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| 00:19.41 | Shaan | is anyone aware of a mobile wifi voip phone, like a linksys or cisco type that have atleast 2 line buttons? |
| 00:20.12 | WIMPy | What are line button good for? |
| 00:20.42 | Shaan | well so lets say you got each line button going out to two seperate providers |
| 00:20.56 | Shaan | so you can use providera for domestic calls and provider b for international |
| 00:20.58 | Shaan | understand? |
| 00:21.22 | WIMPy | I don;t know any wifi phone that supports more than one account. |
| 00:21.48 | Shaan | well it doesn't need to aslong it supports 2 extensions |
| 00:22.22 | WIMPy | What could that be other than two accounts? |
| 00:23.05 | Shaan | not neccessarily |
| 00:23.11 | Shaan | maybe you using one sip account going to asterisk box |
| 00:23.17 | Shaan | and then that you got two extensions |
| 00:23.19 | Shaan | 1 to company a |
| 00:23.23 | Shaan | and 1 to company b |
| 00:23.58 | WIMPy | Sure, but that has nothing to do with the phone then. |
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| 00:29.02 | Shaan | yes it does |
| 00:29.06 | Shaan | when your making outbound calls |
| 00:29.18 | Shaan | for instance you will want one to dispkay caller id of the respective company |
| 00:29.46 | WIMPy | How could that be done other than two accounts? |
| 00:30.01 | WIMPy | Or some dialling prefix perhaps? |
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| 00:31.58 | Shaan | i suppose you can do a dialing prefix |
| 00:32.11 | Shaan | but that is alot harder then just pressing the line button |
| 00:32.53 | WIMPy | And what could that line button do, other than selectiong an account? |
| 00:33.21 | Shaan | lol your answering a question with a question |
| 00:34.15 | WIMPy | You said it need not be multiple accounts and you don't like dialling a prefix. So what other possibilities do you think might exist? |
| 00:35.45 | Shaan | what about softphones |
| 00:35.51 | Shaan | any reliable multi account softphone? |
| 00:36.01 | WIMPy | Lots |
| 00:36.23 | Shaan | i dont mind paying for an enterprise one but, i want stability and no bs |
| 00:43.39 | MadHatter42 | how do I limit chanspy to listen to only three extensions |
| 00:43.39 | MadHatter42 | ? |
| 00:47.12 | Shaan | ??? |
| 00:48.11 | MadHatter42 | chanspy |
| 00:48.20 | MadHatter42 | can listen like everything |
| 00:48.28 | MadHatter42 | i want to limit that range to 3 ext |
| 00:48.31 | MadHatter42 | Shaan, |
| 00:48.46 | MadHatter42 | i've found this guide http://www.untrustedroot.com/blog/2011/08/31/limiting-the-chanspy-function-in-a-freepbx-environment/ |
| 00:49.11 | MadHatter42 | but when i change the context that ext doesnt work anymore |
| 00:50.17 | MadHatter42 | anyone ? |
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| 01:27.03 | superscrat | MadHatter42 |
| 01:27.32 | superscrat | On our wiki, I found this article: https://wiki.asterisk.org/wiki/display/AST/Application_ChanSpy, that led me to this: https://wiki.asterisk.org/wiki/display/AST/Application_ExtenSpy |
| 01:28.46 | superscrat | but, if you are looking for assistance with Freepbx, you will get better answers from their forums. |
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| 06:53.57 | dyn60 | Hi. Anybody did PSAP server for eCall on Asterisk? Msg me. |
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| 07:45.12 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:36.04 | WIMPy | Is there any known issue with DTMF not being relayed from one SIP channel to another? Both using rfc2833. |
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| 09:11.00 | troulouliou_dev | hi when using zrtp is it correct to say that it is impossible to listen calls at the server side ? |
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| 09:36.02 | WIMPy | It's reasonable hard to do. |
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| 10:12.19 | phix | WIMPy!!!! WOOOOOOOO!!!!!!1 |
