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| 00:17.54 | marcozink | Hello, I |
| 00:18.48 | marcozink | I'm having troubles with a "#" character in a dial string to my ITSP, it's showing as %23, i'm using asterisk 11, any ideas? |
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| 05:23.41 | sigmaorion | hi guys! I need to use either G729 or iLBc with FEC enabled... how do I enable FEC on those codecs in Asterisk? |
| 05:23.55 | sigmaorion | is FEC supported at all? |
| 05:26.08 | sigmaorion | as for silk you can set fec=true or fec=false in codecs.conf |
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| 05:38.07 | raspberrypifan | options for plain asterisk billing |
| 05:39.07 | Samot | CDRs/ |
| 05:41.27 | raspberrypifan | ? |
| 05:45.55 | [TK]D-Fender | MS Excel |
| 05:46.32 | raspberrypifan | well sometihing more systemic |
| 05:46.57 | [TK]D-Fender | MS Excel + VBA |
| 05:52.47 | raspberrypifan | sign |
| 05:52.50 | raspberrypifan | sigh |
| 05:52.53 | raspberrypifan | not really what i was thinking |
| 05:53.24 | Samot | We're not sure what to recommend. Most of the options have a lot of text boxes. |
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| 06:01.05 | drmessano | Home ITSP |
| 06:14.35 | Samot | He literally had just said FreePBX has too many text boxes, it felt unpolished and needed improvement. He suggested it all be done with menus. |
| 06:15.57 | Samot | Then TK showed me a video on how to polish a turd. |
| 06:17.16 | Samot | Apparently if you have a lot of meat in your diet your turd will be really shiny when you polish it. |
| 06:17.30 | drmessano | lol |
| 06:17.43 | drmessano | That dude is just.. idk |
| 06:18.09 | drmessano | He wants an ITSP in a box that has like 3 boxes and bills with PayPal |
| 06:19.08 | Samot | Doesn't everybody? |
| 06:20.56 | drmessano | Someone needs to come up with an ITSP Distro thats easy to configure as Snapchat |
| 06:21.06 | drmessano | Maybe actually use Snapchat to configure it |
| 06:21.41 | drmessano | Doesnt Snapchat have a REST API? |
| 06:25.59 | Samot | I've never used Snapchat. |
| 06:26.52 | drmessano | Its all the hotness |
| 06:27.37 | drmessano | Have you ever wanted to take a selfie and put dog ears on yourself? |
| 06:27.43 | drmessano | Well, you cn |
| 06:27.43 | drmessano | can |
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| 07:05.37 | UncleKiwi | hey i have a few spa122 ATA's registering to an Asterisk box over lan |
| 07:05.55 | UncleKiwi | and every few months or so the SPA122 hangs |
| 07:05.58 | UncleKiwi | cant ping it |
| 07:06.07 | UncleKiwi | and have to pull the power and boot it again |
| 07:06.14 | UncleKiwi | and then its fine |
| 07:06.24 | UncleKiwi | just wondering if anyone has seen this |
| 07:06.39 | UncleKiwi | I have just upgraded them to the most recent firmware |
| 07:06.46 | UncleKiwi | to see if this will help |
| 07:07.13 | UncleKiwi | or are the spa122's just garbage |
| 07:13.31 | Samot | SPA122's are pretty decent. |
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| 08:16.08 | UncleKiwi | yeah i just get them failing after like several months |
| 08:16.21 | UncleKiwi | you know they are just not 100% uptime things |
| 08:16.44 | UncleKiwi | so as you can imagine if i have 100 of these devices in service |
| 08:17.01 | UncleKiwi | its just making me look bad |
| 08:17.09 | UncleKiwi | when they fail |
| 08:17.58 | UncleKiwi | its not asterisk letting me down at all |
| 08:18.15 | UncleKiwi | i think its just garbage hardware |
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| 08:21.03 | Alblasco1702 | UncleKiwi, try to make a script that restarts the SPA122's on a time nobody is there. |
| 08:21.16 | Alblasco1702 | befoe they hang. |
| 08:21.55 | UncleKiwi | yeah i guess i could rebooth them once a month |
| 08:22.11 | UncleKiwi | that might .... solve it |
| 08:22.51 | UncleKiwi | but that would assume that they have a min hang time |
| 08:23.19 | UncleKiwi | which i would assume is probably greater than a month |
| 08:23.29 | UncleKiwi | so yeah might work |
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| 08:26.46 | UncleKiwi | i have a school and a hotel im about to put asterisk into |
| 08:26.59 | UncleKiwi | 42 phones |
| 08:27.04 | UncleKiwi | in the hotel |
