| 00:08.38 | justdave | so CDR(userfield)=CHANNEL(peerip) probably, then |
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| 00:32.41 | Phlunk3 | hi guys, I am trying to increase the volume of incoming calls. I attempted by editing /etc/asterisk/extensions.conf and adding " exten => s,1,Set(VOLUME(TX)=10)" and " exten => s,n,Set(VOLUME(RX)=4)" to no effect. can anyone suggest other methods to try to achieve this? |
| 00:37.59 | WIMPy | I can't answer your question, but you should change volume where it's converted to analogue, i.e. on the phone. Doing it on Asterisk can give noticably worse quality. |
| 00:45.08 | *** part/#asterisk wooster (~dan@pozparty.accessaidsnetwork.com) |
| 00:53.30 | Phlunk3 | thanks for the input WIMPy, I don't suppose you can tell me what the s,n / s,1 means btw? |
| 00:54.46 | WIMPy | That's your extension/priority. |
| 00:55.04 | Phlunk3 | thanks |
| 00:55.06 | WIMPy | And if you don't know that, they are probably wrong. |
| 00:55.23 | WIMPy | ~book |
| 00:55.23 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:55.32 | Phlunk3 | yeah haha for sure, I will go do some reading I guess ;) |
| 00:55.33 | WIMPy | Check the dialplan basics. |
| 00:55.40 | Phlunk3 | tx mate |
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| 04:02.55 | maximko | Hello guys. I'm trying to originate call from dialplan: Originate(Local/1000@testcontext,exten,testcontext,0000) but getting warning "No path to translate from SIP/phone-0000003f to Local/1002@testcontext-00000007;2". If I specify SIP/phone instead of Local/1000@testcontext as Originate's tech_data parameter, it works normally. How I can fix it? 1002 extension is Dial(SIP/phone). Asterisk 12.1.1 |
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| 05:41.49 | maximko | CLI originate (originate Local/1002@testcontext extension 0000@testcontext) works well. |
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| 06:09.45 | snadge | permit command.. format is x.x.x.x/255.255.255.0 not /24 right ? |
| 06:10.03 | snadge | when it possibly should support either.. to save confusion.. but whatever ;) |
| 06:13.40 | kaldemar | CIDR notation should also work. |
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| 06:32.55 | snadge | apparently in some situations .. it may not.. but I haven't gone as to research that.. basically.. the wisdom seems to be to use netmask rather than CIDR |
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| 09:01.59 | sarthor | HI, What device I have to use that Can Get normal land line telephone calls, and send to my asterisk computer, also if my asterisk clients want to make call to any land line phone here in the country I want to make them able to use that route? is it possible by linksys spa400 or spa3000 devices? |
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| 09:30.54 | shamelessn00b | Hi guys, can anyone help me out with a SIP config |
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| 09:36.01 | shamelessn00b | http://pastebin.com/TAuVsXh7 |
| 09:36.38 | shamelessn00b | when I send an outbound call from the console like channel originate sip/mtnrw/07883XXXXX application wait 30 |
| 09:36.47 | shamelessn00b | in the From header its sending this |
| 09:37.26 | shamelessn00b | From: "Anonymous" <sip:+2502800XXXXX@anonymous.invalid>;tag=as388f4946 |
| 09:40.58 | freddo | hi, I just took delivery of a Digium 1A4A04F with HW Echo Can :) - awesome. First question though - does it work well in a VM (KVM) or am i better to run it on the bare metal? |
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| 10:02.15 | tparcina | sarthor: You need a SIP provider, or a device that has FXO port. I'm not familiar with Linksys devices. |
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| 10:05.35 | sarthor | tparcina: yea that linksys device have 1 fxo and 1 pxs and 1 ethernet port. |
| 10:06.19 | tparcina | sarthor: Then you should be fine with it. |
| 10:06.36 | sarthor | tparcina: OK. thanks |
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| 10:28.03 | maximko | snadge, no, don't have permit command. |
| 11:05.38 | shamelessn00b | http://pastebin.com/9Cy8rQUd |
| 11:05.46 | shamelessn00b | can anyone please help me sort out this issue |
| 11:06.09 | shamelessn00b | I registered asterisk as a UA and I created a peer definition |
| 11:06.17 | shamelessn00b | in sip.conf |
| 11:06.22 | shamelessn00b | it shows the peer registered |
| 11:06.31 | shamelessn00b | when I send out a call it says chanunavail |
| 11:07.02 | WIMPy | You need to enable sip debug in order to see what happened. |
| 11:08.12 | shamelessn00b | lemme pastebin the sip debug |
