, you really need to get rid of them. If you have transmission errors, that can also influence DTMF detection. |
| 11:35.34 | din3sh | its the same system WIMpy |
| 11:36.20 | WIMPy | IIRC we already looked at your configuration? |
| 11:37.30 | ectospasm | din3sh: HDLC problems (note they are NOTICE level messages, not a WARNING or ERROR) can mean hardware issues. Call Digium Technical Support to have that corrected (be sure to register the TE220 first). |
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| 11:52.53 | TSM | has anyone got a working example of dundi over sip? |
| 11:53.03 | din3sh | WIMPy: http://pastebin.com/ymp0euFw |
| 11:53.07 | din3sh | my settings |
| 11:53.16 | din3sh | Ectospam: thnks, will do |
| 11:59.03 | _zoom_ | fellas, any vicidialers here? my dispo call url is not called when agent disposition? |
| 12:00.45 | WIMPy | din3sh: No crc4? Otherwise that looks ok to me. |
| 12:03.02 | *** join/#asterisk davlefouAMD (~david@197.15.73.137) |
| 12:03.17 | din3sh | i tried setting crc4 |
| 12:03.21 | WIMPy | Didn't you say you've got HWEC? I think on the latest dahdi versions you have to configure echocanceller=hwec. |
| 12:03.22 | din3sh | more errors on the line |
| 12:03.35 | din3sh | i set it to hwec |
| 12:03.40 | din3sh | still had probs |
| 12:03.45 | din3sh | reverted to mg2 |
| 12:03.52 | WIMPy | Not that that should have any influence on the HDLC issue. |
| 12:04.12 | din3sh | i re crimped another cross cable |
| 12:04.19 | din3sh | actually 2 cables |
| 12:04.26 | din3sh | 3rd cable change |
| 12:04.31 | Greenlight | Maybe you have a faulty line |
| 12:04.34 | din3sh | i dont think i screwed up 3 cables |
| 12:04.43 | din3sh | how to test the line |
| 12:04.44 | ectospasm | it's not likely to be a cabling issue |
| 12:04.49 | din3sh | because there is another issue |
| 12:04.53 | Greenlight | You'll need to call the carrier |
| 12:04.59 | din3sh | , i dont kno if its related to the hdlc |
| 12:05.03 | Greenlight | Unless you have the line testing equipment |
| 12:05.09 | WIMPy | You get them only on the 2nd span? |
| 12:05.14 | din3sh | say you phone the DID 4057878 |
| 12:05.24 | din3sh | sometimes you jst get the busy tone |
| 12:05.33 | din3sh | sometimes it goes in to the extension |
| 12:05.39 | Greenlight | How many spans do you have |
| 12:05.44 | ectospasm | din3sh: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-loopback-test-on-a-Digium-E1-T1-card |
| 12:05.53 | din3sh | when it gets the busy tone, i dont see the call actually reaching asterisk or hitting the dialplan |
| 12:05.55 | WIMPy | That sounds really bad. |
| 12:06.04 | Greenlight | Sounds like the carrier is distributing calls accross both spans, and one span is buggy |
| 12:06.17 | din3sh | one span is connected to carreir |
| 12:06.23 | din3sh | the other to a nortel system |
| 12:06.27 | Greenlight | Ahh I see |
| 12:06.37 | Greenlight | Yea, something bad going on then. I'd call the carrier |
| 12:06.48 | Greenlight | They can likely see those errors at their end too |
| 12:06.56 | Greenlight | So it might get moved along quite quickly |
| 12:08.24 | ectospasm | I'd check to make sure the hardware is functioning properly on a basic level first. And the telco will always say,"NOT IT!" |
| 12:08.42 | din3sh | yeah the client reported the problem to the provider |
| 12:08.58 | din3sh | the provider says they did all tests, their line and modem are ok |
| 12:09.17 | ectospasm | as they *ALWAYS* do. |
| 12:10.16 | Greenlight | AT least it's not an intermittant fault, the carriers have great pleasure in coming out and going "it's working, see". |
| 12:10.44 | din3sh | its a random problem |
| 12:10.44 | Greenlight | 4 months it took for us to get them to finally but monitoring equipment on the line /sigh |
| 12:11.35 | Greenlight | WHy not swap the spans around, that'll verify hardware or line issue. (Presuming you're not getting errors on the nortel span ?) |
| 12:12.21 | din3sh | hdlc on both spans |
| 12:12.22 | din3sh | lol |
| 12:12.29 | din3sh | its a card issue then? |
| 12:12.37 | Greenlight | Heh |
| 12:12.55 | ectospasm | sounds like it |
| 12:13.04 | Greenlight | I wonder if errors on the timing source span, could cause errors on other spans, but doubt it |
| 12:13.23 | WIMPy | Might be because the timing is taken form the first port and used on the second that it's a side effect. |
| 12:13.29 | Greenlight | ^^ |
| 12:13.41 | ectospasm | Greenlight: since that timing is passed onto the secondary spans, it could affect those as well. |
| 12:14.04 | Greenlight | Guess you could disconnect from carrier side, and monitor then, with your card as master timing source |
| 12:14.06 | din3sh | dahdi test for 9hours: Best: 100.000% -- Worst: 99.862% -- Average: 99.995729% |
| 12:14.06 | din3sh | Cummulative Accuracy (not per pass): 99.996 |
| 12:14.24 | ectospasm | dahdi_test scores are mostly useless. |
| 12:14.43 | din3sh | thats what i was thinking, since 1st span is slave to telco and 2nd span acts like master to nortel |
| 12:14.55 | din3sh | wouldnt the error from span 1 transfer on span 2? |
| 12:14.59 | Greenlight | Yea, disconnect from telco, and monitor. |
| 12:15.10 | din3sh | aha! |
| 12:15.28 | WIMPy | thinks Greenlights idea might be worth a try. Disconnect the line and then try some massive testing from the Nortel. |
| 12:15.44 | Greenlight | How frequent are the errors ? |
| 12:16.17 | din3sh | i have noted |
| 12:16.20 | din3sh | when there is no call |
| 12:16.24 | din3sh | i mean at night |
| 12:16.28 | din3sh | there is no hdlc error |
| 12:17.07 | Greenlight | Could mean anything though - remember if there are errors, it could be from "local" interferrance, which may not be present a night |
| 12:17.23 | din3sh | hmmm |
| 12:17.26 | WIMPy | that's probably just because the amount of traffic. |
| 12:17.31 | Greenlight | That too |
| 12:18.07 | din3sh | server top is more than decent, dl380 g7, single quad, 8gbmem |
| 12:18.08 | Greenlight | BUt, with that in mind you'll want to make sure you stress test it, when you disconnect from telco |
| 12:18.40 | din3sh | how to stress test the spans? i mean simulate dahdi calls? |
| 12:18.57 | Greenlight | You could use outgoing spool |
| 12:19.12 | WIMPy | Set up a MOH extension an run around to call it from all phones. |
| 12:19.24 | Greenlight | Ahh, yea that makes more sense :) |
| 12:19.38 | WIMPy | Or better, just play a dialtone. Dialtines are great as you can hear if the'res a timing issue. |
| 12:22.21 | din3sh | aha thnks |
| 12:22.29 | WIMPy | Does the Nortel provide it's own dialtone or do you supply it from Asterisk? |
| 12:23.01 | din3sh | own dialtone |
| 12:23.21 | din3sh | because if span 2 fails, the user hears "no free lines available" |
| 12:23.38 | WIMPy | Ok, but you cann just do an extension with playtones or the like. |
| 12:23.47 | din3sh | what about the issue calls from Telco randomly not reaching asterisk? |
| 12:24.24 | WIMPy | Probably just another transmission error. |
| 12:24.51 | Greenlight | You should call the telco see if they'll give you any info what about what they're seeing at their end |
| 12:25.31 | WIMPy | You might still be able to see something in pri debug. |