| 10:14.01 | WIMPy | What have you done? |
| 10:14.53 | phix | Broke cyrus |
| 10:15.03 | phix | upgraded from Debian Wheezy to Jessie |
| 10:15.10 | phix | Testing out asterisk now |
| 10:16.34 | WIMPy | Oh, please spare me with Debian. I spent two whole evenings trying to get things working on a RPi. |
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| 10:40.08 | Greenlight | My carrier apparently send through SIP headers in response to my invites, showing information such as the reason a call may have been declined. Is there a way to grab a list of all SIP headers in the dialplan? |
| 10:41.17 | WIMPy | HANGUPCAUSE_KEYS |
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| 10:43.21 | Greenlight | This is more X headers, not standard SIP ones |
| 10:44.10 | WIMPy | Then I see no way. |
| 10:44.21 | Greenlight | Damn |
| 10:44.38 | Greenlight | The SIPHeader function only grabs from INVITES yea? |
| 10:45.00 | WIMPy | AFAIK INVITE only, yes. |
| 10:45.10 | Greenlight | Yea, that's what I thought. Okay, thanks |
| 10:45.50 | Greenlight | Apparently they pass a SIP X header when a call is rejected due to me exceeding my channel limit, and it would be great to be able to fire some alerting based on that |
| 10:46.43 | WIMPy | Build yourself a proxy :-) |
| 10:47.07 | WIMPy | Sorry, don't have a better idea than that. |
| 10:47.23 | Greenlight | Yea, OpenSIPs is on my "todo" list |
| 10:48.31 | Greenlight | Although I must say I'm highly impressed how well Asterisk has handled being our main gateway, carrying all RTP and doing CDR's for billing running as a VM |
| 10:49.08 | Greenlight | circa 1400 simultaneous calls |
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| 10:50.05 | WIMPy | That's quite something. |
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| 10:55.27 | ldc | Greenlight: asterisk alone handling 1400 calls? |
| 10:55.48 | Greenlight | Yup |
| 10:56.46 | WIMPy | How much hardware does it need? |
| 10:56.55 | Greenlight | Even doing "proper" SRV dns |
| 10:57.21 | Greenlight | Think we only recently pushed it up to 6 cores, was on 4 for years |
| 10:57.36 | Greenlight | Oh, I tell a lie, it's on 8 cores now |
| 10:57.50 | WIMPy | Does it need them? |
| 10:58.10 | Greenlight | Well, load average: 0.91, 0.85, 0.88, but Asterisk running 400% CPU |
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| 11:26.08 | stefan27 | 1400 rtp sessions? |
| 11:27.03 | Greenlight | Yes, althought deliberately not hitting the core, doing p2p bridging |
| 11:27.26 | stefan27 | what's the typical rtp payload, no video? |
| 11:27.39 | Greenlight | All alaw, no video |
| 11:28.23 | Greenlight | If we need to use other codecs we offload that to another box |
| 11:30.46 | stefan27 | i wish i knew more about linux and asterisk's architecture... I have no idea how performance scales when changing number of CPU cores, rtp sessions, encryption/no-encryption, codecA or codecB, passthrough or transcode... |
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| 11:49.21 | Rico | is there a way to choose the file format of the sound file attached to mail when voicemails are sent by email ? |
| 11:50.17 | Rico | found it, error in my search |
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| 11:52.54 | RadJackson | Hi , is there a way to hangup all channel of a given peer ? channel request hang up takes the channel name or ALL , i would like to hang up only SIP/peername calls |
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| 12:28.59 | [TK]D-Fender | RadJackson: No. You have to do them individually. You could always loop them in a grep from OS CLI |
| 12:31.44 | file | I think we actually could have that functionality in 13+... |
| 12:32.51 | [TK]D-Fender | certainly doable |
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| 12:33.05 | [TK]D-Fender | lower care factor, but low effort to match |
| 12:33.05 | file | earlier versions not so much |
