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| 02:00.12 | [[thufir]] | we use cloud based Asterisk (is that what it's called?) and Cisco hardphones. We could switch to softphones quite easily, correct? |
| 02:01.52 | WIMPy | Asterisk doesn't care if the phone application runs on a PC or some appliance that looks like a phone. |
| 02:04.07 | [[thufir]] | ok, thanks. what's the terminology for this, cloud based Asterisk? |
| 02:04.34 | WIMPy | hosted pbx |
| 02:11.40 | [TK]D-Fender | I hate everyone's throwing "cloud" around everywhere today. |
| 02:12.00 | WIMPy | Everything has become cloudy. |
| 02:12.20 | coppice | soon it will rain |
| 02:13.32 | WIMPy | Ain't no sunshine |
| 02:13.48 | Nivex | http://yaycloud.ytmnd.com/ |
| 02:13.59 | coppice | When she's gone |
| 02:16.06 | [[thufir]] | heh |
| 02:16.32 | WIMPy | Telephony is gone. |
| 02:16.46 | WIMPy | It's so funny how we don't talk any more. |
| 02:17.20 | tuvok302{tablet} | Hey all, i've got a feeling I'm about to ask a very stupid question... but I've run out of google-fu. I'm trying to setup asterisk for the first time, and I've followed the instructions up until the end of the hello world test and I don't get any audio |
| 02:18.08 | WIMPy | What kind of phone are you using and how is it connected to your Asterisk? |
| 02:18.13 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 02:18.15 | tuvok302 | The asterisk console says it runs just fine |
| 02:18.42 | tuvok302 | I'm using the zoiper soft phone, |
| 02:18.58 | WIMPy | On the same LAN? |
| 02:19.00 | tuvok302 | Yeah |
| 02:19.15 | WIMPy | On the same PC? |
| 02:19.20 | tuvok302 | No |
| 02:19.39 | WIMPy | ok, sounds good so far. |
| 02:19.42 | tuvok302 | I've got asterisk installed on another machine |
| 02:19.49 | WIMPy | Any firewall on that PC? |
| 02:20.04 | tuvok302 | no firewall on either machine, or firewall between them |
| 02:20.34 | WIMPy | Are you sure your sound devices are configured correctly? |
| 02:21.16 | tuvok302 | Relatively. When I run the audio wizard in zoiper I can hear the noises, and my mic works |
| 02:22.53 | WIMPy | Did you enable any kind of NAT support, like a STUN server. |
| 02:23.18 | tuvok302 | No |
| 02:24.02 | WIMPy | is running out of ideas. |
| 02:27.45 | tuvok302 | Haha, thanks for the suggestions WIMPy. I'll keep prodding google, see if it can shed any more light |
| 02:34.41 | WIMPy | Just wait for the SIP/RTP guys to wake up. |
| 02:35.18 | WIMPy | Although, being on zoiper, you could use IAX2 instead. Known for a lot less networking trouble. |
| 02:35.34 | WIMPy | And with that I'm off to bed. |
| 02:46.07 | tuvok302 | Whelp, IAX2 worked beautifully. |
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| 02:53.58 | [[thufir]] | is it possible to connect (cluster) a local Asterisk version with an online PBX? Or would that not make any sense? |
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| 03:08.40 | Nivex | I have a bunch of disallow=all allow=... directives from way back in the day. Do I still need those or does Asterisk pretty much figure things out on its own now? |
| 03:29.37 | Nivex | ahh, ok, I refreshed my memory |
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| 12:56.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 13:40.40 | DigiDaz | Hi all, I'm having a problem trying to call from a freeswitch box to an asterisk one. I don't want to authenticate, I am just direct dialling a did using a sip uri. Asterisk keeps asking for authentication. I have already added a trunk with insecure=port,invite but to my untrained eye, what apperas to be happening is that asterisk is mationg the peer, ie: "Found peer '601' for '601' from xxxxx". Anyone any idea how to make this happen? |
