| 00:23.00 | *** join/#asterisk bfoote (foobar@eidolon.bnf.net) |
| 00:25.18 | drmessano | updated from what to what? |
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| 02:37.46 | phormulate | hello |
| 02:38.50 | WIMPy | Good morning. |
| 02:39.43 | phormulate | I've been using 11.18, have all dependencies required, debian 7.0... forgot to compile in speex, recompiled and installed and now ice doesn't work?! --enable-uuid to no avail, I can't remember what I did, going nuts compiling on a remote p3 |
| 02:40.36 | phormulate | the internal pjsip stack is standoffish, I compiled it and ice worked perfectly this afternoon, took a nap, recompiled with speex and now ice/stun are a no go |
| 02:40.40 | WIMPy | Is it evailable/enabled then? |
| 02:40.41 | phormulate | google is no help |
| 02:41.15 | phormulate | it isn't in menuselect as it would be in 12+, but everything is |
| 02:41.48 | phormulate | well all but the deprecated stuff |
| 02:42.33 | WIMPy | I guess it makes sense that it's not an Asterisk option any more with pjsip. |
| 02:43.26 | phormulate | that is in 12+, prior in 11.x it is not in the options |
| 02:43.45 | phormulate | but pjsip is in the source, an embedded ver |
| 02:44.02 | phormulate | I'm pulling my beard off looking where it went wrong |
| 02:44.08 | phormulate | everything was working beautifully |
| 02:50.29 | phormulate | well starting from scratch, another 2 hour compileathon |
| 02:50.30 | phormulate | ugh |
| 02:51.12 | WIMPy | Didn't you say PIII? |
| 02:52.04 | WIMPy | Not even the Raspberry Pi too hours. |
| 02:53.30 | phormulate | ya heheh |
| 02:53.54 | phormulate | a remote server, not to mention, on a line currently dropping packets |
| 02:53.55 | phormulate | this is murder |
| 02:54.31 | WIMPy | How is that related to compile time? |
| 02:54.42 | phormulate | entry latency |
| 02:54.44 | phormulate | debugging |
| 02:54.45 | phormulate | etc |
| 02:55.01 | phormulate | wait 10 seconds to get shell control |
| 02:55.02 | phormulate | ugh |
| 02:57.55 | phormulate | I must again summon a beer and a steeled level of patience |
| 02:57.57 | phormulate | what a day |
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| 09:30.37 | jepperl | Hello, i'm trying to set up a simple Asterisk 13.. i got it up and configured so i can register extensions/endpoints to server. My problem is: i (succesfully) registered a SIP trunk provider. But whenever im receiving an INVITE from that provider, my asterisk server immediately responds with 401 Unauthorized, and therefore i can not make any inboun |
| 09:30.37 | jepperl | d calls to my system. In Asterisk 11, i remember setting 'insecure=invite' could resolve this issue, but i can't seem to find such an option for Asterisk 13/pjsip |
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| 09:38.17 | carrar | moof |
| 09:39.11 | phix | foom! |
| 09:39.19 | jepperl | xhip |
| 09:39.57 | phix | lreppej!!!! |
| 09:40.40 | carrar | yummie curry dinner |
| 09:40.50 | phix | Rogan Josh? |
| 09:41.02 | phix | Jal Fargaiids? |
| 09:41.33 | phix | Palak Paneer? |
| 09:41.54 | phix | Vindaloo? |
| 09:42.33 | phix | Butter chicken? |
| 09:42.42 | phix | Chick tikka? |
| 09:44.33 | jepperl | Vindaloo, thank you |
| 09:46.28 | phix | ⤠|
| 09:46.38 | phix | And what else? Garlic cheese Naan bread? |
| 09:46.42 | phix | Somosa? |
| 09:46.49 | jepperl | both |
| 09:46.55 | jepperl | and some lassi aswell |
| 09:46.57 | jepperl | guava |
| 09:47.11 | phix | nom nom |
| 09:47.20 | phix | tastey desert |
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| 09:48.29 | phix | anything else from the Tandoor? |
| 09:49.45 | phix | That best Indian I have has is in AU and UK (alot of Indians in both countries :), mostly pakistany / north india in UK though so different flavours) |
| 09:50.39 | phix | alot of the food in AU is southern india with malasian and other south east asian influences |
| 09:51.39 | phix | and this is relevant as Indians use Asterisk to make international calls :) |
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| 11:06.58 | Darkerr | Hello, I would like to ask, what is the sip parameter name for changing "Outbound reg. retry 403"? |
| 11:09.40 | wdoekes | $ grep retry.*403 configs/sip.conf.sample |
| 11:09.40 | wdoekes | ;register_retry_403=yes ; Treat 403 responses to registrations as if they were |
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| 11:14.39 | Darkerr | oh, thank you. I thought it will be timelimit to wait before retry on 403 response :( |
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| 12:14.24 | aurs | I have bad sound quality on some calls. If I play the call in wireshark with the default settings, it sounds bad there as well, but if I choose "uninterrupted mode" when decoding the rtp stream in the player in wireshark, the sound is ok. Any ideas on the cause of this anyone? (asterisk 1.8.32) |
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| 12:15.24 | aurs | "dropped by jitter buf: 34%" |
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| 13:13.01 | WIMPy | aurs: Your network connection isn't good enough. |
| 13:13.54 | aurs | WIMPy, but it's the same on a capture on the server side as well. And a restart of the sip client (bria) fixes the issue (which is rare in the first place) |
| 13:14.26 | aurs | it only happens on calls via DAHDI though |
| 13:15.58 | WIMPy | You don't get jetter on DAHDI. |
| 13:16.05 | Chainsaw | Jetter? |
| 13:16.38 | WIMPy | Jitter |
| 13:17.57 | Chainsaw | aurs: Bria isn't the best behaved soft phone client in the world. |
| 13:18.20 | Chainsaw | aurs: It has that non-compliant keepalive, and I now have a user that insists on hanging up their first call upon acceptance. iPad-specific. |
| 13:18.21 | aurs | I've captured with tcpdump on the client, and when I play the audio with 50ms jitter buffer in wireshark, it sounds similar to what we hear in the tests, and it drops audio on what sounds like fixed "intervals" |
| 13:18.57 | aurs | Chainsaw, it's on the desktop client for mac we've had this issue |
| 13:19.31 | aurs | the fact that a restart of the softphone solves the issue makes me pretty sure it's a client issue :D |
| 13:19.48 | aurs | but... just wanted to hear if anyone has experienced the same here |
| 13:19.54 | Chainsaw | aurs: Agreed. I'd still consider an upgrade from Asterisk 1.8 (which is in security fixes only) to 11 though. |
| 13:20.14 | Chainsaw | aurs: As if you do find a bug in 1.8 it will not be fixed. |
| 13:20.31 | aurs | Chainsaw, or go all in and move to 13? |
| 13:20.49 | Chainsaw | aurs: Or that. I only have personal experience with 11 at the moment, so I can't vouch for 13. |
| 13:21.11 | aurs | that day will come. But it is not this day! :D |
| 13:23.14 | aurs | it's just strange that all the rtp packets are in the stream, mostly in correct order, and there is soooo many drops. Could it be that it is dropped by jitter buffer between asterisk servers? jb is not enabled in sip.conf though |
| 13:24.20 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:24.36 | aurs | but still... why does restarting the bria client solve it... I'm out of ideas. Hoping that CounterPath can figure something out |
| 13:25.29 | WIMPy | Well, at least chan_sip will only send rtp if it receives any. That might be a cause. |
| 13:26.19 | *** join/#asterisk SupaYoshi (~SupaYoshi@104.223.1.186) |
| 13:26.51 | aurs | I don't think I've seen audio "destroyed" in wireshark like this before though |
| 13:27.06 | aurs | with the voip calls - player function |
| 13:27.29 | aurs | hmm... time to check a "normal" call in wireshark I guess |
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| 13:36.57 | aurs | yeah... that one looks ok in wireshark both with 50ms jb and "uninterrupted mode" |
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| 13:50.12 | *** join/#asterisk Encrypt (~Chuck-nor@AMontsouris-553-1-41-176.w92-151.abo.wanadoo.fr) |
| 13:50.18 | Encrypt | Hello there o/ |
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| 13:52.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:03.55 | WIMPy | Where? |
| 14:05.45 | Encrypt | WIMPy, Here :p |
| 14:07.28 | *** join/#asterisk tzafrir (~tzafrir@2001:7c0:dc15:72:6267:20ff:fe51:8328) |
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| 14:24.04 | Encrypt | So, I would like to set asterisk on a Raspberry Pi |
| 14:24.11 | Encrypt | But I may experiment on my computer first |
| 14:24.19 | Encrypt | Do you know where I should start? |
| 14:24.34 | Encrypt | Any good "tutorial" explaining the basics? |
| 14:24.39 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:24.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:25.02 | Encrypt | Also, I have understood that all asterisk packages are pretty much out of date and asterisk should be built "by hand"? |
| 14:27.10 | WIMPy | ~book |
| 14:27.10 | 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 |
| 14:27.43 | WIMPy | Well, the packages for some Distros, that is. But Usually everything else on that Distros is in a similar state. |
| 14:28.10 | Encrypt | Apparently, the version is 11.13.1 on Raspbian |
| 14:28.34 | WIMPy | And if you want to do it on the RPi, don't make the same mistake I did. Don't use Raspbian. |
| 14:29.05 | Encrypt | WIMPy, Why? |
| 14:29.17 | WIMPy | I got it to do what I want, but it was quite a journey. |
| 14:30.57 | babak | Hi,If 1)we have a shared realtime sip registration databse between multi Asterisk servers or 2) run DUNDI between servers which one is better for call routing ? |
| 14:37.28 | zekoZeko | Encrypt: on RPi you can use asterisk from raspbx |
| 14:37.40 | Encrypt | Yes :) |
| 14:38.30 | zekoZeko | if you don't want to use freepbx and everything else in raspbx just add their sources list and install asterisk11 or asterisk13 package. |
| 14:39.05 | zekoZeko | they generally have the latest versions, I think I'm running 11.18 now on RPi, let me check if it's .19 |
| 14:40.13 | zekoZeko | it's .18 |
| 15:00.27 | Encrypt | zekoZeko, Oh, that would be nice |
| 15:01.04 | Encrypt | Because I plan to reinstall what I currently have on my Raspberry Pi model B (chinese edition, first batch), that is to say my web, mail and printer servers |
| 15:01.43 | Encrypt | zekoZeko, So, it's possible to directly install this package with apt-get? |
| 15:05.06 | Encrypt | Modem reboot, I'll come back right after that |
| 15:09.48 | *** join/#asterisk hfb (~hfb@pool-96-247-47-53.lsanca.dsl-w.verizon.net) |
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| 15:14.09 | Encrypt | I'm back |
| 15:14.47 | Encrypt | zekoZeko, According to the website, the last version is the 13.4.0, right? |
| 15:17.44 | *** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net) |
| 15:22.53 | [TK]D-Fender | That's what it says. It is wrong however |
| 15:22.54 | [TK]D-Fender | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 15:23.09 | [TK]D-Fender | like the topic says..... |
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| 15:31.23 | Encrypt | Ok :) |
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| 16:48.31 | drab | hi, I seem to be running into this problem, but no trace of a solution in any bug report I managed to find: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=783123 |
| 16:49.33 | drab | basically pjsip fails to load with various error messages realated to undefined symbol: ast_sip_presence_xml_create_node |
| 16:50.09 | drab | this is on ubuntu 14.04, asterisk 13 |
| 16:51.30 | drab | in my case asterisk is actually running, but pjsip isn't working and no peers are defined (as per sip show peers) |
| 16:52.05 | [TK]D-Fender | report is old |
| 16:52.11 | [TK]D-Fender | post your own with a current release |
| 16:52.31 | [TK]D-Fender | drabin my case asterisk is actually running, but pjsip isn't working and no peers are defined (as per sip show peers) <- show us this BTW |
| 16:52.33 | [TK]D-Fender | ~pb |
| 16:52.33 | 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. |
| 16:52.34 | [TK]D-Fender | ^^^ |
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| 16:57.52 | drab | [TK]D-Fender: http://pastebin.ca/3104563 |
| 16:58.24 | drab | oh, forgot to add show peers, sorry, updating |
| 16:58.30 | [TK]D-Fender | Looks like you've borked your install of it |
| 16:58.38 | [TK]D-Fender | wrong lib vers probably |
| 16:59.56 | drab | mmmk, I will start over. thanks |
| 17:00.32 | drab | sip show peers |
| 17:00.32 | drab | Name/username Host Dyn Forcerport Comedia ACL Port Status Description |
| 17:00.35 | drab | 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 17:00.38 | drab | doh |
| 17:00.40 | drab | mouse slip, sorry |
| 17:03.16 | [TK]D-Fender | And that is pointless no matter what... |
| 17:03.21 | [TK]D-Fender | that is for CHAN_SIP ONLY |
| 17:13.00 | *** join/#asterisk xpheres (~bodhi64@dslb-088-073-021-077.088.073.pools.vodafone-ip.de) |
| 17:13.09 | xpheres | hello, I have a cisco 7970 behind a nat that does not receive calls and when it calls there is only one way Audio |
| 17:17.00 | xpheres | http://pastebin.com/uEdtMTdc |
| 17:17.08 | xpheres | maybe someone could see what's wrong there |
| 17:19.39 | xpheres | the server is behind a router and the other cisco is under another router |
| 17:19.44 | xpheres | why they do not connect? |
| 17:29.16 | [TK]D-Fender | Contact: <sip:101@192.168.178.25:5070>transmitting #3 (no NAT) to 192.168.178.25:5072: |
| 17:29.23 | [TK]D-Fender | You running a spofphone on * itself? |
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| 17:38.13 | xpheres | I set up NAT on for the extension and the phone configuration file |
| 17:38.15 | xpheres | now I have this |
| 17:38.15 | xpheres | http://pastebin.com/0M2YBh8d |
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| 17:38.37 | xpheres | and the telephone does not register |
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| 18:01.16 | xpheres | http://pastebin.com/0M2YBh8d |
| 18:01.21 | xpheres | please help |
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| 18:08.25 | xpheres | why is this: Retransmitting #6 (NAT) to 192.168.178.25:5072: when I configured the phone to call to port 5070 and just used 5072 as security port? |
| 18:08.31 | xpheres | http://pastebin.com/0M2YBh8d |
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| 19:36.24 | babak | Hi,if there was a shared Realtime db for sip registration for multiple Asterisk servers, is there a way to realize sip phone registered on which Asterisk server ? |
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| 21:06.18 | jordanfeld | Does anyone have suggestions for getting user count for a confbridge via agi ? |
| 21:07.38 | [TK]D-Fender | You don't get it via AGI |
| 21:07.48 | [TK]D-Fender | you get it via something outside. CLI or AMI |
| 21:08.15 | WIMPy | Or via the COONFBRIDGE function? |
| 21:11.59 | jordanfeld | CONFBRIDGE_INFO(parties,bridgeNum) |
| 21:12.08 | jordanfeld | Thanks WIMPy |
| 21:12.17 | jordanfeld | That works via agi |
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| 23:48.44 | Milos | I want to speak SRTP with my Asterisk PBX and my upstream sip provider |
| 23:49.01 | Milos | what do I need to configure? particularly, I should not need to generate a certificate for this purpose |
| 23:49.19 | Milos | just like I don't need to generate a certificate to browse a secure website... |
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| 23:54.44 | Milos | am I right thinking that or what? |
| 23:54.55 | Milos | all the tuts seem to say you need to generate a tls cert |
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| 23:58.56 | cihhan | Hi all! Im trying to build a VoIP system for our office using Asterisk. I have a SIP trunk provider and I can make calls outside; however I cannot get any incoming calls. Any ideas? |
| 23:59.36 | cihhan | And also I would like to ask a question that I am really wondering: Is it a must to have a voip card? I have VoIP phones |
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