| 00:17.49 | igcewieling | Cisco wants you to buy Call Manager and use the phones in SCCP code. They have no interest in you using their phones with any other PBX. |
| 00:18.52 | artyx | THey can want in one hand, and defecate in the other, guess which will fill up first? ;) |
| 00:21.20 | slav3_kitten | it's cisco... so the want hand |
| 00:21.39 | artyx | Will remain sufficiently wanting ;) |
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| 01:18.22 | raden | Katty, ? :) |
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| 04:52.12 | KNERD | is there way to get more detailed SIP headers? |
| 04:54.42 | carrar | turn on sip debuging |
| 04:54.54 | carrar | use ngrep and look at the sip packets |
| 04:55.16 | carrar | capture them with tcpdump and use wireshark |
| 04:55.21 | carrar | lots of options |
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| 05:14.30 | KNERD | that is not enough info... |
| 05:14.52 | KNERD | i mean just like they appear in the CLI, just more details |
| 05:18.30 | igcewieling | sip debug is as good as it will get in the Asterisk CLI |
| 05:18.46 | igcewieling | if you want something more use wireshark/tshark |
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| 05:36.35 | KNERD | ahh...okay thanks igcewieling: |
| 05:38.46 | mitchrodrigues | Quick question, is their a way to debug manager commands |
| 05:38.50 | mitchrodrigues | to the console |
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| 05:50.13 | WIMPy | Same answer: Use wireshark. |
| 05:54.19 | KNERD | i guess that is the only way to knwo if something is getting passed to the server |
| 05:54.59 | KNERD | wireshark for linux? |
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| 06:23.24 | KNERD | no wireshark for linux |
| 06:23.41 | WIMPy | ? |
| 06:26.16 | KNERD | wireshark..linux version |
| 06:27.03 | WIMPy | What's with that? |
| 06:30.54 | KNERD | exactly...don;t exist |
| 06:31.12 | WIMPy | So what's everybody using then? |
| 06:35.12 | KNERD | no idea |
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| 07:03.10 | wdoekes | KNERD / WIMPy: normally tcpdump -w to write pcap files. you can analyze them on your (linux) desktop with wireshark |
| 07:03.35 | KNERD | thanks |
| 07:04.10 | wdoekes | KNERD: e.g. tcpdump -vs0 -w output.pcap port 5060 |
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| 07:49.26 | KNERD | wdoekes: okay//thanks for that info |
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| 08:22.11 | KNERD | wdoekes: guess I need to study tcpdump...that example you gave me was a bit too sterile..it was just like looking at the output of the CLI with sip debug on... |
| 08:22.28 | KNERD | too formatted...yeah that is it... |
| 08:22.37 | KNERD | hoping for more raw data |
| 08:23.37 | KNERD | such was what SIP headers the box is actually receiving, not what asterisk is interpreting and what it is ignoring |
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| 08:54.24 | adnc | hello, I've an iax2 peer I would like to monitor and control a LED. for this I used AMI and also asterisk -r commands, but it does not seem to me a good way executing a command every second in order to trigger the status of my peer. is there a better way I could make use of? |
| 09:00.57 | kaldemar | .. |
| 09:05.21 | eject_ck | how should I understand this message? |
| 09:05.21 | eject_ck | [Aug 2 11:57:32] ERROR[1963]: pbx_impl/ast/ast108.c:737 sccp_wrapper_asterisk18_rtp_write: SEP20BBC092F531: Asked to transmit frame type 2 with no samples. |
| 09:06.24 | eject_ck | I'm wondered if there are any benchmarks to run synthetic load tests for asterisk encoding ? |
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| 12:45.29 | cfrod | Anyone got experience hosting Asterisk in an azure vm? Everything is working perfectly, except registering with a voip provider. Just times out. |
| 12:45.50 | pabelanger | sounds like a network issue, not a vm issue |
| 12:46.15 | pabelanger | are you virtual adapters setup for bridge mode or nat ? |
| 12:46.34 | cfrod | It's just configured for nat |