| 12:30.07 | din3sh | hmm |
| 12:30.58 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 12:34.35 | din3sh | part of pri intense debug--> http://pastebin.com/tnz7GFdK |
| 12:36.19 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 12:36.31 | WIMPy | That one did arrive. |
| 12:36.44 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:38.45 | din3sh | how to track down errors in PRI intense debug |
| 12:38.50 | din3sh | i enabled it on both spans |
| 12:38.54 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 12:39.02 | din3sh | i have loads of debug info |
| 12:39.02 | Greenlight | Is there an argument to Dial that'll cause a part of the dialplan (say a macro) to get run just before the Dial actually happens, on the newly created channel ? |
| 12:39.37 | *** join/#asterisk youjelly (~youjelly@39.47.99.85) |
| 12:39.58 | Greenlight | From what I can see both M and U(x) both get called after connect |
| 12:40.12 | WIMPy | din3sh: If you know when a call didn't get through, look what happened at that time. |
| 12:41.03 | WIMPy | Greenlight: You want something that runs on each dialled channel before dialling? |
| 12:42.05 | Greenlight | Kinda, yea. I'm needing to identify the channel created by Dial, for AMI purposes. But I can't use Local channels. |
| 12:42.39 | WIMPy | Look at the pre-dial handlers. |
| 12:42.45 | Greenlight | yea just this second seen it! |
| 12:42.57 | Greenlight | Serves me right for using voipinfo first |
| 12:42.59 | Greenlight | Dammit |
| 12:43.19 | din3sh | WIMPy: i see hdlc errors at that same time |
| 12:43.27 | Greenlight | Pre-dial handlers looks perfect, cheers WIMPy |
| 12:45.05 | Greenlight | din3sh: The most likely problem is a faulty line. Your card sits nice and safe in your server, and is relavitely new. The line goes underground sometimes using 100 year old cable, and is liable to be poked and prodded by construction workers etc.... |
| 12:45.24 | WIMPy | din3sh: So the message is never received in any decodable form then. What about crc4? Did you just disable it or is it not available? |
| 12:51.19 | *** join/#asterisk din3sh (~din3sh@41.136.248.40) |
| 12:51.27 | din3sh | hello again |
| 12:51.42 | din3sh | WIMPy: i have other dahdi installations with no crc4 enabled |
| 12:51.55 | din3sh | i jst checked logs |
| 12:52.12 | Greenlight | Doesn't CRC try and correct errors though ? |
| 12:52.21 | din3sh | [Feb 14 16:21:56] NOTICE[28802] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 2 |
| 12:52.21 | din3sh | [Feb 14 16:21:56] NOTICE[28801] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 |
| 12:52.31 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 12:52.36 | din3sh | doesnt look good |
| 12:52.37 | din3sh | http://pastebin.com/UBU3a7m0 |
| 12:52.41 | WIMPy | that's the idea. |
| 12:53.09 | din3sh | i did set crc4 on, i had alarms on the line |
| 12:53.16 | Greenlight | din3sh: Why not connect the nortel direct to the telco line, and watch for errors ? |
| 12:53.43 | Greenlight | It really does sound like you've a crappy like - exactly same stuff I used to see on a bad line I had a few years back |
| 12:53.46 | Greenlight | *line |
| 12:53.48 | din3sh | the nortel was actually connected to the line |
| 12:53.50 | din3sh | directly |
| 12:54.04 | Greenlight | And, can you see line errors on the nortel ? |
| 12:54.07 | din3sh | that was 6-9 months back |
| 12:54.18 | din3sh | before i hooked an * box in the middle |
| 12:54.40 | din3sh | the client has been using the line with nortel for many years now |
| 12:54.54 | din3sh | maybe now line condition has gone bad |
| 12:54.55 | WIMPy | You've had that issue for over half a year? |
| 12:55.21 | din3sh | nope |
| 12:55.27 | din3sh | jst popped up in december |
| 12:55.34 | din3sh | now getting critical |
| 12:55.45 | WIMPy | Aha |
| 12:55.50 | din3sh | when i first hooked asterisk in the middle, there wasnt such problems |
| 12:56.04 | Greenlight | I'd bet money that it's the line |
| 12:56.11 | din3sh | though i did have hdlc errors since the beginning |
| 12:56.14 | WIMPy | yes |
| 12:56.48 | WIMPy | Do you have sound output on that box? |
| 12:56.51 | Greenlight | SO, it's gradually got worse, as line issues due. Corrosion, moisture that kinda thing |
| 12:57.04 | din3sh | no sound output |
| 12:57.43 | WIMPy | Anyway, I'm with Greenlight. Sounds like your line is drowning. |
| 12:58.10 | file | throw it a life raft! |
| 12:58.24 | WIMPy | K60 :-) |
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| 12:59.33 | Rico29 | Hi, I'm looking for the latest 1.8 mysql scheme for queues, can somebody tell me where I can find it please ? |
| 12:59.35 | din3sh | they tested the line and said at the moment of the test, evrythg was ok |
| 12:59.47 | youjelly | sometimes ncrc works |
| 12:59.57 | din3sh | if its a random problem, is it possible to have a constant test over a day or 2? |
| 12:59.57 | WIMPy | What kind of thest? |
| 13:00.22 | Greenlight | din3sh: Yea, took us FOUR months to get the carrier to actually do that kinda of testing. Eventually proved it was a line problem the they moved from copper to fibre |
| 13:00.47 | youjelly | telcos are a LOT of red tape.... |
| 13:00.52 | din3sh | what were the errors you had? |
| 13:01.02 | din3sh | WIMPy: no idea |
| 13:01.26 | WIMPy | din3sh: Not a real test. But monitoring is surely possible. A good test would be to take the line down and do a BERT. |
| 13:02.19 | din3sh | BERT? |
| 13:02.33 | WIMPy | Bit Error Rate Test |
| 13:03.46 | din3sh | ok |
| 13:05.50 | din3sh | the frequency of hdlc errors during day when system is fully used is like every 5-6 mins |
| 13:07.23 | jmls | meh |
| 13:07.30 | jmls | here's an odd request |
| 13:07.51 | jmls | I have a device runing windows ce 5.0 and and client has asked me to put a softphone onto it ... |
| 13:07.58 | jmls | is there such a beast ? |
| 13:08.39 | *** join/#asterisk [TK]D-Fender (~Joe@v5208878.static2.cidc.net) |
| 13:09.58 | din3sh | WIMPy: I have dahdi 2.6.1, should i update? --> http://downloads.asterisk.org/pub/telephony/dahdi-linux/ChangeLog-2.6.2-rc1 |
| 13:11.35 | [TK]D-Fender | din3sh, Do you have a reason to? |
| 13:11.45 | WIMPy | din3sh: Doesn't looke like it would make a difference. |
| 13:13.02 | din3sh | i have tried every single solution, trying a desperate one |
| 13:13.50 | din3sh | WIMPy: the server is in a remote location, how to test the span seperately? disable one in /dahdi/system.conf? |
| 13:14.27 | Rico29 | I'm looking for the latest 1.8 mysql scheme for queues, can somebody tell me where I can find it please ? |
| 13:14.38 | WIMPy | I wouldn't relay on anything less than unplugging the cable. |
| 13:15.32 | din3sh | argh! here comes late night onsite testing |
| 13:15.34 | din3sh | :S |
| 13:15.55 | WIMPy | Let the telco do some real testing. |
| 13:16.34 | din3sh | okay |
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| 13:27.39 | _zoom_ | how to override caller id in pri outbound |
| 13:27.42 | _zoom_ | ? |
| 13:28.05 | [TK]D-Fender | _zoom_, "core show function CALLERID" <- |
| 13:33.32 | *** join/#asterisk Sevet (~steveayre@149.241.69.221) |