| 13:53.20 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 13:59.46 | [TK]D-Fender | is [601] the one it should be matching? |
| 14:04.33 | Katty | I"M ALIVE. |
| 14:04.48 | Chainsaw | Katty: This is great news! |
| 14:04.59 | Katty | yes. |
| 14:05.06 | Katty | and my dad's alive too. that's also great news. |
| 14:05.25 | Chainsaw | Katty: Let's celebrate. There can be pie. |
| 14:05.35 | Katty | YES. |
| 14:05.47 | Katty | i should make a pie, it'd be theraputic. |
| 14:05.48 | bewbs | as opposed to being not alive? |
| 14:06.26 | Katty | me, or my dad? |
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| 14:29.51 | DigiDaz | [TK]D-Fender: Sorry was AFK, no 601 is the remote extension on freeswitch but also there is an extension 601 on the asterisk too, the invite is being sent to 44161xxxxxx@asteriskbox the from line is 60a@freeswitch box |
| 14:30.02 | DigiDaz | 601@freeswitchbox |
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| 14:34.33 | [TK]D-Fender | DigiDaz: then you did your peer on * wrong |
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| 14:36.04 | DigiDaz | [TK]D-Fender: I though it was simply hostname=ipoffreeswitch, type=friend,port=insecure,invite and that would be enough? |
| 14:36.20 | DigiDaz | and context=from-trunk |
| 14:37.20 | DigiDaz | Is ther a way of getting any more debug info, I have core set verbose 9 and core set debug 9 but I get no output |
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| 14:39.40 | [TK]D-Fender | DigiDaz: HOST... not HOSTNAME |
| 14:39.58 | [TK]D-Fender | and "type=peer" |
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| 15:08.05 | alami | can't start conference with app_meetme.c:1228 build_conf: Unable to open DAHDI pseudo device |
| 15:08.32 | [TK]D-Fender | dahdi is not loaded... |
| 15:09.32 | alami | [TK]D-Fender: do you mean chan_dahdi.so= |
| 15:10.12 | [TK]D-Fender | that's the config... there is also the entire package itself |
| 15:10.18 | alami | loader.c:840 load_resource: Module 'chan_dahdi.so' already exists. |
| 15:10.24 | [TK]D-Fender | go prove that DAHDI is installed. |
| 15:11.48 | alami | http://pastie.org/9302291 |
| 15:11.55 | alami | [TK]D-Fender> |
| 15:12.18 | alami | i have asterisk 1.8 |
| 15:12.34 | alami | installed befor dahdi, can that be the Problem? |
| 15:13.30 | [TK]D-Fender | yes |
| 15:13.33 | [TK]D-Fender | it can |
| 15:13.48 | alami | [TK]D-Fender: any suggestions? |
| 15:13.55 | [TK]D-Fender | stop * |
| 15:14.10 | [TK]D-Fender | "dahdi_cfg -vvvv" |
| 15:14.15 | [TK]D-Fender | and PB the output |
| 15:15.05 | alami | http://pastie.org/9302299 |
| 15:18.00 | alami | [TK]D-Fender: on /etc/dahdi/ i have only xpp.conf |
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| 15:23.58 | alami | [TK]D-Fender:any idea? |
| 15:36.02 | DigiDaz | [TK]D-Fender: Sorry, my bad, it was already host and also I've tried peer |
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| 15:46.23 | [TK]D-Fender | DigiDaz: show your peers & the call |
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| 17:03.44 | *** join/#asterisk DelphiWorld (~TheVoicer@openvpn/user/DelphiWorld) |
| 17:03.48 | DelphiWorld | hi asteriskers |
| 17:03.52 | DelphiWorld | anyone use jssip? |
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| 17:18.31 | DelphiWorld | guys? |
| 17:20.49 | WIMPy | ~polls |
| 17:20.50 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 17:30.45 | DelphiWorld | LoL |
| 18:05.02 | DelphiWorld | ... |
| 18:06.11 | TazzNZ | DelphiWorld: what do you want to do with jssip or what is wrong |
