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| 03:54.58 | Tod|Home | Hey all. Happy Father's Day to all daddys. I'm running v11.8.1 on a router. I'm seeing this in my console: http://i.imgur.com/hIeRn0R.png I can't figure out where to config to be sure that there aren't extra slashes - have looked in asterisk.conf (directory section), cdr.conf, etc. No joy. Any ideas on what file I need to edit to resolve?. |
| 03:56.04 | Tod|Home | it presents that the .../asterisk//cdr-csv//Master.csv is the issue (note the double slashes). |
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| 04:08.12 | WIMPy | Permissions? |
| 04:09.13 | Tod|Home | was a moron (& non linux savvy)... multiple slashes are treated as a single slash. |
| 04:09.20 | Tod|Home | I manually created the file and all is good. |
| 04:10.12 | Tod|Home | err, created the folder AND the file. |
| 04:10.36 | Tod|Home | thank you, however. |
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| 06:25.03 | tehrabbitt | hey i'm running freepbx and I can't seem to get my inbound calls to go through to my SIP device, however outbound calls work fine |
| 06:25.18 | tehrabbitt | I've forwarded 5060, 10000-20000 to the FreePBX server |
| 06:25.29 | tehrabbitt | outbound calls, I get 2-way audio |
| 06:25.35 | tehrabbitt | inbound calls, I dont' get audio on either side. |
| 06:26.58 | Tod|Home | I am totally ignorant, but do you have stun enabled on the client? I had that kick my butt earlier today. |
| 06:27.06 | tehrabbitt | no stun |
| 06:27.12 | tehrabbitt | or at least i'm not using it |
| 06:27.25 | tehrabbitt | the PAP device STUN is set to "no" |
| 06:27.28 | Tod|Home | yeah, it was enabled by default on one of my soft clients. |
| 06:28.25 | tehrabbitt | I can't figure out why outbound works fine but inbound breaks |
| 06:28.26 | Tod|Home | that is the extent of my knowledge on the matter... ;) sorry to be of no help. |
| 06:28.30 | tehrabbitt | no worries |
| 06:44.03 | tehrabbitt | anyone able to make a test call for me? |
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| 06:46.22 | tehrabbitt | i figured out what the problem is i think |
| 06:46.30 | tehrabbitt | I had to enable advanced nat on pfsense |
| 06:46.34 | tehrabbitt | and set to "STatic Port" |
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| 08:03.36 | TazzNZ | tehrabbitt: did you set externip in the settings ? |
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| 09:49.33 | Ibrahim22 | Hi everyone, I'm trying to make calls between 2 webrtc peers. I have setup their sip accounts with transport=udp,ws,wss. I have set avpf=true and encryption=yes. I have done everything in every guide I could find, but I'm still getting a "process_sdp: Rejecting secure audio stream without encryption details". Every forum post on the first 10 pages |
| 09:49.34 | Ibrahim22 | of Google have no conclusive answer, so I thought I'd ask here. I'm using JsSIP as my JavaScript library. |
| 09:51.09 | Stefan27 | My s300 snom phone registered in asterisk 12 in sip.conf gets error chan_sip.c:10663 process_sdp: We are requesting SRTP for audio, but they responded without it! when it tries to make a call. (But when the same device registers at pjsip.conf there's no error.) 'sip show peer myphones300' shows Encryption:no so why does asterisk request SRTP? |
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| 10:12.01 | beanie | hello, I am having a problem whereby I cannot hear people calling me but they can hear me...please could somebody help :-) |
| 10:12.01 | beanie | <beanie> it's the same on external and internal IP's and I cannot hear freepbx system prompts |
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| 11:11.59 | Chainsaw | file: The fixes for the recent HTTPS DoS have addressed ASTERISK-18345, so you can close that with fix versions 1.8.28.2, 11.10.2 & 12.3.2 |
