| 00:00.45 | navaismo | bad link |
| 00:01.14 | navaismo | well then, do you reload the dialplan? |
| 00:01.48 | Docfxit | Yes. Lets try the fist one again |
| 00:02.10 | Docfxit | http://bin.cakephp.org/view/1230663065 |
| 00:02.59 | navaismo | exten => n |
| 00:03.03 | navaismo | fix that |
| 00:04.07 | Docfxit | Do you want me to make it exten => 130,n, ? |
| 00:04.19 | gusto | i wonder, seeing that DAHDI devices appear here quite often on this channel |
| 00:04.37 | gusto | seems to me like there would be a lot of ppl out there using that HW |
| 00:05.23 | navaismo | Docfxit, yes |
| 00:05.32 | Docfxit | Ok. |
| 00:05.49 | navaismo | gusto, yes many people use dahdi, I use alot dahdi, |
| 00:06.44 | cusco | howelse do you connect to a PRI? |
| 00:07.25 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 00:09.41 | gusto | well, i have no primary rate connection, first of all |
| 00:10.32 | gusto | and second, these cards are not cheap, they are rather expensive, i just looked up how much a 2x FXO 2x FXS would cost ... 600 USD!!! |
| 00:11.03 | navaismo | i think b600 its very affordable |
| 00:11.17 | gusto | well, maybe one buys such a card once in a lifetime, that is an argument that would count |
| 00:11.23 | navaismo | i always try to recommend digium hardware but you have options |
| 00:11.40 | navaismo | openvo--coff-coff |
| 00:12.15 | gusto | i mean, i have a Marantz here, so who am i to judge, but that is something i knew that should last for LONG |
| 00:13.02 | navaismo | so whats happen Docfxit ? |
| 00:13.12 | Docfxit | navaismo: I updated the dial plan |
| 00:13.18 | gusto | and with these DIGIUM cards it may be the same, that once you buy it, you have it as long as you have use for it |
| 00:13.39 | Docfxit | It's in http://bin.cakephp.org/view/1230663065 |
| 00:13.54 | gusto | or when you make money of it, that's another argument |
| 00:14.03 | *** part/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 00:14.08 | cusco | depends on the needs. cheapest solutions are voip |
| 00:14.26 | Docfxit | I updated the cli in http://bin.cakephp.org/view/675566775 |
| 00:14.29 | navaismo | or a modem with tiger cheapset aka x100p |
| 00:15.06 | navaismo | Docfxit, "Invalid extension '13'" you dialed 13 not 130 |
| 00:15.08 | Docfxit | When I made the call I dialed 130 It shows I dialed 13 in the cli. |
| 00:15.15 | gusto | aha |
| 00:15.17 | navaismo | ok press slowly this time |
| 00:15.18 | Docfxit | I did it a few times. |
| 00:15.18 | gusto | a MODEM! |
| 00:15.22 | gusto | that's true |
| 00:15.29 | Docfxit | It came out wrong every time. |
| 00:15.39 | gusto | i had once a modem i could do calls with ... but that was my ISA 33,6 one |
| 00:15.40 | Docfxit | I used two different phones. |
| 00:15.53 | gusto | and it had 1x FXS and 1x FXO |
| 00:15.54 | Docfxit | It still came out wrong. |
| 00:16.02 | navaismo | Docfxit, try to set relaxdtmf in your trunk or dial slow |
| 00:16.05 | cusco | that is one channel only |
| 00:16.29 | navaismo | 1 channel 1 exten |
| 00:17.42 | Docfxit | I'll give it another try. |
| 00:19.31 | Docfxit | As soon as I put in 13 I see Asterisk doesn't wait for the zero. |
| 00:20.01 | Docfxit | Maybe I should change the extension so it doesn't include a zero. |
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| 00:24.49 | gusto | hm |
| 00:25.37 | gusto | there are no modems out there, but the most soundcards/onboard souncards have integrated modems, could they be of use? or there is no way how to connect to them, because there is no connector on a board |
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| 00:27.51 | *** mode/#asterisk [+o file] by ChanServ |
| 00:32.26 | Docfxit | Navaismo: Any ideas as to why it would only accept the 13 ? Maybe it's something else in the dialplan? Would you like to see the complete dialplan? |
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| 00:35.24 | navaismo | pb |
| 00:38.03 | gusto | however, not connection VoIP to PSTN is the most secure option |
| 00:38.17 | navaismo | Docfxit, with what kind of phones are you testing? |
| 00:38.55 | Docfxit | I tried with a cell phone and a land line phone. |
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| 00:39.09 | gusto | for me the question is more like how to cheaply connect old analogue phones to VoIP, because old phones are available in high numbers and VoIP is the future, so ... for me the cheapest solution were PAP2T and SPA112 |
| 00:39.32 | leifmadsen | gusto: those modems will not be of use to Asterisk |
| 00:39.47 | leifmadsen | you can certainly use ATAs to use analog phones |
| 00:39.48 | gusto | leifmadsen: i already thought so |
| 00:40.02 | Docfxit | navaismo: It just came to me that we have voice direction prompts. How does Asterisk know when to dial an extension and when to follow the prompts? |
| 00:40.08 | gusto | leifmadsen: yes, but an ATA isnt cheap either |
| 00:40.09 | leifmadsen | but the cost of the ATA is approaching the cost of an entry VOIP device |
| 00:40.12 | leifmadsen | true story |
| 00:40.17 | leifmadsen | welcome to the world of telephony |
| 00:40.22 | leifmadsen | use softphones if you need free |
| 00:40.22 | gusto | yes |
| 00:40.27 | gusto | nooo |
| 00:40.35 | leifmadsen | then pony up some money |
| 00:40.37 | leifmadsen | those are your options |
| 00:40.45 | leifmadsen | or get into a new line of work |
| 00:40.50 | gusto | i was just saying, because we have SOME old working analogue telephones here |
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| 00:41.55 | leifmadsen | the handsets aren't the issue... it's the connectivity to SIP |
| 00:42.05 | leifmadsen | you either pay in the ATA or in the SIP phone |
| 00:42.08 | leifmadsen | either way, you pay to connect |
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| 00:43.04 | navaismo | Docfxit, asterisk knows based on your dialplan |
| 00:43.04 | leifmadsen | runs away |
| 00:43.27 | Docfxit | navaismo: In voicemenu-custom-5 there is a WaitExten(20) for the extension. |
| 00:43.36 | gusto | leifmadsen: of course |
| 00:43.58 | Docfxit | So I don't understand why it didn't wait for 130. |
| 00:44.15 | leifmadsen | show dialplan section |
| 00:44.25 | gusto | leifmadsen: the best soulution for me would probably be one ATA where i can connect a lot of telephones to, like 16 ports or such but then i would pay in cables :-D |
| 00:44.40 | leifmadsen | facepalms |
| 00:44.42 | leifmadsen | gusto: I say good luck to you sir |
| 00:45.09 | gusto | leifmadsen: dont worry, at the moment, i am fine |
| 00:45.16 | leifmadsen | I'm not worried |
| 00:45.21 | navaismo | Docfxit, yes 20 secons to enter a exten seems fair enough, but waht kind of phone are you using |
| 00:46.33 | Docfxit | I tried with a cell phone and a land line phone. Do you want to know the model phones? |
| 00:47.53 | Molo | ya i'm tired of spending $ on ATAs |
| 00:48.07 | Molo | people just need to quit faxing and use email |
| 00:52.22 | navaismo | Docfxit, nope |
| 00:52.32 | navaismo | Molo, +100000 |
| 00:52.59 | navaismo | why the hell people use faxes, they can scan and email |
| 00:53.54 | navaismo | Docfxit, if you can fix quickly cahnge your exten 130, if not debug your dtmf and check the if relaxdtmf is enable in your trunk |
| 00:56.47 | Docfxit | What file is relaxdtmf in? |
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| 00:57.31 | navaismo | wtf twinkie fingers, i cant write |
| 00:57.52 | navaismo | Docfxit, in chan_dahdi.conf or dahdi-channels.conf |
| 00:58.25 | Docfxit | I just got it working. |
| 01:00.33 | navaismo | great |
| 01:00.35 | Docfxit | In the past the cli was showing it going to voicemenu-custom-5. I just noticed it went to voicement-custom-4. I included Voice_Prompt_That_I_Recorded in voicemenu-custom-4 and it started working. |
| 01:00.58 | Docfxit | navaismo: Thank you for all your help. |
| 01:01.25 | Docfxit | Sorry I'm not more savy with Asterisk. |
| 01:01.34 | navaismo | np |
| 01:02.17 | Docfxit | I think the other guys have left for the evening. If you get a change please thank them for me. |
| 01:03.35 | navaismo | k they back later |