| 13:33.47 | Sevet | hi, is anyone here familiar with the ooh323c library? |
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| 13:50.24 | zamba | what's it called when period from when you start dialing a number until it's automatically "sent"? |
| 13:50.45 | zamba | the period* |
| 13:50.46 | zamba | i mean |
| 13:50.58 | zamba | on some phones the timeout never occurs and you have to press send manually |
| 13:52.11 | din3sh | phone dialplan timeout? |
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| 13:55.26 | zamba | aight |
| 13:58.31 | zamba | anyone got experience with audiocodes or tenor? |
| 13:58.46 | zamba | which one do you recommend? |
| 13:58.53 | zamba | for connecting with lync |
| 13:58.56 | zamba | eh |
| 13:58.57 | zamba | haha |
| 13:58.59 | zamba | asterisk |
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| 14:01.38 | [TK]D-Fender | zamba, Never heard of tenor in here before. |
| 14:01.41 | Neptu_laptop | hej anyone knows a good catalog of sip providers? |
| 14:01.51 | zamba | [TK]D-Fender: but audiocodes you know? |
| 14:02.05 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:03.25 | [TK]D-Fender | zamba, I worked with their 24-port FXS gateway once many years ago. Bit of a curve but it worked. |
| 14:03.36 | [TK]D-Fender | zamba, How many ports do you actually require? |
| 14:03.44 | zamba | [TK]D-Fender: only 2 |
| 14:03.58 | [TK]D-Fender | zamba, Why would you look at either of these 2 brands then? |
| 14:04.20 | zamba | [TK]D-Fender: dunno? |
| 14:04.22 | [TK]D-Fender | zamba, Linksys range covers this just fine, far cheaper, and easier to configure |
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| 14:06.20 | din3sh | spa122 would do |
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| 14:07.25 | zamba | the analoge end here is the PSTN.. does that matter? |
| 14:08.13 | [TK]D-Fender | <zamba> what's it called when period from when you start dialing a number until it's automatically "sent"? <- this is an FXS delay for a phone interface, not POTS -> PSTN side |
| 14:08.22 | [TK]D-Fender | zamba, So which end is it you're actually buying for? |
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| 14:22.59 | din3sh | [TK]D-Fender: can "Notices" in logs be responsible for call drops? |
| 14:23.59 | kannan | hi, on asterisk 1.8 LTS, i am using adaptive ODBC with mysql for cdr. in case i need to alter the column structures, how can we reload the cdr while live calls are running |
| 14:24.31 | [TK]D-Fender | din3sh, No. A notice in a log is a notice in a log. |
| 14:25.20 | kannan | ok i got it - module reload cdr_adaptive_odbc.so |
| 14:28.28 | din3sh | now a notice of HDLC error might be responsible for inbound calls not reaching an * box randomly? i mean one moment the calls goes in, you try 5mins again , it doesnt |
| 14:28.31 | *** join/#asterisk Weezey (~ohno@wap54g-07.loit.ca) |
| 14:29.38 | Weezey | In asterisk 11, for random nomadic peers I previously had nat=yes but should I be now using nat=force_rport,comedia or nat=auto_force_rport,auto_comedia? |
| 14:33.03 | [TK]D-Fender | din3sh, The NOTICE isn't responsible. that CAUSE of the notice is another matter. |
| 14:33.07 | igcewieling | din3sh: no, the notification of an HDLC error will not cause provlems with calls. The actual HDLC error would be the cause. |
| 14:33.10 | igcewieling | looks at [TK]D-Fender |
| 14:33.49 | [TK]D-Fender | igcewieling, MY TELEPATHIC FREQUENCY! |
| 14:33.56 | [TK]D-Fender | calls the FCC on igcewieling |
| 14:34.06 | igcewieling | din3sh: Think of HDLC errors as "static" on the line. i.e. the data coming from the telco to the driver is getting corrupted for some reason |
| 14:34.28 | igcewieling | hey, I'm licensed for that band too! |
| 14:36.49 | [TK]D-Fender | I love it when people say the "Check Engine" light killed their cars.....\ |
| 14:37.06 | din3sh | hmm okay |
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| 14:45.36 | chris_n | in that case, removing the light should fix it right up, heh? ;) |
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| 14:47.00 | Greenlight | Did the trick for my car |
| 14:47.19 | Greenlight | Same with that pesky brake light |
| 14:47.51 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:48.08 | ghost75 | meanwhile in france http://www.theatlantic.com/technology/archive/2013/02/this-guys-car-got-stuck-at-125mph-for-an-hour/273140/ |
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| 14:48.53 | [TK]D-Fender | And somehow he never thought of killing the engine.... |
| 14:49.13 | ghost75 | has to be a fake |
| 14:49.33 | [TK]D-Fender | Turn key. Engine stops. |
| 14:50.09 | Greenlight | That can't be real |
| 14:50.10 | [TK]D-Fender | Guy was in a hurry. He just wanted a police escort to proctect him as he speeded :) |
| 14:50.26 | ghost75 | yes important drugs delivery |
| 14:50.54 | Greenlight | Mind, perhaps the police advised against killing the engine |
| 14:51.09 | ghost75 | why |
| 14:51.34 | Greenlight | You'd loose all the electronics etc, might have been considered unsafe |
| 14:51.42 | Greenlight | *more* unsafe |
| 14:51.50 | Greenlight | BUt, seems like some sort of joke to be honest |
| 14:52.07 | Greenlight | Mind, it's ont he Guardian site too: http://www.guardian.co.uk/world/2013/feb/13/french-driver-200kmh-car-chase?CMP=SOCNETTXT6966 |
| 14:52.16 | ghost75 | if thats really the case on modern cars, then this is questionable |
| 14:52.41 | ghost75 | remember that pedal thing on toyota? |
| 14:53.15 | Greenlight | Nope .. |
| 14:53.53 | ghost75 | i think floor mat pressed down gas pedal and people were loosing control of car lol |
| 14:54.06 | Greenlight | haha |
| 14:54.51 | Greenlight | Pressing brake and car going *faster* seems crazy. Brakes are normaly just power *assisted* so it's still physically applying pressure on the caliper |
| 14:55.23 | ghost75 | i think this car was modified for a disabled person |
| 14:55.39 | Greenlight | DOes it say that ? |
| 14:56.32 | igcewieling | stories like that remind me of problem reports. "I hangup the phone and my computer reboots" |
| 14:56.38 | ghost75 | read comments |
| 14:57.01 | ghost75 | maybe they removed it in article later |
| 14:57.09 | Greenlight | Why didn;t the police just box him in, and slow down gradually |
| 14:57.32 | igcewieling | "My call dropped as soon as the other person answered", I look up the call and the call length is 45 mins |
| 14:59.14 | ghost75 | http://www.youtube.com/watch?v=WVPHblo_QZQ |
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| 15:52.34 | Weezey | what's the best way to get current call count per peer in the dialplan? |
| 15:53.06 | Weezey | ie: I have 23 channels total, if there's only two left, don't make an outbound call with that peer, pick a new route. |
| 15:53.06 | *** join/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 15:53.46 | igcewieling | Weezey: none as far as I know, but calllimit= (or similar) might be useful, or you could hack something up with global variables |
| 15:53.47 | slicknick5181 | asterisk 1.8 need some help with features.conf. I'm not sure I understand exactly how to do dynamic features |
| 15:54.48 | Greenlight | Weezey: Check out GROUP_COUNT |
