| 00:00.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:11.40 | *** join/#asterisk KNERD (~KNERD@23.227.189.101) |
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| 00:26.26 | KNERD | Is a VoIP channel consisting of one way sending, or duplex sending and receiving? |
| 00:26.51 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 00:30.14 | [TK]D-Fender | duplex |
| 00:35.28 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 00:44.21 | *** join/#asterisk Apteryx (~maxim@69-165-208-140.cable.teksavvy.com) |
| 00:44.56 | Apteryx | Hello :) Could anyone help me to enable h264 video in Asterisk with pjsipà |
| 00:44.59 | Apteryx | *? |
| 00:48.25 | Apteryx | I'm loading the format_h264.so module in modules.conf, but when I use h264 in my endpoints allow= definitions, asterisk complains that: WARNING[32218]: channel.c:834 ast_best_codec: Don't know any of (h264) formats |
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| 00:56.00 | KNERD | [TK]D-Fender: thanks...so in using u-law, for example. 64BKBs per send and 64KBs per receive, or 64KBs for both send and receive? |
| 00:59.55 | KNERD | Apteryx: did you compile Asterisk with h264? |
| 01:04.58 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 01:05.02 | [TK]D-Fender | Each |
| 01:05.13 | ndb | hey, how can I debug chan_sip.so loading it via modules with gdb? |
| 01:05.17 | Apteryx | KNERD: I have format_h264.so in /usr/lib/asterisk/modules, so I guess so |
| 01:10.09 | KNERD | [TK]D-Fender: thanks |
| 01:12.14 | KNERD | Apteryx: give us mor elog details |
| 01:28.56 | Apteryx | KNERD: http://pastebin.com/SMVtSPed |
| 01:32.28 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 01:36.02 | Apteryx | Should I define anything in codecs.conf to get h264 working? |
| 01:38.24 | Apteryx | Must be me being stupid again. Just noticed there is a res_format_attr_h264.so module... which I was not loading. |
| 01:38.53 | Apteryx | I'm sorry if I made you loose a hair or two! |
| 01:40.01 | *** join/#asterisk TheSin (~TheSin@gateway.bluefalls.ca) |
| 01:40.53 | TheSin | hey guys I just updated dahdi and it wiped all my configs, none of the kernel modules loaded and even now that i have loaded them I can't get an spans, anyone have any ideas or anyplace I can go read |
| 01:41.06 | Apteryx | The bad news is that I get the exact same WARNING. :( |
| 01:45.30 | Apteryx | KNERD: Also, now I realized that probably having format_h264.so is not enough, and I would require codec_h264.so as well, but I don't have this on my system. Maybe your first question was spot on after all! I'm really sorry for my ignorance. |
| 01:49.31 | *** part/#asterisk TheSin (~TheSin@gateway.bluefalls.ca) |
| 01:51.43 | *** join/#asterisk TodWulff (~TodWulff@unaffiliated/todwulff) |
| 01:52.28 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 01:52.34 | TodWulff | thinks "NAT'g an asterisk server is a real PITA..." |
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| 01:54.00 | [TK]D-Fender | I've never had issues with it |
| 01:54.23 | TodWulff | I am totally ignorant <-- root cause |
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| 01:55.21 | TodWulff | a non-linux guy with asterisk installed on a wrt router trying to learn the ins and outs of iptables in a matter of a few hours |
| 01:55.54 | TodWulff | read as - trying to boil the ocean, ya know. I'll get it, just needed to vent for a sec. |
| 01:56.47 | TodWulff | If anyone has a NAT'd asterisk instantiated on an Asus router and is willing to share their iptables config, i would be appreciative. |
| 01:58.17 | [TK]D-Fender | Does * have a public IP on it, or is it just running on a router that is behind another? |
| 02:00.17 | TodWulff | it has a public IP that is not static. I have a domain name and am using afraid.org as my name servers. however, name resolution typically results in resolving to the isp name vs. my domian. I am sure I am overthinking things too, only because I am ignant. |
| 02:01.05 | TodWulff | * is running on the edge device (Asus RT-AC66U) |
| 02:01.28 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 02:02.17 | TodWulff | and Asus has taken steps, in the interests of security, to lock down thing on the router. I am having to learn a lot more than I had initially thought I was going to, when I took up this endeavor. |
| 02:03.40 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 02:03.48 | TodWulff | I had a working instantiation when all my sip clients were on the lan. But I just got an external DID and when trying to open things up for WAN connectivity, I got in a flat spin and suspect that I will turn it into a smoking hole in the ground. |
| 02:04.25 | *** join/#asterisk TheSin (~TheSin@gateway.bluefalls.ca) |
| 02:04.37 | Penguin | Bind asterisk to the router's private gateway address and then configure it as you would when it is behind a NAT. |
| 02:05.54 | [TK]D-Fender | If * has a public IP, then it doesn't need to look it up |
| 02:06.00 | Penguin | If you have phones on the public internet, you'll also have to forward the pertinent ports. |
| 02:06.34 | [TK]D-Fender | And since it IS the edge router there is you have nothing to forward... only allow input on |
| 02:06.42 | [TK]D-Fender | which is 5060, and your rtp.conf range |
| 02:06.43 | TodWulff | the problems lie in the chains - i.e. input vs. forward vs. output vs. pre/post routing. given that the * is on the firewall device, it is not as simple as port forwarding and what not. I am rethinking my original decision of putting it on the router. I may just spool up a VM and go that way. |
| 02:06.44 | [TK]D-Fender | all UDP |
| 02:07.00 | [TK]D-Fender | NO FORWARDING |
| 02:07.23 | [TK]D-Fender | Check that your INPUT policy allows SIP+RTP |
| 02:07.43 | TodWulff | Yeah, I may be missing RTP all together, I think. |
| 02:08.05 | TodWulff | I've focused on sip only and that may be one of my hurdles. |
| 02:08.22 | TheSin | what does "dahdi: Span WCT1/0 does not support setting type." mean and why don't I have a span |
| 02:08.24 | Penguin | SIP is only signaling. RTP is where your audio is. |
| 02:09.21 | TodWulff | yeah, I suspect that might be part of the issue. I did have it where a call would connect but no audio pass through. got to overthinking things and effed it all up. just reset iptables and am going to start from scratch. |
| 02:09.55 | Penguin | The INPUT chain is for any packets destined for the box where you're issuing the iptables commands. |
| 02:10.24 | Penguin | That's the one that allows or disallows things to reach asterisk. |
| 02:10.37 | TodWulff | if I bind * to my external ip, does it have to be an ip or can it be a fqdn? |
| 02:11.14 | TodWulff | I would like the latter, given my isp doesn't issue static ips w/o a business account. |
| 02:12.14 | Penguin | If asterisk would even accept a hostname in the bindaddr field, you would bind to whatever address the name resolves to at the time chan_sip loads up. |
| 02:12.42 | Penguin | If for some reason your IP address would change, you'd be binding the wrong address until the next time you load chan_sip. |
| 02:13.20 | TodWulff | and with * being inside of the firewall, on an internal ip - i.e. 192.168.1.1 - I find that I have to DNAT and SNAT shites - it is fugly - lol |
| 02:13.28 | TodWulff | yeah, good point Penguin. |
| 02:13.38 | TheSin | is there a pay support I can call at this time? |
| 02:14.23 | TodWulff | ok, I am out. TheSin sounds to have a time-sensitive need. |
| 02:14.52 | TodWulff | goes into lurker mode. |
| 02:15.12 | TodWulff | thanks, btw, [TK]D-Fender and Penguin |
| 02:16.04 | [TK]D-Fender | [22:10]TodWulffif I bind * to my external ip, does it have to be an ip or can it be a fqdn? <- * binds to an IP. |
| 02:16.15 | [TK]D-Fender | FQDN is just a way to resolve to an IP |
| 02:17.24 | [TK]D-Fender | NExt you need to make sure your PEERS are set right |
| 02:18.23 | KNERD | Apteryx: did you look in "make menuselect" then? |
| 02:18.35 | TodWulff | PEERS being lan devices, yes, [TK]D-Fender? |
| 02:19.14 | [TK]D-Fender | being the SIP devices you define in sip.conf for the things * is actually going to talk to. |
| 02:19.16 | TodWulff | i.e. ATAs, IP phones, etc. on the LAN or my itsps out in the cloud |
| 02:19.22 | TodWulff | ah, yeah. |
| 02:19.31 | TodWulff | all of the above. |
| 02:19.38 | [TK]D-Fender | Get those wrong and you in for 1-way audio as well |
| 02:19.44 | [TK]D-Fender | NO REINVITES. |
| 02:19.48 | [TK]D-Fender | directmedia=no <- |
| 02:19.53 | TodWulff | nods |
| 02:20.29 | TodWulff | nat = yes, qualify = yes <-- OK? |
| 02:20.33 | [TK]D-Fender | Go make sure of those bits... and when you actually want to see what's going on... show us a failed call with SIP DEBUG enabled |
| 02:20.36 | [TK]D-Fender | "sip set debug on" |
| 02:21.07 | TodWulff | copy that. thank you. |
| 02:21.32 | TheSin | no one here can help with dahdi at all? |
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| 02:24.22 | [TK]D-Fender | TheSin: APSTEBIN yrou configs, dahdi status dumps, etc. |
| 02:24.25 | [TK]D-Fender | ~pb |
| 02:24.25 | 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. |
| 02:24.27 | [TK]D-Fender | ^^^^ |
| 02:25.37 | TheSin | I have nothing for configs atm cause the error I posted is making it so I no longer have a span |
| 02:25.54 | TheSin | so dahdi_genconf just blanked eveything |
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| 02:32.29 | [TK]D-Fender | forget genconf. |
| 02:32.48 | [TK]D-Fender | Make then yourself |
| 02:32.51 | [TK]D-Fender | them* |
| 02:33.24 | TheSin | I just down graded to 2.7 form 2.10.0 and it's all working |
| 02:33.27 | TheSin | genconf is right |
| 02:33.31 | TheSin | span and everythign is there |
| 02:34.02 | TheSin | that is messed up |
| 02:34.13 | TheSin | I'll call digium in the morn on it |
| 02:34.24 | TheSin | ther eis somethign wrong with 2.10.0 for my HW for sure |
| 02:36.50 | [TK]D-Fender | What is 2.7? |
| 02:37.04 | TheSin | dahdi |
| 02:37.29 | TheSin | if I use the 2.10.0 kernel modules I get that weird msg on boot in dmesg |
| 02:37.44 | TheSin | I reinstalled the kernel modules form 2.7 and all working |
| 02:37.45 | [TK]D-Fender | hrm |
| 02:38.10 | TheSin | maybe compiler? I know debian just updated the compiler, maybe a bin incompat with the libs? |
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| 02:47.39 | Apteryx | KNERD: What h264 should I check in make menuselect? There is no h264 related option in the "Codec Translators" section, and I already built "format_h264" from the "Format Interpretors" section. |
| 02:50.35 | Apteryx | If I understood from a mailing list discussion, there is no codec_h264.so. However, loading format_h264.so should enable some kind of passthrough, am I right? |
| 02:52.13 | Apteryx | Is loading the format_h264.so and setting allow=h264 in pjsip.conf enough to enable h264 passthrough support? Still getting the WARNING[1093]: channel.c:834 ast_best_codec: Don't know any of (h264) formats message. |
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| 03:10.05 | Apteryx | Ok. Gave up about making h264 and Asterisk 12 + pjsip work. |
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| 03:10.55 | Apteryx | My next attempt: getting opus passthrough working with this setup. Is it even possible with trunk Apteryx 12.5 and chan_pjsip? |
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| 03:33.25 | Apteryx | opus pass through is not working in Asterisk 12. I'm using: disallow=all, then allow=opus |
| 03:34.01 | Apteryx | I'm using Jitsi and Bria agents, which both supports Opus, and I have the res_format_attr_opus.so module loaded. |
| 03:34.17 | Apteryx | When I try a call, I get: Media None |
| 03:35.06 | Apteryx | Instant hangup. |
| 03:35.31 | Apteryx | If I use ulaw instead, it works flawlessly. |
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| 03:57.14 | Apteryx | My opus pass through is not working because Jitsi is named "opus" while Bria's is name "OPUS - Full HD". Silly thing. |
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| 06:17.34 | ovoshlook | Hello I need to do some dialplan after called party answer to call. Does Dial provide this functionality with G([[context^]exten^]priority). I mean that I want to call dialplan function when called party onhook. |
| 06:17.50 | ovoshlook | ? |
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| 07:47.28 | madd | Hello |
| 07:52.47 | madd | attention: I will take n00p-speak because I didn't know the right terms. I want to configure "blinklights" on the phone at the info-point when other phones ringing. If no one takes "other phones" then the info-point guy should be able to take this call. |
| 07:53.40 | madd | i /could/ make a call-group, but that would not be the same |
| 07:58.27 | madd | i just want a sign that someone is call - and the possibility to take over a call |
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| 08:14.34 | wdoekes | madd: *n00b*-speak. you want BLF and "hints" (in asterisk) |
| 08:15.48 | wdoekes | https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints |
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| 08:19.59 | madd | wdoekes: thx :-) |
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| 10:13.51 | Zedax | what can cause a sucessful call with srtp to not have sound on both ends upon answering? without srtp it works perfectly |
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| 10:44.45 | phix | Zedax: Some srtp implementations don't work with asterisk, like snom, I have never got srtp to work with them |
| 10:47.31 | Zedax | phix: i'm using linphone :/ |
| 10:47.43 | phix | hmmmmm |
| 10:47.50 | phix | no idea |
| 10:48.11 | phix | brb, wasabi flavoured dried peas are calling me |
| 10:52.10 | Zedax | i get |
| 10:52.13 | Zedax | message: bandwidth usage for call [04A5B868]: audio=[d=0,0,u=1,1] video=[d=0,0,u=0,0] kbit/sec message: Thread processing load: audio=0,000000video=0,000000 message: ms_quality_indicator_update_local(): no packet received since last call message: Changing [server] [INVITE] transaction [065868D8], from state [ACCEPTED] to [TERMINATED] |
| 10:52.27 | Zedax | it automatically hangs after some secs |
| 10:52.41 | Zedax | so it's not transitting anything on srtp it seems.. |
| 10:52.47 | Zedax | transmitting* |
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| 10:57.15 | pbxman | afternoon |
| 10:57.39 | Zedax | hi |
| 11:02.15 | *** join/#asterisk infobot (ibot@rikers.org) |
| 11:02.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.0 (2014/09/24), 1.8.31.0 (2014/09/24); Standard: Asterisk 12.6.0 (2014/09/24); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 11:08.39 | *** join/#asterisk catphish_ (~catphish@unaffiliated/catphish) |
| 11:23.39 | *** join/#asterisk infobot (ibot@rikers.org) |
| 11:23.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.0 (2014/09/24), 1.8.31.0 (2014/09/24); Standard: Asterisk 12.6.0 (2014/09/24); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 11:24.15 | catphish_ | this is the issue i have, my the time i get a call from the customer and look at the sip packets, they're all originating from a public IP, and the phones are re-regisering correctly |
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| 11:25.09 | catphish_ | maybe i'll have wireshark watch for RFC1918 packets for a while |
| 11:25.30 | catphish_ | but i wondered if there was anything in asterisk that might cause the internal IP to be used even when nat=yes |
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| 12:16.38 | Zedax | blink works with asterisk's srtp? |
| 12:35.49 | Aamit | Zedax, Jitsi is bet one |
| 12:35.56 | Aamit | best |
| 12:36.56 | Zedax | i've tried with jitsi and it doesn't seem to work, i enable/force srtp in config, but then when trying to call asterisk logs say the device is not using srtp |
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| 13:02.06 | Zedax | i get ***WARNING[6699][C-00000019] chan_sip.c: Matched device setup to use SRTP, but request was not! *** with jitsi, even if srtp is enabled |
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| 13:04.22 | pecanha | Hello, I'm using monitor to record calls redirected to queues. But I would like to record every extension call. Which is the best approach (including avoiding duplicate records with queues)? |
| 13:05.36 | [TK]D-Fender | just use monitor on all your extensions then |
| 13:06.48 | pecanha | [TK]D-Fender: So I would remove monitor from queues e let monitor working on the extensions only? |
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| 13:08.10 | jww | Hello. |
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| 13:09.32 | [TK]D-Fender | pecanha: just let the initial point in your dialplan initiate it. |
| 13:09.34 | jww | I just get my hands on a debian 4 server running asterisk 1.2.13 . I guess I should update quickly. |
| 13:10.11 | jww | anything I could say to the customer about newers version of asterisk ? |
| 13:10.28 | pecanha | [TK]D-Fender: the context used? |
| 13:11.23 | [TK]D-Fender | jww: "Fixed dozens and dozens of security holes that could get your server raped". "Fixed hundreds of things that can cause your performance to drop and your server to crash". "Oh, and there's new good stuff" |
| 13:11.49 | [TK]D-Fender | jww: Basically the stuff you should already know to tell them without having to ask. |
| 13:12.06 | [TK]D-Fender | pecanha: Extensions are in contexts... |
| 13:12.35 | pecanha | [TK]D-Fender: yes I know, but I didn't get where should I put instead on every extension. |
| 13:12.49 | coppice | [TK]D-Fender: aren't these lists always "101 things which can cause your performance to drop"? |
| 13:14.19 | [TK]D-Fender | coppice: Arbitrary number. Photo is suggested serving. Some settling of the package may have occured. Sold by weight, not volume. Consult a physician. OAC. |
| 13:14.23 | jww | [TK]D-Fender: it was more about the new good stuff. |
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| 13:15.09 | [TK]D-Fender | jww: G.722 support, T.38 faxing support. Conferencing without DAHDI. SIP via TCP. Encryption. 50 million other things. |
| 13:15.58 | [TK]D-Fender | pecanha: there is no "instead". recording is triggered in the dialplan. So like I said... do it on every extension that could possibly need it at the starting point |
| 13:16.48 | coppice | jww: everyone like new good stuff, and closes their eyes to the new bad stuff |
| 13:19.04 | jww | umm that current debian asterisk version is strange. is there something I miss about versioning ? it could not be 1.8.13 only |
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| 13:19.56 | [TK]D-Fender | Yes it can. |
| 13:20.09 | [TK]D-Fender | GLACIERS move faster than Debian |
| 13:20.51 | coppice | Congress moves faster than Debian |
| 13:21.54 | jww | I would say it's a bad start for a job. I feel I gonna have hard time on this one. |
| 13:22.39 | jww | thanks for your answers guys. |
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| 13:24.44 | Penguin | Show them the list of security fixes over the last 10 years. Then say, "This is why we need to update it immediately." |
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| 13:32.10 | jww | Penguin: I'm writing them a whole story, where [TK]D-Fender is the hero. |
| 13:32.41 | [TK]D-Fender | I want a princess. Make sure I get one. |
| 13:33.28 | jww | :) |
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| 13:47.38 | file | meep meep |
| 13:48.38 | file | SIP all the things! |
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| 13:57.49 | bananapie | Anyone know why the beep sound file that comes with asterisk sounds has a bit of white noise and sounds very different than the output of 'play -n -b 16 -c 1 -r 8000 synth 0.35 sin 700 vol 0.30' ? |
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| 14:03.50 | newtonr | bananapie, did you try playing it in a few different formats to see if it is the same with all of them? |
| 14:05.19 | newtonr | bananapie, err, ignore me, I'm sleepy and was thinking of the language sets that we provide various encodings for. |
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| 14:22.20 | Zedax | should asterisk be listening on udp for rtp when doing netstat? |
| 14:22.42 | [TK]D-Fender | RTP is *only* UDP |
| 14:23.40 | Zedax | i know but |
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| 14:24.16 | Zedax | should be asterisk listening to any of the 10000:20000 ports on udp? |
| 14:24.31 | Qwell | Zedax: Only during a call. |
| 14:24.58 | Zedax | Qwell: if i enable srtp it's not listening on udp :S |
| 14:25.43 | Zedax | i have the ports opened on iptables and the clinets send the rtp stream, but asterisk is not getting the stream |
| 14:25.59 | *** join/#asterisk orn (~orn@2a01:8280:101:0:e5c0:8867:7956:b59a) |
| 14:27.13 | orn | has specifying the realm for registration/invites changed recently in asterisk? (i'm using 11.7.0) What I've found in wikis to specify realm (both realm=x and putting it in the registration string itself) doesn't seem to be working -- asterisk itself is always the realm |
| 14:28.40 | orn | [general] has register => udp://username@my.realm.domain:password@my.register.domain/username |
| 14:28.45 | orn | where realm and register domain is the same domain |
| 14:29.03 | orn | and then I also have realm=my.realm.domain in the context for the peer |
| 14:29.31 | orn | also, is realm= only for [general] ? |
| 14:30.16 | orn | the auth realm seems to be fine, it's just when I'm sending invites |
| 14:33.03 | [TK]D-Fender | peer has nothing to do right a register statement |
| 14:33.11 | [TK]D-Fender | s/right/with/ |
| 14:34.36 | orn | never mind -- i figured out what i was doing wrong and it was totally derp :) problem not even on * |
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| 15:03.58 | karamell | It's possible to define multiple anonymous endpoints in pjsip? |
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| 15:07.23 | [TK]D-Fender | wonders how you "define" something that is anonymous ... let alone multiple of them. |
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| 15:10.28 | WIMPy | With multiple bindaddresses, for example? |
| 15:11.00 | karamell | Yes, with multiple transports. |
| 15:16.26 | newtonr | [TK]D-Fender, to allow anonymous inbound calls with PJSIP you need to define an [anonymous] endpoint. |
| 15:16.56 | [TK]D-Fender | newtonr: Multiple anonymous starts sounding kinda "specific" ;) |
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| 15:18.26 | karamell | Anyway, it is possible without patching? |
| 15:20.31 | newtonr | karamell, I'm not sure exactly what you mean yet. Do you want to accept anonymous calls over, for example, udp and ws transports? |
| 15:20.42 | karamell | yes |
| 15:21.05 | karamell | udp and tls with srtp |
| 15:21.11 | file | there can be only one anonymous endpoint |
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| 15:25.52 | newtonr | karamell, and yes , just define the anonymous endpoint and inbound calls over any transport that you have should work |
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| 15:34.42 | file | falls over |
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| 15:39.42 | kpettit | I'm using swift. And doing Swift(Hello world) sort of stuff works fine. But I need to figure out a way to pull data from a URL that returns pure text. Something like Swift(www.website.com/statustext). Any ideas how I can do that or give a variable to swift? |
| 15:41.30 | Qwell | func_curl |
| 15:42.06 | kpettit | Qwell: Oh very cool. DIdn't know about that one. I'm checking into it.... |
| 15:42.43 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CURL |
| 15:43.24 | kpettit | I'm giving that a try right now. Thanks |
| 15:48.39 | skrusty | does ExecIf work with != (not equals) comparisons? |
| 15:49.04 | skrusty | stupid question, but just can't see anything that says it does |
| 15:50.16 | Penguin | It will work. |
| 15:50.27 | skrusty | cheers |
| 15:50.45 | Penguin | $[ xx != xy ] ... |
| 15:51.12 | skrusty | yeah |
| 15:55.20 | kpettit | Qwell: apparently I don't have func_curl. I'm checking asterisk "make menuconfig" and I just see XXX func_curl. I'm assuming it can't add it for some reason. I've got curl libs so I'm not sure what's going on there |
| 15:55.34 | Qwell | you need the dev package |
| 15:55.52 | Qwell | https://wiki.asterisk.org/wiki/display/AST/cURL#cURL-DependenciesandInstallation |
| 15:56.01 | kpettit | thanks. trying that now.. |
| 15:56.04 | Qwell | don't follow any of the res_config_curl bits, obviously |
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| 16:32.38 | kpettit | Qwell: Thanks again for the help. I got it working. Had to recompile but after that it worked great! |
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| 16:48.20 | MasterChen | good morning |
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| 16:49.32 | MasterChen | I am currently getting a "no path to translate from SIP to IAX2" error for a specific peer.I googledand say posts about codec issues, but I don't believe that to be the issue. Ideas? |
| 16:50.26 | pecanha | Hello, how can I set default mixmonitor path before I call " Queue" command? |
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| 17:06.43 | [TK]D-Fender | MasterChen: Show us the call |
| 17:06.45 | [TK]D-Fender | ~pb |
| 17:06.45 | 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. |
| 17:06.46 | [TK]D-Fender | ^^^ |
| 17:07.06 | drewbug | Let's say you have a phone number with Google Voice (or Twilio, or Verizon). When you "port" that number to another provider, what actually happens? What would it take to *be* a provider, and own your number yourself? Is that even a meaningful concept? |
| 17:09.40 | [TK]D-Fender | Go google up "becoming a CLEC" |
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| 17:11.18 | drewbug | [TK]D-Fender: Will do, thanks! |
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| 17:43.27 | justdave | anyone have experience with asterisk services in a VM with 100 or so concurrent users? I know it always used to suck to run asterisk in a VM, but VM technology is better than it used to be these days. |
| 17:43.54 | justdave | I run my home system in a VM, and it works fine, but I never have more than 2 concurrent calls :) |
| 17:44.55 | [TK]D-Fender | tons of people run large numbers of vms for this |
| 17:45.03 | [TK]D-Fender | make sure you give it suitable system resources |
| 17:45.13 | Penguin | If you have enough CPU and RAM assigned to the VM, I wouldn't expect trouble. |
| 17:45.47 | [TK]D-Fender | I've heard better things about ESXi & KVM than I have for Xen |
| 17:45.56 | Penguin | Call quality degrades when system resources and network resources get exhausted. |
| 17:46.18 | justdave | yeah, would be ESX here |
| 17:46.58 | justdave | (my home system is actually Fusion, but that's basically ESX without the fancy admin utilities) |
| 17:50.17 | justdave | our office hardware is hitting end of warranty and was hoping since we have existing ESX hosts available that we could use it instead of buying more hardware :) So that's promising to hear. |
| 17:56.36 | voip | morning |
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| 18:29.51 | karamell | Asterisk crashing with "asterisk: symbol lookup error: /usr/local/lib/asterisk/modules/res_pjsip.so: undefined symbol: pjsip_tls_transport_start2" after adding tls transport to pjsip.conf. Asterisk 12.6.0, pjsip latest from github asterisk/pjproject. |
| 18:31.19 | file | mjordan, huh |
| 18:31.40 | newtonr | karamell, Can you pastebin your pjsip.conf ? |
| 18:32.01 | mjordan | huh |
| 18:32.07 | mjordan | I got that as well at one point. |
| 18:32.21 | mjordan | karamell: do you have IPv6 enabled in your pjproject? |
| 18:32.34 | file | it certainly exists |
| 18:32.54 | mjordan | yeah |
| 18:33.04 | mjordan | I think I ran into this when I enabled IPv6 in pjproject, but then did not re-build asterisk |
| 18:33.19 | mjordan | once I rebuilt Asterisk after enabling IPv6 in pjproject, things worked dandy |
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| 18:34.32 | karamell | newtonr, http://pastebin.com/hfQmPdFp |
| 18:34.44 | karamell | mjordan, how to check it? |
| 18:35.42 | mjordan | well, you can't really "check it". The fact that you are getting this error pretty much means that Asterisk was not compiled against a version of pjproject that supported IPv6 |
| 18:35.51 | mjordan | because that symbol that is missing is only present when IPv6 is enabled |
| 18:36.06 | mjordan | I'd re-build asterisk |
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| 19:09.40 | karamell | ok works after rebuilding, thanks. |
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| 20:01.43 | rrittgarn | Caller A calls in and is blind transferred to a specific parking spot by setting PARKINGEXTEN=71 (for example), this lights up the BLF on the phones and all is happy until caller B calls in, and less qualified receptionist blind transfers Caller B to 71 as well... Now Caller A and Caller B are talking without an easy way to retrieve them... |
| 20:01.49 | rrittgarn | PB of dialplan: http://pastebin.com/yPpUVKMc |
| 20:02.08 | rrittgarn | Is there any way to prevent this (aside from more qualified end users) while maintaining the pick your parking spot functionality? |
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| 21:18.47 | *** part/#asterisk Defraz (~Defraz@205.185.92.236) |
| 21:23.49 | *** join/#asterisk yeahrock (~yeah@p578E7424.dip0.t-ipconnect.de) |
| 21:24.53 | yeahrock | hi everyone, is there somebody willing to help me with getting Google Voice to work? |
| 21:25.13 | yeahrock | is https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google the latest current recommened documentation? |
| 21:26.36 | paulc | rrittgarn: You get any joy with your parking question? |
| 21:28.03 | newtonr | yeahrock, Google dropped support for xmpp with GV, so probably not. That is the latest documentation on wiki.asterisk.org. No idea if things still work or not there. |
| 21:28.33 | newtonr | yeahrock, I've heard some say they can still call through GV from Asterisk, and others say it stopped working. |
| 21:28.52 | yeahrock | newtonr: thanks. so, in the future there will be no way to use asterisk with GV? |
| 21:29.06 | yeahrock | or is it a temporary situation? |
| 21:29.23 | newtonr | yeahrock, I can't forsee the future. No idea. You could probably search Google forums. :) |
| 21:30.11 | yeahrock | ok, thanks a bunch |
| 21:30.14 | yeahrock | how sad :( |
| 21:33.11 | newtonr | yeahrock, np |
| 21:34.04 | yeahrock | then (forgive the off-topicness) which US based sip provider is recommended where I can port my landline number to? (that was my main purpose for trying to get google voice to work) |
| 21:34.27 | [TK]D-Fender | voip.ms |
| 21:34.30 | [TK]D-Fender | vitelity.net |
| 21:34.35 | [TK]D-Fender | flowroute.com |
| 21:34.36 | [TK]D-Fender | etc |
| 21:37.04 | Penguin | I'm still using Google Voice with XMPP on *1.8. |
| 21:37.05 | rrittgarn | @paulc, no joy as of yet |
| 21:37.34 | yeahrock | Penguin: any idea if they disable it for new accounts but handle existing ones gracefully? |
| 21:37.45 | Penguin | I have no way to know that. |
| 21:37.56 | yeahrock | ok thx |
| 21:38.04 | Penguin | I just know that I have three accounts configured on my box and they work as of last night. |
| 21:38.15 | Penguin | I haven't been home today to use/test them. |
| 21:39.48 | yeahrock | so, it seems that google are going to drop xmpp support, right? |
| 21:40.24 | yeahrock | so if I get a bunch of XMPP debug messages with messages from the google server that look okay, itâs likely that support having been dropped is not the case for me, hm? |
| 21:40.49 | Penguin | They already dropped support for it. |
| 21:41.03 | yeahrock | I can even see chat messages in my asterisk debug |
| 21:41.05 | *** join/#asterisk jzaw (~jzaw@community-wifi.dzki.co.uk) |
| 21:41.15 | yeahrock | so, *some* xmpp stuff is still working |
| 21:41.18 | Penguin | Nothing has changed for me so far as far as calling to/from GV/asterisk. |
| 21:42.24 | yeahrock | hm, so if I get this right, we use xmpp for session negociation and motif for the actual voice transmission, right? |
| 21:42.34 | Penguin | I don't. |
| 21:42.49 | Penguin | I'm using gtalk/jingle/xmpp. |
| 21:43.07 | Penguin | I'm also using asterisk 1.8. |
| 21:43.14 | yeahrock | i see |
| 21:44.03 | yeahrock | iâm on 11 |
| 21:44.16 | yeahrock | installed as a deb package via wheezy-backports |
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| 22:05.30 | RazaMetaL | hi |
| 22:10.37 | RazaMetaL | i've a trouble with a &$#@ cisco phones, i |
| 22:10.51 | RazaMetaL | i'm moving to sip for using with asterisk |
| 22:11.13 | RazaMetaL | but, some phones can not be upgraded |
| 22:11.14 | RazaMetaL | :( |
| 22:12.06 | RazaMetaL | i've connected one 7911G phone to my laptop that have isc-dhcp server and tftp-hpa servers configured |
| 22:13.04 | RazaMetaL | the phone request dhcp, the server offer dhcp |
| 22:13.17 | RazaMetaL | but the phone does not accept the dhcp ip |
| 22:13.19 | RazaMetaL | :( |
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| 23:19.34 | Kobaz | interesting |
| 23:19.44 | Kobaz | the most outbound calls happen on wednesday or thursday |
| 23:19.52 | Kobaz | for me anyway... |
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| 23:43.36 | Apteryx | Hello :) Could someone give tell me how to turn pjsip debug on? |
| 23:43.53 | Apteryx | I've tried setting debug=yes in my pjsip.conf but it didn't do anything. |
| 23:46.56 | roswell | try issuing 'core set debug 3' in `asterisk -r`. tho this will give you with lots of other lines |
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| 23:58.53 | ruben23 | <PROTECTED> |