| 00:11.04 | bari | Samot, it is longer output... i can send you txt file with content |
| 00:11.13 | *** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta) |
| 00:14.06 | bari | Samot, or through the web ... https://justpaste.it/130np |
| 00:19.25 | *** join/#asterisk vandyk (~vandyk@187.183.4.107) |
| 00:24.46 | *** part/#asterisk kharwell (kharwell@nat/digium/x-bjvvclbenxmzfwzi) |
| 00:31.37 | [TK]D-Fender | we are clearly not seeing debug for a call going out |
| 00:32.01 | [TK]D-Fender | that does not look like you ran the command I gave you but rather one restricting iP |
| 00:38.01 | bari | [TK]D-Fender, sip debug was set to on |
| 00:39.41 | [TK]D-Fender | go prove DNS is working from OS CLI |
| 00:39.47 | [TK]D-Fender | for that domain you are calling |
| 00:44.54 | bari | i can check only DNS A record from OS.... (only nslookup installed)... |
| 00:46.29 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 00:46.52 | snadge | can asterisk be configured to include a hostname in the from: field of an invite, instead of the ip address? |
| 00:46.54 | bari | and A record is working... i also tried to set IP istead of domain |
| 00:47.14 | snadge | i was going to dig into the spec for this.. but i wondered if someone might know off hand |
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| 00:56.06 | [TK]D-Fender | fromdomain <---- subtly named parameter sitting right there.... |
| 00:57.13 | bari | i have to leave.. thank you for advices... my asterisk openwrt build looks strange.. i will try another build for standard linux like debian and i will see |
| 00:58.03 | *** part/#asterisk bari (~bari@2a01:8c00:ff00:80d5:7f98:4fc6:3e8:d181) |
| 01:00.07 | snadge | im such a dumbass |
| 01:00.25 | snadge | nobody has ever complained that the hostname wasn't in the from domain before though |
| 01:00.44 | snadge | and i was about to just fire back.. thats not how it works, just deal with it |
| 01:01.20 | snadge | theres also the possibility that changing that setting for every peer.. may confuse the majority who are expecting an ip there, i don't know |
| 01:01.39 | snadge | im not comfortable with just changing the fromdomain for the one person in 3 years who has complained about it |
| 01:04.14 | snadge | so im going to reject that request anyway.. i just wanted to learn something first ;) |
| 01:04.43 | Snwspeckle | General question, is it possible to send a request to a SIP URI you may not be in a call with? |
| 01:04.47 | snadge | its a client setting though.. so theoretically i could do it just for his peer.. but i don't want to, and cant be bothered |
| 01:05.17 | snadge | theres no clicky gui setting for it.. so it would have to be done in the back end.. bzzt |
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| 02:00.46 | Primer | [TK]D-Fender: https://ceregatti.org/asterisk.txt <-- sip show settings |
| 02:03.19 | Primer | From what I can tell, I'm simply not getting any packets back from outbound.vitelity.net |
| 02:16.07 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
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| 02:25.46 | ChannelZ | well I guess that depends on what IP, outbound is a big pool |
| 02:29.34 | ChannelZ | and I don't actually see your original question in my scrollback so can you summarize what's up? One-way audio, or..? |
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| 04:16.31 | Primer | If that was in regards to what I wrote: I'm making a call from my phone to asterisk, which is behind NAT and setup to go out via vitelity. I'm running tcpdump and seeing no traffic coming back from them. |
| 04:16.53 | Primer | Yet I can receive calls from them just fine via the registration to them, but that uses a different host. |
| 04:17.08 | Primer | Is there no upnp support in asterisk? |
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| 04:58.41 | igcewieling | there is no upnp support in asterisk. however there is STUN (and maybe TURN). Generally it is easier to set up nat correctly. |
| 04:58.50 | igcewieling | Primer: I use vitelity. |
| 05:01.46 | igcewieling | set nat=comedia,force_rport in sip.conf. If Asterisk is behind NAT, then set externip or externhost to whatever is the public IP. Log into your Vitelity Portal, click on Support, Asterisk Support. It might be less confusing to click on Support / Generic SIP support. |
| 05:03.01 | igcewieling | make sure your firewall allows all oubound connections to port 5060 as well as ports 10000 - 20000 are allowed. If you like to live life on the edge, feel free to turn off the firewall while experimenting |
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| 08:32.17 | Primer | Yeah, I've done everything you've mentioned. I have port 5060 and 10000-20000 forwarded to the machine behind the NAT, rtp.conf set to those ports, but it's just not making it past the SIP handshake |
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| 09:02.58 | ChannelZ | You get SIP Retransmission notices in sip debug? |
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| 09:34.23 | Primer | indeed |
| 09:35.23 | Primer | I don't know what changed. I used to use this system all the time, until my polycom soundstation stopped working. Since then I've upgraded the distro from mint 18 to 18.1 and replaced the router, which is provided by the ISP (I don't have an option for a bridge) |
| 09:35.56 | Primer | But suffice it to say that back then it was working perfectly |
| 09:37.45 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 09:45.30 | ChannelZ | Well if you can't even get SIP replies back you've got firewall problems |
| 09:45.55 | ChannelZ | Or the original packets are never even making it off the box |