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| 01:06.06 | Docfxit | Great. |
| 01:06.09 | gusto | so |
| 01:06.11 | Docfxit | Thanks a bunch. |
| 01:06.22 | gusto | of apples |
| 01:11.55 | navaismo | I preffer cranberries |
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| 02:44.17 | Russ | so I noticed today that t-mobile supports 'hd voice', and my voip provider already supports hd voice |
| 02:44.21 | Russ | what has to happen for the two to work together? |
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| 03:13.08 | hebber | Hi, I have a problem loading acl's into ARA using mysql - asterisk don't want to load the table without the presence of the acl.conf file. When its presence it loads some info from the file and some info from mysql. Does someone have an idea of what goes wrong? |
| 03:14.38 | hebber | yes, the acls is defined in extconfig.conf : acls => odbc,asterisk,ast_acltable |
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| 03:33.43 | leifmadsen | hebber: acl.conf => odbc,asterisk.ast_acltable |
| 03:33.54 | leifmadsen | because you can only load it statically, not dynamically |
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| 03:38.36 | hebber | Hi Leif, thanks for reply, I will see how I get it to load in statically, instead of dynamically |
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| 03:39.25 | hebber | any hints in your book? :-) |
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| 03:41.44 | hebber | I see now that your corrected me already |
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| 03:46.50 | hebber | leifmadsen: however I did follow the documentation here: https://wiki.asterisk.org/wiki/display/AST/Named+ACLs using the lastest branch of Asterisk 11 |
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| 03:48.07 | leifmadsen | hebber: I can't speak to that -- I just looked in my 1.8 configs, but possible ACLs are dynamic in 11 now |
| 03:48.19 | leifmadsen | Kobaz: ^^^^^^^^ |
| 03:48.23 | leifmadsen | I think he wrote that stuff |
| 03:49.02 | leifmadsen | ah ya, looks like named acl's exist in asterisk 11 now which can be loaded dynamically |
| 03:49.07 | leifmadsen | I can't speak to it unforatunetly |
| 03:49.09 | leifmadsen | and now I'm off to bed |
| 03:49.33 | hebber | ok, thank you Leifmadsen - will continue to try and fail :) |
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| 04:24.27 | hebber | Kobaz: may I ask you a question regarding realtime ACL - realtime using the documentation only works if the acl.conf is loaded together with realtime configuration. |
| 04:25.08 | hebber | without the presence of acl.conf - realtime acls fails to load |
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| 04:54.00 | MooJuice | anyone had issues with call transfers randomly not succeeding with asterisk 10.11.1? I get a SIP Notify from the receiving party (SIP client) but both parties just hear MOH and then the call drops? |
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| 06:04.54 | rue_bed | so I have this mgcp problem, I can only get one gateway to work |
| 06:05.12 | rue_bed | is mgcp.conf supposed to be able to handle more than one? |
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| 06:41.23 | Russ | are the dahdi TDM400P boards limited to 8khz? |
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| 06:45.20 | ChannelZ | I think the answer is basically yes |
| 06:45.58 | Russ | hmm..suppose it wouldn't matter anyway, I'm guessing my cordless phones are at 8kHz anyway |
| 06:46.09 | Russ | what's a home user to do |
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| 06:46.40 | ChannelZ | well it's analog so it's more to do with the D/As |
| 06:47.11 | ChannelZ | Although internally I'm sure it's treated similarly to ulaw |
| 06:47.22 | Russ | my cordless phones are digital |
| 06:48.11 | ChannelZ | but if they are POTS they would still get it analog from the TDM card |
| 06:48.27 | Russ | right, so both the phones and the fxs would need to support 16khz |
| 06:48.42 | Russ | hmm, wonder what the att and vtech marketing speak means 'best sound quality by extending the frequency band' |
| 06:48.48 | Russ | (on their cordless phones) |
| 06:50.32 | ChannelZ | At the end of the day I don't think it matters much since the whole public phone network isn't wideband |
| 06:51.02 | Russ | t-mobile is now |
| 06:53.15 | ChannelZ | great if you have direct access to their network |
| 06:53.56 | Russ | I'm curious if some of that is already happening, like the termination between my provider (teliax) and t-mobile, does it already happen over the internet? |
| 06:55.19 | Russ | it'd also be nice if I make a SIP call to home, that it'd be wideband |
| 06:55.19 | ChannelZ | I suppose it'd be more likely but probably not widespread, it would depend on teliax's termination with the rest of the world |
| 06:55.43 | Russ | and some countries are wideband, yes? no? |
| 06:58.03 | ChannelZ | I don't know specifically |
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| 06:58.29 | Russ | I think AU is, since Telstra is |
| 06:58.41 | ChannelZ | I would imagine still only with mobile networks though, since they have control over it and it would be easier to do given the nature of the devices using the network |
| 06:58.51 | Changos | Hi guys, I need to help |
| 06:59.01 | Russ | ok, you can help me! |
| 06:59.15 | Russ | hook me up with wideband goodness |
| 06:59.23 | ChannelZ | http://en.wikipedia.org/wiki/Wideband_audio#Deployment |
| 07:02.40 | Changos | I've one server on Gentoo with Asterisk, I've E1 card and this work fine, but I can't find who I can split all E1 (30 Channels) by line number, e.g. I have 4 lines number for my ISP and all this input from E1, but how I can split the E1 for, line 1 max 5 channel, line 2 max 10 channels, line 4 all remaining |
| 07:03.08 | Changos | I already seach the Internet several times, and found nothing about it :s |
| 07:03.56 | ChannelZ | Well the channels are virtual. |
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| 07:04.37 | ChannelZ | In the sense that you don't typically only get calls from a particular DID on a specific channel |
| 07:04.38 | schmidts | good morning |
| 07:05.37 | ChannelZ | mo-nan |
| 07:06.20 | Changos | ChannelZ: yes, I know |
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| 07:06.57 | Changos | ChannelZ: have you some example code/guide how I can create this virtual channels ? |
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| 07:08.33 | ChannelZ | They already are. If you are wanting to control outgoing calls, you could do it with channel groups but I don't think you have any real control over incoming... besides writing a semi-complicated call counter system or something that would track how many calls are active for a given DID and then reject new calls if that limit had been reached or something |
| 07:09.25 | ChannelZ | Is there a reason you want to be so rigid with it? |
| 07:13.28 | ChannelZ | Say you had 2 companies sharing the same PRI and wanted to allocate 10 channels to company A and 10 channels to company B, you could just make 2 different channel groups, one for channels 1-10 and one for 11-20 for instance, and then only dial out through the respective groups. But if company A got 11 incoming calls simultaneously, I'm not sure there's any easy built in way to refuse that. (Perhaps your telco can limit channels per DID? No idea honestl |
| 07:13.29 | ChannelZ | y, I don't really deal with PRI..) |
| 07:14.06 | Changos | ChannelZ: Well, the issue is, line 1 is for administrative, line 2 call center, when a lot people calling to company, call center receive those calls. But if incoming calls to Call Center e.g. is 30 simultaneously, line 1 (administrative) can't output call because not have channel available |
| 07:16.07 | hebber | Changos: I think you can limit the ACD queues to a certain limit of people waiting, hence disconnect more calls to the queue |
| 07:16.14 | ChannelZ | I think you'd have to do it manually with a call counter |
| 07:16.50 | Changos | so I guess that this example is very common, and I hope that exist some solution for somehow split the E1 Channel |