| 11:08.36 | WIMPy | You can likely read that yourself. |
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| 11:17.41 | shamelessn00b | I've been trying to |
| 11:17.52 | shamelessn00b | I was running a tcpdump side by side |
| 11:17.55 | shamelessn00b | http://pastebin.com/fCArcXz5 |
| 11:18.31 | andi | Hi |
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| 11:18.53 | shamelessn00b | SIP/2.0 488 Not Acceptable Here this is the only interesting bit I found |
| 11:19.09 | WIMPy | SIP/2.0 488 Not Acceptable Here |
| 11:19.24 | WIMPy | Yes, unfortunatly they are not very helpfull. |
| 11:19.51 | andi | I have an asterisk setup based on centos 6. I'm using sip trunking with sipgate and I have a snom 370 which is registered as client 6001 I'm trying to redirect a call from the trunk to SIP/6001 but I get Unable to create channel of type 'SIP' (cause 20 - Subscriber absent). |
| 11:20.02 | andi | Could the problem be that my phone is behind a nat? |
| 11:20.21 | shamelessn00b | so what now lol |
| 11:20.26 | andi | I googled a bit and tried to set nat=yes and qualify=no for the client 6001. |
| 11:21.22 | WIMPy | shamelessn00b: You cann make wild guesses or ask them if they have any better ideas. |
| 11:21.40 | StaRetji | TK D-Fender gave me tip on how to prevent IVR with AMD, but I don't see option to prevent this call from answering at all. |
| 11:21.49 | Neoti | Would anyone be able to point me in the right direction for help on astmanproxy, specifically filtering only 1 extension to 1 user account in the astmanproxy.users file. or to only have 1 context shown to a user in the astmanproxy.users file? |
| 11:21.52 | WIMPy | andi: Subscriber absent sounds like the phone isn't even registered. |
| 11:22.06 | StaRetji | I need option to prevent if alerting 0 to 2 seconds and calls is answered |
| 11:22.18 | andi | WIMPy: How can I check if it is? |
| 11:22.35 | StaRetji | or 0 to 1 seconds, ,cause ivr machine answers at once |
| 11:22.45 | WIMPy | andi: Sip show peers |
| 11:23.38 | shamelessn00b | they say its working when they register from a sip client |
| 11:24.10 | shamelessn00b | on eyebeam client |
| 11:24.14 | WIMPy | shamelessn00b: Not helpfull, either. |
| 11:24.22 | shamelessn00b | I know right |
| 11:24.34 | shamelessn00b | I'll ask what codecs they support |
| 11:24.46 | shamelessn00b | right now that guy is away |
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| 11:31.30 | andi | Hmm, strange, after a reload the client is showing up as unmonitored and asterisk can dial to SIP/6001 but then gets a Retransmission timeout reached on transmission ... seqno 102 error. |
| 11:31.47 | andi | The old sup show peers lines are scrolled out I cannot find them anymore. |
| 11:32.21 | andi | I think the problem is my setup. I have an asterisk server with a public IP and my snom 370 is behind nat. |
| 11:32.21 | WIMPy | So now you know you have a network issue. |
| 11:33.00 | WIMPy | Qualify can help to keep the NAT port open. But it's more efficient to enable keep-alives on the phone. |
| 11:33.28 | andi | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions I think number 9 is my case. |
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| 11:36.23 | andi | Ah, here we go. Now it's ringing, perfect. |
| 11:37.17 | andi | Is there an easy eay of centralizing the main configuration settings of the clients? |
| 11:37.40 | WIMPy | What does that mean? |
| 11:40.05 | andi | Let's say I have multiple of the snom 370 phones and I would like to centralize their basic configuration settings like this keep-alive setting to deploy this configuration to all phones, is there a way to do that easily? |
| 11:40.34 | andi | I'm using puppet for my linux configuration which works great and i wondered if there's something like that for phones, too. |
| 11:40.55 | WIMPy | I'm still not sure about "centralized". Do you want something like the teplates you can use for multiple peers? |
| 11:41.48 | WIMPy | Or do you want some GUI? |
| 11:42.17 | andi | I don't know if this question is still asterisk related. It's rather a question about configuring the phones, give them the correct identities and so on. |
| 11:42.23 | andi | No, a gui is not needed. |
| 11:42.55 | WIMPy | You can provision them. See the Snom wiki for details about that. |
| 11:43.55 | andi | Ah, cool, provisioning was the buzzword, thanks. :) |
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| 12:00.33 | tparcina | How to setup daylight time saving on Grandstream GXP1450, if the change date is last sunday in March? |
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| 12:44.45 | tparcina | Can I register and edit this manual? - http://www.bacula.org/5.2.x-manuals/en/main/main/Installing_Bacula.html |