| 15:56.01 | Greenlight | Wheezy: http://www.astblog.com/2008/09/17/count-and-limit-number-of-calls-under-asterisk/ |
| 15:56.21 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 15:58.21 | Weezey | Greenlight: woo, that's perfect |
| 15:58.25 | slicknick5181 | Can anybody help with dynamic features in asterisk 1.8. Providing me with an example on one might be enough |
| 15:59.07 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 16:00.15 | Weezey | igcewieling: calllimit works for my customers switches but if I set the limit too low, I think it would block the incoming calls I'm trying to allow in by failing over to another provider for outbound dialing. |
| 16:00.29 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 16:06.26 | chris_n | igcewieling: thanks for the headsup on the CHANNEL function; fail2ban now blocks ips from anonymous sip calls |
| 16:13.21 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:13.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:13.54 | slicknick5181 | Can anybody help with dynamic features in asterisk 1.8. Providing me with an example on one might be enough |
| 16:14.44 | igcewieling | chris_n: glad I could help. It is often handy to do a "core show applications" and "core show functions" just to explore what is available. |
| 16:15.16 | igcewieling | slicknick5181: does features.conf.sample in the asterisk source code not helping? |
| 16:15.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:16.41 | slicknick5181 | igcewieling: no it's the one file that I just can't seem to understand, What I'm really looking for is DTMF in0call xfer to the parking lot |
| 16:18.15 | slicknick5181 | I did figure out ;Incoming |
| 16:18.15 | slicknick5181 | [from-callcentric] |
| 16:18.15 | slicknick5181 | exten => s,1,set(__DYNAMIC_FEATURES=home-feat) |
| 16:18.15 | slicknick5181 | exten => s,n,Dial(SIP/101&SIP/102,55,tTkK) |
| 16:18.15 | slicknick5181 | exten => s,n,Voicemail(101,u) |
| 16:18.21 | Greenlight | ~pb |
| 16:18.21 | 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. |
| 16:20.23 | slicknick5181 | I Apologize greenlight |
| 16:20.46 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 16:22.57 | Greenlight | No need to apologise :) |
| 16:24.54 | igcewieling | slicknick5181: set dtmf on your console => line in logger.conf then logger reload in the CLI just to confirm Asterisk is actually seeing the DTMF in the call. |
| 16:26.57 | slicknick5181 | I already did that but I found my problem |
| 16:28.04 | slicknick5181 | igcewieling: I already did that but I just not figured what that issue was but I ran accross another one, Once I park a call I can't get it back |
| 16:29.39 | [TK]D-Fender | slicknick5181, Show us the park, the failure, and your configs |
| 16:31.55 | slicknick5181 | I'm so sorry for bothering you guys but I figured out that I did not have include => parkedcalls in my context |
| 16:32.00 | *** part/#asterisk gauravp (~gaurav@c-68-80-206-60.hsd1.pa.comcast.net) |
| 16:32.13 | slicknick5181 | everything works now! :) I love asterisk |
| 16:32.41 | *** join/#asterisk stmaher (~stephen@80.68.89.200) |
| 16:32.43 | stmaher | hi guys.. |
| 16:32.57 | jsmith | Howdy, stmaher |
| 16:33.08 | stmaher | Im trying to get the DNID but it keeps showing the register auth number and not the TO: number in the sip header.. |
| 16:33.12 | stmaher | what am I doing wrong ? |
| 16:33.32 | [TK]D-Fender | stmaher, Thinking it's either |
| 16:33.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:33.43 | [TK]D-Fender | stmaher, Its the INVITE line |
| 16:34.04 | stmaher | Invite line contains register auth number |
| 16:34.22 | [TK]D-Fender | stmaher, If they use the TO for something else useful, then parse it our with the SIP_HEADER function |
| 16:34.26 | stmaher | header to: contains right number.. |
| 16:34.35 | stmaher | how can I extract that into a variable for use as a DNID? |
| 16:34.41 | [TK]D-Fender | ^ |
| 16:35.01 | [TK]D-Fender | stmaher, This is unfortunately necessary for some providers |
| 16:35.22 | stmaher | [TK]D-Fender thanks for the help on this.. |
| 16:35.29 | [TK]D-Fender | stmaher, np |
| 16:35.56 | stmaher | Ill explain what im trying to do.. |
| 16:36.09 | stmaher | Its a simple Trunk setup.. |
| 16:36.16 | stmaher | Ive never used nextiva before.. |
| 16:36.30 | stmaher | I place a call in bound.. hits their sip trunks and forwards to our asterisk box.. |
| 16:37.02 | stmaher | the calledID or DNID is always my auth register number.. and not the true DNID however the TRUE dnid is contained in the TO address of the sip header.. |
| 16:37.17 | stmaher | is there any way for me to use this in an asterisk variable i can allocate direct dials with? |
| 16:38.26 | [TK]D-Fender | stmaher, I jsut gave you the function. Parse it out |
| 16:38.35 | stmaher | what function? |
| 16:38.55 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 16:38.56 | [TK]D-Fender | <[TK]D-Fender> stmaher, If they use the TO for something else useful, then parse it our with the SIP_HEADER function |
| 16:38.56 | stmaher | do you have a link? |
| 16:38.57 | [TK]D-Fender | .... |
| 16:39.10 | [TK]D-Fender | You have the name. you have asterisk. Therefor you have the INSTRUCTIONS |
| 16:39.11 | stmaher | strange.. that didnt make my console |
| 16:39.19 | jsmith | stmaher: Type "core show function SIP_HEADER" at the Asterisk CLI |
| 16:39.23 | stmaher | thank you |
| 16:39.43 | stmaher | sorry [TK]D-Fender thats not on my chat history.. |
| 16:39.45 | stmaher | strange.. |
| 16:39.48 | stmaher | Thank you all |
| 16:39.53 | stmaher | Esp jsmith and [TK]D-Fender |
| 16:40.00 | Weezey | Greenlight: works perfectly, Set(GROUP(carrier)=${CHANNEL:4:-9}) for incoming |
| 16:41.04 | Weezey | Greenlight: and Set(GROUP(carrier)=${CUT(DIALSTRING,/,2)}) for outgoing |
| 16:42.10 | Weezey | then before I send out calls I do GotoIf($[${GROUP_COUNT(maincarriername@carrier)} > 19]?nextcarrier) |
| 16:42.23 | teloniusz | hi. I'm trying to use 'r' option to Dial() to make asterisk generate ringtone for me. How to control the frequency of this ringtone? |
| 16:43.37 | [TK]D-Fender | teloniusz, indications.conf <- |
| 16:45.18 | leifmadsen | teloniusz: you can specify a specific indication with later versions of Asterisk. For the duration, its for the entire time that Asterisk is attempting to take a call. In most cases you should't actually need to use the 'r' flag. Ringtone should be indicated during call setup when you get the appropriate messages. |
| 16:45.22 | teloniusz | [TK]D-Fender: http://www.voip-info.org/wiki/view/Asterisk+config+indications.conf says "In short, the settings in indications.conf affect only the tones generated by the Playtones command." - is it wrong? |
| 16:45.23 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 16:45.29 | *** part/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 16:45.36 | leifmadsen | yes. The voip-info wiki is almost always wrong now. |
| 16:45.45 | leifmadsen | try wiki.asterisk.org or asteriskdocs.org |
| 16:46.04 | teloniusz | leifmadsen: thx, will try from now on |