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| 13:48.22 | RadJackson | What is the best way to run two asterisk instances in the same BOX ? i'm thinking about docker containers , here is the use case : we would like to have two asterisk one for incoming calls one for outgoing calls , actually we have one asterisk box, and due to the huge amount of simultenous outgoing calls , asterisk CPU gets > 90% , also the memory gets high, so in such case we need to reboot |
| 13:48.22 | RadJackson | Asterisk , but this has an impact on the current incoming calls. So having an independant box only for outgoing calls can let us manage the box without any effect on the incoming calls. |
| 13:48.33 | RadJackson | Are VMs good enough to run Asterisk ? |
| 13:48.50 | Qwell | RadJackson: What happens when you want to transfer an incoming call out? |
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| 13:52.10 | Jayk123456789 | Hello, I have a question regarding Caller Line Identification within asterisk. I have recently moved to a new VOIP provider and I have just purchased 10 DDI's. I need to know how to I can get asterisk to recognise which number was dialled so that I can route the call internally to the correct department depending on which number was called. I have the line "same => n,GotoIf($["${EXTEN}" == "123456"]?s4)" in the extensions.co |
| 13:52.25 | RadJackson | Qwell all incoming calls run on one box via ivr , so that case does not exists on our current business plan |
| 13:53.17 | Jayk123456789 | could somebody tell me if the above line will let me correctly route calls? (ran out of characters) |
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| 13:55.13 | RadJackson | same => n,GotoIf($["${EXTEN}" = "123456"]?s4) |
| 13:56.54 | [TK]D-Fender | Jayk123456789: That line doesn't tell us anything. It is part of some other extension whose pattern we can't see and we have no idea where that jumps to or what that does. |
| 13:57.16 | RadJackson | My advice try managing and modelising did/department entities outside your dialplan , if not everytime you get a new DID , you will have to add an extension on your extensions.conf file. |
| 13:57.32 | [TK]D-Fender | huh? |
| 13:58.34 | RadJackson | and its "=" not "==" |
| 14:00.25 | Jayk123456789 | d-fender, what information would you like me to provide? |
| 14:01.49 | [TK]D-Fender | Jayk123456789: I just told you how little we see. We have no idea where you are jumping, what other things may have been set. You asking about the flow of the equivalent of a computer program based on showing us 1 line. |
| 14:02.19 | [TK]D-Fender | PASTEBIN is your friend |
| 14:02.24 | [TK]D-Fender | ~pb |
| 14:02.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:02.25 | [TK]D-Fender | ^^^ |
| 14:03.53 | Jayk123456789 | http://pastebin.com/9FZaLwrg |
| 14:04.52 | Jayk123456789 | That is the beginning of my context where that line resides and also the first instance of it being called. |
| 14:05.47 | [TK]D-Fender | Jayk123456789: well as you were told, it's = not == |
| 14:05.57 | [TK]D-Fender | Jayk123456789: Then... as to where it goes... we don't see the target in there |
| 14:06.56 | Jayk123456789 | http://pastebin.com/9QtKG7Ym |
| 14:07.13 | Jayk123456789 | Ok, this is the whole extensions config. |
| 14:07.21 | RadJackson | lol |
| 14:07.57 | Jayk123456789 | This config works, so the == is fine... |
| 14:08.41 | [TK]D-Fender | Looks OK except for the error that was stated twice so far |
| 14:09.13 | [TK]D-Fender | But it messy and would be better written with explicit extension #'s and use of a gosub/macro for the common holiday checks, etc |
| 14:10.35 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:10.48 | Jayk123456789 | We are re-doing it all so i will definitely look into cleaning up the new one. |
| 14:11.22 | RadJackson | [TK]D-Fender According to your experience, how many simultenous calls can an asterisk box support running theses configs (MEM/asterisk/cpu) : http://pastebin.com/4AfgQszn ? |
| 14:11.36 | [TK]D-Fender | RadJackson: What do you need? |
| 14:12.36 | RadJackson | Such server CPU and MEM gets high after 4 hours of non stop 400/simultenous/outgoing/calls and < 50 simultenous incoming calls |
| 14:12.38 | RadJackson | 450 sum |
| 14:12.59 | RadJackson | I want know if this is normal , or must be a lack of server optimization |
| 14:13.07 | [TK]D-Fender | RadJackson: Very likely. |
| 14:13.11 | *** join/#asterisk kharwell (kharwell@nat/digium/x-jxebpsbcafbtzmtn) |
| 14:13.46 | RadJackson | Ok thanks. So i need to have multiple boxes in such case. |
| 14:17.41 | [TK]D-Fender | Or optimize properly |
| 14:17.47 | Jayk123456789 | [TK]D-Fender: are you familiar with Caller Line Identification? |
| 14:18.06 | [TK]D-Fender | RadJackson: You haven't said anything about transcoding, recording, etc all of which impact the load. |
| 14:18.23 | [TK]D-Fender | Jayk123456789: CallerID is as boring a topic as it gets. |
| 14:18.49 | [TK]D-Fender | Jayk123456789: "core show function CALLERID" <- |
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| 14:21.00 | Jayk123456789 | I know but the only issue I am having is getting my new asterisk box to show me which number I dialled so that I can then take the number and route it internally. |
| 14:21.46 | [TK]D-Fender | Jayk123456789: The number that was DIALED .... is CALLED Line, not CALLING. |
| 14:21.52 | [TK]D-Fender | or Caller |
| 14:21.56 | [TK]D-Fender | that'd be CALLEE |
| 14:22.14 | [TK]D-Fender | Call'ed line is tyically in the EXTEN. |
| 14:22.26 | [TK]D-Fender | Of course this means you should be looking at the actual call. |
| 14:26.29 | Jayk123456789 | ok thanks |
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| 15:42.42 | ldc | hello |
| 15:42.57 | ldc | my provider requires me to send the caller-id in the From: part of the sip invite message |
| 15:43.22 | ldc | is a Set(CALLERID(ALL)="123") before the Dial enough? |
| 15:53.24 | ldc | apparently I also need so set fromuser |
| 15:58.47 | [TK]D-Fender | No. |
| 15:59.09 | [TK]D-Fender | * will use the callerID as the "from" unless overriden by fromuser |
| 15:59.21 | [TK]D-Fender | So you don't need both typically. |
| 15:59.27 | [TK]D-Fender | depending on how they work |
| 15:59.50 | [TK]D-Fender | CallerID is by default in ... the CALLERID() |
| 16:01.40 | ldc | ah thanks |
| 16:02.19 | ldc | apparently to get it to work with this provider I need to set CALLERID(all) to "123123" <123123> |
| 16:02.40 | ldc | and to delete fromuser= in sip.conf after setting the trunk as peer |
| 16:03.43 | [TK]D-Fender | Either can do the job |
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| 16:15.45 | ldc | http://pastebin.com/vDi1WF6p |
| 16:15.52 | ldc | it doesn't seem to process the directive here |
| 16:16.14 | ldc | from sip debug I can see it sends the callerid for the extension set in sip.conf |
| 16:16.23 | [TK]D-Fender | Ther is no "seem" ... because we aren't looking at the call with SIP debug for proof... |
| 16:16.42 | ldc | so my outgoing calls appear as +101 |
| 16:16.55 | ldc | From: "Lorenzo" <sip:101@ip>;tag=as5414a234 |
| 16:16.56 | ldc | To: <sip:callednumber@sip.didlogic.net>;tag=as75ced832 |
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| 16:19.04 | [TK]D-Fender | show us the full call |
| 16:24.59 | ldc | [TK]D-Fender: fixed, was missing closed ) |
| 16:25.14 | ldc | it would only pass up to Callerid(all |
| 16:25.44 | [TK]D-Fender | Good |
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| 16:42.34 | tsuriu | good afternoon guy |
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| 17:57.30 | WIMPy | pa: I met Jolly yesterday, so he's still alive. But he's got a job on th other coast so is busy travelling instead of coding. |
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| 20:17.26 | nixhr | hi |
| 20:17.44 | *** join/#asterisk Stokkeland (~milton@static-96-230-72-2.bstnma.fios.verizon.net) |