| 12:49.10 | [TK]D-Fender | Think you're in the wrong channel for that... |
| 12:52.26 | StaRetji | [TK]D-Fender: thank you for yesterday's info on AMD |
| 12:52.35 | StaRetji | I appreciate it |
| 12:53.20 | StaRetji | unfortunately, it seems it can't drop calls if alerting time is 0 |
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| 12:55.19 | [TK]D-Fender | StaRetji: "core show application waitforsilence" |
| 12:55.41 | StaRetji | wow, thanks dude |
| 12:55.47 | StaRetji | will check it out |
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| 13:00.47 | tparcina | [TK]D-Fender: Yes, I see. Sorry. :) |
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| 13:51.06 | alami | hello, i want to create a conference bridge, i have not found a lot of documentation about confbridge and i found it difficult so i decide to use meetme application |
| 13:51.31 | alami | i'm using asterisk 11.8.1 and meetme require dahdi wich i can't install |
| 13:51.33 | alami | http://pastebin.com/P4nf8kD6 |
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| 13:52.12 | [TK]D-Fender | alami: You didn't install the pre-requisites for it |
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| 13:56.15 | vassilux | Hi all, is there a possibility re restart a channel for a incoming call when chan_dahdi send Cause is 44 ? |
| 13:57.39 | vassilux | sorry just restart :-) |
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| 14:11.36 | WIMPy | Nice one. Don't know what's causeing this, but when that box is restarted, IAX registrations won't work. |
| 14:12.24 | WIMPy | 'iax2 show registry' shows the correct IPs with port 0. And tcpdump shows that it sends to port 4520. |
| 14:14.02 | WIMPy | When I do 'iax2 reload' it changes to port 4569 as expected. |
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| 14:19.32 | shamelessn00b | hey WIMPy it was a codec negotiation issue |
| 14:19.45 | shamelessn00b | I removed the disallow and allow rules and it started working |
| 14:19.46 | shamelessn00b | weird |
| 14:20.41 | WIMPy | perfectely normal. |
| 14:23.23 | alami | i have install dahdi-linux package, and i still can't recompile asterisk 11.8.1 with app_meetme |
| 14:24.10 | WIMPy | alami: That's ok. Confbride is the way to go anyway. |
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| 14:31.41 | ipengineer | Does anyone know of a way to write to a custom CDR field once the call has completed? I am basically trying to track the time a call is monitored by taking the billsec at the beginning and subtract them at the completion of the call.. Here is my example: https://gist.github.com/zconkle/30733dbecb39e03df31b |
| 14:34.22 | WIMPy | There is a config option. Either you can write to CDR in the h extension _or_ you can read all values, IIRC. Not sure what exactely would be missing. |
| 14:35.03 | pabelanger | vassilux, you want to reset a specific channel in dahdi? |
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| 14:35.34 | ipengineer | WIMPy: Right.. So I have no problems writing to that field.. I am just wanting to make sure that the values dont get written until the end of the call.. |
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| 14:57.58 | lewzer | hey guys - I'm thinking of putting in an elastix server for my phones. how do i interface them with my server? |
| 14:58.04 | lewzer | just voip phones? |
| 14:58.24 | WIMPy | ~elastix |
| 14:58.24 | infobot | elastix is, like, a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 14:58.36 | WIMPy | And Asterisk can interface to all kinds of phones. |
| 14:59.00 | lewzer | sorry im looking at both |
| 14:59.42 | lewzer | Asterisk how do i interface with the voip lines coming into my office, i have 5 - I'm guessing a modem wont do it eh? |
| 15:02.38 | WIMPy | VOIP has some sort of accounts. PSTN has lines. What is it you have? |
| 15:02.53 | WIMPy | Oh, abd both come in very different flavours. |
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| 15:06.59 | lewzer | well we've had 2 voip lines from our provider, we just got another 2 (so 4 lines in total /w fax) so we have phones with 2 lines and I'm not really sure how to move forward |
| 15:07.39 | lewzer | im feeling that a pbx / self hosted solution internally is better then say an older phone system like avaya nortel kinda thing |
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| 15:10.49 | [TK]D-Fender | Avay & Nortel... make PBX's |
| 15:11.45 | [TK]D-Fender | [10:57]lewzerhey guys - I'm thinking of putting in an elastix server for my phones. how do i interface them with my server? <- what kind of phones? |