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| 12:51.04 | phpboy | Hey all, I recently upgrade from 1.8.* to 12.3.0 and it seems it's logged all calls twice in CDR (100% identical)... anybody got an idea of why this would be? |
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| 13:03.08 | malcolmd | Chainsaw: you tested and it worked for you? re: ASTERISK-18345 and the recent security fixes? |
| 13:03.33 | Chainsaw | malcolmd: They're still reeling from your 11.10.1 attempt I'm afraid, it'll be after hours before I get to try again. |
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| 13:03.49 | Chainsaw | malcolmd: But seeing what your fix does and seeing what the original patch did... it's fairly obvious. |
| 13:04.38 | [TK]D-Fender | phpboy: Redundant CDR backend entries |
| 13:04.46 | malcolmd | Chainsaw: whurps ;) drop a note on the issue please once you've confirmed. |
| 13:05.39 | Chainsaw | malcolmd: Will do. |
| 13:07.22 | phpboy | [TK]D-Fender: where would this be? cdr.conf? |
| 13:07.32 | phpboy | it's only since the upgrade :\ |
| 13:07.35 | malcolmd | Chainsaw: many thanks |
| 13:08.39 | phpboy | I only have an entry in cdr_mysql.conf |
| 13:09.59 | [TK]D-Fender | "CDR <tab>" |
| 13:10.08 | [TK]D-Fender | Don't forget ODBC, etc... |
| 13:13.06 | phpboy | [TK]D-Fender: as I've come to expect from you 95% of the time, you are right! |
| 13:13.08 | phpboy | thanks |
| 13:13.23 | [TK]D-Fender | You're welcome |
| 13:13.25 | phpboy | although, it's quite lame that ODBC also logs to CDR when it's not set up? :\ |
| 13:13.45 | phpboy | I'll be an ass |
| 13:13.47 | phpboy | it is :( |
| 13:13.47 | [TK]D-Fender | It shouldn't. |
| 13:14.11 | phpboy | weird that I didn't have this issue with 1.8 but I do with 12.30 |
| 13:14.19 | phpboy | *12.3 |
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| 13:21.02 | [TK]D-Fender | probably noload-ed it in modules.conf or removed. the .so |
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| 13:40.14 | phpboy | <PROTECTED> |
| 13:40.18 | phpboy | thanks a lot man |
| 13:43.06 | [TK]D-Fender | Which was it? |
| 14:05.04 | file | moo |
| 14:06.03 | phpboy | [TK]D-Fender: I commented out the config in cdr_odbc.conf |
| 14:06.10 | phpboy | will clean up properly later |
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| 14:35.33 | Ibrahim22 | Hi, anybody have any tips on solving my websockets problem in which I get 'Rejecting secure audio stream without encryption details' even though I have set everything correctly in my sip.conf |
| 14:39.14 | newtonr | do you have "encryption=yes" for that peer? |
| 14:39.16 | [TK]D-Fender | You'd need to provide real details for us to work with. |
| 14:40.17 | Ibrahim22 | What kind of details? sip.conf? sip debug? |
| 14:41.18 | [TK]D-Fender | Clearly those are the minimum starting point |
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| 14:44.13 | Ibrahim22 | just a moment, im preparing it now on pastebin.com |
| 14:49.48 | Ibrahim22 | http://pastebin.com/uXMhFAdq for sip.conf |
| 14:50.52 | Ibrahim22 | http://pastebin.com/k4EsLePj for sip debug |
| 14:51.33 | Ibrahim22 | i have removed my ip and other real references to my actual install, but the settings are real |
| 14:51.41 | Ibrahim22 | i am running asterisk 12.3.2 |
| 14:52.50 | Ibrahim22 | I have installed libsrtp, which is also recognized by asterisk during make menuselect during last update this afternoon |
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| 14:54.46 | Ibrahim22 | And I am using JsSip.net as my JavaScript library |
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| 15:20.10 | vedic | Hi, I have freshly installed dahdi. After 'make' step, it showed that make was successful and should go ahead with make install. I have done make install but dahdi_cfg -vvvv is giving error: Unable to open master device /dev/dahdi/ctl |
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| 15:23.29 | WIMPy | load it. |
| 15:23.36 | WIMPy | modprobe dahdi |
| 15:23.46 | WIMPy | And the driver you need for your card as well. |
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| 15:31.31 | vedic | WIMPy: got dahdi_cfg -vvvv working |
| 15:31.47 | vedic | WIMPy: I had not changed user and group for dahdi config file |
| 15:32.03 | vedic | WIMPy: But device driver for TDM card is not giving error |
| 15:37.38 | vedic | WIMPy: I am using sangoma wanpipe driver and its giving error: af_wanpipe_src.c 1740:33:error: macro "sk_for_each" requires 3 arguments, but only 2 given |
| 15:38.19 | tuxx- | vedic: sangoma is releasing a new wanpipe driver specifically for that compilation error |
| 15:39.15 | vedic | tuxx: Do you know from where to get that? I just download the driver from here: http://wiki.sangoma.com/wanpipe-linux-drivers |
| 15:39.31 | tuxx- | its still in QA, so i hope within a couple of days/weeks. |
| 15:40.08 | vedic | tuxx: Ah! thats very bad |
| 15:40.22 | vedic | I see people posting this error since early January 2014 |
| 15:40.36 | tuxx- | you can just add the `node` parameter to the sk_for_each function in the af_wanpipe_src.c files |
| 15:40.58 | tuxx- | it will compile after that :) |
| 15:41.09 | tuxx- | i think i even have a patch somewhere, lemme check |
| 15:41.09 | vedic | tuxx: Will the older version of dahdi work or older version of wanpipe? |
| 15:41.15 | tuxx- | im not sure of that vedic |
| 15:41.48 | vedic | I think I will have to stay live with patch then |
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| 16:00.28 | vedic | tuxx: Any other way then the patch? For me previous series will also do fine |
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| 16:00.50 | vedic | tuxx: I just need to run asterisk today. May be next month when I will run production I can get new drivers |
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| 16:14.30 | vedic | tuxx: ftp://ftp.sangoma.com/linux/custom/7.0/ |
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| 16:39.30 | rrittgarn | ~book |
| 16:39.31 | 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 |
| 16:39.39 | rrittgarn | (for my own linking purposes) |
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| 16:48.44 | tuxx- | vedic: sorry, was commuting home. Anyway, if the old drivers work for you now i'd suggest get those :) |
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| 20:02.39 | pringlescan | hello all, I'm new to dialplans and am having an issue. I need to conditionally do something when the caller presses 0. if a variable is set to -1, this is a no op, and nothing should change state wise (they're in a queue)… if the variable is not -1, it needs to execute a macro. no matter what I do, even a noOp or a verbose, the call gets disconnected on 0… any ideas? |
| 20:03.59 | [TK]D-Fender | "core show application Gotoif" <- |
| 20:08.13 | Dovid | Is there anyone here from Germany? |
| 20:08.46 | WIMPy | Yes, and even more in #asterisk-de |
| 20:09.31 | WIMPy | Apart from that, |
| 20:09.36 | WIMPy | ~polls |
| 20:09.36 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 20:10.05 | Dovid | WIMPy: Are you in de? |
| 20:10.19 | WIMPy | yes |
| 20:10.25 | Dovid | may i Pm? |
| 20:10.48 | WIMPy | Did you read what infobot wrote? |
| 20:11.34 | Dovid | WIMPy: I need a test call done from a German Mobile phone. I have special routign for Mobile and PayPhone and nee to see if I set it up right in my Asterisk system |
| 20:13.07 | WIMPy | What kind of number? |
| 20:13.13 | Dovid | locla number |
| 20:13.22 | WIMPy | checks his balance. |
| 20:13.42 | Dovid | WIMPy: I pm'd the number |
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| 21:05.23 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 21:46.43 | TazzNZ | that was a quick trip home Dovid :) |
| 21:47.17 | Dovid | TazzNZ: Didn't notice my computer decided to re-connect. gona try this once more ;) |
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| 21:55.51 | mbrit | hi people, im using an asterisk server to communicate using a custom softphone, i've noticed that i can't receive video because asterisk sends my packets to the port i'm using as a source and not to the one specified in the SDP, do i need to configure something to avoid that behavior? |
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| 22:09.13 | ipengineer | Is there a way from the CLI to list all channels with mixmonitor running? |
| 22:22.14 | TazzNZ | ipengineer: not that I know of |
| 22:22.18 | TazzNZ | you will have to script that |
| 22:22.58 | ipengineer | TazzNZ: That is no problem. Is there a command that shows all channels and if they are currently being monitored or how would I access that data |
| 22:24.12 | TazzNZ | sip show channels, then sip show channel <chan_id>, then mixmonitor list <owner_channel_id> |
| 22:24.32 | TazzNZ | will show something like this |
| 22:24.34 | TazzNZ | gsr1sip01*CLI> mixmonitor list SIP/90072-00002b61 |
| 22:24.36 | TazzNZ | MixMonitor IDFileReceive file Transmit File |
| 22:24.38 | TazzNZ | ========================================================================= |
| 22:24.57 | ipengineer | Ahh. Ok Makes sense |
| 22:25.00 | ipengineer | Thanks |
| 22:25.29 | TazzNZ | this will show something like |
| 22:25.45 | TazzNZ | mixmonitor list SIP/94336-00002b71 |
| 22:25.47 | TazzNZ | MixMonitor IDFileReceive file Transmit File |
| 22:25.49 | TazzNZ | ========================================================================= |
| 22:25.51 | TazzNZ | 0x7f73652a0f60 |
| 22:25.53 | TazzNZ | whoops |
| 22:26.00 | TazzNZ | the ID got cut off |
| 22:26.06 | TazzNZ | sorry - filename |
| 22:26.16 | TazzNZ | but you get the idea |
| 22:26.21 | ipengineer | yea I do thanks |
| 22:26.55 | jmmills | Does anyone have any links to benchmarks on Asterisk running on bare metal vs lxc vs vmware? |
| 22:27.12 | TazzNZ | jmmills: that question is way to general |
| 22:27.25 | TazzNZ | and, tbh, I wouldn't run it inside a VM |
| 22:27.33 | TazzNZ | but that is me |
| 22:28.02 | jmmills | I'm just trying not to repeat work, I have some IT manager that is trying to say I can run this deployment in virts, I'm doing 40k calls a day, and I want to shove a benchmark in his face |
| 22:28.06 | cusco | what is lxc ? |
| 22:28.10 | jmmills | linux container |
| 22:28.20 | cusco | and baremetal ? |
| 22:28.24 | jmmills | like a chroot in a kernel |
| 22:28.27 | cusco | ah |
| 22:28.36 | jmmills | bare metal == running directly on the host kernel |
| 22:28.39 | ipengineer | We run on xenserver and have around 400 active calls peak per day |
| 22:28.50 | cusco | in lxc the kernel is the same in case of a chroot |
| 22:28.50 | jmmills | I'm doing about 40k calls a day |
| 22:28.53 | TazzNZ | jmmills: are you doing direct media ? |
| 22:29.01 | jmmills | I'm not sure what my peak concurrent is right now |
| 22:29.22 | jmmills | TazzNZ: G729 and G711 from handsets to asterisk |
| 22:29.26 | jmmills | multi-tenant pbx |
| 22:29.27 | cusco | wonders about how many he's doing |
| 22:29.36 | jmmills | so no transcoding at that level |
| 22:29.53 | jmmills | uplinks via sip to some audiocodes hardware that distributes over PRIs |
| 22:30.10 | jmmills | I guess one could say the audiocodes hw acts as an SBC |
| 22:30.18 | TazzNZ | jmmills: yeah - you *could* in theory do it in VM's |
| 22:30.41 | jmmills | It's a horribly designed system, a bunch of agi (not fastagi) |
| 22:30.44 | TazzNZ | but the problem with VM's are time slicing |
| 22:30.47 | jmmills | I don't want it on a vm |
| 22:31.00 | jmmills | I'm trying to push for bare metal running inside docker as a migration scenario |
| 22:31.12 | TazzNZ | so, when you are dealing with *anything* real time (like....funny that....RTP) a VM is not a good idea |
| 22:31.17 | jmmills | that way I can push it around like it was on a MaaS |
| 22:31.29 | jmmills | there is definitely a lot of media going on |
| 22:31.45 | jmmills | just not a lot of transcoding from what I'm seeing |
| 22:31.49 | TazzNZ | yeah - all your * box is doing is "flipping" the packet on the NIC |
| 22:31.50 | cusco | well if a call is a line in cdr, i'm on 25.000 today |
| 22:31.58 | cusco | so thats about 12.000 calls :p |
| 22:32.17 | jmmills | cusco: how many registrations? |
| 22:32.19 | jmmills | i.e. sip users |
| 22:32.24 | cusco | how plenty |
| 22:32.31 | jmmills | while we are comparing muscles :) |
| 22:32.31 | cusco | counts |
| 22:32.41 | cusco | no, we're weak |
| 22:32.54 | cusco | 128 |
| 22:32.56 | cusco | atm |
| 22:32.56 | TazzNZ | flex's......yeah....ok.......maybe not :) |
| 22:33.47 | cusco | and you? |
| 22:33.49 | jmmills | I've got some plans to do some cool stuff, but currently I'm just trying to get this platform to a place where it's not so annoying to manage |
| 22:34.05 | jmmills | ~70 active channels perl host |
| 22:34.06 | jmmills | per host |
| 22:34.12 | jmmills | 5 production hosts |
| 22:34.17 | jmmills | just doing asterisk |
| 22:34.35 | TazzNZ | jmmills: nice |
| 22:34.47 | TazzNZ | we doing 24k of calls/day on our office * |
| 22:35.01 | jmmills | 7 in total, federated using dundi, with kamailio running as a proxy/packet mutator between a couple audiocodes mediant 2000s |
| 22:35.07 | jmmills | that's not bad |
| 22:35.26 | TazzNZ | we have another cluster of * boxes doing about 2k of calls per day.....for now....there is major plans for them |
| 22:35.27 | cusco | We would like to implement something like kamalio |
| 22:35.30 | jmmills | I could make this junk scream if I ever get a chance to upgrade the OS |
| 22:35.35 | cusco | if I had the time to read about it ... |
| 22:35.37 | cusco | :( |
| 22:35.41 | jmmills | and then off load agi to dedicated hw |
| 22:35.55 | jmmills | kamailio is dark magic :) |
| 22:36.04 | jmmills | they didn't set this up the way it should be |
| 22:36.31 | TazzNZ | jmmills: any call recording going on ? |
| 22:36.52 | jmmills | it should be kamailio in front handing phone registrations, and then lightweight asterisk (or FreeSWITCH) acting as an SBC, then asterisk cluster for dialplan features, and another asterisk cluster for voicemail and conferencing |
| 22:36.59 | jmmills | TazzNZ: yeah, some - but not that much |
| 22:37.10 | jmmills | majority of channel->disk I/O is voicemail |
| 22:37.21 | jmmills | stored on a dumb nfs server |
| 22:37.21 | TazzNZ | I still don't understand why people want to offload the reg's to another box |
| 22:38.01 | TazzNZ | you could have used that to avoid VM's - since it will (iirc) transcode the audio to record it |
| 22:38.02 | jmmills | because it's signaling traffic |
| 22:38.07 | TazzNZ | it does however |
| 22:38.11 | TazzNZ | depend on the format |
| 22:38.32 | TazzNZ | jmmills: ok....but you could do that with Asterisk too |
| 22:38.37 | TazzNZ | why openSER |
| 22:38.57 | jmmills | TazzNZ: that's the current setup |
| 22:39.13 | TazzNZ | jmmills: it's not an un-common one is what I am saying |
| 22:39.23 | TazzNZ | and I don't understand why |
| 22:39.38 | jmmills | because it's thinner, and (I think) allows you to handle phones that don't do things correctly more easily - also it's just basic load balancing theory - I don't want my * boxes selected via SRV |
| 22:39.55 | jmmills | There are a couple presentation that give some good theory on it |
| 22:40.04 | jmmills | personally I've never implemented it |
| 22:40.08 | TazzNZ | link them ? :) |
| 22:40.12 | jmmills | um |
| 22:40.14 | jmmills | looks |
| 22:40.21 | TazzNZ | I am not attacking you, but just talking in general :) |
| 22:40.32 | jmmills | TazzNZ: http://www.tmcnet.com/tmc/videos/default.aspx?vid=5824 |
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| 22:41.17 | jmmills | It seems like the right path for me to go if I want "elastic" scaling of asterisk, so I can bring up more instances or drop them as media processing load goes |
| 22:41.38 | jmmills | For example, I don't need all of my instances during the middle of the night |
| 22:41.49 | jmmills | high traffic is 8am-12pm |
| 22:41.58 | TazzNZ | sure - but if you are doing that with openSER, why not Asterisk |
| 22:42.01 | jmmills | but then, what if a customer runs a popular ad campain? |
| 22:42.04 | TazzNZ | that is the part I don't get |
| 22:42.27 | jmmills | Well, your phone registrations are staying pretty consistent, but your media volume is in flux |
| 22:42.51 | jmmills | so it makes sense to separate them, also if you end up having an asterisk box crash, the phone registration doesn't drop |
| 22:43.12 | cusco | how do you bring up more instances? like more asterisk boxes? |
| 22:43.21 | jmmills | The theory jives with the same idea as running a lightweight HTTP server in front of a cluster of FastAGI boxen |
| 22:43.28 | cusco | or more resources to the current one? |
| 22:43.48 | TazzNZ | jmmills: but if that box crashes, you loose all of the reg's |
| 22:44.00 | TazzNZ | same problem as *if* asterisk crashes |
| 22:44.23 | jmmills | cusco: that's the part I'm working on - I've been doing more and more research on it, and I'm thinking that tieing in coreos+etcd+confd with dundi, so I can use an orchestration tool with openstack to just say "gimme 3 more asterisk nodes" and then they automatically enroll themselves into dundi, and then SBC dispatch |
| 22:44.28 | jmmills | and start taking calls |
| 22:44.49 | jmmills | TazzNZ: it's a lot more lightweight, like I said only signaling traffic |
| 22:44.59 | jmmills | and I've been thinking about that as well |
| 22:45.11 | jmmills | how to share registration session information between kamilio |
| 22:45.22 | jmmills | then you can just use layer 2/3 HA in front of that |
| 22:45.29 | TazzNZ | right - so the "only" reason is that openSER will handle say 10000 reg's per server, while in the same space/hardware, Asteisk will do say 4000 |
| 22:45.38 | jmmills | I'm not saying it's a solved problem, but certainly an interesting one |
| 22:45.59 | jmmills | TazzNZ: and those registrations won't impact media performance |
| 22:46.02 | TazzNZ | jmmills: see - that is where I would use Asterisk Realtime |
| 22:46.11 | jmmills | you don't want a misbehaving group of phones impacting other customers call quality |
| 22:46.25 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:46.32 | jmmills | I use realtime static for sip users and dialplans et cetera |
| 22:46.40 | jmmills | I don't write registration information to the database |
| 22:47.07 | jmmills | and I'm in the process of porting all the dialplans to use dundi lookups so phones can be registered to any pbx and still find each other |
| 22:47.23 | TazzNZ | have you looked into DUNDi ? |
| 22:47.33 | jmmills | It's currently implemented :) |
| 22:47.40 | jmmills | just a lot of dumb dialplans need to be fixed |
| 22:47.42 | TazzNZ | sweet |
| 22:47.56 | TazzNZ | and on DAHDI, I am currently looking into SS& |
| 22:47.59 | TazzNZ | LOL...again |
| 22:48.02 | TazzNZ | SS7 |
| 22:48.03 | jmmills | that's one of my hurdles into making this a service I can run in a docker container |
| 22:48.09 | jmmills | No SS7 :) |
| 22:48.16 | jmmills | leave that to legacy switches |
| 22:48.28 | TazzNZ | well.....it works :) |
| 22:48.46 | jmmills | Our switch engineering team has been talking to me about implementing a SS7 service |
| 22:49.10 | TazzNZ | you know that * supports SS7 right ? |
| 22:49.23 | jmmills | Yes, via TDM circuits |
| 22:49.53 | TazzNZ | you are looking at linking all this via DAHDI right ? |
| 22:49.56 | jmmills | the idea is that I would be writing a caching CID proxy/dispatcher that accepts SIP info requests and then delegates to other services |
| 22:50.31 | jmmills | TazzNZ: yes that would be a part of it - but you'll have to excuse me - I haven't finished reading my SS7 book |
| 22:50.42 | TazzNZ | jmmills: all good :) |
| 22:51.02 | jmmills | I guess it's always interesting when you throw a DevOps guy at a voice platform :) |
| 22:51.31 | TazzNZ | jmmills: yip - there is that :) |
| 22:52.31 | TazzNZ | jmmills: when you finished your reading, talk to me again about SS7 |
| 22:52.46 | TazzNZ | would be good to get your take on it |
| 22:53.05 | jmmills | Will do. |
| 22:54.05 | WIMPy | How well does it work? |
| 22:56.19 | *** join/#asterisk imcdona (~imcdona@75-151-101-98-Washington.hfc.comcastbusiness.net) |
| 22:56.32 | TazzNZ | WIMPy: most telco's use it these days for signalling, every time you port a number, chances are, SS7 is doing it |
| 22:56.53 | WIMPy | With Asterisk, off course. |
| 22:57.15 | TazzNZ | ah - still researching that :) |
| 22:57.38 | WIMPy | And I don't know how things work in your area, but for porting numbers that's be a temporary thing here. |
| 22:58.37 | TazzNZ | hhhhmmm - didn't know that |
| 22:58.43 | TazzNZ | for us, it's "perm" |
| 22:58.59 | TazzNZ | it's almost like a 301 in http |
| 22:59.30 | TazzNZ | what I want to do with SS7 is what jmmills is looking for |
| 23:00.05 | TazzNZ | I want phones to register on an * server, then use SS7 to route the call from the provider to the correct server |
| 23:00.14 | WIMPy | tries to find out what that is. |
| 23:00.20 | jmmills | SS7 is pretty much defacto signally for legacy circuits |
| 23:00.26 | jmmills | it's still a big part of the PSTN |
| 23:00.34 | WIMPy | aye |
| 23:00.46 | WIMPy | For what's left of the PSTN? |
| 23:00.49 | jmmills | TazzNZ: that only makes sense if you are interlinking with TDM circuits |
| 23:01.14 | TazzNZ | jmmills: welllllllll - you can run cross-overs between servers |
| 23:01.16 | jmmills | Well, lets just say that the carrier I work for is phasing out SS7 internally |
| 23:01.38 | jmmills | Well, if you are using SIP there isn't much need for SS7 |
| 23:02.06 | jmmills | especially not when you have DUNDi and Enum |
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| 23:02.41 | WIMPy | If there weren't those evil side effects. |
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| 23:49.59 | jmmills | TazzNZ: another use case for kamailio that I wasn't thinking of, policy based routing of calls like LCR |
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| 23:55.31 | TazzNZ | jmmills: yeah - I would have thought those would be in your asterisk configs |
| 23:55.46 | TazzNZ | the way I see things, if I must user openSER, is something like this |
| 23:56.13 | jmmills | Do you really want to manage say 15 different peers with different cost structures from each asterisk instances? |
| 23:56.19 | TazzNZ | openSER (reg's) --- call ---> Asterisk --- LCR/Other routing/etc ---> Provider |
| 23:56.31 | TazzNZ | you going to do that in openSER |
| 23:56.45 | TazzNZ | and I would AGI it |
| 23:57.20 | WIMPy | AGIs are evil |
| 23:58.08 | TazzNZ | WIMPy: only when they break or "you don't know what they do |
| 23:58.23 | WIMPy | No. Always. |
| 23:58.30 | TazzNZ | why are they evil ? |
| 23:58.32 | WIMPy | They fork and forks are evil. |
| 23:58.36 | ChkDigit | Just installed my AEX2400, and it is not hanging up incoming lines. I've checked my ring and tip, and also made sure my carrier is a loop starter... Where do I look next? |
| 23:59.23 | ChkDigit | incoming = FXO for me... sorry. |
| 23:59.30 | TazzNZ | bbib |