| 07:17.09 | ChannelZ | and then look at the counter and reject the 30th incomg call to try and leave a channel open |
| 07:18.05 | Changos | hebber: yea, just right now I've this configuration, but I think that this form is not the best. |
| 07:18.37 | Changos | I hope that exist other form "more clear" |
| 07:19.53 | hebber | Changos: ok |
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| 07:21.18 | ChannelZ | Look at the GROUP and GROUP_COUNT functions |
| 07:21.51 | Changos | sincerely, I guess that limit the ACD queues is not a good solution, I think that this is "dirty", but work, xD |
| 07:22.08 | Changos | ChannelZ: thanks, I'll try |
| 07:22.50 | kaldemar | Changos: you can't control in asterisk how the calls come in from the provider. |
| 07:23.03 | ChannelZ | this might give you an idea: http://www.astblog.com/2008/09/17/count-and-limit-number-of-calls-under-asterisk/ |
| 07:23.23 | Changos | i thought that there was a simpler way, e.g. Channel 1-10 DID XXXXX channels 11-15 DID XXXXX and not more |
| 07:23.51 | kaldemar | that's something the provider would have to do. |
| 07:24.30 | Changos | kaldemar: yeah, I'll calling to ISP and ask about it |
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| 07:27.56 | Changos | thanks guys ! |
| 07:27.58 | Changos | ChannelZ: |
| 07:27.59 | Changos | hebber: |
| 07:28.01 | Changos | kaldemar: |
| 07:29.31 | ChannelZ | good luck |
| 07:31.33 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.186.231) |
| 07:36.55 | ChannelZ | hmm well looking at it, that example I posted is totally broken |
| 07:39.09 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-055-203.pools.arcor-ip.net) |
| 07:39.26 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 07:39.32 | kaldemar | ,n,GROUP(${EXTEN}) |
| 07:40.59 | ChannelZ | yeah it's a function not an application |
| 07:41.26 | ChannelZ | the idea is sort-of partly right anyway :) |
| 07:44.55 | ChannelZ | untested but something more like this http://pastebin.com/j4MBXNtY |
| 07:45.24 | kaldemar | that's a perfect example for why random googling for examples is bad. |
| 07:45.31 | ChannelZ | heh yeah |
| 07:45.48 | ChannelZ | Broken Since 2008(tm) |
| 07:46.20 | kaldemar | i'd put the count check first. why bother putting the channel in a group if the group is full anyway? |
| 07:47.55 | kaldemar | bloggers should pay more attention to the crap they write. |
| 07:48.22 | ChannelZ | But then they'd be writers, not bloggers. |
| 07:49.10 | ChannelZ | Although that's even getting questionable these days |
| 07:49.17 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:50.34 | ChannelZ | Are you the guy from Lost? |
| 07:50.56 | ChannelZ | Oh wait.. that was a John. |
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| 08:06.05 | ghost75 | i was ever wondering why blogs are so popular ( in my eyes its crap) |
| 08:09.17 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 08:21.51 | ChannelZ | mostly |
| 08:22.36 | WIMPy | Just like forums. |
| 08:23.01 | WIMPy | AKA the concept of newsgroups made unusable. |
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| 08:24.15 | PbxMan | morning |
| 08:24.46 | WIMPy | Cool. Here as well. |
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| 09:38.11 | nunne_ | Refer/SIP Response 302 "Moved Temporarily" ... It doesn't send the call via my __TRANSFER_CONTEXT ... Is this the normal behaviour? Or is it any other way to see if it's a transfered called? (need to make sure the channels is not answered again) |
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| 09:42.13 | kaldemar | nunne_: 302 is not transferring, but forwarding. |
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| 09:42.33 | wdoekes | nunne_: I set a __var before dialing. if I see that that var is set in my phone context, I know it's a redirect |
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| 09:43.26 | nunne_ | wdoekes: smart workaround! thanks :) |
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| 09:47.36 | rox | Hello, I am trying to end a SIP call with a specific code (i need to distinguish this call ending from others). I was trying to set the HANGUPCAUSE variable, but no matter what i set to it, i always get 19 on the caller's side. I do a Hangup(${MyCode}). How would one go about ending a SIP call with some kind of identifiable information? |
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| 09:50.45 | rox | Here is the code: http://pastebin.com/cPzRpai2 |
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| 10:08.14 | kaldemar | rox: what do you see in CLI? |
| 10:08.45 | rox | kaldemar: on the other side i get DIALSTATUS=CONGESTION and HANGUPCAUSE=19 |
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| 10:09.16 | rox | kaldemar: i don't need to send this specific HANGUPCAUSE over, any meaningful way to reject the call would be sufficient for me |
| 10:09.20 | kaldemar | sure, but what do you see in CLI? |
| 10:09.33 | rox | kaldemar: what would be the canonical way to meaningfully reject a SIP call in asterisk=? |
| 10:10.09 | rox | kaldemar: a second |
| 10:10.32 | kaldemar | enable sip debug too while you're at it. |
| 10:11.19 | rox | CLI is silent, apart from Verbose messages |
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| 10:11.39 | rox | how about the general wuestion, how would one go about meaningfully rejecting a call in asterisk? |
| 10:11.42 | rox | how would you do it? |
| 10:11.59 | kaldemar | with a cause code, just like you're trying to. |
| 10:12.05 | rox | did i do something completely inapropriate? |
| 10:12.21 | rox | ok, so i got that right |
| 10:12.23 | kaldemar | the verbose messages are part of the interesting stuff. |
| 10:12.55 | rox | hmm, i guess the other side is translating my cause code into 19 even though i am sending 16 |
| 10:13.52 | rox | in reality, this is a loop call, i call somebody and they call me back, then i have to reject the call, but i have to distinguish my call rejection from theirs |
| 10:14.26 | *** join/#asterisk schmidts (~schmidts@vie-086-059-105-021.dsl.sil.at) |
| 10:14.31 | schmidts | hello |
| 10:14.31 | rox | i need to distinguish between me rejecting the call and the called SIP trunk actually being congested |
| 10:15.05 | schmidts | does anyone of you have allready tried using asterisk with openfire for distributed device state (PUBSUB) ? i allways get a 403 error when asterisk tries to send a device state |
| 10:15.13 | schmidts | register and even jabbersend works fine |
| 10:18.56 | wdoekes | rox: sending hangup 16 doesn't make sense.. NORMAL_CLEARING = hangup after the call was picked up |
| 10:19.57 | rox | wdoekes: ok, then it makes sense that the other side is overriding this code |
| 10:19.59 | wdoekes | rox: I don't know which codes you already pass on, but you can take a look in include/asterisk/causes.h and cause2sip in chan_sip.c |
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| 10:20.40 | rox | wdoekes: ok, I will try some different status codes |
| 10:20.45 | wdoekes | (pro-tip: Hangup(USER_BUSY) and any other identifier found in causes.h works) |
| 10:22.27 | kaldemar | that should be documented in the app if the feature is meant to be permanent. |
| 10:22.30 | *** part/#asterisk deo_ (~deo@222.127.13.226) |
| 10:23.50 | kaldemar | then again, with SIP you need to look at the source anyway since that seems to be the only place that states what values mean in SIP terms. |
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| 10:25.58 | rox | kaldemar: i think the other side overrides my code with DIALSTATUS=CONGESTION and HANGUPCAUSE=19 no matter what i send, so i guess i'll have to work around it |
| 10:26.25 | rox | any other reasonable ways to reject calls, other then simple Hangup(${Code}) |
| 10:26.28 | rox | ? |
| 10:27.55 | wdoekes | Hangup(code) is the right way.. and it works.. try a couple of common ones: USER_BUSY (sip/486), UNALLOCATED (sip/404) |
| 10:29.11 | rox | wdoekes: overridden, i've tried the list |
| 10:29.40 | wdoekes | what kind of sip code gets sent then? |
| 10:29.54 | kaldemar | if the other end overrides what you set, there's nothing you can do is there? |
| 10:30.11 | kaldemar | i asked for the CLI output to see if you're setting it properly. |
| 10:30.46 | rox | kaldemar: i put the code in pastebin |
| 10:31.43 | rox | kaldemar: the actual CLI output will take me about 10 minutes to grep out, it's a production machine with quite some traffic |
| 10:32.33 | kaldemar | rox: the dialplan proves nothing. you're using a variable that is not set in those lines. |
| 10:33.29 | wdoekes | and the hangupcauseclear() app that doesn't exist, as well as setting the HANGUPCAUSE variable.. which you shouldn't. Hangup() does that |
| 10:34.18 | kaldemar | HangupCauseClear does exist. |
| 10:34.45 | wdoekes | it does? ok.. |
| 10:36.52 | kaldemar | it comes from func_hangupcause.c in asterisk 11. |
| 10:37.15 | wdoekes | I see. but it's irrelevant for the ${HANGUPCAUSE} var.. |
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| 10:38.29 | rox | i added the Hangupcauseclear call after a few hours of trying to figure out how to send the code over, the code fails to be delivered with or without the Hangupcauseclear call |
| 10:40.04 | rox | oh, whole debugging i also thought of the problem with the variable, so i tried to set the cause code numerically, i.e. Hangup(17), but it also didn't get delivered on the other side |
| 10:40.10 | kaldemar | asterisk -vvvr | grep dialednumber@context <-- makes getting output a little easier in a busy system. |
| 10:46.33 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 10:47.35 | wdoekes | .. and -C4 to get surrounding context |
| 10:51.27 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 10:51.42 | gavimobile | one of my peers doesn't get the clid info logged properly |
| 10:52.44 | gavimobile | the field for clid comes out like this "Iphone" <112> which is what is set in sip.conf however for 1 of my peers it comes out like this "110" <0000FFFF0000> and its not set like that in sip.conf |
| 10:53.03 | gavimobile | this is what's set in sip.conf Office Portable <110> |
| 10:53.14 | rox | kaldemar: http://pastebin.com/mtc8F4Xn there is the CLI output, there are two call IDs, 4036 for my original call, 4040 for when i get the call back from caller |
| 10:54.09 | rox | so in 4036 i get the call from a phone, then i call this other SIP trunk and get the call back from them in call ID 4040, then i reject that call and check for status back in 4036 |
| 10:54.29 | kaldemar | -- Executing Hangup("SIP/From1010-9e764230", "") <-- you're not passing any code. |
| 10:55.02 | rox | this is the final hangup |
| 10:55.32 | kaldemar | also, CLI output is better than snippets from logs. |
| 10:55.35 | rox | i don't know, why the hangup in 4040 is not recorded |
| 10:56.06 | kaldemar | not much use for that output then. |
| 10:56.18 | rox | in 4040 i only get this: == Spawn extension (from-detel, 980051258508, 4) exited non-zero on 'SIP/5060-9e744190' |
| 10:56.35 | rox | there should be a hangup there with code 17 |
| 10:58.30 | kaldemar | looks like your previously pasted extension is not even used. |
| 11:00.30 | wdoekes | kaldemar: I think he's showing us the other end :-/ |
| 11:00.32 | rox | kaldemar: http://pastebin.com/sntjn1Lj there, another try, this time the hangup with code 17 is clearly recorded |
| 11:00.34 | gavimobile | help |
| 11:00.44 | rox | and the overriden cause code on teh other side, which is 19 |
| 11:01.44 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 11:02.09 | rox | kaldemar: so this: -- Executing Hangup("SIP/5060-00f008c0", "17") in new stack |
| 11:02.10 | rox | should result in getting HANGUPCAUSE 17 , but i get this: DETEL Call ended with DS: CONGESTION HC: 19 |
| 11:06.14 | wdoekes | gavimobile: pastebins of sip.conf and dialplan and execution might help |
| 11:06.54 | kaldemar | rox: now you're just pasting stuff without any context whatsoever. what you see in your verbosity depends on when it is executed and where. |
| 11:07.35 | rox | kaldemar: it's OK i'll work around it, i'll just set some variable and be done with it |
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| 11:15.00 | hebber | exit |
| 11:15.09 | hebber | ops :) |
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| 11:23.41 | Greenlight | I'm having an issue with outbound callerid on an ISDN30 circuit. I've done a PRI trace via the Asterisk CLI and I can see the outbound CLI that I want to use in the SETUP packet, however when the call comes through it's always replaced with the main CLI for the circuit. Virgin Media have tested the line and assure me it's setup to allow presentation of any number. Any ideas what this could |
| 11:23.42 | Greenlight | be, or are there known issues? |
| 11:27.00 | kaldemar | your telco does that. they restrict what you can send. |
| 11:28.04 | Greenlight | We should have "Type 5 CLI" and be able to present any number we want, and they confirm that this is the case |
| 11:28.17 | kaldemar | if you see the caller id going out of your box in the signaling and it still is replaced, then your telco is to blame. |
| 11:29.10 | Greenlight | So, if when i enable "pri debug on span 1" and I see the outgoing CLI in there, I can be guarenteed that nothing else at the Asterisk/DAHDI/LibPRI side will alter or change that? |
| 11:30.53 | kaldemar | that is how it goes out. |
| 11:31.46 | Greenlight | Ok, thanks - guess back to shouting at Virgin Media :) |
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| 11:46.47 | BorjaGVO | Hi people. A simple question: What is the purpose of having to different log files (full & messages). Is it having different debugging levels? |
| 11:47.50 | kaldemar | BorjaGVO: yes. look at logger.conf and you'll see. |
| 11:48.59 | BorjaGVO | kaldemar: yep, just checking..It might had some hidden purpose that I was not seeing.. |
| 11:49.00 | BorjaGVO | thanks |
| 11:49.03 | kaldemar | but rather logging levels than debugging levels. |
| 11:51.05 | *** join/#asterisk af_ (~getsmart@78.134.0.72) |
| 11:51.29 | BorjaGVO | yeah, that is more precise. Few days ago I had a security issue and I found myself looking for information that it wasn't available. In order to have it, I enabled "sip set debug". I found this emasure as good enough...any advice about this? |
| 11:51.34 | BorjaGVO | (kaldemar) |
| 11:51.39 | BorjaGVO | (anyone) |
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| 12:10.43 | TobSnyder | I have a problem using SIP Trunk - an incoming sipgate call is disconnected after 10s - any idea why this can happen/has someone already had such a problem? |
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| 13:04.16 | powerunits | hello every one |
| 13:05.10 | powerunits | please can any one guide me. if we have panasonic PBX which has sip support . can we communicate it with asterisk? |
| 13:06.27 | powerunits | so asterisk SIP users can call any extension number on panasoic pbx |
| 13:07.15 | [TK]D-Fender | powerunits, If it speaks SIP ..... can you think of any reason why not? |
| 13:07.49 | powerunits | well i know it should .. but i have not tried it yet |
| 13:08.03 | powerunits | so i thought i should ask asterisk expert if they have done |
| 13:08.09 | powerunits | any sort of integration |
| 13:11.09 | *** part/#asterisk nunne_ (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 13:11.11 | [TK]D-Fender | powerunits, You should probably just go try. |
| 13:17.02 | powerunits | sure thanks |
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| 13:24.59 | Greenlight | When starting MixMonitor from the CLI (or via AMI "Command" action) is there a way to pass the options available normally (such as "a" for append) ? |
| 13:26.06 | Greenlight | As in> MixMonitor start SIP/4712-0000094b /test.wav |
| 13:26.14 | Greenlight | But make it append and not overwrite |
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| 13:44.17 | kaldemar | Greenlight: "core show help mixmonitor start" |
| 13:45.20 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 13:45.37 | Greenlight | Yea, but that doesn't show any place where the options can be specified, unless "args" means the filename AND any options? |
| 13:47.21 | kaldemar | filename.extension[,options[,command]] are all arguments. |
| 13:48.43 | Greenlight | Ahh cool, so I can just comma seperate the "args" there, exellent - thanks! |
| 13:55.37 | *** join/#asterisk serafie (~erin@nat/digium/x-izgvuhzopenvgmvl) |
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| 14:16.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:19.49 | Katty | hello my asterisk does not work at all how to fix plz??/ is urgen thx |
| 14:21.39 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 14:26.00 | Greenlight | Have you tried turning it on ? |
| 14:27.06 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 14:27.42 | chuckf | Katty: install it first, then try again |
| 14:29.51 | jacekowski | how can i tell which echo canceller i'm set to at the moment? |
| 14:31.34 | [TK]D-Fender | jacekowski, What card do you have? |
| 14:31.39 | Katty | hugs chuckf |
| 14:31.47 | [TK]D-Fender | Katty, Mew. |
| 14:32.12 | Katty | g'morning fender bender. |
| 14:33.22 | schmidts | good evening katty ;) |
| 14:33.56 | jacekowski | [TK]D-Fender: i've found it |
| 14:34.26 | jacekowski | <PROTECTED> |
| 14:34.44 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 14:35.16 | Katty | schmidts: ello. |
| 14:38.21 | chuckf | hugs Katty |
| 14:39.22 | chuckf | Katty: Are you all unpacked and settled in the new place? |
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| 14:41.21 | Katty | kind of. |
| 14:41.28 | Katty | we can talk about it in /query if you want (= |
| 14:41.36 | rue_house | wow this is a busy channel |
| 14:41.55 | rue_house | overflow my scrollback buffer in just one night |
| 14:42.37 | *** join/#asterisk bchia (~Adium@nat/digium/x-pdrbhnbjvxviiiby) |
| 14:42.59 | rue_house | I have a mgcp.conf problem, I cant specify more than one gateway, anyone know if this is a known issue? |
| 14:43.01 | chuckf | rue_house: you need a bigger buffer |
| 14:44.07 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 14:45.13 | jacekowski | i've just had asterisk "crash" again |
| 14:47.08 | jacekowski | it stopped handling calls |
| 14:47.32 | jacekowski | and core restart when convenient had done nothing |
| 14:47.38 | jacekowski | even though there was no calls in progress |
| 14:48.56 | [TK]D-Fender | rue_house, considered showing us what you're doing and what's actually happening? |
| 14:50.14 | rue_house | I'll post the mgcp.conf, either gateway will work alone, but if I specify two of them, only the last one works |
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| 15:01.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:02.20 | ipc9 | i recently updated to the newest asterisk and, in turn, had to update my chan_capi driver. Before when i restarted the asterisk service it would drop all active calls, but now when i restart the service it does not drop the calls and asterisk says there are no active calls. |
| 15:04.07 | *** join/#asterisk elico (~Thunderbi@109.66.109.73) |
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| 15:07.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:09.39 | rue_house | is there a virtual vu meter that could be added to a conference call to get an idea of realtime rtp audio levels? |
| 15:10.37 | rue_house | (STILL wrestling with audio levels on the polycom phones (my current oppinion is that polycom sucks)) |
| 15:10.44 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 15:11.06 | carrar | polycom phones work awesome |
| 15:11.29 | rue_house | no, they output no audio levels, the aastra we got work fine |
| 15:11.54 | carrar | I just use the default audio levels that are set in the example config from Polycom |
| 15:11.56 | rue_house | on the polycom phones nobody can hear anything, and there is NO tech support for the xml config |
| 15:11.58 | carrar | and they work fine |
| 15:12.06 | rue_house | not on our phones |
| 15:12.31 | rue_house | nobody can tell us what the ranges are for the XML audio levels |
| 15:12.40 | carrar | using the default XML configs from polycom? |
| 15:12.43 | rue_house | or how the different gains are configured |
| 15:12.47 | rue_house | right |
| 15:12.55 | carrar | did you read the admin guide? |
| 15:13.07 | rue_house | yep, says nothing about the audio gain settings |
| 15:13.41 | rue_house | and our vendor cant understand my question about the ranges for values for them, and they wont pass the question on to polycom |
| 15:14.03 | rue_house | (williams communications in canada) |
| 15:16.19 | rue_house | the only question about the polycome phones I think they would be able to answer is "what did I buy" |
| 15:16.28 | rue_house | and I'm sure their answer would not be "a brick" |
| 15:17.18 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:19.05 | *** join/#asterisk elico (~Thunderbi@109.64.211.97) |
| 15:20.02 | [TK]D-Fender | <rue_house> (STILL wrestling with audio levels on the polycom phones (my current oppinion is that polycom sucks)) <- it's always just you. Years and years of "just you". But sure, try to sell that to the rest of us :) |
| 15:24.03 | *** join/#asterisk pejman_ (~pejman@37.63.176.39) |
| 15:28.11 | *** join/#asterisk artistic (~pejman@37.63.176.39) |
| 15:29.19 | carrar | heh |
| 15:29.54 | artistic | HELP please, lsdahdi: FXO FXSKS (EC: OSLEC - INACTIVE) |
| 15:31.17 | artistic | why Inactive? how to active? I've got elastix, it says my FXO "Not detected by Asterisk" |
| 15:31.38 | [TK]D-Fender | artistic, that has nothing to do with your Elastic issue |
| 15:31.49 | [TK]D-Fender | x* |
| 15:32.11 | artistic | [TK]D-Fender: ok, I know, what should I do? |
| 15:32.47 | [TK]D-Fender | #elastix <- |
| 15:32.55 | [TK]D-Fender | For their GUI detection and setup bits. |
| 15:32.59 | [TK]D-Fender | It's not supported here. |
| 15:33.16 | [TK]D-Fender | Your device seems to be there. If their scripts are supposed to handle it then they will have to support thwm |
| 15:33.19 | [TK]D-Fender | them* |
| 15:34.06 | *** join/#asterisk luke0512 (~eric@HSI-KBW-091-089-021-231.hsi2.kabelbw.de) |
| 15:34.09 | *** join/#asterisk Cubber (~ronny@cpe-74-71-250-35.twcny.res.rr.com) |
| 15:34.16 | luke0512 | hello |
| 15:36.38 | luke0512 | i'm new to asterisk and am trying to make a connection to isdn net via my phonebox with internal S0 |
| 15:37.05 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:37.10 | *** join/#asterisk artistic (~pejman@37.63.176.39) |
| 15:37.21 | luke0512 | i'm using a pci card with cologne chip and trying to use the chan_capi module |
| 15:37.55 | luke0512 | make install runs fine but loaading the module results in error |
| 15:38.05 | luke0512 | WARNING[4776]: chan_capi.c:8286 cc_init_capi: CAPI not installed, chan_capi disabled! |
| 15:38.40 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:39.00 | luke0512 | someone out there with some hints? |
| 15:45.11 | *** join/#asterisk elico (~Thunderbi@109.67.200.232) |
| 15:46.19 | *** join/#asterisk sustav (~vpp@nat/digium/x-tghswqtzdhjqmchu) |
| 15:49.54 | *** join/#asterisk artistic (~root@208.117.34.136) |
| 15:51.25 | artistic | HELP please, lsdahdi: FXO FXSKS (EC: OSLEC - INACTIVE) |
| 15:51.28 | artistic | I got disconnected from channel, really sorry, did anyone answer? |
| 15:51.54 | [TK]D-Fender | <artistic> [TK]D-Fender: ok, I know, what should I do? |
| 15:51.54 | [TK]D-Fender | <[TK]D-Fender> #elastix <- |
| 15:51.54 | [TK]D-Fender | <[TK]D-Fender> For their GUI detection and setup bits. |
| 15:51.54 | [TK]D-Fender | <[TK]D-Fender> It's not supported here. |
| 15:52.04 | WIMPy | luke0512: I'd try to use LCR. |
| 15:54.09 | *** join/#asterisk VultureZ (~Chuck@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 15:55.15 | artistic | core show channels: 0 active channels I think the problem is with asterisk config |
| 15:58.01 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:58.01 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:58.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:58.30 | [TK]D-Fender | artistic, Elastix builds the configs. Not you. You need to go through their means for confuiguring it. |
| 15:58.37 | luke0512 | mmmh...i just read about chan_capi to work on isdn but chan_lcr i have not tried yet |
| 15:59.39 | WIMPy | I guess capi should work as well, but I never tried that. You could also use dahdi. |
| 16:02.02 | *** join/#asterisk blizzow1 (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 16:03.22 | luke0512 | i want to use the asterisk behind the phonebox, therefore it is connected via the internal S0 on my phonebox, on the phonebox there are 3 analog telephones and 1 anlalog fax connected |
| 16:04.04 | *** part/#asterisk hurdman (~ygcheny@r2d2.r0b0t.fr) |
| 16:04.13 | luke0512 | mmmh..so ihave to search for the other methods and give them a try |
| 16:08.34 | *** join/#asterisk elico (~Thunderbi@109.64.169.6) |
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| 16:08.51 | *** part/#asterisk nixofortune (~ishevtsov@cpc1-croy17-0-0-cust180.croy.cable.virginmedia.com) |
| 16:09.24 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 16:10.09 | luke0512 | some additional infos of my system http://fpaste.org/e0dv/ |
| 16:11.46 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:20.09 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
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| 16:42.45 | *** join/#asterisk mm_tcooper (62671948@gateway/web/freenode/ip.98.103.25.72) |
| 16:43.12 | mm_tcooper | ey all, quick question i am using OpenSIPs as a proxy between asterisk and our carrier level3, sip.conf is configure to send calls to the opensips IP, which then proxies the request to the level3 IP, everything works from a call prospective, but if I TCPdump on the asterisk box I see that we are sending BYE messages directly to level3 not to opensips to be forwarded. is this a configuration problem with opensips or asterisk? |
| 16:45.51 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
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| 16:52.34 | luke0512 | ok capi is now loaded with chan_capi the permissions on /dev/capi20 have been wrong |
| 16:53.17 | luke0512 | now i have to take a look at the exten file(s) |
| 16:58.11 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:00.11 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 17:12.16 | luke0512 | to early...there is still a prob the card is ignored |
| 17:18.31 | *** join/#asterisk ClintGoudie-Nice (~cgoudie@smtp.callware.com) |
| 17:21.52 | ClintGoudie-Nice | Hello. I've got a device that's attempting a switch to switch transfer. In the Refer-To line, it has what appears to be the correct info for the destination endpoint, but when the device sends it's refer, I'm getting a notify back from asterisk saying "481 Call leg/transaction does not exist" |
| 17:23.11 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 17:23.11 | ClintGoudie-Nice | How can I diagnose what is going wrong? |
| 17:23.41 | file | chan_sip requires all legs to be present within it, if this is not true then it won't work |
| 17:23.47 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 17:24.13 | ClintGoudie-Nice | file: so Asterisk cant handle a conceptual switch to switch transfer? |
| 17:24.33 | ClintGoudie-Nice | unless the second call leg is established through the asterisk. |
| 17:24.34 | oej | file: THat's not fully correct. chan_sip can send invite/replaces to another host during transfer. |
| 17:27.59 | file | it still has some knowledge about the dialogs |
| 17:31.48 | ClintGoudie-Nice | file: Thanks for that info. I will ensure the other call leg is established through the asterisk. |
| 17:31.55 | ClintGoudie-Nice | That info was invaluable |
| 17:32.18 | file | ClintGoudie-Nice, sorry for the inconvenience! |
| 17:45.17 | *** join/#asterisk k3asd` (~k3asd`@static-94-32-127-180.clienti.tiscali.it) |
| 17:45.42 | k3asd` | hi |
| 17:48.25 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:48.25 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:52.12 | ghost75 | exten => h,1,ExecIf($[ ${GROUP_COUNT(intern)} < 1 ]?Gosub(throttleoff,s,1)) |
| 17:52.16 | ghost75 | returns: -- Executing [h@phones998780:1] ExecIf("SIP/10-00000030", "0?Gosub(throttleoff,s,1)") in new stack |
| 17:52.47 | ghost75 | what this means: "0?Gosub(throttleoff,s,1)" |
| 17:54.52 | [TK]D-Fender | ghost75, It means it's not doing it. |
| 17:55.37 | ghost75 | so that group is >=1 at this point? |
| 17:56.31 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 17:58.19 | *** part/#asterisk JustinAiken (~JustinAik@justinaiken.com) |
| 17:58.48 | ghost75 | yes i just nooped it |
| 18:02.15 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 18:04.20 | Mon|A|rch | anyone worked with the asterisk sms feature? |
| 18:05.41 | leifmadsen | asterisk sms functionality is practically non-existant |
| 18:05.52 | leifmadsen | app_sms is tied to a pretty specific network |
| 18:09.26 | *** part/#asterisk dr0ck (~dr0ck@c-67-172-153-201.hsd1.co.comcast.net) |
| 18:09.58 | Mon|A|rch | bummer |
| 18:10.04 | Mon|A|rch | which network? |
| 18:10.21 | Mon|A|rch | if it's at all usable I'd like to explore it |
| 18:10.36 | Mon|A|rch | I'm willing to do email-relay shenanigans if i need to |
| 18:10.56 | _Corey_ | Mon|A|rch: Have you heard of Twilio? |
| 18:11.33 | ipc9 | i recently updated to the newest asterisk and, in turn, had to update my chan_capi driver. Before when i restarted the asterisk service it would drop all active calls, but now when i restart the service it does not drop the calls and asterisk says there are no active calls. |
| 18:11.47 | Mon|A|rch | i have not _Corey_ |
| 18:12.51 | luke0512 | mmmh seems not to work with chan_capi |
| 18:13.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:13.09 | _Corey_ | Mon|A|rch: Yeah, have a look there or Voxeo... I think you'll find it "shananigan" free |
| 18:13.36 | Mon|A|rch | well |
| 18:13.43 | Mon|A|rch | we used voxeo prophecy a while back |
| 18:14.08 | Mon|A|rch | it broke about three months into usage, and the expensive support they sold us never responded to calls or emails |
| 18:14.16 | Mon|A|rch | so I'm not keen on going the voxeo route |
| 18:14.22 | Mon|A|rch | I'll look at twilio though |
| 18:14.35 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 18:14.37 | luke0512 | service capi is running but when i type capiinfo the system tells me capi not installed??? |
| 18:16.41 | ghost75 | <PROTECTED> |
| 18:16.48 | *** join/#asterisk dr0ck (~dr0ck@c-67-172-153-201.hsd1.co.comcast.net) |
| 18:16.52 | ghost75 | under which conditions this error shows? |
| 18:16.58 | ghost75 | i get it from ami |
| 18:17.19 | pabelanger | ghost75: fix your script |
| 18:17.26 | ghost75 | disconnect without logoff? |
| 18:17.30 | pabelanger | you are closing the socket before asterisk is finished sending you info |
| 18:17.31 | ChannelZ | Your script terminated unexpectedly |
| 18:17.45 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:17.51 | ghost75 | the info i have received |
| 18:18.21 | ChannelZ | AMI sends things without being asked if you' |
| 18:18.43 | ghost75 | i tried also logoff before script end |
| 18:19.03 | luke0512 | now i get the Warning message again |
| 18:19.07 | ChannelZ | oops. .... if you're monitoring events, etc. So it could be a bad case of timing where you were done and logging off but it was spitting something out. More likely if the system is quite busy |
| 18:19.33 | ghost75 | i get it every time executing the script |
| 18:19.47 | ghost75 | lets say 95% of the time |
| 18:20.50 | ChannelZ | Do you attempt to clear the read buffers after you issue the logoff? |
| 18:21.17 | ghost75 | there is not much to control, its a perl module |
| 18:21.36 | ghost75 | i tried also to undefine the variable containing the ami connection |
| 18:22.31 | ChannelZ | well just ignore the error then if you can't fix the perl module to close cleanly |
| 18:22.47 | ChannelZ | it's not a catastrophic error |
| 18:22.53 | *** join/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 18:22.54 | ghost75 | hmm i remember something its possible to disable buffers in perl |
| 18:23.13 | TheKernel[work] | Hi, is dtmfmode=rfc2833 still valid in 10.4? |
| 18:23.30 | TheKernel[work] | 10.4.2 |
| 18:24.12 | ChannelZ | should be |
| 18:24.28 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 18:25.13 | ghost75 | not nice to show this as "error" |
| 18:25.54 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 18:25.58 | ChannelZ | Well it's not nice to terminate connections either but I think everyone will live. |
| 18:26.09 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 18:26.27 | raden | Katty, :) |
| 18:26.35 | raden | anyone use asterisk for SMS ? |
| 18:27.15 | ChannelZ | I got it working via XMPP through Vitelity but only to see if it worked |
| 18:29.18 | luke0512 | bye |
| 18:29.21 | *** part/#asterisk luke0512 (~eric@HSI-KBW-091-089-021-231.hsi2.kabelbw.de) |
| 18:31.41 | Mon|A|rch | what does the error "everyone is busy/congested at this time" mean exactly? |
| 18:31.56 | ChannelZ | Everyone is busy. |
| 18:32.11 | Mon|A|rch | lol |
| 18:32.11 | ChannelZ | Whatever device(s) or whatever you called said "no" |
| 18:32.25 | Mon|A|rch | okay, that's what i needed to know |
| 18:32.41 | ChannelZ | Sometimes it means "you dialed a number I have no idea what it means" and the remote end will return that. Sometimes it means "whatever you dialed is busy" |
| 18:32.43 | Mon|A|rch | it's not asterisk saying it can't start the call |
| 18:32.52 | Mon|A|rch | it's that the call wasn't picked up or was busy with another call |
| 18:33.01 | ChannelZ | Yeah, it's a response from the other end |
| 18:33.11 | Mon|A|rch | cool |
| 18:33.11 | Mon|A|rch | thanks |
| 18:34.11 | [TK]D-Fender | <Mon|A|rch> what does the error "everyone is busy/congested at this time" mean exactly? <- could mean nothing. You'd have to see the complete call to be sure. |
| 18:35.42 | Mon|A|rch | in this context, what ChannelZ said makes perfect sense |
| 18:36.15 | wltjr | is there a better way to ring lots of phones at the same time without having a bunch of &s in dial? |
| 18:37.34 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 18:37.39 | navaismo | maybe a Queue with ringall |
| 18:38.12 | *** join/#asterisk MarKsaitis (~MarKsaiti@82-71-61-117.dsl.in-addr.zen.co.uk) |
| 18:38.35 | ChannelZ | define "better" ? |
| 18:39.09 | [TK]D-Fender | <ChannelZ> Everyone is busy. <ChannelZ> Whatever device(s) or whatever you called said "no" <- not necessarily |
| 18:39.11 | [TK]D-Fender | Mon|A|rch, ^ |
| 18:39.30 | [TK]D-Fender | <Mon|A|rch> it's not asterisk saying it can't start the call <- it CAN be |
| 18:39.32 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 18:39.52 | Mon|A|rch | [TK]D-Fender, the situation is that we're calling multiple people simultaneously with that script i was working on yesterday |
| 18:40.02 | Mon|A|rch | so, they called my phone, and then someone else called my phone |
| 18:40.13 | [TK]D-Fender | Mon|A|rch, same answer.... |
| 18:40.14 | Mon|A|rch | i didn't pick up the second line, instead i cancelled it |
| 18:40.20 | [TK]D-Fender | Mon|A|rch, Still ahve to look at it in full |
| 18:40.28 | Mon|A|rch | fair enough |
| 18:42.21 | Mon|A|rch | http://pastebin.com/eTJwwvv7 |
| 18:42.27 | ghost75 | flushed it, still error |
| 18:43.09 | *** join/#asterisk adeel (~adeel@216.183.80.220) |
| 18:43.17 | [TK]D-Fender | Mon|A|rch, that means "SIP debug" <- |
| 18:44.26 | Mon|A|rch | are you saying i should paste the sip debug, or that's what that message is saying |
| 18:45.28 | wltjr | navaismo: I was thinking about something like that |
| 18:45.32 | ChannelZ | if you REALLY want to know, SIP debug.. but you've pretty much explained what happened. |
| 18:45.33 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 18:45.38 | ChannelZ | You rejected a call from your phone. |
| 18:46.21 | Mon|A|rch | I'll wait for the testers to tell me if horrible things happened |
| 18:46.24 | Mon|A|rch | then I'll know if there were issues |
| 18:46.35 | Mon|A|rch | it's great having other people to test your crap |
| 18:47.46 | *** join/#asterisk elico (~Thunderbi@bzq-109-66-88-247.red.bezeqint.net) |
| 18:53.02 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:53.25 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:04.08 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
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| 19:06.17 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 19:11.41 | Mon|A|rch | So, it appears that asterisk can only send SMS to landlines, and mobiles via bluetooth? |
| 19:12.02 | Qwell | from mobiles via bluetooth |
| 19:12.19 | Mon|A|rch | i see |
| 19:12.38 | Mon|A|rch | i guess i have to find somewhere else to route texts through |
| 19:16.42 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 19:17.02 | Mon|A|rch | anyone know of software that could do this for free? |
| 19:17.06 | navaismo | chan_mobile or chan_dongle |
| 19:17.08 | Mon|A|rch | or do i have to pay for something like twilio |
| 19:17.22 | Mon|A|rch | I'll check out chan_dongle |
| 19:17.23 | navaismo | in my country there is no sms for free |
| 19:17.36 | Mon|A|rch | chan_mobile doesn't support what i need |
| 19:17.47 | Mon|A|rch | i don't need it for free, we'll pay for the sms |
| 19:17.57 | navaismo | you need to pay service, but i was read alot of good things about twilio |
| 19:18.23 | Mon|A|rch | how much is the service? |
| 19:18.25 | navaismo | also a gsm gateway can help you |
| 19:19.18 | navaismo | http://www.twilio.com/sms/pricing |
| 19:20.13 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 19:20.27 | navaismo | In US is 1Cent for outbound in myne 8.3Cents |
| 19:24.26 | rogers- | We just got a new cable modem from Time Warner after an electrical surge took out the old one. Now we are having sporadic service with outr external SIP provider. All outbound calls seem to work, however inbound work sporadically. Does this sound like a double NAT issue or Sip ALG? I called TW and they said the new cable router is in bridge mode, but I have no way to confirm. |
| 19:24.33 | ghost75 | OT: is there 4k tv also coming in US? |
| 19:24.53 | rogers- | When calls are not working, if I call outbound, then try to call inbound, it works. Like a channel has been opened and it works until it times out |
| 19:25.42 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 19:28.43 | *** join/#asterisk Fastidius (~quark@unaffiliated/quark) |
| 19:30.19 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 19:40.50 | timholum | Has anyone used Activa for asterisk before? ( An asterisk tapi interface ) |
| 19:42.05 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 19:42.48 | navaismo | rogers-, you need a sip debug or cli output |
| 19:42.58 | _Corey_ | timholum: I thought that project died years ago |
| 19:44.02 | rogers- | navaismo, what should I look for in the debug |
| 19:51.01 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 19:52.57 | navaismo | the call and messages baout it XD |
| 19:56.10 | *** join/#asterisk district (district@falseprophet.ca) |
| 20:13.03 | *** join/#asterisk omani (~omani@unaffiliated/omani) |
| 20:13.24 | omani | my menuselect.makeopts is not getting honoured when upgrading to current/latest lts release 18.19.1 |
| 20:13.30 | omani | is this a known bug? |
| 20:13.49 | Kobaz | what the heck is 18.19.1 |
| 20:14.04 | Kobaz | oh 1.8 |
| 20:14.44 | Kobaz | if you ./configure again then makeopts can get cleaned |
| 20:16.19 | ChannelZ | Some makeopts you put in your home directory or /etc/asterisk (I forget) don't get honored, I know that. |
| 20:16.23 | ChannelZ | Like sound package selections |
| 20:16.26 | *** join/#asterisk psilikon (~joel@mail.vicimarketing.com) |
| 20:16.44 | omani | Kobaz: my steps are: ./configure, cp oldversion/menuselect.makeopts ., make menuselect |
| 20:16.51 | Kobaz | ah |
| 20:16.53 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:17.04 | omani | I tried it with ~/.asterisk.makeopts |
| 20:17.12 | omani | after configure. |
| 20:17.22 | omani | but still make menuselect with default values |
| 20:17.23 | ChannelZ | yeah that's one. Perhaps some other things get ignored as well. |
| 20:17.41 | omani | I'll try .asterisk.makeopts in home prior to configure now. |
| 20:17.43 | ChannelZ | I know there's a bug about it somewhere |
| 20:17.48 | omani | then ./configure, then make menuslect |
| 20:17.52 | omani | okay |
| 20:17.56 | omani | good to know |
| 20:18.14 | omani | so what can I do? |
| 20:18.20 | omani | just copy old makeopts and run make instead? |
| 20:18.28 | omani | by skipping make menuselect? |
| 20:19.14 | ChannelZ | Well if you copied them over it should work |
| 20:19.50 | ChannelZ | It's just the integrating-makeopts-from-user-config-files bit that doesn't seem to work reliably. |
| 20:19.52 | ChannelZ | https://issues.asterisk.org/jira/browse/ASTERISK-18137 |
| 20:20.21 | omani | ok |
| 20:20.32 | omani | I'll just run make |
| 20:20.39 | ChannelZ | although this is a similar one I ran into: https://issues.asterisk.org/jira/browse/ASTERISK-11556 |
| 20:20.40 | omani | hope it works with custom makeopts copied in. |
| 20:20.53 | ChannelZ | Mostly I use defaults but always go in and deselect the sound and MOH packs |
| 20:21.35 | omani | ok |
| 20:21.43 | ChannelZ | it's annoying |
| 20:23.05 | Katty | waves at raden |
| 20:25.36 | omani | hmm doesn't work |
| 20:26.10 | omani | seems I have to copy both menuselect.makeopts and menuselect.makedeps to get a working "make" |
| 20:26.31 | omani | trying. |
| 20:27.38 | omani | no. didn't work |
| 20:27.51 | omani | man I gotta go through all values by hand. |
| 20:27.53 | omani | :/ |
| 20:35.50 | Qwell | glomps Katty |
| 20:46.46 | Merlin | anyone aware of a asterisk 1.2 backport of res_speech.so ? |
| 20:49.26 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 20:52.11 | [TK]D-Fender | That's not a backport. |
| 20:52.18 | [TK]D-Fender | that's a WAYWAYWAYWABACKport |
| 20:52.50 | [TK]D-Fender | Merlin, So why are you still running 1.2 at all? |
| 20:53.57 | Merlin | because i hate myself |
| 20:53.59 | Merlin | haha |
| 20:54.09 | Merlin | no, it's because fonality trixbox |
| 20:54.17 | Merlin | i'm working on someone else's system |
| 20:54.45 | Merlin | fonality hasn't provided an update past 1.2 for trixbox pro customers |
| 20:54.48 | Merlin | for no good reason |
| 20:55.48 | dr0ck | well they did say they added millions of lines of code, must just take a while to update it |
| 20:58.03 | *** part/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 20:59.21 | [TK]D-Fender | Merlin, Not having to care is a great reason. If they are not obligated to then... oh well |
| 20:59.44 | [TK]D-Fender | Merlin, Don't expect car parts for your model forever |
| 21:00.15 | [TK]D-Fender | Merlin, And they did provide an update... that involves a complete R&R |
| 21:00.47 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 21:01.25 | Qwell | [TK]D-Fender: at this point, I think maybe a DNR is in order |
| 21:02.39 | [TK]D-Fender | Qwell, Not hostile enough. More like "Terminate Immediately With Extreme Prejudice" |
| 21:09.04 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 21:13.39 | Merlin | haha |
| 21:14.18 | Merlin | so, i'm out of luck with res_speech, right? |
| 21:15.22 | [TK]D-Fender | Merlin, maybe you could find a coder you could pay enough for such a thing to be done for you.... |
| 21:15.33 | Qwell | or |
| 21:15.35 | Qwell | ~upgrade asterisk |
| 21:15.35 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 21:15.56 | [TK]D-Fender | 1.2 dead |
| 21:15.58 | [TK]D-Fender | 1.4 dea |
| 21:16.02 | [TK]D-Fender | 1.6.0 dead |
| 21:16.03 | Qwell | 10 dead |
| 21:16.08 | [TK]D-Fender | 1.6.1 dead |
| 21:16.15 | [TK]D-Fender | 1.6.2 dead |
| 21:16.26 | [TK]D-Fender | Qwell, Isn't 10 still technically getting bug fixes? |
| 21:16.33 | Qwell | [TK]D-Fender: security only |
| 21:16.38 | [TK]D-Fender | OH |
| 21:16.49 | [TK]D-Fender | 10 dead :) |
| 21:16.52 | Qwell | quite |
| 21:16.58 | [TK]D-Fender | Till when? |
| 21:17.05 | Qwell | it just moved |
| 21:17.07 | Qwell | so about a year |
| 21:18.03 | Qwell | well, I guess there's technically one release still pending |
| 21:22.22 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 21:32.17 | [TK]D-Fender | Checkout time, BBIAB |
| 21:35.15 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
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| 21:51.23 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 21:52.45 | fireman_biff | Some of the "channel" and "dstchannel" fields in my CDR have the format "DAHDI/i1/5551234-1234". What does the "i1" refer to? |
| 21:53.02 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 21:57.21 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 21:58.28 | feeshon | Currently have a conference bridge issue where very often the first person into the conf gets MOH and when the next person joins the first person is not in the conf |
| 21:59.02 | feeshon | Well it says that person is in the bridge but the MOH on hold is still given to the first caller |
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| 22:09.47 | *** join/#asterisk elico (~Thunderbi@109.66.88.247) |
| 22:10.37 | navaismo | and what show your cli? |
| 22:15.55 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
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| 22:25.12 | phunguy | i'm trying to call a certain international number, and i'm just getting a busy signals. the number matches my dialplan, but i'm not showing anything on my DID provider side for termination, so its something on my asterisk end. |
| 22:30.24 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 22:33.23 | raub | Stupid question: when I try to start asterix I get general protection error messages in the log file: http://pastie.org/private/sganbzhw0uizo1d4vx8g |
| 22:38.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:38.41 | navaismo | phunguy, cli output and sip debug if apply |
| 22:43.19 | phunguy | navaismo I am looking at my asterisk log now |
| 22:43.30 | phunguy | i'm just not sure how/why the call is being terminated |
| 22:44.27 | navaismo | you need to see your sip debug |
| 22:44.31 | navaismo | of that call |
| 22:45.20 | navaismo | if your call is rejected see if you have permission to dial international numbers, |
| 22:45.41 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 22:46.03 | gavimobile | can I use goto to goto a label rather than a context? |
| 22:46.05 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 22:46.11 | AkkerKid | heya all! |
| 22:46.31 | navaismo | gavimobile, labesl only works in the same context of your goto |
| 22:46.48 | phunguy | i know i have permissions |
| 22:46.48 | gavimobile | navaismo: thanks |
| 22:46.50 | phunguy | let me check a sip debug of the call |
| 22:47.24 | AkkerKid | so I imaged my asterisk box to new hardware and once it coots up, the audio is choppy. moving from core4quad to quad xeon and i don't think anything was combiled on the source machine. what could be the issue? |
| 22:48.16 | AkkerKid | boots, core2quad, compiled* |
| 22:48.28 | navaismo | AkkerKid, you mean like copy instead fresh install (compile) |
| 22:48.51 | AkkerKid | navaismo: I imaged the HDD using DD |
| 22:49.06 | AkkerKid | might as well have put my existing HDDs into the new box and turned it on |
| 22:49.50 | navaismo | eww my personal opinion is to make a fresh install and copy config files |
| 22:50.02 | AkkerKid | source box > dd > gzip > ftp > gzip > dd > new box |
| 22:50.16 | AkkerKid | unfortunately, reinstall is not an option |
| 22:51.09 | AkkerKid | i've gotta get this working as is. I would like to reinstall as well... |
| 22:51.23 | AkkerKid | but there's way too much customization on this box to start over. |
| 22:52.33 | AkkerKid | is there any reason why a more powerful machine would sound bad on identical software? |
| 22:57.15 | navaismo | nope |
| 22:59.57 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
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| 23:20.35 | timholum | Does anyone use activa tsp? |
| 23:20.56 | timholum | I am having issue's getting it installed on windows 7 without having ninja soft phone turned on |
| 23:21.49 | *** join/#asterisk CunningPike (~CunningPi@ip-64-134-160-143.public.wayport.net) |
| 23:36.58 | navaismo | maybe asking in their forum/maillist/developer |
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| 23:44.09 | ghost75 | http://resource.zayoenterprise.com/docs/phone-general/poe_chart.pdf more power than i thought |