| 16:46.44 | stmaher | [TK]D-Fender Additionally.. I just want to thank you in general.. I was on here about 6-7 years ago with SIP interoperability with an IVR system.. Its great to see your still helping the community.. Thank you again! |
| 16:49.52 | stmaher | <PROTECTED> |
| 16:50.00 | stmaher | ;-) |
| 16:52.17 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 16:54.10 | Greenlight | Weezey: Glad it's working for you! :) |
| 16:54.19 | Weezey | so perfect |
| 16:54.39 | Weezey | always wanted that counter. |
| 16:55.16 | Weezey | plus the CLI group show channels is nice and tidy and gives an accurate count of all calls to/from carriers now |
| 16:55.40 | Weezey | as opposed to before where inter-switch calls are displayed with core show channels |
| 16:56.18 | Greenlight | Sounds like it's made a good positive difference then |
| 16:56.37 | Greenlight | It's nice when things "just work" with Asterisk :) |
| 16:57.24 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 16:58.36 | Weezey | definitely. the big thing for me is keeping that PRI low use when its channel count is high. I guess next I'll need to add an AGI call to SMS me to let me know it did that. |
| 16:59.36 | Greenlight | Is this to keep "space" for incomming calls ? |
| 17:00.20 | Weezey | yeah |
| 17:00.34 | Greenlight | I wonder if you could have a second count for incomming calls, and then factor that into the calculation for the outbound (like limit=incomming_usage - 2) |
| 17:02.20 | Weezey | one of my customers is a florist so today and yesterday have been nuts and another decided today was a good day for a conference call, so usage peaked higher than I was ready for. Ordered another 23 channels but until that comes in, leaving 3 free when dialing out should be fine. |
| 17:02.57 | Weezey | adding a call centre in the next 30 days so I needed it anyway. |
| 17:03.14 | Greenlight | Why PRI rather than SIP ? |
| 17:03.49 | Weezey | they sell PRI service delivered via SIP. |
| 17:04.27 | Weezey | you can get any number of channels but you have to have one 23 channel to one of their switches. |
| 17:04.48 | Greenlight | I see |
| 17:05.11 | Weezey | it's a very antiquated approach. |
| 17:05.33 | Weezey | quality is great though, my customers do a lot of faxing and it works over IP. |
| 17:06.15 | Greenlight | Well, if it works, and the cost is okay, why not |
| 17:06.36 | Weezey | exactly |
| 17:06.49 | Weezey | ... and my channel upgrade is complete |
| 17:13.02 | Weezey | anyone sucessfully done mwi from one asterisk box to another? |
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| 17:22.25 | Greenlight | I'm trying to decide if I should play the "music" to agents inbetween calls, or just silence. Thinking it may just drive them insane... |
| 17:23.33 | igcewieling | Greenlight: Personally I'd play a silent audio file with some sort of pleasant tone every X seconds/mins to remind them they are in the queue and off hook |
| 17:24.46 | Greenlight | Yea, that's not a bad idea. Will go with that, thanks |
| 17:28.53 | jsmith | Greenlight: We have a DJ in our call center that mixes our "between call" music |
| 17:29.02 | jsmith | Greenlight: Our agents seem to like that |
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| 17:49.44 | igcewieling | I want to set up MoH with George Carlin stuff (but only for local calls, not customer calls) |
| 17:49.47 | igcewieling | 8-) |
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| 18:40.41 | glaz | I'm working on an API that talks with Manager, is there a way to log what Manager gets? |
| 18:41.37 | jsmith | glaz: Since it's a plain-text protocol, you could log it with tcpdump |
| 18:42.40 | ChannelZ | or 'manager set debug' ? |
| 18:43.29 | glaz | I did manager set debug on |
| 18:43.37 | glaz | nothing shows but logged on and logged off |
| 18:43.42 | glaz | jsmith: true. |
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| 18:49.30 | glaz | FIY: tcpflow -i lo -C -e port 5038 |
| 18:49.37 | glaz | this is an amazing tool. |
| 18:51.59 | Katty | HI LADS. |
| 18:55.27 | ChannelZ | mmmheh hhehhhmmheh... tool.. hehmm mmmheh |
| 18:58.36 | glaz | utility I should say |
| 18:58.54 | glaz | it pints STDIN vs STDOUT in different colors |
| 18:59.09 | glaz | it's "less raw" than tcpdump I should say heh |
| 18:59.54 | psykon | While doing and apt-cache for 'tcpflow' I came across 'tcpslice' extract pieces of tcpdumps... i could have used that last week when I had to send a portion of a dump to a carrier. Couldn't get wireshark to export a portion of it. |
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| 19:01.32 | ChannelZ | looks like there are a lot of fun tcp* tools |
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| 19:43.34 | chris_n | tcpslice is *very* cool |
| 19:44.46 | jsmith | Don't forget ngrep as well :-) |
| 19:45.09 | psykon | ngrep-sip is pretty slick too |
| 19:45.57 | glaz | finally got my API working after upgrading from 1.4 to 11.2.1 |
| 19:46.16 | glaz | glad to see they finally implemented the command "queue pause" |
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| 20:47.31 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 20:48.04 | TheCompWiz | does anyone know how to get the mailbox context of the calling device from within dialplan (if available) |
| 20:50.40 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 20:51.19 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 20:51.20 | jsmith | TheCompWiz: ${SIPPEER(mailbox)} would get you the mailbox and context |
| 20:51.28 | jsmith | TheCompWiz: (something like 1234@default) |
| 20:51.39 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 20:51.39 | jsmith | TheCompWiz: Then you could use the CUT function |
| 20:52.22 | jsmith | TheCompWiz: So, off the top of my head, something like ${CUT(${SIPPEER(mailbox)},@,2)} |
| 20:52.26 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:52.27 | jsmith | And he's gone :-( |
| 20:52.38 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 20:52.42 | jsmith | TheCompWiz: So, off the top of my head, something like ${CUT(${SIPPEER(mailbox)},@,2)} |
| 20:52.45 | TheCompWiz | ... no clue what happened there. |
| 20:52.49 | davlefou | hi, i have something strange. I have twice include A and B, with context c-A and c-B. My asterisk look the context only in the firts of my include! |
| 20:52.57 | TheCompWiz | jsmith: exactly what I was thinking ... thanks 10-million times :D |
| 20:53.12 | jsmith | davlefou: I'm confused ... can you be more clear? |
| 20:53.45 | davlefou | jsmith: Sorry, i ll try. |
| 20:53.53 | SuperNull | hey guys, im getting a near constant stream of NOTIFYs from ATAs and asterisk just keeps spitting '489 bad event'. |
| 20:54.08 | igcewieling | CUT uses a variable not a value |
| 20:54.17 | jsmith | igcewieling: Ah, yeah... forgot about that |
| 20:54.19 | igcewieling | <PROTECTED> |
| 20:54.21 | jsmith | igcewieling: Good call |
| 20:54.26 | jsmith | should know that |
| 20:54.32 | igcewieling | jsmith: easy to forget when you don't use it all the time |
| 20:54.51 | jsmith | igcewieling: Yup... I haven't used CUT() recently |
| 20:55.10 | jsmith | SuperNull: Then tell your ATA to stop sending NOTIFYs :-p |
| 20:55.17 | igcewieling | I have scripts which CUT, slice, dice and julienne data |
| 20:55.21 | rogers | anyone familiar with linksys spa3102s? PSTN to VoIP device |
| 20:55.35 | jsmith | igcewieling: I do too, but I haven't touched them in years :-) |
| 20:55.42 | jsmith | rogers: I used one a few years ago -- why? |
| 20:56.00 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 20:56.09 | SuperNull | yeahhh im gonna have to figure that out jsmith ;) old linksys SPA-2102s. |
| 20:56.10 | TheCompWiz | I think freenode hates me today |
| 20:56.50 | rogers | jsmith, trying to work the impossible with it. We have a temperature monitoring device that needs to make an outbound call. So I have it hooked up to the linksys connected to asterisk. It dials the call, and starts ringing. Then the device makes its announcement before the call is answered. Any idea what setting that might be where the device thinks the call was picked up? |
| 20:57.18 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 20:57.36 | rogers | as soon as it starts the ring process, the device thinks its connected |
| 20:58.18 | jsmith | Well, if it's analog signaling, there's not a lot you can do |
| 20:58.29 | davlefou | I have creatte an file for register context from ovh, i have tree group, ovh-1, ovh-2 and ovh-3. Ovh-1 are the main sub include à sip.conf. In the bigining of the file, i have include the sub include ligne, ovh-2 and ovh-3. So, i a include ovh-2 in firt, asterisk ll look for ovh-2 for ovh-2 and ovh-3 context. |
| 20:58.30 | rogers | anything I could try? |
| 20:58.43 | jsmith | rogers: Depending where in the world you live, you might get battery-reversal when the far-end answers |
| 20:58.48 | SuperNull | jsmith i'm retarded. its the linksys keep-alive messages.. i have a bad feeling if i turn it off some NAT stuff will break. |
| 20:59.02 | rogers | USA |
| 20:59.34 | jsmith | rogers: Yeah, no idea what your temperature monitoring device is looking for -- |
| 20:59.40 | rogers | yeah |
| 21:00.15 | jsmith | davlefou: OK, I *think* I'm following you |
| 21:00.46 | davlefou | it is difficulte to explain |
| 21:00.48 | jsmith | davlefou: You need to understand how Asterisk follows include=> links |
| 21:01.01 | jsmith | davlefou: Asterisk will first look in the first context |
| 21:01.14 | jsmith | davlefou: Only if it doesn't find any match will it follow the first include |
| 21:01.43 | jsmith | davlefou: Then it looks in that included context. If there is no match, it looks for an include. If there's an include there, it follows that (depth-first search) |
| 21:02.06 | jsmith | davlefou: If there's no include there, it goes back up and tries the second include in the previous context (if a second include exists) |
| 21:02.15 | jsmith | davlefou: Once it finds anything that matches, it doesn't follow any other includes |
| 21:02.37 | SuperNull | jsmith your some kind of superman asterisk channel respondant. |
| 21:02.42 | [TK]D-Fender | davlefou, Short answer : it finds the first hit in order. |
| 21:03.09 | [TK]D-Fender | davlefou, It won't search everywhere. It'll find the first match it can, not the "best" amongst multiple possible matches |
| 21:03.26 | jsmith | SuperNull: Well, I don't hang out here very often -- because when I do, I tend to hog the channel. I taught Asterisk classes for something like six or seven years, so I have a pretty good grasp on Asterisk |
| 21:03.27 | [TK]D-Fender | davlefou, Which is for the best because your includes are processed in order |
| 21:04.06 | jsmith | used to live, eat, breathe, and drink Asterisk |
| 21:04.12 | davlefou | i have found the problem. Have pre include before on file and made asterisk crazy. |
| 21:04.20 | *** join/#asterisk youjelly (~youjelly@39.47.99.85) |
| 21:05.13 | davlefou | Thanks jsmith [TK]D-Fender for you help. I made an very very stupid error! |
| 21:05.47 | mjordan | jsmith == the man |
| 21:06.17 | jsmith | mjordan: I ain't all that and a bag of chips -- and unfortunately, I usually don't have time to sit in the channel and answer questions. Today I just happen to be sick in bed :-/ |
| 21:07.02 | mjordan | yuck. I blame travel. Look what it did to russellb |
| 21:07.38 | TheCompWiz | jsmith: any other ideas? ${SIPPEER(mailbox)} is empty... |
| 21:07.39 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 21:08.07 | jsmith | TheCompWiz: This is a registered SIP peer calling into Asterisk? |
| 21:08.12 | TheCompWiz | of course. |
| 21:08.22 | SuperNull | jsmith what do you do now ? |
| 21:08.52 | TheCompWiz | ah... I think I know why.... it's ${SIPPEER(xxxx,mailbox)} |
| 21:09.00 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 21:09.04 | TheCompWiz | xxxx=peer in question |
| 21:09.09 | jsmith | SuperNull: I left teaching to go be the Fedora Project leader for two years, and now I do open source advocacy for a large web hosting firm |
| 21:09.10 | gavimobile | sup folks!!! just saying hi |
| 21:09.18 | gavimobile | was in the area ;-p |
| 21:09.32 | jsmith | SuperNull: But I still play with Asterisk a lot |
| 21:10.57 | jsmith | TheCompWiz: Ah, right... and if you set "setvar=mypeername=blah" in sip.conf, then you'll have a channel variable named ${mypeername} that you can reference to get the name of the peer |
| 21:11.14 | TheCompWiz | jsmith: yup... I can bang on it from here :D |
| 21:11.27 | TheCompWiz | thanks... I didn't know 'bout the func SIPPEER before. :D |
| 21:12.03 | davlefou | Include, one include is ok, several include of file, crazy asterisk |
| 21:12.12 | igcewieling | TheCompWiz: it is often helpful to occasionally do a "core show applications" and "core show functions", especially when installing major updates to Asterisk |
| 21:12.22 | TheCompWiz | lol |
| 21:12.50 | WIMPy | The CHANNEL function should give you the peer name. |
| 21:13.00 | TheCompWiz | igcewieling: probably... but sometimes I get so focused on making it work... I forget to look at what's available. |
| 21:14.53 | igcewieling | TheCompWiz: the time to do this is not when you are trying to resolve an issue |
| 21:15.02 | TheCompWiz | LOL |
| 21:15.10 | TheCompWiz | ... and when that might be? :D |
| 21:15.28 | igcewieling | ssh into asterisk from your smart phone from the bathroom? |
| 21:16.04 | igcewieling | when you say to yourself "I wonder if asterisk can...". or any number of times. |
| 21:25.34 | [sr] | WIMPy: need your help |
| 21:25.38 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 21:26.32 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 21:28.53 | *** join/#asterisk Xliff (~clifton@208.58.82.22) |
| 21:30.47 | Xliff | Hello. I represent a company that has been using an Altigen system for about 2 years. Due to the fact that we have chosen to switch telecom providers from Megapath to RCN, we are discovering that the existing Altigen system can't connect to their SIP lines. We are fed up with Altigen and were wondering if we could replace the Altigen server with an Asterisk replacement but still keep use of the Altigen IP 720 phones. Does anyone know the likelyhood of this |
| 21:30.48 | Xliff | ? |
| 21:31.27 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:32.24 | navaismo | if your phones has the native SIP protocol I guess asterisk can use it |
| 21:33.18 | [sr] | who can help me here on a thing, after a core reload, dahdi stop's working |
| 21:33.35 | [sr] | i only can make/receive call's after dahdi restart |
| 21:33.57 | [sr] | i have this bug request: https://issues.asterisk.org/jira/browse/PRI-154#comment-202978 |
| 21:35.09 | navaismo | <PROTECTED> |
| 21:36.20 | [sr] | navaismo: when do you want me to load it, after the reload? |
| 21:36.59 | [TK]D-Fender | Obviously not |
| 21:37.22 | [TK]D-Fender | Doing a load after a reload... would be loading a module after it should already be loaded and just generate an error |
| 21:40.15 | [sr] | think i got it, after the reload it seems to crashed, and dahdi restart makes the load |
| 21:40.18 | [sr] | we then |
| 21:41.52 | igcewieling | Xliff: RCN is giving you a SIP handoff and not analog or PRI? |
| 21:42.56 | [sr] | navaismo: http://pastebin.com/WATjDFut |
| 21:43.12 | [sr] | navaismo: no call's are possible at this moment (no dahdi restart made yet) |
| 21:43.45 | navaismo | thats what [TK]D-Fender was talking about |
| 21:44.09 | navaismo | try unloading and then load without restart dahdi |
| 21:45.31 | [sr] | navaismo: http://pastebin.com/3abZmQXZ |
| 21:45.53 | [sr] | i can make call's now |
| 21:48.37 | [sr] | navaismo: what can i do to debug/help on the fix? |
| 21:48.37 | igcewieling | heh, I thought [sr] was restarting dahdi there for a moment. |
| 21:49.24 | navaismo | [sr] why you are using core reload? |
| 21:49.56 | igcewieling | if you "service dahdi restart" (or whatever is correct for your OS) while chan_dahdi is loaded in Asterisk expect bad things to happen. |
| 21:50.29 | [sr] | igcewieling: no! dahdi restart, in asterisk off course!! |
| 21:50.44 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 21:50.57 | igcewieling | [sr]: saying chan_dahdi instead of dahdi would lessen confusion |
| 21:50.58 | [sr] | navaismo: :$ it's freepbx when i make changes, i think it does a core reload after it (am i wrong?) |
| 21:51.14 | [sr] | igcewieling: i admit, you're right |
| 21:51.54 | [sr] | navaismo: there's the pri debug on the #bug, don't if that info helps on anything |
| 21:54.18 | navaismo | yes freepbx reload all |
| 21:56.32 | navaismo | [sr] you can pb all the reload output |
| 21:56.53 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 21:57.50 | Xliff | igcewieling: Yes. SIP and not analog/PRI. We already have PRI through Megapath. |
| 21:57.58 | [sr] | navaismo: yes, sec |
| 21:58.48 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 21:58.53 | polysics | hello! |
| 21:59.00 | Xliff | I am thinking about grabbing a small server, taking one of our phones and trying to connect. These phones are also IP phones and the hope is we can still use those through Asterisk as well. Otherwise we're going to have to sink a lot of money into replacing the entire phone infrastructure. |
| 21:59.10 | Xliff | Which may just be unavoidable. |
| 21:59.11 | polysics | I managed to get UniMRCP working with Asterisk and Lumenvox |
| 21:59.23 | polysics | but apparently only the Speech family of applications works |
| 21:59.35 | polysics | since I need to pass grammars inline, I need MRCPRecog |
| 21:59.45 | polysics | but it really looks like it's not working |
| 22:00.04 | [sr] | navaismo: http://pastebin.com/NmL3HQde |
| 22:00.08 | polysics | the only (cryptic) error I get is this: RECOGNIZER channel error! |
| 22:00.47 | polysics | dialplan here: https://gist.github.com/polysics/5bfc86d103cbe1310d8e |
| 22:01.02 | polysics | anyone has has seen this before? |
| 22:01.17 | polysics | Lumenvox reports no error so I might be inclined to blame the app itself |
| 22:02.42 | *** join/#asterisk Fubard (~brawr@vpn.bctconsulting.com) |
| 22:02.46 | navaismo | [sr], seems like your system.cof is lost or unconfigured |
| 22:03.17 | [sr] | navaismo: for dahdi? |
| 22:03.55 | [sr] | navaismo: for dahdi, it's there |
| 22:03.58 | navaismo | at line 71 start chan_dahdi complaining about missing signalling |
| 22:04.37 | navaismo | 'Attempt to configure channel 1 with signaling Unknown signalling -1 ignored because it is already configured to be FXS Kewlstart.' |
| 22:04.58 | navaismo | the owner of the files in /etc/asterisk are asterisk? |
| 22:05.04 | [sr] | navaismo: but it is set, bri_cpe_ptmp |
| 22:06.05 | [sr] | navaismo: yes, all of them , but i'm going to make a amportal chown |
| 22:06.27 | igcewieling | Xliff: better vendor and community support may easily make replacing the phones the logical thing to do |
| 22:07.01 | [sr] | navaismo: same thing, freepbx reload, dahdi stops working after it |
| 22:07.05 | Xliff | igcewieling: I agree, wholeheartedly but I do have to do due dillgence so that the bean counters are willing to part with the $$$ |
| 22:07.25 | navaismo | can you pb the system.conf and chan_dahdi.conf?? |
| 22:07.26 | Xliff | The only problem is that this system isn't even 2 years old. |
| 22:07.47 | pabelanger | [2013-02-14 15:44:01.824] WARNING[6179]: res_fax_spandsp.c:368 spandsp_log: WARNING T.30 ECM carrier not found |
| 22:07.54 | pabelanger | any feedback on what that means? |
| 22:08.21 | Fubard | Has anyone ever had a SIP phone fade in/out randomly? It is definately not cutting in/out (as in dropped packets) but the volume is bounces between normal and very low for a fractions of a second at a time then goes back to normal for the rest of the call. |
| 22:08.34 | igcewieling | pabelanger: looks like a non-fax call? |
| 22:08.35 | Fubard | I put the phone on an Online UPS to rule out power fluctuation and I put a laptop on a hub one hop downstream of the phone, ran a packet capture, pulled the audio out of the rtp stream and hear the fading there. |
| 22:09.11 | pabelanger | igcewieling, it actually was a fax |
| 22:09.13 | polysics | no one has ever seen a situation where Speech works and MRCPRecog does not? |
| 22:09.24 | [sr] | navaismo: http://pastebin.com/cqDJseL0 |
| 22:09.33 | pabelanger | fax came through, but will the above warning |
| 22:10.05 | [sr] | navaismo: the rest of the include files are empty |
| 22:10.45 | navaismo | [sr], and the system.conf? |
| 22:11.10 | [sr] | ops forgot |
| 22:11.58 | [sr] | navaismo: http://pastebin.com/CQJHK8kz |
| 22:12.15 | Fubard | pabelanger: Error Correction Mode (ECM) not found on the carrier side perhaps? |
| 22:14.50 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 22:15.13 | navaismo | after the reload the /proc/dahdi still exist? |
| 22:15.18 | navaismo | [sr], ^ |
| 22:15.24 | Fubard | pabelanger: if the fax came through okay then it sounds like during the handshake it was determined that ECM would not be used. If you have a large number of failed faxes then it might be worth investing time into findout out why its not being used. |
| 22:17.06 | Fubard | pabelanger: otherwise just let it be. if you can get ECM to work 'some' of the time then chances are that the reason it didn't work this one time is due to the far-end's fax machine. when in doubt, pass the buck. :) |
| 22:17.52 | [sr] | navaismo: yes, check: (on this moment cant make call's) http://pastebin.com/9cvzV1yp |
| 22:20.18 | navaismo | thats weird... how else... |
| 22:20.26 | igcewieling | Often people see a post on the internet from some idiot which recommends disabling ECM on fax over ulaw, which sort of helps a horrible situation be just terrible, but does nothing for T.38 |
| 22:21.03 | pabelanger | Fubard, that is possible. We are just in the testing phase right now, using 1 fax machine so it might be far end |
| 22:21.12 | [sr] | navaismo: ya, i even thought that was some bug from 2.11x but no, have no ideia what could be this |
| 22:21.13 | pabelanger | but ya, will keep an eye on it and see why we are getting the warning |
| 22:21.20 | pabelanger | might have to just read the source and see |
| 22:22.05 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 22:23.44 | navaismo | [sr], core show locks show something? |
| 22:23.59 | [sr] | navaismo: in the state where i cant make calls? |
| 22:24.23 | igcewieling | [sr]: must be tired again, he is mixing up FreePBX verisons and Asterisk versions |
| 22:25.23 | navaismo | [sr], yes when you cant make calls |
| 22:25.37 | navaismo | brb going to eat |
| 22:25.38 | [sr] | navaismo: dont have "core show locks" :) |
| 22:26.14 | navaismo | oh yes that option need to be enabled on compiler flags |
| 22:26.16 | navaismo | bbl |
| 22:26.30 | [sr] | let me recompile then |
| 22:26.31 | [sr] | w8 |
| 22:30.18 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 22:31.35 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 22:31.58 | nny | https://issues.asterisk.org/jira/browse/ASTERISK-7396 (and from the snom wiki) A solution would be to use the "P-Asserted-Identity" header. If this header is present, the phone will display its content instead of the one from the "From" header. (in Asterisk 1.8 you must set "sendrpid=pai" in sip.conf to send the "P-Asserted-Identity" header). |
| 22:32.14 | nny | Is the above information still the best practice for retaining callerid through attended transfer |
| 22:32.16 | nny | ? |
| 22:32.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.18.2) |
| 22:33.19 | Xliff | Thanks for the help, everyone. More research and then I think we may have to bite the bullet. |
| 22:33.25 | nny | looks like I need to enable connected line updates in 1.8 |
| 22:36.35 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 22:36.51 | nny | so it looks like setrpid is set to "no" by default. What process outside of setting to yes is needed to get attended transfers to update the caller ID? |
| 22:38.10 | WIMPy | sendrpid=pai and rpid_update=yes should do it. |
| 22:42.20 | nny | WIMPy: will test on 2 peers in the network, thank you |
| 22:44.17 | nny | WIMPy: if i cna ask another question: I have my SIP definitions for two systems setup with fromuser=XXX and it's essentially breaking callerid/callback through the two boxes (from phone to phone). The callerid says "XXX" instead of "Extension 1234". I did this because the system would elsewise complain that the peer auth user name was wrong. Can I set the trunk to jsut allow all traffic from point A to point B via IP and ignore the user nam |
| 22:45.10 | WIMPy | Use the IP if it is static. |
| 22:46.27 | nny | OK. when Peer 1 calls Peer 2 it usually gets the called party information, with the SIP bridge it's not. (shows calling "109" instead of "109, John Doe". Any of what I said related to that? |
| 22:46.29 | WIMPy | But I think RPID/PAI shoudl also work in that case. |
| 22:46.50 | nny | WIMPy: interesting will do some testing, just asking questions as I have to test during off hours and gathering some base intel |
| 22:47.18 | WIMPy | Do you have trustrpid? |
| 22:47.29 | nny | WIMPy: not right now |
| 22:47.39 | nny | WIMPy: default settings for those, just learning about them |
| 22:47.54 | WIMPy | NFI what the default is. |
| 22:48.28 | igcewieling | if you have trustrpid=yes then Callerid will be set by Reporte-Party-ID header, same for trustrpid=paid |
| 22:48.33 | nny | yeah voip-info doesn't either. What would be the preferred setting in this case? |
| 22:48.42 | igcewieling | ..er..trustrpid=pai |
| 22:48.54 | igcewieling | nny: what does sip.conf.sample say about it? |
| 22:49.09 | nny | sounds like this header would fix a lot of these issues |
| 22:49.28 | nny | igcewieling: i'll check, i usually don't do make samples when installing |
| 22:49.42 | nny | but i have a new backup of them here |
| 22:49.49 | WIMPy | Would be great if both could be used. |
| 22:50.42 | nny | ;trustrpid = no ; If Remote-Party-ID should be trusted |
| 22:50.52 | nny | doesn't say what it is by default |
| 22:51.06 | nny | can i check via cli on a system where it isn't defined? |
| 22:51.25 | nny | i have never looked to see if asterisk will spit back variable values for a config file from cli -_- |
| 22:54.13 | igcewieling | nny: /path/to/asterisk-src/configs/sip.conf.sample |
| 22:54.17 | WIMPy | Some |
| 22:54.53 | WIMPy | Would be a good idea to make it configurable WHERE from/rpid/pai end up instead of only which one to use. |
| 22:56.47 | nny | igcewieling: yeah that's the line in the sample config |
| 22:57.51 | igcewieling | I use sendrpid=yes trustrpid=yes rpid_update=yes |
| 22:58.06 | nny | igcewieling: WIMPy I will test all three and see what happens |
| 22:58.07 | igcewieling | actually sendrpid=pai |
| 22:58.39 | nny | I don't need to set the PAI in the dialplan then? |
| 22:58.41 | igcewieling | there we go: trustrpid=yes sendrpid=pai rpid_update=yes |
| 22:58.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:59.13 | igcewieling | sendrpid=pai appears to generate the PAI from CID on outgoing calls |
| 22:59.44 | nny | igcewieling: gotcha. Ok I'll test it along with changing the sip peer defintions between the two boxes to static IP instead of auth/registration as well |
| 23:00.58 | WIMPy | wonders why his status subscriptions don't work any more. Did I pick a bad Asterisk verison? |
| 23:01.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.54 | igcewieling | WIMPy: maybe you have too many? We get that all the time. Users want to have 5 phones monitor 20 extensions each and the polycoms get sort of upset about that |
| 23:03.25 | WIMPy | Nah. It used to work. |
| 23:04.20 | WIMPy | I have hints for call forwarding states. But when the CFU is activated by cron, all others switch off. |
| 23:04.21 | nny | for my bridged systems I went with setting the phone to ext@box 1 or ext@box 2 for the BLF subsscriptions (and adding the peer definition to both boxes). The cisco phone just works that way |
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| 23:06.53 | WIMPy | tries todays asterisk to see if there's a difference tomorrow. |
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| 23:08.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:20.27 | WIMPy | I like the lightshow effect on BLF in the minutes after restarting Asterisk. However that may work. |
| 23:20.56 | ChannelZ | Das blinkenlights |
| 23:21.21 | WIMPy | Extremely slooooow, but yes, somehow. |
| 23:22.56 | ChannelZ | http://www.plzi.com/kuvia/files-omat/achtung_iso.gif |
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| 23:23.39 | nny | heh |
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