| 18:06.31 | TazzNZ | or are you simply wanting to know how many people are using it |
| 18:06.40 | DelphiWorld | TazzNZ: i have been trying to click a click to call button for month but no luck using jssip |
| 18:07.36 | TazzNZ | ok - so it's broken - what parts are broken ? Do you see anything on Asterisk that can help us help you debug it ? |
| 18:07.48 | TazzNZ | what about the logs or debug info created by jssip |
| 18:07.56 | DelphiWorld | TazzNZ: not ast issue but javascript i'm not well on that |
| 18:08.30 | TazzNZ | ok....you are aware that you are in an Asterisk channel, not a JavaScript channel |
| 18:09.05 | DelphiWorld | TazzNZ: i knowthat. but asterisk have a web socket, so discution is not limited to sip.conf |
| 18:09.22 | file | erm that's a stretch |
| 18:10.51 | TazzNZ | yeah...quite a stretch - you still not telling us what is wrong, or giving us any debug info to help you |
| 18:11.19 | DelphiWorld | TazzNZ: i want guidance on what do i need to create a simple click to call button |
| 18:11.57 | TazzNZ | I am going to say you are in the wrong channel for that |
| 18:12.06 | TazzNZ | We can tell you what you need done on the Asterisk side |
| 18:12.22 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 18:12.22 | DelphiWorld | TazzNZ: ... |
| 18:12.58 | TazzNZ | but it seems that your issue is related to the Web site of things (e.g. the button, the jssip lib etc) |
| 18:16.40 | qakhan | i am getting followin message when i run .php script from dialplan |
| 18:16.42 | qakhan | utils.c:1328 ast_carefulwrite: write() returned error: Broken pipe |
| 18:17.14 | TazzNZ | qakhan: how are you running this file, agi ? |
| 18:17.26 | qakhan | yes |
| 18:18.09 | TazzNZ | your agi is either dying before asterisk is done writing data to it or you are closing stdin |
| 18:18.51 | qakhan | here is my .php http://pastebin.com/8S5yc762 |
| 18:19.17 | qakhan | and extensions.conf http://pastebin.com/YeD44Hr3 |
| 18:19.55 | TazzNZ | you know that stdin/out/err is opened by default in PHP ? |
| 18:20.19 | qakhan | no i dont know |
| 18:20.35 | TazzNZ | they are STDIN/OUT/ERR |
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| 18:21.21 | qakhan | can you give me a link which cover this topic? |
| 18:21.29 | *** part/#asterisk DelphiWorld (~TheVoicer@openvpn/user/DelphiWorld) |
| 18:21.46 | TazzNZ | qakhan: php.net - search for IO Streams |
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| 18:24.03 | TazzNZ | qakhan: if you put verbosity at 5 (or above) what do you see in the Asterisk console when a call is made that calls this agi |
| 18:29.22 | qakhan | ERROR[11430][C-00000025]: utils.c:1328 ast_carefulwrite: write() returned error: Broken pipe |
| 18:29.27 | qakhan | same thing |
| 18:29.37 | TazzNZ | there should be more |
| 18:29.45 | TazzNZ | like *way* more |
| 18:29.51 | qakhan | AGI Rx << SET VARIABLE agent3 Local/9303@agent |
| 18:29.51 | qakhan | <SIP/3289-00000046>AGI Tx >> 200 result=1 |
| 18:29.51 | qakhan | [Jun 18 14:28:52] ERROR[11430][C-00000025]: utils.c:1328 ast_carefulwrite: write() returned error: Broken pipe |
| 18:31.55 | TazzNZ | and your debug log from the php script - does that show any reason why it died ? |
| 18:33.39 | qakhan | where that log saves? |
| 18:34.48 | TazzNZ | that might be the issue.... |
| 18:35.14 | TazzNZ | what do you set $log as? |
| 18:35.18 | qakhan | where is that debug log file locates? |
| 18:35.45 | TazzNZ | qakhan: uhm, this is *your* script hey, not mine |
| 18:37.51 | qakhan | i know but is there any specific location where php log file creates? |
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| 18:38.17 | TazzNZ | qakhan: no |