| 12:46.37 | eject_ck1 | cfrod: how much it cost ? |
| 12:46.47 | cfrod | I've posted the configs+logs here: http://forums.asterisk.org/viewtopic.php?f=1&t=87536 |
| 12:46.50 | pabelanger | cfrod: that is your issue |
| 12:46.52 | pabelanger | ~sipnat |
| 12:46.52 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 12:46.52 | eject_ck1 | azure vm hosting |
| 12:47.09 | cfrod | eject_ck1: Right now it's an extra small instance, which is about 10$ iirc |
| 12:47.20 | cfrod | pabelanger: Thank you :) |
| 12:47.31 | eject_ck1 | cfrod: check STUN |
| 12:49.08 | cfrod | eject_ck1: On it. |
| 12:49.21 | cfrod | Also, the first link (the quick guide) is dead. |
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| 13:11.55 | Katty | morning |
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| 13:14.33 | igcewieling | 'morning katty |
| 13:14.59 | Katty | waves to igcewieling |
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| 13:24.03 | igcewieling | my cat has cancer 8-( |
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| 14:01.12 | colinwielga | Good morning, I have been reading the Asterisk The Definite Guide 4th Edition and am learning about modules…I current have access to an enterprise level asterisk environment and use a test lab to learn. The question I have is, when I unload app_voicemail.so on the asterisks box, I still somehow have access to voicemail on the phones....Access to leave messages as well as access to check messages |
| 14:01.16 | colinwielga | How is this possible? |
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| 14:03.19 | igcewieling | colinwielga: after unloading the module does "module show like voicemail" show anything else? |
| 14:04.04 | igcewieling | voicemail is tied into directory so you may have to unload that as well. modules can and do depend on each other |
| 14:04.12 | colinwielga | shows 0 modules loaded |
| 14:04.28 | igcewieling | colinwielga: pastebin the CLI output if a call which shows the problem. |
| 14:05.09 | colinwielga | Module Description Use Count |
| 14:05.10 | colinwielga | 0 modules loaded |
| 14:05.24 | colinwielga | is that what you are requesting? |
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| 14:07.01 | colinwielga | I guess I'm just trying to learn how exactly it works…If I unload the app_voicemail.so module, how is it that i'm still able to to access voicemail? |
| 14:07.29 | colinwielga | COuld it be because we have a datacenter and the app_voicemail.so may be running on a server somewhere? |
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| 14:10.10 | igcewieling | colinwielga: no, that is not why |
| 14:11.03 | colinwielga | ok, well that is helpful…Thought I was missing something…So essentially that means that there is some other module that may be handling the voicemail? |
| 14:11.04 | igcewieling | colinwielga: for the most part trying to unload modules is a silly and pointless exercise, with a few exceptions. If you don't want people to use voicemail then don't allow that application in your dialplan. |
| 14:11.50 | igcewieling | but without the CLI output of a call where this happens, I cannot help you further and likely nobody else can either. |
| 14:12.06 | colinwielga | Standby for cli output |
| 14:13.05 | igcewieling | to a pastebin |
| 14:13.06 | igcewieling | ~pb |
| 14:13.06 | 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. |
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| 14:19.45 | colinwielga | http://pastebin.com/9QR5Y5YJ |
| 14:21.22 | igcewieling | colinwielga: the call is going off server. You didn't say you were sending calls off-server |
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| 14:24.32 | colinwielga | hmmm I think I understand…so what you mean is that the voicemail is being handled by some other server right? |
| 14:25.10 | colinwielga | Well I think that pretty much answers my question. Thank you for your time! |
| 14:25.25 | igcewieling | It means your call is being transferred to another PBX, what happens after that we don't know. |