| 08:27.39 | UncleKiwi | dont worry i dont be using any more spa122's |
| 08:30.36 | Alblasco1702 | UncleKiwi, or it barfs on a configuration setting don't ask me why. |
| 08:30.55 | UncleKiwi | maybe |
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| 10:01.18 | ronybeck | When a UAS sends a 401 unauthorized (authentication challenge) does the To and From fields need to match the invite (especially the IP address)? |
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| 11:12.37 | bounceman | Hi, can anyone provide me with some ideas on this issue. We experience that asterisk do not answer to INVITEs but 8-10 seconds later. When I look at the sip debug CLI the INVITE does not even show. If I run tcpdump port 5060 I see them quite well. |
| 11:13.01 | bounceman | The response times are excellent, there are no network gaps and the server has 8 cores with 0.8 load on each and 1gb of free space. |
| 11:13.12 | bounceman | This ALWAYS occur on high loads when queued calls reach 50-60 |
| 11:13.15 | shido6 | compile from source? |
| 11:13.31 | shido6 | +did you |
| 11:13.41 | shido6 | oh!!! |
| 11:13.54 | shido6 | so you are processing 50-60 concurrent calls or total calls? |
| 11:14.02 | bounceman | currently 20 ongoing calls and 52 in queue |
| 11:14.26 | bounceman | 100 active sip dialogs |
| 11:14.41 | shido6 | kewl beans, so thats a lot of call data to paste in pastebin.. soo standby |
| 11:15.11 | bounceman | Asterisk 11 btw |
| 11:15.13 | bounceman | latest |
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| 11:55.58 | shido6 | brb breakfast for the fam |
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| 12:14.58 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 13:08.24 | bounceman | What can cause asterisk to not recieve INVITES on high load? The invites register in tcpdump but no in the asterisk cli |
| 13:09.46 | Samot | Then it's receiving INVITES. |
| 13:10.31 | file | if chan_sip is in use then blocking operations (database access, slow DNS) can cause UDP traffic to be delayed |
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| 13:19.22 | bounceman | Samot: yeah, I've seen that asterisk do answer after about 8-9 seconds, however which it so late since the sip dialog has been destroyed. |
| 13:20.01 | Samot | What are the specs on the PBX? |
| 13:20.18 | bounceman | 8 cores and 2gb of ram |
| 13:20.28 | bounceman | 4gb* |
| 13:20.59 | Samot | Are you recording calls? |
| 13:21.08 | bounceman | It occurred some hours ago when the calls went from 0 calls in queue to 60 :D |
| 13:21.14 | bounceman | opening hours to say |
| 13:21.17 | bounceman | Yes |
| 13:21.54 | Samot | So you have 20 active inbound, 52 in the queue (which I'm sure is playing music) |
| 13:22.26 | Samot | So roughly 75 calls on top of all the other inbound calls you might have but recording all the audio.... |
| 13:22.36 | Samot | I'd raise your resources. |
| 13:22.38 | Samot | This a VM? |
| 13:23.10 | bounceman | 20 active calls and 50-60 in queue |
| 13:23.18 | bounceman | around 100 channels |
| 13:23.24 | bounceman | Sometimes 80 calls in queue |
| 13:23.31 | bounceman | Playing music correct |
| 13:23.40 | Samot | Calls in the queue are active calls. |
| 13:23.44 | bounceman | yeah |
| 13:24.21 | Samot | Call recording can cause delays. |
| 13:24.25 | bounceman | It's a vmware |
| 13:24.43 | bounceman | The cpu load is only about 0.8 and there is plenty of free ram |
| 13:24.49 | Samot | Because right now you are actively recording 60 calls. |
| 13:25.01 | bounceman | They do only record once reaching an agent, queue is not recorded |
| 13:26.39 | bounceman | I might wanna start logging slow queries in the mysql, see if there is any holdup, but why would that stop asterisk from sending a TRYING? |
| 13:37.00 | [TK]D-Fender | DB lookups will freeze * |
| 13:37.34 | bounceman | Is the lookup being done before a trying get sent? |
| 13:39.17 | file | I do believe so, and because chan_sip is single threaded a request can block all others |
| 13:39.21 | bounceman | I would expect atleast to see the INVITE in the cli |
| 13:43.41 | [TK]D-Fender | chan_sip = blocked. |
| 13:43.51 | [TK]D-Fender | You shouldn't see a thing... because nothing got the packet yet |
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| 13:48.36 | bounceman | I see, do you have any tips of how to troubleshoot this Fender? |
| 13:48.40 | bounceman | Or anyone else for that matter |
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| 14:15.00 | catphish | i'm looking at using a proxy in front of asterisk, is it possible to have sip peers that require authentication normally, but are trusted if they come via the proxy? |
| 14:17.42 | catphish | i'm a little confused about how it will work, i don't really want my proxy to be a peer itself, i want it to be able to tell asterisk which peer a request is originating from, and have asterisk trust that information, unfortunately i don't 100% understand sip proxying yet |
| 14:25.46 | jkroon | bounceman, pjsip or chan_sip? |
| 14:26.12 | jkroon | i'm better if you run netstat -nuap | grep 5060 you'll see the receive queue has stuff in it. |
| 14:26.20 | bounceman | jkroon: how can I see which one is being used? |
| 14:26.38 | jkroon | if you're asking that question it's chan_sip :) |
| 14:26.41 | jkroon | http://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests |
| 14:26.48 | jkroon | how many SIP endpoints do you have? |
| 14:27.04 | bounceman | asterisk 11 is chan_sip if am not out in the blue |
| 14:27.20 | bounceman | Usually around 20 active, 30 in total |
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| 14:27.22 | jkroon | you're correct. read the above. |
| 14:27.28 | jkroon | your problem is *disk* IO related. |
| 14:28.44 | bounceman | You think so |
| 14:29.04 | jkroon | i spent a week trying to track down the source before I wrote that. |
| 14:29.15 | jkroon | chan_sip is single-threaded on RX. |
| 14:29.43 | jkroon | on register it does an fsync() to disk by way of sqlite ... which can take *seconds*. |
| 14:29.57 | bounceman | we use mysql, maybe not any different |
| 14:30.19 | jkroon | not your problem. chan_sip uses astdb which uses sqlite. you have no choice in that matter. |
| 14:30.37 | bounceman | Do you have any tips how we can monitor the IO? |
| 14:30.39 | jkroon | mysql is very smart about caching etc in-memory, much more so than sqlite, but sqlite is more suited for the task here. |
| 14:30.40 | bounceman | Verify that it is causing it |
| 14:30.48 | jkroon | iotop and iostat -dmx 1 |
| 14:30.58 | jkroon | also, that netstat command |
| 14:31.11 | bounceman | I am looking for high utilization%? |
| 14:31.20 | jkroon | yes, and long queue times. |
| 14:31.51 | jkroon | high utilization % values is not *always* a problem (NCQ queues up to 31 concurrent commands, so even a value of 100% doesn't mean you're maxing out on disks). |
| 14:32.08 | bounceman | how do I read netstat -nuap | grep 5060 ? I see udp followed by 0 |
| 14:32.11 | bounceman | meaning queue is 0 ? |
| 14:32.38 | jkroon | Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name |
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| 14:32.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:33.10 | jkroon | so yes, if the value direction after udp ever goes >0 you've probably got a problem with backlog in chan_sip. it's normally very fast and sporadic >0 values is OK. |
| 14:33.32 | jkroon | meaning netstat just happens to sample at the exact moment the packet arrived but before asterisk did a read() on the socket. |
| 14:33.47 | bounceman | Ok |
| 14:34.54 | jkroon | read my blog entry that I gave you the link for. |
| 14:35.13 | jkroon | it doesn't give ways to confirm that you've got the problem, only the fix. |
| 14:35.21 | bounceman | I will, I already started it. |
| 14:35.24 | jkroon | which is a definite shortcoming of the write-up. |
| 14:35.43 | bounceman | Great help |
| 14:36.11 | bounceman | I will be a frequent visitor of your blog, looking for more fixes for disguisting issues. |
| 14:37.18 | jkroon | lol, i don't write that often at this stage. |
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| 14:37.28 | jkroon | but thanks for the implied compliment. |
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| 14:40.57 | bounceman | jkroon: what do you mean by RX? |
| 14:41.06 | file | receive |
| 14:42.29 | jkroon | indeed. i'm off. i'll be back tomorrow if you'd like to get more info. |
| 14:42.49 | jkroon | I'm reasonably sure I've got the patches I used to measure those timings around still as well. |
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| 14:48.39 | bounceman | That would be cool |
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| 14:51.45 | carp3 | Hi, when i put 10 files in spool/asterisk/outgoing queue, asterisk tries to call them at once. so 9 of them fail. is there anyway to tell asterisk to call them one by one ? |
| 14:53.59 | [TK]D-Fender | no |
| 14:54.17 | [TK]D-Fender | You need to feed themthat way yourself by whatever means you come up with for that |
| 14:54.45 | [TK]D-Fender | including triggering the next at the end of each via dialplan, etc |
| 14:55.23 | wdoekes | or by requeueing themselves after failure, with a increasing or random delay |
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| 15:26.38 | catphish | is it possible to have asterisk skip password authentication of peers for certain IP addresses? (chan_sip) |
| 15:28.06 | catphish | i have a proxy between end users and asterisk, and i'd like it to be trusted to authenticate peers, but i'd like end users connecting directly to asterisk be authenticated by asterisk |
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| 15:45.51 | xai | Hello. Why "Scheduling destruction of SIP dialog" happens? |
| 15:46.48 | xai | (possible reasons) |
| 15:52.41 | shido6 | catphish, yes |
| 15:52.50 | [TK]D-Fender | because * will not let lagging comms stay open |
| 15:52.55 | [TK]D-Fender | this does not mean it's a CALL |
| 15:53.02 | catphish | shido6: how does one do this? |
| 15:53.10 | [TK]D-Fender | so don't take it that this is related to any issue you are currently having |
| 15:53.10 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP130-04-1177588791.dsl.bell.ca) |
| 15:53.18 | shido6 | host=IP of that host and remove secret |
| 15:53.45 | shido6 | Im assuming you have all the other details in sip.conf for that peer |
| 15:53.53 | catphish | shido6: oh yeah, i was aware of that, what i was hoping for was more a way to bypass the secret for a trusted proxy |
| 15:53.58 | shido6 | there I go making an ass of myself, again. |
| 15:54.07 | catphish | i wasn't very clear |
| 15:54.27 | catphish | i could just do that on a peer-by-peer basis as i migrate them to the proxy |
| 15:54.45 | *** join/#asterisk sigmaorion (b5e41945@gateway/web/freenode/ip.181.228.25.69) |
| 15:54.48 | catphish | ideally i wanted people to be able to move from "direct to asterisk" to "proxied" in their own time |
| 15:55.20 | sigmaorion | hi guys... any idea if RFC5109 was or is planned to be supported by Asterisk? |
| 15:56.11 | shido6 | ok quick question - are you looking to deploy "Upper Registration" ? where all customers reg to the public facing SBC and the SBC forwards to registration to asterisk but the client has no clue about the SBC, just the domain? |
| 15:56.22 | shido6 | err |
| 15:56.48 | shido6 | ok - its too early, i can't type |
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| 16:10.55 | catphish | shido6: yes, that's what i want to do, the SBC will pass registrations through to asterisk, clients will only know about the domain |
| 16:11.21 | shido6 | ah ha! |
| 16:11.42 | shido6 | my psychic abilities and spidy senses are not as off as I once thought. |
| 16:12.08 | catphish | kamailio likes to do auth itself and pass unauthenticated requests to asterisk, though perhaps this is unnecessary, might be able to move things around to asterisk always still does the auth and the proxy doesn't care |
| 16:12.21 | shido6 | unauth shouldn't touch asterisk |
| 16:12.28 | shido6 | thats why kam is there |
| 16:12.35 | *** join/#asterisk x-fak (~WinGuru@reactos/tester/x-fak) |
| 16:12.37 | catphish | what? |
| 16:12.49 | catphish | oh right yeah i see what you're saying |
| 16:13.02 | shido6 | i should say unauthorized clients shouldn't touch asterisk |
| 16:13.05 | catphish | so you think kam *should* do the auth before anything gets to asterisk |
| 16:13.23 | shido6 | isn't it talking to asterisk's db? |
| 16:13.39 | catphish | in which case, asterisk itself doesn't need to check the auth, which is the recommended way to set it up, the problem is that i have clients right now that connect direct to asterisk and authenticate, i didnt want to have to move them all to the proxy at once |
| 16:13.53 | catphish | yes, it consults asterisk's database |
| 16:13.56 | catphish | and does the auth |
| 16:14.05 | catphish | but what it doesn't do it send any auth to asterisk |
| 16:14.23 | catphish | it's designed to work with a totally insecure asterisk as far as i can tell |
| 16:15.00 | shido6 | dam to Asterisk is insecure but kam should auth using asterisk's db - so no unauth traffic should ever hit asterisk. |
| 16:15.08 | shido6 | kam not dam |
| 16:15.38 | shido6 | do u have topology hiding enabled, too ? |
| 16:15.54 | file | wobbles |
| 16:16.48 | catphish | shido6: topology won't really be a secret, since asterisk will be doing media |
| 16:17.23 | catphish | so i guess the answer is "no", i cant skip password auth on a per-IP basis |
| 16:20.03 | *** join/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
| 16:20.42 | jrun | can SIPShowpeer return all peers without specifying the peer's name? |
| 16:23.37 | jrun | SIPpeers |
| 16:25.12 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 16:28.03 | [TK]D-Fender | Contact: <sip:0884640052@119.252.14.202:5060> |
| 16:28.15 | [TK]D-Fender | oops |
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| 17:10.15 | jgo | ahh it's a wonderful day |
| 17:11.28 | file | it ain't bad |
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| 19:34.41 | *** join/#asterisk st2 (~st2@92.255.195.145) |
| 19:37.20 | st2 | hello, i had issue with Dial command inside GoSub subroutine. |
| 19:38.09 | st2 | when Dial command exit with non zera status, executing of subroutine interrupts too |
| 19:38.31 | st2 | Dial(SIP/${USERNAME}_${EXTEN},60,g|M(logger)) |
| 19:39.21 | [TK]D-Fender | | is NOT valid |
| 19:39.31 | st2 | for example Dial command. But when callee hanguo before connect i've got interrupt of my extension scenario |
| 19:39.46 | [TK]D-Fender | show us the actual dialplan and the cactual call attempt |
| 19:40.21 | st2 | should i use comma instead of | ? |
| 19:42.40 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1:a423:8e46:49a5:e17) |
| 19:43.51 | [TK]D-Fender | neither |
| 19:43.58 | [TK]D-Fender | no extra chars at all |
| 19:52.23 | st2 | sorry for your waiting |
| 19:52.44 | st2 | i try to fix dial command but no success |
| 19:52.53 | st2 | http://pastebin.com/1aVmqAtZ here is my call log |
| 19:53.39 | st2 | i try to migrate my extensions from realtime database to traditional config files throught GoSub using |
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| 19:55.51 | st2 | so in pickup extension i want use just GoSub call |
| 19:56.02 | st2 | mysql> select * from sip_extensions where context='new-pops_pickup' and exten='s'; |
| 19:56.02 | st2 | +--------+-----------------+-------+----------+-------+-----------------------------------------------------------------------------------+ |
| 19:56.02 | st2 | | id | context | exten | priority | app | appdata | |
| 19:56.02 | st2 | +--------+-----------------+-------+----------+-------+-----------------------------------------------------------------------------------+ |
| 19:56.02 | st2 | | 441132 | new-pops_pickup | s | 1 | GoSub | sub-call,incomming,1(new-pops,SIP/new-pops_102&SIP/new-pops_101&SIP/new-pops_103) | |
| 19:56.03 | st2 | + |
| 19:59.10 | st2 | and i want to put all old login of call processing in subroutine |
| 20:05.36 | st2 | here is my part of dialplan according to incomming call |
| 20:05.37 | st2 | http://pastebin.com/giD6vVLg |
| 20:10.25 | *** join/#asterisk st2 (~st2@92.255.195.145) |
| 20:10.50 | st2 | sorry, bad connection |
| 20:11.18 | st2 | so what's wrong with my dialplan? |
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| 20:57.51 | jonah | Hi can anyone please help me out. My asterisk setup was working great but suddenly today I noticed asterisk wasn't anwsering the phone any more and in the asteris cli when I ring in it gives this error: [Jul 5 21:54:39] WARNING[8698][C-00000002]: chan_dahdi.c:1225 my_get_callerid: read returned error: Invalid argument |
| 20:58.13 | jonah | any fix for this would be really appreciated thanks |
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| 23:04.59 | kippi | hey |
| 23:05.53 | kippi | I am trying to up the amount of pseudo channels, however /sys/module/dahdi/parameters/max_pseudo_channels is missing |
| 23:06.38 | kippi | how can I get dahdi to create these |
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