| 09:45.58 | Primer | Yet zoiper on my phone on the same network works fine |
| 09:46.07 | Primer | when connecting to vitelity in the exact same manner |
| 09:46.13 | GeneralSpongebob | Which version of linux is best for Asterisk in a VM? Centos or Ubuntu? |
| 09:46.13 | ChannelZ | That's sort of a different path though. |
| 09:46.22 | Primer | tcpdump shows the packets leaving, just not coming back |
| 09:46.24 | ChannelZ | I assume Zoiper is running on another machine, NAT'd |
| 09:46.34 | Chainsaw | GeneralSpongebob: Whichever one you have most operational experience with. |
| 09:46.34 | ChannelZ | Asterisk is running on Linux, which is the firewall, yes? |
| 09:47.01 | Primer | It's running on my android phone, and configured to use the same vitelity account for outbound that asterisk is using |
| 09:47.08 | Primer | no, the Linux box is behind the NAT |
| 09:47.26 | Primer | ISP supplies a NAT router, not a bridge |
| 09:47.28 | ChannelZ | Is the firewall a hardware router or.. ? |
| 09:47.30 | ChannelZ | ok |
| 09:47.34 | Chainsaw | GeneralSpongebob: If you're used to CentOS, you're going to find Ubuntu frustrating and vice versa. |
| 09:47.43 | Primer | And again, this was all working just fine a few months ago |
| 09:48.10 | Primer | I didn't have to specify things like externip or localnet in sip.conf (both things I've tried today) |
| 09:48.13 | ChannelZ | Is asterisk set to register to vitelity? |
| 09:48.19 | Primer | yes, and that works |
| 09:48.24 | Primer | I can receive calls just fine |
| 09:48.30 | *** join/#asterisk ntz (~dpecka@static-84-42-228-122.net.upcbroadband.cz) |
| 09:48.33 | ntz | hello |
| 09:48.36 | ChannelZ | Then that makes no sense |
| 09:48.49 | Primer | I agree |
| 09:48.56 | ntz | is there some include() function for sip.conf please ? |
| 09:49.03 | ChannelZ | #include |
| 09:49.08 | Primer | I've forwarded ports 5060 and 10000-20000 UDP on the router to the machine |
| 09:49.16 | ntz | ChannelZ: but for sip.conf, not for ael |
| 09:49.36 | ntz | this is ael and I use it https://wiki.asterisk.org/wiki/display/AST/AEL+including+other+files |
| 09:49.55 | ntz | but I need something like etc/asterisk/sip.conf.d/ |
| 09:49.59 | ChannelZ | that should be true of pretty much all asterisk config files |
| 09:50.14 | ntz | oh really, nice, thanks !!!! didn't know ut |
| 09:50.16 | ntz | **it |
| 09:53.07 | Primer | the odd thing is that tcpdump does show one packet coming back: 09:52:42.389579 IP 64.2.142.190.5060 > 10.1.1.1.5060: SIP: SIP/2.0 404 Not Found |
| 09:53.41 | Primer | But that's after it times out: [Feb 1 09:52:39] WARNING[6776]: chan_sip.c:4047 retrans_pkt: Retransmission timeout reached on transmission 0a3fd94473249b9c0bdfdc2330ca14d5@47.150.137.82:5060 for seqno 102 (Critical Request) |
| 09:55.04 | ChannelZ | ntz - finally found it.. https://wiki.asterisk.org/wiki/display/AST/Using+The+include%2C+tryinclude+and+exec+Constructs |
| 09:56.45 | ChannelZ | does your router have some sort of SIP ALG function that's on mucking with things? |
| 09:56.59 | Primer | I suppose I should check that |
| 09:57.46 | GeneralSpongebob | Thanks for the reply Chainsaw. I was speaking to my colleague at the same time and I think we will go with Ubuntu 16.04LTS for production |
| 09:58.11 | ntz | ChannelZ: wow, thanks again .... that's .... that's just bombastic :D .. and I was wronging to asterisk thinking a long time that #include is just only for ael |
| 09:58.45 | ChannelZ | nope it's part of the standard config parser, pretty handy |
| 09:58.56 | ntz | yeah ..... |
| 09:59.56 | ntz | ChannelZ: btw, I have also something interesting - haha: # file asterisk |
| 09:59.57 | ntz | asterisk: ELF 32-bit MSB executable SPARC32PLUS Version 1, V8+ Required, dynamically linked, not stripped |
| 10:00.36 | ntz | I guess not many ppl has it because I had to patch it .... otherwise, it's mess and couldn't be built on solaris |
| 10:00.52 | ntz | ofc I released my patches already |
| 10:01.14 | ChannelZ | hmm |
| 10:01.22 | ntz | but they are kinda dirty, I guess possibly breaking things on linux ... the purpose was to make it working not to fix a stupid code |
| 10:02.44 | ChannelZ | Any particular reason? Or just to see if it could be done? |
| 10:03.31 | ntz | yes, I had a readon and I have it already in prod several months .... now I'm only polishing my solaris packages |
| 10:03.38 | ntz | *reason |
| 10:03.46 | ntz | if the money could be a reason |
| 10:04.55 | ChannelZ | ok just curious |
| 10:05.05 | ntz | ChannelZ: just adding a small little things like improving etc/sysconfig/$svcname svc config file (yes, I use this on solaris too for the most of my services) and few more things |
| 10:06.26 | ntz | actually in my todo list is to fix a broken implementation of T38 protocol in asterisk since it's really wtf ..... it works on the surface, but below the surface ... |
| 10:13.39 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 10:15.09 | Primer | ChannelZ: thanks for the help. I'll try again tomorrow. |
| 10:15.52 | TandyUK | oh T38 nearly cost me a £25k/yr contract last year |
| 10:16.23 | ChannelZ | Primer: Sure, sorry I wasn't more helpful, kind of been in and out tonight and am off to bed myself |
| 10:16.24 | TandyUK | existing setup was a bunch of analog/digital phones, through an avaya ipoffice 500, and some fax machines, all going out over isdn |
| 10:16.43 | TandyUK | new hosted pbx, everything works beautifully, _except_ the 2 fax machines |
| 10:17.11 | TandyUK | they couldnt even fax each other, on multiple different ATA's |
| 10:17.32 | TandyUK | i was told i was crazy by one of my suppliers, but someone _really_ needs to invent a VOIP fax machine |
| 10:18.15 | TandyUK | that is a fax with an ethernet port, native config for sip, and/or email, and give end users the feeling of faxing, but really use sensible means to tramnsmit the data behind the scenes |
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| 10:48.42 | MacroMan | Chrome is giving me a SSL error on a cert generated by the ast_tls_cert script: `ERR_SSL_SERVER_CERT_BAD_FORMAT` |
| 10:48.55 | MacroMan | Ive googled but can't find a solution. Anyone come accross thjs? |
| 10:51.32 | MacroMan | Fixed it. I had made an exception in chrome for a different cert at the same address before. |
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| 13:10.10 | Northcode_ | Anyone else having trouble accessing http://wiki.asterisk.org/ ? |
| 13:13.21 | Northcode_ | Well anyways, I'm having some trouble configuring opus on asterisk 13.10, the clients can connect with it just fine but it doesn't seem to respect my configuration in codecs.conf. I don't seem to be able to enable dtx |
| 13:15.35 | Northcode_ | http://pastebin.com/WuNxfYmz this is how I tried to set it up, any help would be appreciated |
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| 13:23.19 | igcewieling | TandyUK: sounds more like your sales person almost cost you 25k/hr by poorly understanding the customer requirements |
| 13:30.44 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:30.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:44.04 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:58.11 | GeneralSpongebob | Do we know when the wiki will be back? |
| 13:59.35 | file | the ISP that hosts the wiki (which is also the ISP that provides internet to Digium itself) is currently experiencing problems, it'll be fixed asap |
| 13:59.41 | [TK]D-Fender | Looks fine tto me |
| 13:59.56 | file | [TK]D-Fender: it's sporadically wonky |
| 14:00.15 | file | routing related it seems |
| 14:03.06 | dan_j | Hi. Please can someone explain what is happening here? This endpoint has been working for over 2 years and is suddenly showing this: http://pastebin.com/Hvb08exy |
| 14:03.19 | dan_j | Endpoint dead? |
| 14:04.02 | file | the To header seems to contain garbage at the end |
| 14:04.32 | dan_j | Weird that its only the To header and nothing else. I'll get the endpoint restarted to see if it helps. |
| 14:05.57 | Samot | Why are those tags looking so weird? |
| 14:06.35 | Samot | tag=22620e1f-8d74-42fe-b95f-830a3fbf6fb8 <-- never seen a FROM or TO tag look like that |
| 14:07.45 | dan_j | Not a clue. It's a Gigaset deskphone. Been working for a couple of years without any issue. |
| 14:07.58 | Samot | Where the packet it sent? |
| 14:08.09 | Samot | s/Where/Where's/ |
| 14:08.28 | dan_j | The Gigaset sent that packet to Asterisk. |
| 14:08.45 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:09.03 | Samot | It's an OK to an OPTIONS request. |
| 14:09.13 | Samot | Where the original OPTIONS |
| 14:09.18 | Samot | That only has a TO Tag? |
| 14:09.28 | dan_j | Just realised what you meant. |
| 14:10.28 | file | how is the wiki now? |
| 14:11.44 | Samot | Looks fine? |
| 14:12.03 | file | HE tweaked their routing |
| 14:12.28 | file | methinks there were some saturated 10Gbps links |
| 14:12.36 | dan_j | https://www.irccloud.com/pastebin/uYPG1nDT/ |
| 14:12.42 | dan_j | Samot: there you go. |
| 14:14.03 | Samot | To: <sip:emotions_201@92.XXX.XXX.16:5060>;tag=d1dbbg4`,4814,525e,8518,237bee1c6d52 < BAD |
| 14:14.19 | Samot | 1) I'm not sure about the comma's. |
| 14:14.27 | Samot | 2) the backtick is the bad thing. |
| 14:15.01 | dan_j | Ok. I'll factory reset the unit. |
| 14:18.31 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:24.01 | GeneralSpongebob | wiki's back |
| 14:37.36 | *** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 14:43.34 | GeneralSpongebob | if I want to start from scratch can I delete everything in /etc/asterisk and write new configs or are there any other files I'd have to delete? |
| 14:44.42 | igcewieling | /var/lib/asterisk /usr/lib/asterisk |
| 14:47.18 | *** join/#asterisk chendy (~alexc@121.34.144.242) |
| 14:48.13 | Samot | Wait.. |
| 14:48.22 | Samot | Start from scratch as in, just redo all the configs |
| 14:48.38 | Samot | Or reinstall Asterisk completely? |
| 14:51.52 | [TK]D-Fender | Thre is also voicemail, astdb, DAHDI if using compatible hardware,etc |
| 14:53.12 | Samot | Yeah, there's a few things. |
| 14:53.28 | Samot | But that is why I asked for clarification. |
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| 15:08.16 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:09.19 | GeneralSpongebob | the configs. Thanks igcewieling |
| 15:10.39 | GeneralSpongebob | Forgive me if I've missed something from the setup guide but am I supposed to do something to get Asterisk to run as 'asterisk' instead of root? Is there any guide for this on the wiki? |
| 15:11.56 | [TK]D-Fender | yes |
| 15:12.08 | [TK]D-Fender | And in the book |
| 15:12.12 | [TK]D-Fender | ~BOOK |
| 15:12.12 | 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 |
| 15:13.15 | GeneralSpongebob | Any hint where in the wiki? |
| 15:15.08 | igcewieling | you mean like this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Main+Configuration+File |
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| 15:20.12 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-rfiawgmmpffyvhvg) |
| 15:22.44 | GeneralSpongebob | possibly but that doesn't look to be specific to getting asterisk to run as a none-root user |
| 15:22.49 | *** join/#asterisk cch123 (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 15:23.27 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 15:29.04 | igcewieling | look closer |
| 15:29.51 | igcewieling | "; User to run asterisk as (-U) NOTE: will require changes to ; directory and device permissions " and " Group to run asterisk as" |
| 15:30.04 | igcewieling | it doesn't get more obvious. |
| 15:30.55 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-dqqegdtkkjpzzuth) |
| 15:30.55 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:32.51 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 15:33.08 | [TK]D-Fender | There are init scripts to update as well |
| 15:35.19 | *** join/#asterisk rwb (~Thunderbi@204.13.43.166) |
| 15:35.29 | igcewieling | that is mostly a distro specific thing though. |
| 15:35.49 | [TK]D-Fender | no, from source : make config |
| 15:36.12 | igcewieling | [TK]D-Fender: I thought that installed the distro specific scripts? |
| 15:36.43 | [TK]D-Fender | yeah I suppose in that sense. I intterpreted you to an AIW distro like PBX/*NOW |
| 15:36.46 | [TK]D-Fender | but year |
| 15:36.52 | igcewieling | in that case, GeneralSpongebob can just edit /etc/sysconfig/asterisk |
| 15:36.57 | [TK]D-Fender | and it is a common thing though tto use the provided initt scripts |
| 15:42.22 | GeneralSpongebob | igcewieling, thanks for pointing me toward that file but that wasn't enough information. I ended up finding a forum post which did include the steps. |
| 15:43.26 | GeneralSpongebob | Asterisk is now running as 'asterisk' after having created the user, editting the init scripts, and chown'd the directories but I can't connect to the console now which is probably a permissions issue somewhere I'd guess |
| 15:44.28 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 15:44.46 | [TK]D-Fender | you didn't mention having sttarted itt |
| 15:44.59 | GeneralSpongebob | Turns out I just had to "sudo asterisk -r" I keep forgetting about sudo |
| 15:45.10 | GeneralSpongebob | Asterisk was running (I'd checked with /etc/init.d/asterisk status) |
| 15:45.25 | GeneralSpongebob | and is running as 'asterisk' as checked with ps |
| 15:48.53 | igcewieling | good luck. |
| 15:49.37 | *** join/#asterisk chendy (~alexc@113.116.62.60) |
| 15:51.05 | GeneralSpongebob | I've checked the locations you mentioned earlier, igcewieling, but it looks like they are literally the meat of the applications rather than just configuration files. If I clear /etc/asterisk and write new files would that be all of the 'old' config gone? |
| 15:51.17 | GeneralSpongebob | I'm asking this questions so I know how to safely wipe out the samples |
| 15:52.54 | igcewieling | removing /etc/asterisk will remove all of the important Asterisk configs. |
| 15:53.02 | GeneralSpongebob | thanks |
| 15:54.35 | [TK]D-Fender | keep in mind just how many files do have to be there for * to function in what anyone would consider to be a functional state |
| 15:55.16 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 15:55.16 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 15:55.22 | *** join/#asterisk phrearch (~spindle@34.tbb.spindle.osso.nl) |
| 15:55.25 | phrearch | hello |
| 15:56.24 | GeneralSpongebob | Good day phrearch |
| 15:57.15 | phrearch | im new to asterisk. learning by reading asterisk the definitive guide. i'm trying to play a music on hold file, but i don't get any audio. i can see in the cli that the moh is playing but cant hear anything. im having an openbts installation in-between. i get some warnings like `ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave`. does asterisk moh plays the audio on the system using alsa by default? |
| 15:57.21 | phrearch | Hi GeneralSpongebob! |
| 15:58.32 | phrearch | i suspect that everything else is fine, because when i dial from a mobile to the extension, i get a proper message like https://paste.kde.org/pnybjokr4 |
| 15:59.09 | [TK]D-Fender | look at your MOH config |
| 15:59.29 | [TK]D-Fender | because you define how & where it gets it from |
| 16:01.10 | phrearch | i have a very basic musiconhold.conf https://paste.kde.org/pwteo5lig |
| 16:01.24 | phrearch | default dir is /var/lib/asterisk |
| 16:01.41 | phrearch | and /var/lib/asterisk/moh contains some example wavs from make samples |
| 16:02.16 | igcewieling | no, it does not use alsa by default. |
| 16:02.20 | [TK]D-Fender | we don't see the warning in your PB you pasted earlier |
| 16:02.34 | [TK]D-Fender | as for not getting audio, actually answer the channel first |
| 16:02.38 | [TK]D-Fender | Answer() |
| 16:03.44 | phrearch | sorry, the warning is https://paste.kde.org/peuqsqauw |
| 16:04.36 | phrearch | I actually got this example from the book: https://paste.kde.org/pgp5tuak8 |
| 16:05.03 | phrearch | ill see if it Answer works. thanks |
| 16:05.10 | [TK]D-Fender | nothing about that says anything about MoH |
| 16:05.44 | [TK]D-Fender | You are probably loading incorrectly configured channel drivers for chan_alsa, or something similar |
| 16:06.10 | [TK]D-Fender | chan_console, etc |
| 16:07.49 | phrearch | awesome. Answer fixes it! |
| 16:07.56 | phrearch | thanks |
| 16:08.21 | [TK]D-Fender | You're welcome |
| 16:08.34 | phrearch | save to disable alsa and still use moh i suppose? |
| 16:08.54 | igcewieling | phrearch: almost nobody uses chan_alsa for anything. |
| 16:08.59 | [TK]D-Fender | Always remember that when you intend to throw audio at a caller before anything that would just bridge (like Dial), you should probably be explicitly answering |
| 16:09.58 | phrearch | Thanks, will try to remember that! |
| 16:11.59 | phrearch | the unable to open slave warning doesnt seem to come from chan_alsa, because it's added as noload in modules.conf |
| 16:13.06 | [TK]D-Fender | chan Oss, and a few others, |
| 16:13.10 | [TK]D-Fender | concloe included |
| 16:13.13 | [TK]D-Fender | console |
| 16:14.03 | igcewieling | phrearch: my modules.conf: http://pastebin.ca/3762695 |
| 16:14.04 | phrearch | aha ok. ill see if i can fix the alsa error. probably can't find the right device for the asterisk user |
| 16:14.22 | igcewieling | use that and I expect most of your errors to go away |
| 16:14.35 | phrearch | igcewieling: thanks! |
| 16:15.50 | phrearch | yea that fixes the alsa error, nice! |
| 16:16.28 | phrearch | ill disable the calendar and mp3 modules. doesn't seem to be compiled by default |
| 16:17.30 | igcewieling | my sample already disables the calendar modules |
| 16:24.11 | *** join/#asterisk anonymouz666 (bd191ff7@gateway/web/freenode/ip.189.25.31.247) |
| 16:25.24 | anonymouz666 | hi, anyone already had problems trying to register a VVX 300 phone within asterisk? |
| 16:25.36 | anonymouz666 | the damn phone always send the digest username empty |
| 16:27.15 | anonymouz666 | that's a polycom model |
| 16:27.37 | igcewieling | anonymouz666: nothing different from all other polycom phones |
| 16:28.44 | igcewieling | my standard phone config: reg.1.auth.userId="609" reg.1.address="609" reg.1.auth.password="shhhhhi" reg.1.server.1.address="10.0.0.254" |
| 16:28.56 | igcewieling | yes, you need both userid and server |
| 16:29.13 | anonymouz666 | which firmware? |
| 16:29.30 | igcewieling | anonymouz666: anything paste 3.3.0 i.e. ALL VVXs |
| 16:29.46 | igcewieling | Everytihng later than 3.30 should work |
| 16:30.07 | anonymouz666 | igcewieling: do you think that could be due I am configuring through WEB? |
| 16:30.35 | anonymouz666 | there's an option there. Auth credentials. |
| 16:30.36 | igcewieling | anonymouz666: I can't help you with the web config. |
| 16:30.37 | anonymouz666 | it's filled. |
| 16:30.59 | anonymouz666 | igcewieling: I have no problems at all, switching to another way |
| 16:32.40 | igcewieling | There are two kinds of people. People who use centralized provisioning for Polycom phones People who will start using Polycom's central provisioning some day. |
| 16:36.44 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 16:46.43 | anonymouz666 | I am not against centralized provisioning, but it there's an web option, it should work. |
| 16:47.28 | anonymouz666 | it makes no sense to offer something with issues and in this case, basic SIP auth. |
| 16:54.17 | *** join/#asterisk Brixius (~Brixius@69.195.155.219) |
| 17:01.16 | igcewieling | The web config works, I just can't tell you how to do it that way. |
| 17:04.21 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 17:04.25 | wasanzy | Hi |
| 17:04.57 | wasanzy | can asterisk convert say "one" to 1? |
| 17:05.05 | Samot | For the VVX phones.... |
| 17:06.02 | Samot | You use Identification, which should have "Display Name", "Address", "Label" |
| 17:06.11 | Samot | "Address" = extension. |
| 17:06.18 | Samot | Or your SIP user |
| 17:06.52 | Samot | In the Authentication section, User ID/Password = SIP User/Password |
| 17:07.17 | Samot | "Server 1" is where you put the server information. |
| 17:08.42 | Samot | It's the same for all VVX series. |
| 17:11.28 | igcewieling | wasanzy: your question is confusing. |
| 17:12.50 | wasanzy | igcewieling: one is a word form of 1. and my question is does asterisk hv a function that can take one and return it's figure form "1"? |
| 17:14.59 | Samot | So a string to an integer. |
| 17:15.50 | igcewieling | There is nothing in Asterisk to convert the string "one" into the digit "1", but you could do it easily with a lookup table in the dialplan or astdb |
| 17:18.06 | igcewieling | If you have a bunch of global variables like DIGITS[one] = "1" and DIGITS[two] = "2", etc. To look up, if the variable DIGIT="one", then ${DIGITS[${DIGIT}]} should return "1" |
| 17:19.15 | *** join/#asterisk bayan (~dootdoot@unaffiliated/bayan) |
| 17:19.25 | igcewieling | those are NOT really arrays, btw, but we treat them like an array here. |
| 17:19.55 | igcewieling | quotes might be needed. |
| 17:20.32 | *** join/#asterisk Snwspeckle (268c1fd3@gateway/web/freenode/ip.38.140.31.211) |
| 17:20.40 | Snwspeckle | Anyone here have experience with sending messages to a SIP URI that you're not currently in a dialog with? |
| 17:21.03 | igcewieling | Snwspeckle: define "messages" |
| 17:21.53 | Snwspeckle | igcewieling: Sending metadata to inform a URI about application level state information, but it needs to be outside a dialog. |
| 17:22.28 | Samot | What type of message? |
| 17:23.05 | Samot | NOTIFY, SUBSCRIBE? |
| 17:23.12 | Samot | OPTIONS? |
| 17:23.23 | Snwspeckle | Two examples I have is when a user on a call mutes themselves, I need to inform other users about this so they can update their UI to reflect that. |
| 17:23.24 | igcewieling | Snwspeckle: This applies to Asterisk 11: The only was I can think of is to send a SIP SIMPLE (text message) to your application. If your application needs arbitrary SIP packets, then Asterisk cannot help you. Use something like sipp or sipsack |
| 17:23.45 | igcewieling | Snwspeckle: Asterisk manager spits out that information |
| 17:24.28 | Samot | How is a user muting themselves? |
| 17:24.37 | Snwspeckle | I'm not using asterisk. I'm here because there's no other IRC channels for SIP. |
| 17:24.52 | Snwspeckle | Samot: Users are in a conference call and they're muting themselves by disconnecting from the conference ports. |
| 17:24.59 | igcewieling | Snwspeckle: Asterisk cannot help you with your issue. |
| 17:25.37 | Samot | So they are sending a feature code to the conf.? |
| 17:25.42 | Samot | Via dialing digits? |
| 17:25.45 | igcewieling | If you were running everything on an Asterisk box then there are a few ugly options like using Asterisk manager. |
| 17:26.39 | igcewieling | (12:24:37 PM) Snwspeckle: I'm not using asterisk. <-- Asterisk is not involved in any of this in any way? |
| 17:26.42 | wasanzy | igcewieling: Thank you |
| 17:27.13 | igcewieling | wasanzy: try the global vars without the quotes. rememebr in Asterisk quotes are often literal. |
| 17:27.26 | Samot | Snwspeckle: What UI has to be updated? How are they muting themselves, as in what do they do to execute the action of muting? |
| 17:27.45 | Samot | What type of SIP messages are you looking to send? |
| 17:29.00 | Samot | And what "users" need to be informed? |
| 17:31.02 | Snwspeckle | Samot: Users are in a party. They opt in to a conference call that is following the mesh topology. A user mutes themselves by pressing the mute button which disconnects them from the conference ports. Doing so removes all audio but I need to send arbitrary data to the other users in the party that they are muted. |
| 17:31.29 | Samot | OK. I think some things need to be cleared up here. |
| 17:31.43 | Samot | First, when you mute yourself in a conf call. You do not disconnect. |
| 17:32.02 | Samot | You are still in the conf. call, you hear the other parties.. |
| 17:32.17 | Samot | You just stop your mic from picking up audio. |
| 17:32.25 | WIMPy | Can easily be done via AMI on *12+. |
| 17:32.58 | Samot | Let's not talk about how to make it work without understanding how it should work first. |
| 17:33.02 | *** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp130-03-74-14-78-25.dsl.bell.ca) |
| 17:33.04 | Snwspeckle | No I cannot hear them. With conference ports you're able to disconnect the outgoing and incoming ports. So I could mute my microphone and-or mute incoming audio. |
| 17:33.20 | Samot | Snwspeckle: What are you using? |
| 17:33.24 | Snwspeckle | I'm using PJSIP |
| 17:33.27 | WIMPy | used to send a messge to the users phone when they (un)nuted themseves whe I was on 12. |
| 17:33.28 | Samot | If you're not using Asterisk, what platform? |
| 17:33.45 | Samot | WIMPy: Not saying it cant be done. |
| 17:33.50 | Samot | Its how he wants to do it. |
| 17:33.58 | Samot | Plus, he's not using Asterisk. |
| 17:34.15 | Samot | He wants to inject a SIP message to a URI while it's in dialog. |
| 17:34.20 | WIMPy | Maybe I should look in to backporting that now that 11 is dead. |
| 17:34.24 | Snwspeckle | I'm using ejabbrd as my SIP server. |
| 17:34.26 | Samot | Related to that dialog but actually "out of dialog" |
| 17:34.55 | [TK]D-Fender | drmessano, Just picked up another 10 x 7941G's @ $7.50 USD/Each |
| 17:35.01 | Snwspeckle | I want to send a SIP message to a URI not in a dialog. This is because people in a party may not be in the SIP call but should be informed about when a user mutes themself. |
| 17:35.33 | Samot | OK. |
| 17:35.40 | Samot | There's a lot of steps to that. |
| 17:35.46 | WIMPy | sipsak |
| 17:36.18 | Samot | Because this sounds like there are web GUIs involved. |
| 17:36.25 | Samot | Since the term "UI" is being used |
| 17:36.36 | Samot | and "Not in the call but in the part so they need to know |
| 17:37.09 | Snwspeckle | Yeah. This is a mobile application but exactly. Party is related to my application. A user can "opt-in" to the conference call and I need to inform the other users about this action. |
| 17:37.50 | Samot | Snwspeckle: You're going to need to hire someone. |
| 17:37.59 | Samot | This is not something you are going to resolve over IRC. |
| 17:38.03 | Samot | There are too many variables.. |
| 17:38.22 | Samot | Without even considering the platform |
| 17:38.55 | Samot | Bring in the platform and it's not Asterisk or anything close, it's almost impossible. |
| 17:39.07 | Snwspeckle | Samot: Lol not an option. It's almost all in place. I think I should possibly look at notify/subscribe and presence? |
| 17:39.57 | Samot | Are the people you are sending this to in a call? |
| 17:40.08 | WIMPy | Are you even sure you want to send the information using SIP? |
| 17:40.11 | Samot | Or they are just an endpoint at this point that gets updates? |
| 17:40.39 | Samot | In order to have a "Dialog" you must be in a call. |
| 17:40.46 | Samot | That's why it's called "Dialog" |
| 17:40.52 | *** join/#asterisk mknooihuisen (~mike@12.150.48.70) |
| 17:41.10 | Snwspeckle | Samot: I have two ways of going about this. I can have users in a call but have their audio disconnected from the conference which seems to work okay. My primary worry was bandwidth but it seems fine. |
| 17:41.31 | Samot | It's not what I'm getting at. |
| 17:41.39 | Samot | If I have you app and there's a call going on.. |
| 17:41.44 | Samot | And I'm not in it.. |
| 17:41.54 | Samot | I'm not part of a dialog.. |
| 17:42.03 | Snwspeckle | Samot: I understand that to have a dialog you must be in the call. |
| 17:42.09 | Samot | So messages to me would not be treated as "out of dialog" |
| 17:42.24 | Samot | That would just be standard transactions. |
| 17:42.38 | Snwspeckle | So what I've done is once you join a party you auto call everyone but the conference ports are not connected. |
| 17:43.11 | Samot | You are going to need to hire someone |
| 17:43.16 | mknooihuisen | Hi all. Iâm attemping to configure my new DID from FlowRoute with asterisk, but Iâm seeing a âchan_sip.c:23850 handle_response_invite: Received response: "Forbidden" fromâ¦â error. Anyone willing to help me debug for a sec? Iâm fairly new to this. |
| 17:43.17 | Samot | Or ones |
| 17:43.36 | Samot | They are going to need to understand your app, your platform, how it interacts now.. |
| 17:43.45 | Samot | In theory, you can do what you want. |
| 17:44.46 | Samot | In practice, there are too many factors to randomly get support in random rooms because they are SIP/VoIP related. |
| 17:44.47 | Snwspeckle | Samot: The out of dialog support I think is above what I need right now. As a compromise I can support doing this within a dialog but I've run into a problem where the INFO method I'm sending is failing during transport every time. |
| 17:45.17 | WIMPy | And as we know that SIP stateless it doesn't really matter if there's a call or not. |
| 17:45.26 | igcewieling | If you are using Asterisk's ConfBridge or MeetMe I suggest you write a small script to connect to the manager port and output what it gets. then you can see all the events you can choose from. |
| 17:45.58 | igcewieling | I especially like the PeerStatus messages. |
| 17:46.41 | Samot | Yes it does. |
| 17:46.49 | WIMPy | From 12 on there a (un)muted messages. |
| 17:47.19 | *** join/#asterisk stux|work (stux@cosmo.lunarshells.com) |
| 17:48.13 | Samot | RFC 6665 <-- is about handling in-dialog NOTIFYs |
| 17:49.18 | Samot | SIP is stateless when it's over UDP because UDP is stateless. |
| 17:49.28 | Samot | SIP is not stateless when it's over TCP/TLS |
| 17:50.21 | Snwspeckle | Samot: Is this possibly why my INFO request within dialog are failing? It seems to be sending them over UDP. |
| 17:50.31 | Samot | I don't know. |
| 17:50.39 | Samot | I don't know what you are using. |
| 17:50.47 | *** join/#asterisk sekil (~sekil@cable-89-216-220-115.dynamic.sbb.rs) |
| 17:50.49 | Samot | I don't know your logic as to handling SIP messages. |
| 17:51.21 | Samot | This isn't about not wanting to help. |
| 17:51.37 | igcewieling | It is for me! 8-| |
| 17:51.37 | Samot | It's about in order to help you would have to provide A LOT of information.. |
| 17:51.53 | Samot | Before we could even start to figure out why it is or isn't working. |
| 17:52.20 | Samot | This isn't an "issue" it's a project. You need someone that can do the project. |
| 17:58.43 | *** join/#asterisk quentusrex (~quentusre@freeswitch/developer/quentusrex) |
| 17:59.51 | mknooihuisen | Hi all. Iâm attemping to configure my new DID from FlowRoute with asterisk, but Iâm seeing a âchan_sip.c:23850 handle_response_invite: Received response: "Forbidden" fromâ¦â error. Anyone willing to help me debug for a sec? |
| 18:02.33 | igcewieling | mknooihuisen: that error means they are rejecting you. it doesn't mean anything more. |
| 18:03.52 | mknooihuisen | igcewieling: Okay, makes sense, but who is âtheyâ FlowRoute? |
| 18:04.00 | igcewieling | yes. |
| 18:04.24 | igcewieling | Well, I think yes. you chopped off the important part of that message. |
| 18:04.41 | mknooihuisen | Sorry, didnât want to make too long of a message |
| 18:04.47 | mknooihuisen | One sec |
| 18:05.28 | mknooihuisen | [Feb 1 12:57:06] WARNING[2991][C-00000023]: chan_sip.c:23850 handle_response_invite: Received response: "Forbidden" from '"GRAND RAPIDS;MI " <sip:+16167175769@sip.flowroute.com>;tag=as41680f5f' |
| 18:06.05 | *** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp130-03-74-14-78-25.dsl.bell.ca) |
| 18:06.28 | mknooihuisen | For context, the 616 number is the one Iâm calling from |
| 18:08.25 | mknooihuisen | igcewieling: See above :) |
| 18:09.04 | igcewieling | yes, that is flow route saying Go Away! BTW, I'm from Holland MI |
| 18:11.10 | mknooihuisen | Oh, nice. Weâre kinda neighbors. Well, feel free to call that number if youâre looking for a new website, or a wedding DJ (I have an eclectic mix of skills :P ) Anyways⦠|
| 18:11.26 | igcewieling | *sigh* I've been troubleshooting a fax issue. The machine never answered the call. Finally I let ring and ring and ring. 2 mins 30 secs later the damn fax machine answers. |
| 18:11.35 | igcewieling | I live in Florida now. |
| 18:11.40 | igcewieling | no snow. |
| 18:11.57 | mknooihuisen | Iâm jealous⦠FL, or some place warm, is in my 5 year plan |
| 18:12.04 | mknooihuisen | Actually, itâs the whole plan pretty much |
| 18:12.30 | igcewieling | I moved out of MI about 20 yrs ago for warmer places. |
| 18:12.47 | mknooihuisen | I totally get it. |
| 18:14.11 | Samot | How is your trunk connected to Flowroute |
| 18:14.13 | Samot | ? |
| 18:14.31 | Samot | Are you using registration or IP auth? |
| 18:15.01 | mknooihuisen | Samot: IP Auth, I whitelisted my serverâs IP. |
| 18:15.09 | *** join/#asterisk overyander (~Jeff@209.141.208.197) |
| 18:15.15 | Samot | OK, then you're sending the call wrong. |
| 18:15.32 | Samot | TECHID*1NXXNXXXXXX@ |
| 18:15.57 | Samot | So yeah, forbidden. |
| 18:16.39 | Samot | You have to prepend your 8-digit techid with an * to the digits you send. |
| 18:17.01 | Samot | So 12345678*1NXXNXXXXXX@sip.flowroute.com:5060 |
| 18:17.32 | Samot | Even then your fromuser= needs to have that id as well. |
| 18:18.36 | mknooihuisen | Iâm misunderstanding something⦠Iâm sending digits by entering them into a phone and pressing âcallâ |
| 18:18.49 | Samot | Yes, you are. |
| 18:19.00 | mknooihuisen | Unless you mean in my Answer() dialplan |
| 18:19.02 | Samot | But they get as far as ASterisk. |
| 18:19.15 | Samot | Then Asterisk figures out (based on what you told it to do) with those digits. |
| 18:19.23 | Samot | In this case, it will Dial() to Flowroute. |
| 18:19.54 | Samot | Now Flowroute says if you use IP Auth, you must prepend what I said above to the call request. |
| 18:21.01 | Samot | So Asterisk needs to take the digits you sent and if they are to go out Flowroute, prepend this stuff. |
| 18:24.44 | mknooihuisen | Samot: Almost positive Iâm misunderstanding something. I tried âexten => ${TECHPREFIX}*18887204767,1,Answer()â but the extension is not recognized when I call |
| 18:25.09 | mknooihuisen | Congrats, you now have my work number, and the one Iâm working on :P |
| 18:26.07 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:29.17 | Samot | What is setting ${TECHPREFIX} ? |
| 18:29.29 | Samot | You have to get that from your Flowroute portal. |
| 18:31.42 | Samot | Are you setting that variable somewhere else and just calling on it? |
| 18:32.15 | Samot | exten => ${TECHPREFIX}*18887204767,1,Answer() <--- Wait, what is this for? |
| 18:32.34 | mknooihuisen | Fixed it, though Iâm not sure why it works |
| 18:32.55 | Samot | What did you use? |
| 18:32.57 | Samot | Show it. |
| 18:33.34 | mknooihuisen | In my [outgoing] context |
| 18:33.36 | mknooihuisen | exten => _1NXXNXXXXXX,1,Dial(SIP/${TECHPREFIX}*${EXTEN}@flowroute) |
| 18:33.38 | Samot | Mask the id with all 1111111 |
| 18:34.00 | Samot | OK, did you set ${TECHPREFIX} elsewhere? |
| 18:34.01 | mknooihuisen | Oh, the techprefix variable is set in [globals] |
| 18:34.05 | Samot | There you go. |
| 18:34.08 | Samot | That's why it works. |
| 18:34.23 | mknooihuisen | But thatâs for outgoing calls |
| 18:34.27 | Samot | Right. |
| 18:34.42 | Samot | Incoming are handled differently. ARe you have a problem with those? |
| 18:35.16 | mknooihuisen | Not anymore, but that was the issue I was debugging |
| 18:35.31 | mknooihuisen | which is why I donât understand how changing something in [outgoing] fixed it |
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| 18:42.36 | mknooihuisen | Samot: Essentially, I have this http://pastebin.com/4KhDWMEn working, but Iâd love to understand why |
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| 18:46.06 | mknooihuisen | Alright, so Iâve just discovered that I probably havenât actually fixed it |
| 18:46.17 | mknooihuisen | But it might make more sense now |
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| 20:18.40 | lpl | hi everybody |
| 20:19.02 | lpl | I have a question about ari and stasis |
| 20:21.51 | lpl | I use python ari-py as a lib, and when I start my script for the first time with a ari listener inside and an app with a given name, the app is register to stasis, when I stop my script or kill it I have a message that the app is destroy. But if I start the script again the app doesn't register and any command then get a asterisk log saying the app isn't register. |
| 20:22.16 | pjensen00 | you can also ask in the #asterisk-ari channel for more ARI specific people |
| 20:22.36 | lpl | ok thank you |
| 20:22.52 | pjensen00 | Also, you should be able to use the same app name |
| 20:23.13 | pjensen00 | I do it frequently when I restart my main ARI listener |
| 20:23.40 | pjensen00 | Can you post the connection code in a pastebin? |
| 20:23.59 | pjensen00 | and/or the logs showing this behavior? |
| 20:25.01 | lpl | there is nobody is the #asterisk-air chan |
| 20:26.33 | [TK]D-Fender | ARI, not AIR |
| 20:26.59 | lpl | yes sorry I use the right one just a typo in my message |
| 20:27.10 | [TK]D-Fender | I jsut went and saw more than a dozen |
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| 20:33.12 | lpl | here is the pasterbin |
| 20:33.13 | lpl | http://pastebin.com/f6hcRfcm |
| 20:34.18 | [TK]D-Fender | youhaven't asking in there yet |
| 20:34.27 | [TK]D-Fender | haven't asked* |
| 20:35.23 | lpl | yes I did and I just added the pasterbin there too |
| 20:35.37 | pjensen00 | No, I'm in that channel along with a lot of other people |
| 20:35.58 | pjensen00 | Your handle didn't show up but D-Fender's did |
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| 20:38.09 | [TK]D-Fender | lpl, You did not join the right channel |
| 20:38.48 | pjensen00 | what's the output you get from your python script when you run it the first time and when you run it the second time? |
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| 20:42.52 | pjensen00 | Are you certain your python script is actually trying to connect to the ari websocket? |
| 20:47.07 | lpl | Sorry I added a # when a type the channel name, do you guys want to continue there ? |
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| 20:59.37 | Centinel | One of our users is unable to make international calls on his phone. Calls ring but then end unexpectedly. Here's the sip.conf entry along with a call record, which shows 503 Unavailable: http://pastebin.com/0ubTV1jY |
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| 21:27.03 | n_byrnes | Hopefully this is a no brainer that I'm just overlooking the obvious, but, I'm troubleshooting a work-around to SIP-ALG in an off-prem extension. I've finally gotten 2 way audio working by setting rtp_symmetric to yes. But, when I call a cell from this remote extension, talk a bit, then the cell hangs up, my phone (the off-prem extension) now doesn't receive any notification that the call is over. What might cause that last message not to be getting |
| 21:27.03 | n_byrnes | there ? |
| 21:28.35 | n_byrnes | BTW, the SIP-ALG work-around was actually to set up a new transport on 6060 that the endpoint registers upon rather than 5060 |
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| 22:04.19 | n_byrnes | Turns out the test endpoint didn't have rewrite_contact set to yes ..... false alarm... |
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| 23:42.48 | yorick | igcewieling: no, I'm pretty sure the release tarball still has sounds/asterisk-core-sounds-en-gsm-1.5.tar.gz |
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