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| 14:26.25 | MikeH | having issues with sip trunk, phones just say busy here but sip trunks appear to be up |
| 14:26.44 | MikeH | Where can I find out why the call is failing - there don't seem to be any messages with asterisk -vvvvvvr |
| 14:27.11 | kpettit | lo. I've got a group of phones connecting to asterisk server through a router. Phones work, but attended transfers and presence doesn't work. At other sites it does but not at this one. |
| 14:27.54 | kpettit | So I'm guessing the issue is with the router at the site. It's a cheapy home router thing. Wasn't sure if there is any settings or ports I need to check for with presence and attended transfers |
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| 14:29.26 | colinwielga | port blocking? |
| 14:29.52 | kpettit | I'm not sure. presence just uses SIP right? |
| 14:30.45 | colinwielga | can you bypass the router for troubleshooting or is this production |
| 14:31.13 | kpettit | it's in production but I was going to see what I can do about testing the router. Wasn't sure if this is a common problem or not. |
| 14:31.34 | kpettit | Didn't want to tell them it's their router and get a new one if it's something simple like a port |
| 14:31.39 | kpettit | all normal phone calls work |
| 14:32.56 | colinwielga | let me check something |
| 14:34.16 | colinwielga | Ok, so in our company, they use different ports for SIP and our presence applications…Might want to verify what ports you are using |
| 14:34.47 | kpettit | ah ok. |
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| 15:18.35 | protocoldoug | Is it possible to see the channel state (busy,ringing,up,etc) on an originated call (via call file, or, "channel originate <tech> ..." at the CLI) ?? When I do it right now, all I see is "down" until the call times out (on a busy) or when I pick up. |
| 15:19.00 | protocoldoug | I see down in either "core show channels" or an AGI "CHANNEL STATUS <chan>" or an AMI "CoreShowChannels" |
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| 15:19.11 | protocoldoug | This happens in both 11.2 and 1.8 |
| 15:20.16 | protocoldoug | I never actually see busy or ringing, Just "down" and "up" |
| 15:20.51 | WIMPy | If you're opn AMI alreaady, you should see a newstate event telling you the channel is now ringing. |
| 15:21.20 | WIMPy | If it's busy it will go away immediately anyway. |
| 15:21.29 | WIMPy | You're not using POTS, do you? |
| 15:22.07 | protocoldoug | It's SIP (to a gateway to the PSTN [I control the gateway, it's not an ITSP]) |
| 15:22.31 | protocoldoug | I'll definitely look into the newstate event, that sounds like a great avenue to explore |
| 15:22.34 | WIMPy | Does that GW give you the information? |
| 15:24.12 | protocoldoug | Y'know............ I made the assumption it did, but..... now I am questioning myself. |
| 15:24.21 | protocoldoug | I just tried with a plain dial() |
| 15:24.31 | protocoldoug | thanks for the insight :) ...i'm gonna hafta pile over a manual |
| 15:25.13 | protocoldoug | with the plain dial() looks the same, though it might've been the originate, but, mannn, I just got stuck in the rabbit hole here... again, appreciate the extra eyes/ears |
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| 15:29.22 | malaphus | Anyone happen to know the length and other restrictions of sip secrets? Such as can I use special characters (any I shouldn't use), maximum length, etc? |
| 15:33.03 | WIMPy | protocoldoug: turn on sip debug to see what your GW tells you. |
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| 15:35.28 | protocoldoug | it's not sending a 180 ring, nor a 486 busy, sooooo.... that's spot on right there :) *hands WIMPy a beer* |
| 15:36.29 | colinwielga | show hints? |
| 15:36.58 | colinwielga | oops nvm. read more and realized it was answered |
| 15:37.03 | ChannelZ | malaphus: normal chars/numbers/punctuation I imagine, but I don't know that I've seen it documented |
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| 16:27.29 | vlad_starkov | Question: Anyone know is it possible to make Gigaset C610A IP phone auto answer with some special SIP-header sent to it? Like in Cisco SPAXXX SIP-header "Call-Info:\;answer-after=0". |
| 16:33.06 | igcewieling | vlad_starkov: that would be a question for the manufacturer. |
| 16:33.58 | vlad_starkov | igcewieling: Just thought somebody know the answer |
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| 18:00.50 | Katty | does it really matter which version of asterisk i get if i want to virtualize it |
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| 18:02.50 | Chainsaw | Katty: Yes, you want 11 so you're not stuck with the old DAHDI timing. |
| 18:03.06 | Katty | mmmk. |
| 18:03.28 | WIMPy | 1.8 din't need that, did it? |
| 18:03.57 | WIMPy | But you want 11 anyway. |
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| 18:04.12 | igcewieling | Katty: You want Asterisk 11 so people don't point and laugh at you. That is the main reason. |
| 18:13.16 | navaismo | can i receive some tips about asterisk 12 and the res_sip.conf, Cant register yet, i understood that i need to disable chan_sip and work only with res_sip. This is the output-->http://pastebin.com/XCNf5iCx |
| 18:14.50 | WIMPy | Well, you can;t have both on the same port, but I dont see why you need to disable one of them. |
| 18:15.24 | navaismo | well i disabled the chan_sip and set the res_sip to 5060 |
| 18:15.30 | navaismo | but all i get is forbidden |
| 18:19.17 | Chainsaw | Well, mostly because 11 is that sweet spot where it is still actively developed enough that they might fix bugs you report... |
| 18:19.35 | Chainsaw | But not quite so actively developed that it breaks every other week. |
| 18:20.05 | Chainsaw | 1.8 bugs are not going to get fixed, 10 was never that stable... |
| 18:20.29 | [TK]D-Fender | [14:19]Chainsaw1.8 bugs are not going to get fixed, 10 was never that stable... <- pardon? |
| 18:20.45 | WIMPy | didn't have more issues with 10 than 1.8 or 11. |
| 18:21.09 | Chainsaw | As long as you didn't use TLS or SIP over TCP, sure. |
| 18:21.53 | WIMPy | No, I think I tried that with 1.8. |
| 18:22.23 | Chainsaw | Now granted, it still takes 3 patches on top of 11 to make it go, but it is stable there. |
| 18:24.39 | navaismo | ... ... |
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| 18:24.57 | Chainsaw | And as navaismo is trying to tell you, 12 doesn't work yet. |
| 18:25.14 | WIMPy | It does work. |
| 18:25.31 | WIMPy | But the latest is very slow. |
| 18:25.44 | file | In your specific situation. |
| 18:26.44 | file | navaismo, change [siptest] to [mysip2000] |
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| 18:41.03 | mjordan | 1.8 is an LTS release that is still supported. Bugs in it are fixed just the same as bugs in 11. |
| 18:41.38 | mjordan | Off the top of my head, I can't think of any bug we've rejected in 1.8 that we fixed in 11. The only ones we do that with are typically CDR bugs, but that's because people got tired of us 'fixing' CDRs. |
| 18:42.30 | mjordan | Since both 1.8/11 are branched and released, at this point, we would almost *never* skip a bug found in 1.8 and fix it in 11. It would either get fixed in both - if it was a problem in both - or it would get fixed in neither and be fixed only in trunk. |
| 18:42.41 | mjordan | The latter situation would only occur if there was major risk associated with the change. |
| 18:43.10 | lorsungcu | anyone know if its possible to capture dtmf in CEL? |
| 18:50.07 | navaismo | file, so endpoint & aors must be the same? |
| 18:50.43 | file | the included endpoint identifier matches an endpoint using the user portion of the From, most clients put the authentication username there |
| 18:54.01 | navaismo | ok let me try it |
| 18:54.07 | navaismo | thanks btw |
| 18:54.15 | file | yw |
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| 19:08.59 | navaismo | file, yes that do the trick. Now how can I set a flag to know which phone answer? I.e. I have two endpoints my softphone and my IP phone how can know which one answer the call? |
| 19:09.14 | file | I don't understand the question |
| 19:10.55 | navaismo | when I dial using chan_gulp--->Dial(${GULP_DIAL_CONTACTS(mysip2000)}) my softphone and my Yealink phone start to ring, then how can i know in the cdr which phone take the call? |
| 19:11.18 | file | you can't, the only differentiating thing would be the contact URI - but that doesn't tell you which, really |
| 19:11.30 | file | if you need to know *precisely* you have to have separate endpoints |
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| 19:12.07 | navaismo | i was thinking that, so same endpoints must be used for one single peer and not for a "ring group" |
| 19:12.29 | file | correct, if you want a ring group then do it using normal Asterisk dialplan |
| 19:12.49 | file | if you want your desk phone, mobile phone, soft phone to all share the same endpoint because they are the same person - then use the above |
| 19:13.32 | file | or don't, up to you! |
| 19:13.38 | navaismo | right, still a great function |
| 19:13.56 | navaismo | let me digg in more features of this asterisk 12 |
| 19:15.53 | file | just remember - it's still a work in progress ^_^ |
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| 19:17.05 | navaismo | yes |
| 19:19.35 | navaismo | lorsungcu, all my searches seems to give negative answer about dtmf and cel, but maybe im just using wrong terms |
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| 19:20.32 | lorsungcu | yeah doesn't do it stock, not familiar enough with it to know what i'd need to do it myself. |
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| 20:05.44 | yoavz | Hey all, I'm trying to get asterisk up and running on rackspace small cloud instance, I've tried to install asterisk following this guide: http://blogs.digium.com/2012/11/05/how-to-install-asterisk-11-on-centos-6/ skipping libpri and dahdi (no need for hardware cards) and trying to get it running turned up to be problematic... |
| 20:06.18 | yoavz | I'm getting this error every second in /var/log/messages - Aug 2 23:05:52 monitor kernel: asterisk[3063] trap invalid opcode ip:55ba88 sp:7fff0da68350 error:0 in asterisk[400000+207000] |
| 20:07.25 | WIMPy | That Asterisk version is not compatible to your hardware. |
| 20:07.32 | WIMPy | Or the emulation? |
| 20:08.03 | WIMPy | Did you build it yourself? |
| 20:08.19 | yoavz | :/ Is there anything I can do to fix it? |
| 20:08.42 | yoavz | Anyone here have any insights on running asterisk on public clouds? |
| 20:09.10 | WIMPy | Did you build it yourself? |
| 20:09.51 | yoavz | yes... |
| 20:11.17 | WIMPy | I don't remember the detaisl, but there was an issue with certain implementations of virtual machines. I think it should work if you disable build_native in the compiler flags section. |
| 20:21.53 | yoavz | WIMPy, I'm looking into it now.. |
| 20:21.58 | yoavz | I'll give it a try and I'll update... |
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| 20:34.21 | yoavz | WIMPy, while I'm waiting for it to recompile, maybe you or anyone else in the channel can help me get started |
| 20:34.47 | yoavz | Do you know any company that'll charge me by the minute without monthly charges? |
| 20:34.56 | yoavz | I need to make very few calls from this server.. |
| 20:35.24 | WIMPy | Actually I don;t know any with monthly charges. What country? |
| 20:35.56 | *** part/#asterisk malaphus (malaphus@carbon.collisionpoint.net) |
| 20:36.03 | yoavz | I'm from Israel |
| 20:36.36 | yoavz | but calling Israel is fairly cheap from the US and from most of EU so the provider's location doesn't really matter |
| 20:37.04 | WIMPy | For the US we have |
| 20:37.09 | WIMPy | ~itsplist-us |
| 20:37.09 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:37.10 | yoavz | Hell yeah! :) It's working, disabling build_native and disabling the console option did the trick :) |
| 20:37.29 | WIMPy | In Europe, sipgate is still the standard choice. |
| 20:37.56 | WIMPy | Not neccessarily the cheapest, tho. |
| 20:38.29 | yoavz | And for the US? which one is the most simple to get started with? (I'm aware that this question might be totally irrelevant, but I have no idea what I'm doing) |
| 20:38.58 | WIMPy | See the list from infobot. |
| 20:56.19 | leifmadsen | WIMPy: good to know |
| 20:56.34 | leifmadsen | WIMPy: is that for large scale, or just personal use? |
| 20:56.44 | WIMPy | Err, wat? |
| 20:56.46 | WIMPy | what |
| 20:56.47 | leifmadsen | sipgate* |
| 20:56.56 | WIMPy | Ah. Both. |
| 20:57.08 | leifmadsen | we're expanding our operations into Europe, so will look into it |
| 20:57.13 | leifmadsen | or at least pass onto those looking into it |
| 20:57.32 | WIMPy | I'm sure you can find better prices. |
| 20:57.42 | leifmadsen | it's not the price we care so much amout |
| 20:57.44 | leifmadsen | about* |
| 20:57.50 | leifmadsen | but rather the reliability and interoperability |
| 20:58.24 | mic_ | leifmadsen: which countries will you start with? Or will it full EU from day one? |
| 20:58.33 | mic_ | s/will it/will it be/ |
| 20:58.44 | WIMPy | If you want high availability, 1&1 is the choice. But they do it only bundled with DSL lines. |
| 20:59.05 | leifmadsen | mic_: likely major countries for the most part, I think we have like... Sweden, Germany, France, etc |
| 20:59.13 | mic_ | leifmadsen: DK? |
| 20:59.22 | leifmadsen | mic_: not sure to be honest |
| 20:59.34 | leifmadsen | we're a hosted pbx tenant, is that what you'd be looking for? |
| 21:00.05 | WIMPy | Tough business. |
| 21:00.12 | mic_ | well, mostly some good alternative in terms of SIP trunks |
| 21:00.15 | mic_ | and international calls. |
| 21:00.22 | WIMPy | There;s one who did a lot of TV ads. |
| 21:00.59 | mic_ | leifmadsen: if you go mobile in DK - you can get international calls almost free, the moment you do that via SIP it's not that funny any more ;) |
| 21:01.03 | WIMPy | And so far I know of noone who has ever used such a service except for temporary while movin office. |
| 21:01.45 | leifmadsen | WIMPy: we do a lot of UC stuff, and have mobile clients, desktop clients, analytics and a lot of things build on top. It's not just hosted pbx, which I agree, is a tough business :) |
| 21:01.49 | leifmadsen | we have value added services |
| 21:02.12 | mic_ | yeah, today that's the value added service and customer support |
| 21:02.17 | leifmadsen | aye |
| 21:02.19 | mic_ | that really counts. Everybody can get a low price |
| 21:02.27 | leifmadsen | been doing hosted tenants for years now |
| 21:02.30 | mic_ | but not everybody can have good customer support ;) |
| 21:02.32 | leifmadsen | probably 8-9 years now |
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| 21:05.38 | mic_ | but back to the hosted pbx - one of the guys doing sales & 1st line support for a common project |
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| 21:06.00 | mic_ | he had a company A - and these guys just started hitting the bottom in terms of service |
| 21:06.12 | mic_ | took him a few days to quickly move all his customers to a company B. |
| 21:06.33 | mic_ | It's a tough market, but still there's plenty of reckless people around |
| 21:06.48 | mic_ | so I would assume the flow is there - as well as some dynamics. |
| 21:06.53 | mic_ | (but I am also very sleepy) |
| 21:08.20 | WIMPy | leifmadsen: With Telefonica and Vodafone you can run in to that nice SIP interoperability issue. (re AOC) |
| 21:09.08 | leifmadsen | hawt :) |
| 21:09.18 | leifmadsen | we've been talking with Colt I think since we already do stuff with them |
| 21:09.37 | leifmadsen | but I know nothing about who we're looking at to be honest. We opened a shop in Amsterdam and someone is dealing with all that. |
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| 22:02.09 | deegen | ?q |
| 22:02.49 | deegen | Nice. A typo AND a wrong window. |
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| 23:01.23 | WIMPy | wonders where in AMI the key "time" occours. |
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