| 20:20.47 | nixhr | I noticed that in asterisk 1.8 when the remote PBX returns 403 Forbidden, asterisk sets disposition to FAILED and hangupcause to 34. In asterisk 11, when remote side returns 403 Forbidden, ast11 sets disposition to NO ANSWER and hangupcause to 21. I would say that ast1.8 handled that properly. Any information why was that changed? |
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| 20:24.53 | nixhr | Is it possible for ast11 to behave like ast1.8 in setting the disposition and hangupcause? |
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| 20:31.06 | iulhk | able to get udp packets at server but unable to get hit on asterisk console when trying to make call, can anybody take a look pls "http://paste.linuxassist.net/view/64f38e64"? |
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| 20:35.59 | Stokkeland | maybe this belongs in #asterisk-dev - its related to seanbright asterisk-opus work; is there or will there ever be a format_opus.c? ..and the next questions which shows I am a noob, is there a possibility to use app PlayBack(file.opus) or some way to play back to opus clients without transcoding? |
| 20:36.35 | seanbright | Stokkeland: short answer - not until digium's lawyers are comfortable with it |
| 20:37.08 | Stokkeland | okay - not even with the patch? |
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| 20:37.20 | nixhr | iulhk: seems like a nat issue |
| 20:38.22 | seanbright | Stokkeland: the patch won't get you the ability to read the file off of disk and not transcode |
| 20:38.43 | seanbright | you could certainly write a format_opus. typically opus files are in ogg containers. |
| 20:38.46 | nixhr | iulhk: this capture you sent, where is it recorded, remoteserver or local?? |
| 20:39.25 | Stokkeland | seanbright: thank you - i did see a comment on that, I dont have the experience to write such myself - was hoping someone had :) |
| 20:39.57 | seanbright | if i remember correctly, the problem is that the opusfile library gives you back PCM data |
| 20:40.29 | seanbright | and i don't know enough about the file format to be able to write something to read the raw opus data out and send it |
| 20:40.47 | seanbright | well... without putting effort in |
| 20:40.54 | seanbright | and ask around - i'm not known for effort |
| 20:40.58 | file | he's not |
| 20:41.03 | seanbright | ^^^^ |
| 20:41.08 | Stokkeland | okay thank you, hehe - |
| 20:42.36 | iulhk | <nixhr>: its on remote server |
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| 20:45.36 | MRH2 | hi wondering if anyone has a polycom ip4000? |
| 20:46.28 | jasonwert | MRH2: Having and issue with it? |
| 20:46.36 | nixhr | iulhk: ok. as i can see in the SDP part your local address is 192.168.something |
| 20:46.55 | nixhr | your router probably does not support sip connection tracking |
| 20:47.12 | nixhr | uh oh, maybe i'm wrong |
| 20:47.14 | iulhk | <nixhr>: i have 2 different asterisk servers, when i use imsdroid with Remote-Server-A, i am getting this issue, when i use linphone at Remote-Server-A, it works fine, and if i use imsdroid with Remote-Server-B , it works fine, both remote asterisk server are same asterisk-11 and the configuration are same |
| 20:47.29 | MRH2 | yeah i am sure i can hear the speaker active even when there is no call active |
| 20:48.24 | nixhr | iulhk: and both imsdroid and linphone are in the same local network? |
| 20:48.36 | iulhk | <nixhr>: yes |
| 20:49.10 | MRH2 | just wondering if anyone has one and this is normal |
| 20:49.14 | nixhr | so the tcpdump says the packet arrives but asterisk does not handle it whatsoever? |
| 20:49.26 | nixhr | if i understood correctly? |
| 20:49.38 | iulhk | <nixhr>: yes |
| 20:50.34 | iulhk | <nixhr>: even i tested with different network, i mean linphone1 using 3g and linphone2 using dsl, it works fine, |
| 20:51.23 | nixhr | so it occurs only with imsdroid? did you try imsdroid from other network? |
| 20:51.41 | iulhk | <nixhr>: but when using imsdroid at the same server , getting this issue, and if i shift to server B , imsdroid works fine by using same or different networks |
| 20:52.37 | nixhr | do the sip packets appear in the console with set sip debug on? |
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| 20:52.48 | carrar | moo |
| 20:52.55 | iulhk | even i tested with Remote-Server-A if caller is linphone and receiver is imsdroid then it works fine, but if caller is imsdroid and receiver is linphone then same issue |
| 20:52.56 | jasonwert | MRH2: I've worked with a lot of them and never saw anything like that. |
| 20:53.44 | MRH2 | just a slight hum like it is still powered on |
| 20:54.10 | iulhk | <nixhr>: if i set sip set debug on, there is no packet, i just took udp dump |
| 20:54.39 | MRH2 | but thanks for the feedback jason :) |
| 20:55.54 | Stokkeland | MRH2: what is the power source? |
| 20:55.56 | nixhr | iulhk: is it possible that some kind of packet filter on the server is in place? |
| 20:56.02 | nixhr | iptables or sth |
| 20:56.04 | nixhr | because |
| 20:56.23 | iulhk | <nixhr>: i have stopped firewall completely |
| 20:56.53 | nixhr | if you see packets with tcpdump, all the packets show there, regardless of firewall rules. however, these packets never reach asterisk because firewall drops them |
| 20:57.10 | nixhr | that is just an assumption |
| 20:58.14 | iulhk | shoudl i test by enabling firewall rules and then try? |
| 20:58.28 | jasonwert | MRH2: Good catch, Stokkeland . The 4000 uses that triangle powersupply. I have seen feedback, buzzing from those |
| 21:00.27 | MRH2 | cool - even when not on a call |
| 21:00.32 | Stokkeland | or even if it is PoE - if that PoE power is a bit dirty it could cause similar hums.. 50 to 60hz hums usually come from power anyway - leftover sinus curves |
| 21:00.38 | MRH2 | i assume there is always some power to the speaker |
| 21:01.01 | iulhk | <nixhr>: regarding your assumption, coz by using linphone at the same server, it works fine :) |
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| 21:08.15 | Stokkeland | iulhk: dont know if its an option but to troubleshoot and eliminate network-to-sys issues for sure, try stop asterisk - then use netcat to check if data is received. nc -l -u 5060 |
| 21:09.06 | iulhk | <Stokkeland>: thank you , let me check |
| 21:11.07 | seanbright | Stokkeland: create an issue on the github.com project and i will see if i can whip up something in my free time |
| 21:11.11 | seanbright | no promises though |
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| 21:13.09 | iulhk | <seanbright>: ok, thank you for your co-operatoin |
| 21:14.55 | Stokkeland | seanbright: Awesome - I will thank you |
| 21:16.01 | iulhk | <Stokkeland>: thank you :) |
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| 21:26.21 | malcolmd | MRH2: we've got some of the polycom sound point ip 4000 conference phones here at digium. we, too, experienced failures with the power supplies over time that lead to buzzing on the phone itself. the phones remained functional and operated normally after we replaced the power supplies. so, it's not entirely uncommon. |
| 21:26.38 | malcolmd | sorry, sound station |
| 21:48.14 | MRH2 | been hunting for a compatible power brick.... no luck |
| 21:52.53 | MRH2 | seems to be a different triangle 'power interface module' for PoE |
| 22:00.40 | MRH2 | ok thanks jasonwert Stokkeland malcolmd i'll explore power options :0 |
| 22:00.42 | MRH2 | :) |
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| 23:13.31 | Katty | CRUMPETS |
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| 23:20.51 | carrar | PICS!! |
| 23:36.39 | phix | Missing '/sys/bus/astribanks/drivers/xppdrv/sync' |
| 23:37.25 | phix | [May 14 09:35:21] WARNING[10888][C-00000001]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 23:37.56 | WIMPy | Looks like you need to install some drivers. |
| 23:38.30 | phix | sudo m-a a-i dahdi |
| 23:38.46 | phix | sudo service dahdi restart |
| 23:39.55 | phix | 4 is not a valid channel number. |
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