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| 19:14.48 | jeffasinger | I'm trying to call an asterisk box using pjsip, however I keep getting a 404 response. I'm assuming I need to do something to my sip.conf to fix this, any ideas of what to check? |
| 19:15.21 | WIMPy | Or your extensions.conf. |
| 19:17.13 | jeffasinger | I think my extensions.conf is okay, as other clients are able to call in fine |
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| 20:12.07 | litn | how sensitive is voip traffic in terms of network quality? the sip provider pings at around 80 miliseconds. I'm pinging 20 packets every minute and I see a 1/20 packet loss every few hours |
| 20:12.20 | litn | could that small of packet loss affect call quality or cause dropped calls? |
| 20:14.31 | litn | or let me put it another way- if I'm having some call quality issues and dropped calls, can I rule out the network based on the above analysis? |
| 20:17.24 | WIMPy | Ping doesn't mean much. |
| 20:17.40 | WIMPy | And it's very unlikely to be related to calls dropping. |
| 20:18.12 | litn | our network connection is unlikely to be related to calls dropping? |
| 20:18.17 | litn | or ping rates |
| 20:21.07 | WIMPy | If you have audio issues, keep an eye on jitter. |
| 20:23.12 | litn | but not dropped calls? that would be totally unrelated? |
| 20:24.04 | WIMPy | The network situation needs to be pretty bad for calls to drop. |
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| 20:33.25 | cosmicwombat | Anyone know a CLI method of determining if a Hardware EC board is installed on a Digium TM800P card ? |
| 20:34.10 | cosmicwombat | sorry.. TDM800P ^ |
| 20:34.42 | WIMPy | Only in dmesg when loading the driver. |
| 20:35.09 | cosmicwombat | dmesg showed dahdi_echocan_oslec: Registered echo canceler 'OSLEC' |
| 20:35.23 | cosmicwombat | So I am guess no EC card |
| 20:35.25 | WIMPy | That's software. |
| 20:35.29 | litn | WIMPy: so this leaves the sip provider, the phones themselves, and the asterisk server then? |
| 20:35.51 | WIMPy | You can have multiple ECs. |
| 20:36.15 | litn | I have heard about duplicating rtp/sip traffic to another server so you can record the traffic there as to not affect your production resources, do you know what term I can google to see how to do that so I can get a capture of a dropped call? |
| 20:36.20 | WIMPy | litn: Look at your logs and if they don't help start debugging. |
| 20:37.08 | WIMPy | It's unlikely that you have to look at RTP. |
| 20:38.44 | litn | to begin debugging, would I first grep the asterisk full log file for the phone number that dropped, or is there another way? |
| 20:39.28 | WIMPy | Have you looked in to the logs? |
| 20:40.01 | WIMPy | If you're lucky, you can find the channel IDs. |
| 20:41.01 | litn | if I'm so lucky, what do I do with that? |
| 20:41.09 | WIMPy | It helps to have some Verbose() at strategic laces in your diallan, like the h extension. |
| 20:41.19 | WIMPy | Read. |
| 20:44.57 | litn | so when I get the channel ID, with that, I would grep for the channel ID to get the full details in the logs? |
| 20:45.10 | litn | I;ll start with that |
| 20:50.26 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.76.92.218) |
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| 21:11.39 | jeffasinger | Is there any logging done by asterisk when it rejects a call with a 404 |
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| 21:14.05 | [TK]D-Fender | Depends what 404 |
| 21:15.20 | jeffasinger | I just get 404 (Not Found) back |
| 21:16.46 | mic_ | barely related, but maybe someone: VoIP + D-Link switches - good mix? |
| 21:19.11 | [TK]D-Fender | Work fine |
| 21:21.30 | [TK]D-Fender | [17:15]jeffasingerI just get 404 (Not Found) back <- to WHAT...... |
| 21:48.04 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 21:52.40 | litn | WIMPy: I was able to get the channel for a dropped call, and compared the grep o f that channel id side by side with a normal call hangup, and there were no differences |
| 21:52.47 | Greek-Boy | By default Polycom phones use DNSnaptr as transport in their config files. Should I change this to UDPonly for asterisk? |
| 21:53.00 | litn | WIMPy: could this be a case of the logs not being verbal enough? |
| 21:55.02 | WIMPy | Either that or there is no problem at your end. |
| 21:55.28 | WIMPy | If one of the other parties thinks there's an issue, it might just cause a normal hangup. |
| 22:00.09 | litn | WIMPy: the situation is that we switched from a broadsoft hosted voip system to an inhouse asterisk system and upon switching have experienced around 15 dropped calls a day. I think this is maybe 1% of our calls but we didn't have this before switching |
| 22:00.42 | litn | we got a new sip host, new internet connection for it, and new phones, so there are a lot of variables I guess |
| 22:01.28 | WIMPy | Indeed |
| 22:02.11 | WIMPy | The what has changed question usually has the best success rate when trouble shooting. |
| 22:04.54 | litn | WIMPy: but you believe if it were the asterisk system itself causing the issue, then it would be evident in the logs, so I can rule that out? |
| 22:06.03 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 22:11.31 | WIMPy | It might. You never know for sure unless you debug the signalling traffic. |
| 22:12.36 | litn | and to do that, I need to put Verbose() in the dialplan? |
| 22:13.02 | WIMPy | No, that just helps you find your way in the logs. |
| 22:13.28 | WIMPy | Use Asterisks debugging facilities or those of e.g. wireshark. |
| 22:14.21 | litn | so at this point in debugging I need to dump sip/rtp traffic? I heard there was a way to send rtp/sip to another server to do the logging. I tried once to tcpdump all of the traffic but it was resource intensive |
| 22:15.16 | WIMPy | Leave out the rtp part unless you need it. |
| 22:18.29 | litn | alright... and then once I have these logs, I'm looking to see if I received a normal BYE from the other end when the call "drops", right? And if so, I should I guess show these logs to my provider? |
| 22:18.34 | litn | thank you for being patient with me btw I really appreciate it |
| 22:23.41 | mic_ | in wireshark |
| 22:23.49 | mic_ | there is a cool function called "Show VoIP calls" |
| 22:23.56 | mic_ | then you can just pick them |
| 22:24.00 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 22:24.03 | mic_ | and it will even draw the flow for you. |
| 22:25.21 | WIMPy | First of all, you want to find out who terminated the call. |
| 22:39.49 | Greek-Boy | [TK]D-Fender you around? |
| 22:39.57 | Greek-Boy | maybe you're watching some football... lol |
| 22:48.09 | Greek-Boy | just wanted to let you know I resolved the problem with Polycom config. Was a syntax error. :D |
| 23:03.00 | [TK]D-Fender | was what I was expecting... |
| 23:15.33 | Greek-Boy | Yeah well, what can I say... I'm a complete Polycom newbie |
| 23:15.56 | WIMPy | Don't they have a log? |
| 23:16.15 | [TK]D-Fender | Shouldn't have had a syntax error when I asked if you reprovisioned with stock configs and said "yes" |
| 23:16.23 | Greek-Boy | They do but funny enough it wasn't showing in the log |
| 23:16.33 | WIMPy | bad |
| 23:16.41 | Greek-Boy | syntax error was my fault |
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| 23:57.28 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |