| 00:11.46 | gevans | file: thanks! confirmed 13.6.0 is working perfectly (so far) |
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| 01:19.53 | snadge | using logrotate with csv files.. im seeing some 0 byte files on rotation.. and i can see its got something to do with asterisk having the files open |
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| 01:20.19 | snadge | so im just wondering if i should use the postrotate option with asterisk logger reload.. or whether to use the copytruncate option |
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| 03:53.31 | WIMPy | monsterco: YOu don't pass arguments there. |
| 03:54.25 | WIMPy | That's what I call timing. |
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| 05:11.56 | moe` | WOW skype support just fixed my SIP profile |
| 05:12.05 | moe` | only took 3 weeks :) |
| 05:12.24 | moe` | so yeah, plugged into flowroute, twilio, skype |
| 05:13.01 | moe` | flowroute+++ twilito++ skype+ |
| 05:13.17 | moe` | twilio, rather |
| 06:07.26 | moe` | I'm almost amazed really |
| 06:07.40 | moe` | that MS fixed my Skype SIP profile |
| 06:07.53 | moe` | still trying to digest that |
| 06:08.10 | igcewieling | buy a lottery ticket! |
| 06:10.29 | moe` | LOL |
| 06:10.30 | moe` | indeed |
| 06:17.44 | moe` | hey, here's a question |
| 06:17.51 | moe` | suppose I have 3 SIP trunks |
| 06:18.38 | moe` | is there a syntax that I can have a call outbound try one provider, then secondary, then a terteriary SIP trunk? |
| 06:19.18 | moe` | obviously as I said I have flowroute, twilio, and skype now. |
| 06:19.38 | moe` | so an outbound failure could fallover to another trunk? |
| 06:20.33 | WIMPy | Just do them in sequence. |
| 06:21.02 | moe` | so that's exten definitions? |
| 06:21.12 | moe` | well yeah I guess it would be eh |
| 06:21.13 | WIMPy | And better check DIALSTATUS/HANGUPCAUSE in between. |
| 06:22.26 | moe` | well my outbound is pretty simple |
| 06:22.47 | moe` | ; --- outbound SIP calls |
| 06:22.47 | moe` | exten=>_9.,1,Dial(SIP/flowroute/${EXTEN:1}) |
| 06:22.47 | moe` | exten=>_7.,1,Dial(SIP/skype/${EXTEN:1}) |
| 06:22.47 | moe` | exten=>_5.,1,Dial(SIP/twilio/+${EXTEN:1}) |
| 06:23.28 | moe` | sorry on the paste but it was stupid to make it pastebin or a unixninja.net/ link |
| 06:26.07 | moe` | I know those statement preclude the possibility of extensions that start with 9, 7, or 5. |
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| 06:34.09 | moe` | any thoughts on this? |
| 06:35.03 | WIMPy | On what? |
| 06:35.16 | moe` | how to rollover on outbound failure |
| 06:35.29 | WIMPy | I told you. |
| 06:35.55 | moe` | noted. I'll read and test some more. |
| 06:38.42 | moe` | it would be a nice thing, most resilient if it would tolerate SIP trunk failures transparent to users |
| 06:39.01 | moe` | now as I have it, users select a SIP trunk based on dial prefix |
| 06:39.06 | WIMPy | Make it happen. |
| 06:39.37 | moe` | well now that I have 3 trunks, that'd be an amusing thing. |
| 06:40.11 | moe` | and I've upgraded the box at the datacenter to a reasonable box |
| 06:40.52 | moe` | the datacenter has triple links, the box itself has dual links to a dual core |
| 06:41.16 | moe` | dual power supplies, right back to UPS and generators (dual all the way) |
| 06:41.50 | moe` | so I'm building a voip.unixninja.net :) |
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| 06:42.57 | WIMPy | How many IPs in how many ASs? |
| 06:43.12 | moe` | many class C's in several AS's |
| 06:43.38 | moe` | we actually have a couple class B's on top of that |
| 06:43.40 | moe` | :) |
| 06:44.28 | WIMPy | Talking about Classes makes you look old :-) |
| 06:44.40 | moe` | so it happens bytimes that we BGP one of the class C's to another data center |
| 06:45.01 | moe` | ... makes me look old, or make you understand that I'm not a newbie. |
| 06:45.16 | moe` | grins stupidly |
| 06:45.22 | WIMPy | What I said :-) |
| 06:45.28 | moe` | onya |
| 06:46.14 | moe` | so it comes back to the asterisk syntax that always confounds me |
| 06:46.16 | WIMPy | But I see you really meant everything. |
| 06:47.21 | moe` | now that I have 3 trunks, outbound trunk failover |
| 06:47.36 | moe` | inbound DIDs are tied to specific trunks of course |
| 06:47.50 | moe` | but outbound is not so tied |
| 06:48.38 | moe` | see above how I let the user select |
| 06:48.54 | moe` | with a 9, 7, or 5 prefix to select outbound SIP trunk |
| 06:49.01 | moe` | I'd like to eliminate that |
| 06:50.40 | moe` | perhaps leave the possibility to select for users, but maybe outbound prefix in general would be 1, and to pick a specific SIP would be 9 (flowroute), 7 (skype), 5 (twilio) |
| 06:50.44 | moe` | for example, that is |
| 06:53.28 | moe` | so can you give me a simple example of outbound SIP trunk failover? |
| 06:53.40 | moe` | <--- lazy idiot |
| 06:53.58 | WIMPy | Have you tried to ask google. You're not the first with that idea. |
| 06:54.46 | moe` | of course I'm not the first with that idea, but I thought I'd ask the asterisk masters if they cared to answer. |
| 06:55.13 | WIMPy | REinvent the wheel? |
| 06:55.45 | moe` | cat - | grep -iE "wheel" |
| 06:55.55 | moe` | return code != 0 |
| 06:59.01 | moe` | well, ok..... cat - | grep "wheel" && echo "wheel already invented" || echo "wheel is new invention" |
| 06:59.05 | moe` | something like that? |
| 06:59.22 | WIMPy | Stop chatting and start working. |
| 06:59.37 | moe` | you do have a point there. |
| 07:05.36 | moe` | hey is there any suck thing as a library statement on asterisk to detect latency/lag? |
| 07:05.59 | WIMPy | Where? |
| 07:06.13 | moe` | since I have 3 trucks, I could route all calls via one SIP trunk |
| 07:06.22 | moe` | leaving 2 trunks to test |
| 07:07.00 | moe` | and if I artifically fail (via BSD ipfw) fail one, for testing, is there an asterisk directive that I can use to detect that? |
| 07:07.18 | WIMPy | Turn on qualify. |
| 07:07.38 | moe` | I have qualify on, but it's not enabled for outbound |
| 07:07.53 | moe` | would that be the simple answer? |
| 07:08.11 | WIMPy | That's what it does. |
| 07:08.15 | moe` | duh |
| 07:08.18 | moe` | ok |
| 07:08.38 | moe` | ok, so now what's the nature of the outbound exten for that? |
| 07:08.40 | WIMPy | It checks regularly and if it doesn't receive an answer in time, it won't even try to call out that peer. |
| 07:09.02 | moe` | yeah but ok, how to syntax use that? |
| 07:09.15 | moe` | that sounds awesome, really |
| 07:09.22 | WIMPy | There is no syntax. Just switch it on. |
| 07:09.39 | moe` | no but for outbound I have a different prefix per SIP trunk |
| 07:10.06 | moe` | how do I roll that into one and make it use qualify times |
| 07:10.08 | moe` | ? |
| 07:10.39 | moe` | this is why I bother you asterisk yoda masters. |
| 07:10.47 | WIMPy | You don't have to. Qulify means the the Dial will fail immediately without trying if the last qualify failed. |
| 07:11.12 | moe` | how do I define my outbound then? |
| 07:12.00 | WIMPy | You still didn't ask google for examples? |
| 07:12.39 | moe` | no I thought I'd ask the yoda masters first, perhaps that's my bad. |
| 07:13.21 | WIMPy | I'm not going to copy&paste for you. |
| 07:13.46 | moe` | no, and I would not expect you to |
| 07:13.52 | moe` | I'll look into it. |
| 07:14.15 | WIMPy | And I gave you two hints in the very beginning. |
| 07:14.24 | WIMPy | And that's really all there is to it. |
| 07:15.55 | moe` | noted. I'll dig into it and I'll get back to you with further questions. Thanks man. |
| 07:16.18 | moe` | I see where its going. |
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| 09:10.07 | napnap | Hi all |
| 09:10.15 | napnap | Since few weeks/month I have *blank* during call. No sound for few seconds then we can continue to speak... Where can i search to see what happens ? |
| 09:10.30 | napnap | When I read asterisk log no seems to be anormal.. |
| 09:10.54 | wdoekes | napnap: set rtp debug on |
| 09:11.20 | wdoekes | it will flood your logs, but you'll see who isn't sending any audio |
| 09:11.41 | wdoekes | alternately you can capture pcaps of all your calls and analyze them afterwards |
| 09:13.28 | wdoekes | (analyze with wireshark, capture with: tcpdump -niany -s0 -C 100 -c 100 -w udp.pcap. udp ) |
| 09:23.34 | napnap | wdoekes, thanks for you anwser I will try to follow your advice.. |
| 09:24.39 | napnap | (strange I'm root and I got permission denied to write in a file with tcpdump... |
| 09:26.21 | napnap | good start.. lol |
| 09:28.02 | napnap | ok with "-Z root" it's work |
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| 09:40.43 | WIMPy | Oh, I had such an issue, where the session-timers would lead to very noticable drop outs. |
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| 10:10.38 | tulga | how to configure like IVR click 1 -> play announcement A, then dial extension 500 in freepbx? |
| 10:11.24 | WIMPy | Ask in #freepbx |
| 10:22.27 | tulga | is it possible to execute AGI every click of IVR? |
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| 11:07.26 | jasiek | hey. i have a concern about astcanary and having that running on a flash memory-backed file system. from my understanding, it writes file metadata to disk once a second. is it likely to wear out an SD card quickly? |
| 11:09.10 | jasiek | this is a tiny setup, less than 10 phones, so the "server" is never under any load |
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| 11:43.22 | casdude | hi |
| 11:47.52 | casdude | I have recently inherited a call recorder that is built in asterisk |
| 11:48.00 | casdude | It has some interesting things in the chan-dahdi.conf file i was wondering if someone could shed some light. |
| 11:48.08 | casdude | http://pastebin.com/H8eLeBui |
| 11:48.20 | casdude | At the top of the file it mentions group=1 does this mean that the settings below for example immediate=no and overlapdial=yes only apply to groups that equal 1? |
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| 14:41.44 | bkruse | Hello friends. Anyone have any experience tweaking stasis performance with stasis.conf? I am using the default config values, but wanted to see what other people are up to |
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| 15:52.41 | acidfoo | moo |
| 15:53.26 | acidfoo | mmm, does anyone know what that means ? cl.c:748 ast_apply_acl: SIP Direct Media ACL: Rejecting 'xxx.xxx.xxx.xxx' due to a failure to pass ACL '(BASELINE)' |
| 15:53.35 | acidfoo | do I need ot add an ACL somewhere ? |
| 15:54.13 | [TK]D-Fender | You have an ACL defined. The caller is failing it |
| 15:54.27 | acidfoo | yeah that I can read english :) |
| 15:54.31 | acidfoo | but I have nothing in acl.conf |
| 15:54.46 | igcewieling | acidfoo: see sip.conf |
| 15:54.53 | [TK]D-Fender | maybe it's not in an external file. |
| 15:55.01 | [TK]D-Fender | You should look at your complete call and the rest of your configs |
| 15:55.21 | acidfoo | I tried permit=0.0.0.0/0.0.0.0 in sip.conf (for the sake of it...) it hasn't change anything |
| 15:56.55 | igcewieling | acidfoo: grep "deny" /etc/asterisk/sip.conf |
| 15:57.30 | [TK]D-Fender | with permit/deny ORDER MATTERS |
| 15:57.45 | acidfoo | right now I have no "deny" in my sip.conf |
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| 15:57.59 | [TK]D-Fender | show us actual call debnug & configs |
| 15:58.07 | igcewieling | acidfoo: I recommend an exorcist. |
| 16:01.38 | catphish | i'm dong some experiments with asterisk 11, i've noticed that my config takes approx 5 seconds to reload, and asterisk doesn't respond to SIP packets during this period, is there any way to do an atomic reload in a separate thread, or have any improvements been made in this direction in asterisk 13? |
| 16:02.22 | [TK]D-Fender | Improvments are always happening and you are 2 branches behind |
| 16:03.54 | catphish | that is almost certainly true, but i'm wondering specifically about config reloads |
| 16:03.57 | file | chan_sip has had no changes to it in that regard, as it's a pretty fundamental change |
| 16:04.09 | file | chan_pjsip doesn't do configuration in the same way and doesn't have that problem |
| 16:04.35 | catphish | actually i need to check whether it's the sip reload that's causing the problem or the dialplan reload, that's probably an important distinction |
| 16:04.58 | igcewieling | "dialplan reload" and "sip reload" |
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| 16:05.37 | igcewieling | catphish: often slow reloads can be caused by rogue AGIs or DNS timing out. |
| 16:05.48 | catphish | "sip reload" is actually almost instant |
| 16:06.16 | catphish | dialplan reload takes about 3 seconds, core reload about 6, i think i should unload unnecessary modules, stop using "core reload" and rethink my question |
| 16:06.16 | catphish | thanks |
| 16:06.27 | file | sip reload doesn't block until the config is done, it informs another thread to do the reload |
| 16:07.15 | catphish | ok, so the issue is definitely with "dialplan reload" |
| 16:07.26 | catphish | that seems to block SIP for several seconds |
| 16:08.56 | igcewieling | catphish: in case it is helpful, here is a list the noload => s I use. http://pastebin.com/uABVNZiy This is from an internal program, not intended for directly copying into modules.conf |
| 16:09.38 | catphish | igcewieling: thanks, i was actually planning to load modules individually as needed, since i probably require quite a small subset |
| 16:09.56 | catphish | but the issue definitely seems to be with the dialplan reload (my dialplan is fat) |
| 16:10.04 | igcewieling | catphish: that is a very good way to be driven to crazy. |
| 16:10.25 | catphish | igcewieling: interesting, i haven't tried yet, it gives the false impression it's going to be simple ;) |
| 16:10.32 | igcewieling | Unless you know Asterisk very well you'll have a lot of trouble figuring out what the min set of modules is |
| 16:11.54 | catphish | so at risk of asking the same question again: does anyone know if there's a way to improve dialplan load time, or if this has been backgrounded in v13? |
| 16:12.07 | *** join/#asterisk paraxor (~paraxor@unaffiliated/prazial) |
| 16:13.45 | igcewieling | I understand. If my Asterisk blocked calls for 6 seconds when doing a reload, we'd have thrown it out. |
| 16:14.59 | igcewieling | sorry, 5 seconds |
| 16:15.04 | *** join/#asterisk petris (~petris@2604:180:0:ad7::73db) |
| 16:15.28 | catphish | right now we actually use realtime, for exactly that reason, but i'm looking to drop realtime, because it's causing other issues |
| 16:15.38 | catphish | one option would be to use realtime for the dialplan only |
| 16:16.58 | catphish | i'm testing with a dialplan of 34046 extensions (146252 priorities) |
| 16:16.59 | igcewieling | oh! |
| 16:18.51 | igcewieling | Do you need that many extensions? Often times people have simple dialplans and do the heavy lifting in an AGI |
| 16:19.46 | catphish | igcewieling: that's actually probably the correct solution in my case, though it'll be a major redevelopment |
| 16:20.00 | catphish | there's a huge amount of duplication in my static dialplan |
| 16:20.04 | igcewieling | ah. |
| 16:20.26 | igcewieling | I run 10,000 numbers and 250,000 calls through a server with 64 extensions. |
| 16:20.45 | igcewieling | 250,000 calls /month |
| 16:20.50 | catphish | using AGI? or just a simple setup? |
| 16:21.09 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 16:21.38 | catphish | i run a web managed end user configurable hosted PBX setup |
| 16:23.39 | igcewieling | AGI assisted with Dialplan, no end users, just routing calls. FreePBX only has 2365 extensions |
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| 16:26.09 | catphish | seems clear i need to do some more reengineering, maybe with AGI |
| 16:27.53 | igcewieling | catphish: the method I use is for the AGI to set a bunch of channel variables, then do the actual dialing, etcin the dialplan using AEL Running Dial from inside an AGI can be.....interesting. |
| 16:31.08 | igcewieling | FreePBX uses a similar design |
| 16:31.26 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 16:32.51 | catphish | is there a specific problem with running applications (or dial specifically) from AGI? |
| 16:34.48 | igcewieling | catphish: when the dial completes you are back in the dialplan, not back to the AGI. |
| 16:35.25 | igcewieling | so for things like hunting, rollover, Call Forwarding of any sort things get a lot more complicated when dialing from AGI |
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| 16:56.42 | *** join/#asterisk pecanha (bff0e270@gateway/web/freenode/ip.191.240.226.112) |
| 16:57.23 | pecanha | Hello, I'm using asterisk 1.8. I would like to know how can I execute an AGI script when a agent/extensions answer the call from queue? |
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| 17:06.46 | acidfoo | in sip.conf I set directmediaacl=directm , then in acl.conf I do [directm] permit=0.0.0.0/0.0.0.0, |
| 17:07.23 | acidfoo | I sill have a peer that, when I do "sip show peer the_peer" still output DirectMedACL : No |
| 17:07.24 | acidfoo | strange |
| 17:07.34 | [TK]D-Fender | pecanha: "core show application queue" |
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| 18:49.12 | Glanzmann | Hello; I have asterisk 1:11.13.1~dfsg-2+b1 on Debian stable and two sip phones. When I talk to from one sip phone to another locally I have very high latency: 2 seconds. When I talk to the outside world using iax or isdn using an eicon diva everything is fine. I'm looking for the cause of the high latency internally and how to get rid of it. Any hints? |
| 18:50.28 | robmal | Shitty network. |
| 18:51.39 | Glanzmann | Not really. It is gigabit and I have a latency of 0.3ms as expected on gigabit ethernet. |
| 18:51.56 | *** join/#asterisk Dovid (~dovid@173.63.105.210) |
| 18:52.04 | robmal | Shitty QoS |
| 18:52.04 | Dovid | how do I find out what ticket this is part of ? https://issues.asterisk.org/jira/secure/attachment/51698/bt.txt |
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| 19:07.06 | Glanzmann | It was 'jbenable = yes'. I set it to know, latency is gone. |
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| 20:58.53 | pjensen00 | I'm trying to figure out why Asterisk is 400ing my REFER |
| 20:58.53 | pjensen00 | http://pastebin.com/vf8m5PB2 |
| 20:59.36 | pjensen00 | The refer-to header has the correct callID/tags (which was a problem we had before) since we're using a B2BUA |
| 21:00.19 | pjensen00 | The one worrying line is "The inv session does NOT have an invite_tsx", but other than that I don't know what would point to something causing the REFER to fail. |
| 21:09.04 | pjensen00 | ... apperas to be an unencoded semi color in the REFER-TO header. BLARG |
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| 23:27.57 | mro_work | aloha pbx peoples. Openvox cards of any use these days? I have a bunch I just found in a box |
| 23:38.59 | WIMPy | Depends. |
| 23:40.23 | mro_work | we have like 6 lines but never more than 3 users |
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