| 15:13.48 | lewzer | ok - wrong terminolgy |
| 15:13.52 | drmessano | Can anyone tell me what would cause a mismatch in media and signalling paths between devices... Like out of the blue, calls for 102 start ringing at 104, 116 calls 108 and the audio from the call is heard on another call between two other extensions, etc |
| 15:14.00 | lewzer | but i have lots of systems i could configure a linux ditro on |
| 15:14.02 | drmessano | It's almost like there's an ARP table out of whack or something |
| 15:14.26 | drmessano | Restarted the Asterisk instance and it persists |
| 15:14.30 | lewzer | im not confused as how i took that box to the phone lines |
| 15:14.54 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 15:15.31 | WIMPy | drmessano: Interesting. I'm about to see a box where users report that when picking up a 2nd call, they both hear the 1st call. |
| 15:15.56 | drmessano | WIMPy, any idea which * version is on the box? |
| 15:16.19 | WIMPy | 11.8.1 |
| 15:16.29 | drmessano | Interesting |
| 15:16.31 | drmessano | 11.5.1 here |
| 15:16.45 | drmessano | I was wondering if it was an issue with 11.5.1 and was thinking about updating |
| 15:16.50 | drmessano | But maybe not |
| 15:17.02 | drmessano | Seems to occur after about 30 days |
| 15:17.12 | WIMPy | I haven't heard about anything the like so far. |
| 15:17.20 | [TK]D-Fender | [11:14]lewzerim not confused as how i took that box to the phone lines <- FIRST YOU SAID PHONES... THEN YOU SAID LINES. pLEASE START OVER AND BE CLEAR ABOUT PRECISELY WHAT IT IS YOU'D LIKE TO INTERFACE WITH. |
| 15:17.28 | [TK]D-Fender | darn caps.... |
| 15:17.48 | drmessano | WIMPy, I think one of my scenarios is identical to the report you have. |
| 15:18.29 | WIMPy | I have absolutely no idea what might be going on there. |
| 15:18.53 | WIMPy | But I will go there to try to find out in a few minutes. |
| 15:19.16 | drmessano | If you find anything interesting, please pass it along |
| 15:19.26 | WIMPy | sure |
| 15:20.02 | drmessano | I rebooted the switches here just in case |
| 15:20.29 | drmessano | But after restarting the phones and Asterisk, it continued. Kinda feel like that ruled out most of the "unstable" bits |
| 15:21.09 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.34) |
| 15:29.36 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:29.54 | ipengineer | After a little more playing I think I have narrowed down my problem.. What is the most effective way to write something to the CDR at call completion? It has to be at the end because it is {billsec} - {somevalue} |
| 15:32.34 | *** part/#asterisk lewzer (~lewzer@s72-38-118-62.static.comm.cgocable.net) |
| 15:40.04 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 15:40.17 | gusto | hi WIMPy |
| 15:40.21 | gusto | WIMPy: how are you? |
| 15:40.32 | gusto | Penguin: hi, what's up? are your mirrors OK? |
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| 16:20.48 | *** join/#asterisk mcrownover (~mcrownove@remote.gawest.com) |
| 16:21.10 | mcrownover | hi everyone |
| 16:25.51 | mcrownover | i am about to upgrade from 1.6 to 1.8 and I am running the g.729 codecs - will I have to contact digium to get them reauthorized or should they not be affected when I upgrade? no hardware will be changed |
| 16:26.49 | [TK]D-Fender | You'll need the new module for your * version, but your licen file should still work |
| 16:28.40 | mcrownover | [TK]D-Fender - thanks! i'll dig that up too |
| 16:29.32 | StaRetji | [TK]D-Fender: I am sorry man, I've read WaitForSilence but I can't understand how it works. - WaitForSilence(500|2) will wait for 1/2 second of silence, twice...I Don't undertand how to use it to make hangup if no ring, if answer at once |
| 16:30.49 | [TK]D-Fender | StaRetji: You should also clarify your needs as far as "answer at once" is concerned. |
| 16:31.57 | StaRetji | well, I have problem that sim card is out of credit sometimes and in that case call that is made is being answered without ring, by operator ivr, saying no enough credit |
| 16:32.11 | StaRetji | I need to hangup(34) and failover that call |
| 16:32.28 | StaRetji | so, this call is called and alerting time is 0 |
| 16:32.40 | [TK]D-Fender | You could also be calling a number that is on a digital circuit and answers immediately as well... |
| 16:32.53 | [TK]D-Fender | This sounds like you're risking losing possibly legimate calls |
| 16:33.12 | StaRetji | hm, real person will have to hear at least half a ring , right? |
| 16:33.12 | [TK]D-Fender | If you don't care about that... just check the DIAL TIME in a Macro/Gosub |
| 16:33.27 | StaRetji | this is: press call, answered next moment |
| 16:33.30 | [TK]D-Fender | [12:33]StaRetjihm, real person will have to hear at least half a ring , right? <- Not necessarily |
| 16:33.49 | [TK]D-Fender | I call from my PRI here and to another digital subscriber with an IVR and it grabs instantly, no ring |
| 16:34.08 | StaRetji | got you |
| 16:34.21 | [TK]D-Fender | [12:33][TK]D-FenderIf you don't care about that... just check the DIAL TIME in a Macro/Gosub <- I'd add some protection logic to this as well |
| 16:34.23 | StaRetji | well, I want any ivr / instant answer to be dropped |
| 16:34.47 | [TK]D-Fender | IVR is a legit call... |
| 16:34.51 | StaRetji | okay, thank you once again |
| 16:35.03 | StaRetji | well, I will failover it to another trunk |
| 16:35.08 | StaRetji | so, no harm done |
| 16:35.46 | *** join/#asterisk Q_Continuum (~qlaras@166.70.234.238) |
| 16:36.12 | Q_Continuum | asterisk on CentOS 6 64-bit VM sizing: office of less than 25, 2C/1GB/20GB should be plenty? |
| 16:36.31 | Q_Continuum | (haven't worked with it, got a request for a VM for someone to set it up on and they have no clue) |
| 16:36.44 | [TK]D-Fender | sure. Make sure it has timing priority though. |
| 16:37.28 | Q_Continuum | Thanks! Will do, the host there isn't loaded at all so shouldn't be a problem, but I'll toss a note in there for it. |
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| 17:20.49 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-gvizaoibzyiagair) |
| 17:20.50 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 17:35.10 | StaRetji | [TK]D-Fender: I read gosub and cmd dial, but seems my brain capacity is simple not enough. I am willing to pay for extension lines which will hangup calls with 34 if alerting time is 0. Thanks |
| 17:39.51 | navaismo | newtonr, I have posted my confs and debugs to the ASTERISK-23425 Issue to prove its working |
| 17:41.03 | newtonr | navaismo, thanks a ton, i'll check them out when I get chance. Maybe there is something slightly different about the other reporters scenario. |
| 17:42.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:44.28 | navaismo | updated seems missconfiguration from the reporter |
| 17:55.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
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| 18:20.05 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 18:20.18 | Katty | good afternoon |
| 18:28.10 | navaismo | o/ |
| 18:28.27 | MaliutaLap | Ms. Katty |
| 18:28.56 | Katty | hugs navaismo and MaliutaLap |
| 18:29.27 | Chainsaw | Evening Katty :) |
| 18:29.29 | MaliutaLap | Katty: you have squirrel conspiracy people over here ... what have you been doing? Spreading lies about non-existant animal species all over globe! |
| 18:29.48 | MaliutaLap | Katty: had a huge thing about it on a friends FB |
| 18:29.48 | Katty | ^_- |
| 18:30.10 | Katty | Chainsaw: good evening, mister Chainsaw! |
| 18:30.22 | MaliutaLap | I still deny the existence of squirrels until I see one IRL |
| 18:30.29 | Katty | pats MaliutaLap |
| 18:30.34 | Katty | and right for you to do so! |
| 18:32.14 | MaliutaLap | Katty: you started it. With your "squirrel" feeding thing - it's now a global conspiracy :) |
| 18:33.05 | *** join/#asterisk willwh (~willwh@unaffiliated/willskills) |
| 18:33.12 | willwh | hey guys :) |
| 18:33.15 | willwh | moy: you around? |
| 18:33.25 | willwh | I was just trying out your team/moy/webrtc-12 branch |
| 18:33.37 | willwh | I compiled pjproject from source with asterisk 12.1.1 and it worked np |
| 18:33.58 | MaliutaLap | great, go to sign up to bands mailing list - yet another one using some service that can't handle greylisting, and doesn't have SPF records that are usable |
| 18:34.08 | willwh | doesn't seem to with your branch, I don't have working audio in 12, with sipml5 as a client, without res_pjsip stuff enabled |
| 18:34.12 | willwh | moy: any ideas? :) |
| 18:34.45 | willwh | I was hoping this would give me working wss:// :) |
| 18:34.54 | willwh | without the need for webrct2sip |
| 18:35.09 | MaliutaLap | swear I'm going to track down all these people with malfeasant SMTP servers and remove certain parts of their bodies before I kill them |
| 18:35.29 | willwh | well, that's not very friendly |
| 18:35.29 | willwh | :) |
| 18:36.11 | MaliutaLap | I can be like that when I have to deal with retards ... also known as "people" (not indiciduals) or "constituents" |
| 18:36.40 | MaliutaLap | s/indiciduals/individuals/ |
| 18:38.19 | MaliutaLap | hmm, 4 hours to wind down from attending a gig - must be a new record for me. Especially since I only had 4 drinks all night |
| 18:47.46 | *** join/#asterisk CoreyIrwin (~coreyirwi@69-165-255-164.cable.teksavvy.com) |
| 18:49.18 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 18:51.18 | boom^time | Hey guys. I have an asterisk server sitting behind NAT with a port range of 12000-13000 UDP forwarded to it. I have the range in rtp.conf changed to reflect it. I have directmedia=no for my ITSP and my externalip set appropriately. For some reason the call will terminate but I hear no audio. With rtp debug on I see that my server is sending packets but not receiving any. I don't hear audio in either direction. Does |
| 18:51.18 | boom^time | <PROTECTED> |
| 18:52.50 | WIMPy | drmessano: I wasn't able to reproduce the issue. |
| 18:53.48 | boom^time | Here is a pastebin of a call with sip debug on in case anyone is interested. |
| 18:53.49 | boom^time | http://pastebin.com/FNc6NWAT |
| 18:54.26 | drmessano | WIMPy, no issues since I rebooted my switches. I am thinking I have a new culprit |
| 18:55.02 | WIMPy | That's quite an andvanced feature for a switch. |
| 18:55.33 | drmessano | Corrupt ARP table? |
| 18:56.09 | WIMPy | That would only make the packets get lost. |
| 18:56.35 | drmessano | I'm guessing at this point.... Restarted * and the phones independant of the switch and it still acted up. Rebooted the switches and all is quiet now |
| 18:57.00 | drmessano | Seems like when this occured 60 days ago or so, same steps |
| 18:57.08 | drmessano | Switch rebooting ultimately resolved it |
| 18:57.18 | drmessano | WHY? I dunno |
| 18:57.45 | WIMPy | Well, Asterisk isn't the only thing that fails. |
| 18:58.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:58.13 | drmessano | I'm sure the switches are far behind on firmware updates. Maybe a bug. Who knows |
| 18:58.26 | drmessano | Right now I dont think I can get into them |
| 18:58.33 | WIMPy | Must be a very interesting bug. |
| 18:58.44 | drmessano | Again, I am guessing |
| 18:59.12 | drmessano | Nothing to go on except the result of specific steps |
| 18:59.31 | WIMPy | Looks like theres soe peer matching hiccup in iax. *sometimes* calls go the the context defined for guest calls, but still show with the correct peer name in 'core show channels'. |
| 19:10.40 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 19:11.00 | ipengineer | Does hangup_handler_push no longer have access to the CDR(billsec) variable? |
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| 19:40.17 | davlefou | hi, do you know if there problem with asterisk 1.8 and Cirpack/v4.56? i have that message : SIP/2.0 401 Unauthorized.. |
| 19:40.50 | [TK]D-Fender | That reallys says nothing |
| 19:41.04 | [TK]D-Fender | that is a standard challenge message, not just a final refusal |
| 19:41.16 | [TK]D-Fender | ERVERY device I have gets a 401 FIRST as a challenge. |
| 19:41.30 | [TK]D-Fender | And then... if your fails... that's because you'd have an auth mismatch in the end |
| 19:41.42 | [TK]D-Fender | it isn't the product that is bad... it's the configuration if that is the case |
| 19:41.43 | davlefou | ok |
| 19:42.45 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 19:43.16 | davlefou | i was never authorized. I don't undestand why it don't work with my asterisk. |
| 19:43.27 | davlefou | How can i find the problem? |
| 19:44.12 | [TK]D-Fender | look at the complete debug |
| 19:47.18 | newtonr | davlefou, a complete debug = https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:47.45 | newtonr | more specifically the "sip set debug on", but the other things are often helpful as well |
| 19:47.54 | newtonr | davlefou, you'll also want to read up on SIP messaging |
| 19:48.01 | davlefou | http://pastebin.com/FJCDm9cV |
| 19:48.09 | davlefou | from ngrep on 5060 |
| 19:49.41 | [TK]D-Fender | davlefou: Aterisk SIP debug from CLI... not 3rd party |
| 19:51.54 | davlefou | yes, i do it |
| 19:54.23 | davlefou | http://pastebin.com/7czkLwbf |
| 19:55.27 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 19:56.01 | [TK]D-Fender | Your auth is clearly no good |
| 19:57.08 | davlefou | yes, but why? the user work on over asterisk. So what is bad with this authentification? |
| 19:58.19 | davlefou | the id and passwd works. Under my asterisk 1.4, all is ok. |
| 19:59.03 | davlefou | I test serval asterisk with vm and it works, only this one have problem. |
| 19:59.37 | davlefou | In vm i put exactly the same linux and asterisk, it work! |
| 20:05.30 | newtonr | davlefou, it doesn't look like your asterisk system is even trying to authenticate, asterisk isn't sending the far end a REGISTER with Authorization header in response. post your sip.conf with the passwords and account names scrubbed |
| 20:05.40 | newtonr | davlefou, pastebin the sip.conf |
| 20:05.50 | *** join/#asterisk dirt_diver__ (sid25018@gateway/web/irccloud.com/x-vrbdgqslsmzhtyil) |
| 20:08.43 | davlefou | http://pastebin.com/y8ScnuQR |
| 20:08.47 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 20:10.58 | willwh | newtonr: I meant to bug you about this: https://reviewboard.asterisk.org/r/3248/ |
| 20:11.02 | willwh | I wondered if you'd had a crack |
| 20:11.12 | willwh | I have sipml5 working fine with asterisk 12.1.1, but only using ws:// |
| 20:11.44 | willwh | when I compile moy's /team/moy/webrtc-12 branch, I get no res_pjsip modules in makeselect |
| 20:11.58 | willwh | and that's the only way I've been able to get working voice with sipml5 |
| 20:12.14 | willwh | newtonr: maybe ping me if you want to blab about it :D |
| 20:14.26 | davlefou | i have purge my asterisk et reinstall with orignal file. But no result. |
| 20:15.06 | newtonr | willwh, lol, "if you'd had a crack" |
| 20:15.59 | willwh | :D |
| 20:16.02 | willwh | I'm a Scot |
| 20:16.09 | willwh | sorry for the colloquialisms |
| 20:16.33 | newtonr | willwh, I figured what you meant, it just sounded funny when I interpreted it in american english |
| 20:16.51 | newtonr | willwh, you are using chan_sip right? |
| 20:16.58 | willwh | s/have a crack at it/try it |
| 20:17.03 | newtonr | right :D |
| 20:17.26 | newtonr | i did get a chance to lab it up again with chan_pjsip, but ran into a crash and another issue |
| 20:17.30 | willwh | well, I think I was using res_pjsip with 12.1.1 |
| 20:17.49 | willwh | I'd have to go and reconfigure and makeselect to see :P |
| 20:17.52 | willwh | as I don't remember |
| 20:18.04 | willwh | I'm running webrtc-12 from moy atm |
| 20:18.09 | newtonr | willwh, I filed ASTERISK-23572 after I tried it with pjsip |
| 20:19.03 | newtonr | willwh, otherwise, calling from a ws endpoint to a udp endpoint worked occasionally, but calling the other way around didn't work at all for me |
| 20:19.31 | willwh | oh, yeah, it works great for me with my 11.6-cert build ;) |
| 20:19.38 | willwh | but, honestly, to use it in prod, I want wss:// |
| 20:20.07 | willwh | and I don't really want to compile webrtc2sip, mainly because that's a hell of a job I've heard |
| 20:20.17 | willwh | it's reliant on some old ffmpeg 1.x libs I think |
| 20:20.33 | willwh | and I'd prefer to have asterisk handle this all natively, I don't care about video either ;) |
| 20:22.24 | willwh | honestly, I'm not even too worried about calling back to sipml5.... I've been using it to call outbound for a week |
| 20:22.29 | willwh | not a single call problem |
| 20:22.30 | willwh | :D |
| 20:22.50 | willwh | my little phone client is hilarious too; input field and 2 buttons >.> |
| 20:23.27 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 20:23.54 | newtonr | yeah i'm swamped with stuff to look at right now, all I can say is try SVN versions as fixes come out, and if you still have any issues be sure to file it in the tracker |
| 20:24.22 | newtonr | i haven't messed with wss at all recently |
| 20:26.36 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 20:27.02 | Dovid | Hi. When I set Set(CALLERID(name)=) it will set it to the number. is there any way of making it NULL? |
| 20:28.57 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 20:29.00 | WIMPy | What is "it"? |
| 20:31.43 | newtonr | davlefou, just now looking at your config |
| 20:33.04 | davlefou | ubuntu 12.04 x64 |
| 20:33.13 | willwh | newtonr: there is this too: https://issues.asterisk.org/jira/browse/ASTERISK-21930 |
| 20:33.29 | davlefou | newtonr, what do what i look about my config? |
| 20:33.30 | willwh | I'll keep an eye out for 12.2.0 :) |
| 20:35.38 | newtonr | davlefou, config looks fiine. I'm not sure why asterisk is not responding to your providers 401 challenge with a REGISTER with Authorization header |
| 20:36.11 | newtonr | davlefou, have you tried with a newer version of asterisk than 1.8.10 ? That is a pretty old version by now. |
| 20:36.53 | davlefou | it was on default on my ubuntu. |
| 20:37.43 | newtonr | davlefou, ah, yeah you might try building the latest release from source and seeing how that works for you. http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.26.1.tar.gz |
| 20:40.11 | marceloamorim | guys, anyone here can help me with file with a debug, [tk]D-Fender tried help me yesterday but the log file wasn`t full, now didn`t remove anything from the file. |
| 20:40.19 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 20:40.48 | marceloamorim | The problem is, sometimes during the call I can`t listen the audio |
| 20:49.51 | Kobaz | http://www.ooma.com/how-ooma-works/voice-quality |
| 20:49.52 | Kobaz | sooo |
| 20:50.07 | Kobaz | what codec are they using for those 'redundant packets' |
| 20:53.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:55.52 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:57.13 | *** join/#asterisk FuriousGeorge (182c2966@gateway/web/freenode/ip.24.44.41.102) |
| 20:57.21 | FuriousGeorge | howdy all |
| 20:58.36 | Chainsaw | It is the George. Good evening. |
| 20:58.49 | FuriousGeorge | yes Chainsaw it is I |
| 20:58.52 | FuriousGeorge | :P |
| 20:59.06 | FuriousGeorge | The angriest George in all IRC |
| 21:13.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:28.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:56.48 | *** join/#asterisk undecided (~AndChat66@36.37.236.199) |
| 21:57.56 | undecided | Hi |
| 21:59.53 | undecided | On my VPS running asterisk I also have android VM installed |
| 22:00.29 | undecided | On the Android VM there is an application which accepts call s |
| 22:01.40 | undecided | How can I reroute that call to some extension on my asterisk ? |
| 22:02.50 | undecided | I'm looking for some hack here obviously |
| 22:03.34 | bsdice | undecided you mean you Dial() to your Android and this should redirect the call? |
| 22:04.06 | bsdice | WHEN should it redirect, unconditionally? |
| 22:04.38 | undecided | No. Call lands on Android application |
| 22:05.28 | undecided | I want to forward it to asterisk |
| 22:06.08 | bsdice | how does it reach the Android VM, via UMTS? what app is running there? |
| 22:06.35 | undecided | I know this should be difficult to do but assuming everything possible on Linux |
| 22:07.08 | bsdice | bridging a 3G call to asterisk will be very hard |
| 22:08.01 | undecided | It's an app which gives free DID s and the application accepts calls |
| 22:08.21 | undecided | No 3g it's on vm |
| 22:08.23 | bsdice | no odea ;) |
| 22:08.26 | bsdice | no idea ;) |
| 22:08.36 | bsdice | obscure |
| 22:09.12 | undecided | Android VM exists on the same Linux server with asterisk |
| 22:32.27 | navaismo | undecided, and in your android app how do you handle the call? |
| 22:32.55 | navaismo | in your android app can you sent to another number? |
| 22:33.42 | undecided | No option to forward from there |
| 22:33.57 | navaismo | then how do you want to do it? |
| 22:34.58 | undecided | Need somehow to hack it into transferring the call to asterisk |
| 22:35.21 | undecided | Which dials on the same machine |
| 22:35.25 | navaismo | then you are in the wron channel |
| 22:35.30 | navaismo | wrong* |
| 22:35.57 | navaismo | when you have the redirection/forward then we can help you. |
| 22:36.28 | navaismo | If your app has register sip data use it with asterisk |
| 22:36.43 | undecided | The forward is the problem |
| 22:36.56 | navaismo | yeah and its in the android side not in the asterisk |
| 22:37.11 | undecided | No |
| 22:37.28 | undecided | It's Linux side |
| 22:37.30 | navaismo | No |
| 22:38.05 | navaismo | if you android app use sip use tcpdump to reverse the sip data |
| 22:38.18 | undecided | I can get the call auto answered on Android |
| 22:39.05 | undecided | I just need to take the audio and get it on some asterisk. extension |
| 22:39.08 | navaismo | Here in sterisk we can help you if: 1.- You have the registration data, 2.-Know how the incoming call works 3.-Create the peers etc |
| 22:39.44 | navaismo | but we cant help you if you have a closed app that cannot send/redirect the call to another peer |
| 22:39.55 | undecided | No registration data here man |
| 22:40.58 | undecided | I think my problem is more related to audio pipes in Linux |
| 22:41.30 | navaismo | still in worong channel |
| 22:41.32 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 22:42.07 | undecided | But the best chance to find someone who can understand me, and mightt have done something similar is here |
| 22:42.26 | navaismo | No, i dont think so |
| 22:42.36 | navaismo | or at least with the details that you provide |
| 22:43.09 | navaismo | did you know what technology use your app to get the call? How is connected? etc etc |
| 22:43.10 | undecided | Well, I think my question is clear |
| 22:44.47 | navaismo | nothing to do here....*fly away* |
| 22:45.10 | undecided | No I haven't checked that because I need to bypass the app |
| 22:46.15 | undecided | But they don't give sip details or anything that can be used by asterisk directly |
| 23:07.17 | *** join/#asterisk petris (~petris@192.184.93.147) |
| 23:16.28 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 23:22.41 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |