| 00:07.40 | igcewieling1 | How many surrealists does it take to screw in a lightbulb? One to hold the giraffe and one to fill the bathtub with brightly colored power tools. |
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| 00:08.17 | WIMPy | While connected to the mains? |
| 00:09.07 | igcewieling1 | WIMPy: I believe all giraffe in the USA are required to have a GFCI, just in case. |
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| 01:29.04 | WIMPy | Has there been a serious change in how chan_dahdi processes dialling information? It looks like you now need immediate=no, but I am very sure that used to be a non working configuration. I also documented it that way. |
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| 02:36.12 | kfife | + |
| 02:36.46 | WIMPy | That's only 2/3 * |
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| 05:15.21 | igcewieling1 | If there was, you'd think it would be in the DAHDI changelog. |
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| 06:18.41 | samneo | hi buddies |
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| 06:42.03 | ChannelZ | o/ |
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| 11:40.50 | Hypfer | so i have one sipgate account which routes into music on hold and then onto voicemail. Is it possible for me to use BLF and transfer the call to a sip phone with the blf button? |
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| 12:15.14 | WIMPy | Yes |
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| 12:15.46 | todwetsprock | join #asterisk |
| 12:16.50 | WIMPy | igcewieling1: It's probably not dahdi, but chan_dahdi. But that's always part of the challenge. There are so many parts. I usually faild to locate the correct component when reporting bugs. |
| 12:19.57 | todwetsprock | hello ! i try to increase atxfernoanswertimeout = 30 (default 15) , but after asterisk restart changes do not apply. whats wrong ? |
| 12:22.33 | WIMPy | Are you even using that feature? |
| 12:23.33 | todwetsprock | *2 works fine, but if unanswered for 15 secs the call is finished. that should be 30 seconds |
| 12:24.17 | todwetsprock | so one has enough time to go to the phone |
| 12:24.59 | WIMPy | I cancelled after 24 sec here which happens to be what I configured. |
| 12:25.18 | todwetsprock | i put it in features_general_custom.conf, that should be oki |
| 12:25.46 | todwetsprock | features_general_custom.conf is included inside features.conf |
| 12:25.50 | todwetsprock | thats oki too |
| 12:26.11 | WIMPy | Oh, that sounds like a FreePBX question. Maybe you should ask in #freepbx. |
| 12:26.43 | todwetsprock | ...or put it inside mysql: |
| 12:27.12 | todwetsprock | mysql->asterisk->features show me all features enabled |
| 12:27.48 | todwetsprock | but i have no clue where to put atxfernoanswertimeout inside mysql. |
| 12:27.55 | WIMPy | Oh, wow. I didn't even know something like realtime features existed. |
| 12:27.59 | todwetsprock | Wimpy: do you have an idea where to put atxfernoanswertimeout inside mysql ? |
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| 12:29.02 | todwetsprock | ...if i put changes there and restart my freepbx, freepbx takes notices of my changes, i experienced that behavior |
| 12:29.43 | WIMPy | Or is that only their databese they use to generate configs? |
| 12:30.01 | todwetsprock | nope |
| 12:30.42 | todwetsprock | its where elastix/freepbx reads n write to, and asterisk is reading too -- to my experience |
| 12:33.37 | todwetsprock | wimpy: you have atxfernoanswertimeout=24 inside your features.conf (or its includes), and thats it ? |
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| 12:36.19 | WIMPy | Yes. Just a single file. |
| 12:36.41 | todwetsprock | wimpy thx, so im at the right direction at least |
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| 13:22.07 | [TK]D-Fender | todwetsprock: Asterisk does not use MYSQL in FreePBX. That's all GUI for the purpose of generating flat configs |
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| 14:28.09 | Ybsen_lucero | good morning! |
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| 14:47.57 | Ybsen_lucero | Should someone help me? |
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| 14:50.02 | [TK]D-Fender | ~ask |
| 14:50.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 14:51.41 | Ybsen_lucero | well... |
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| 14:56.07 | Ybsen_lucero | i need to deploy a Open Source management interface for an Asterisk server who shows in real time the used channels and the status of each logged agent. I used ViciDial but it hasn't graphics to show... what do you recomend? |
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| 14:59.59 | [TK]D-Fender | Depends on your definition of "agent" |
| 15:00.08 | [TK]D-Fender | And "logged" |
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| 15:05.32 | Ybsen_lucero | Agent: the person who handles incoming or outgoing customer calls for a business |
| 15:06.24 | Ybsen_lucero | logged: connected to phone central, no matter the condition (default, available, busy, break...) |
| 15:08.44 | [TK]D-Fender | Those things are concepts you have to invent and define. |
| 15:08.56 | [TK]D-Fender | Because "break" doesn't really exist |
| 15:09.03 | [TK]D-Fender | and "connected is vague |
| 15:09.35 | [TK]D-Fender | Asterisk doesn't talk to "people". It talks to devices. Which is where you need to give a VERY clear definition about what being "logged in" actually means |
| 15:10.03 | [TK]D-Fender | Is that to say they are using some sort of SIP phone that will actually register/deregister at the user's request? |
| 15:10.12 | [TK]D-Fender | Is that exactly how you are doing this? |
| 15:10.46 | [TK]D-Fender | You also need to look at how you define these other states |
| 15:11.54 | *** join/#asterisk theron (~theron@2620:10d:c090:200::9:e078) |
| 15:12.36 | stefan27 | what governs what format (alaw vs ulaw) that the moh class 'ringing' will be played in? on two different asterisks i join a similar queues by sending an offer sdp, where the order of the offreed codecs in both cases are the same '8 0 ...', but one asterisk plays the moh in u-law and the other in a-law, this must be because the two asterisks have different confs, which conf might that be? |
| 15:13.22 | stefan27 | both machines have both files ringing.al and ringing.ul in /var/lib/asterisk/moh/ringing |
| 15:13.58 | [TK]D-Fender | it's chosen based on the codec of the channel |
| 15:14.06 | [TK]D-Fender | The file does not chose. The file is chosen |
| 15:15.55 | stefan27 | but both channels support both codecs? |
| 15:22.58 | igcewieling1 | have you considered naming them with a .ulaw and .alaw extension? |
| 15:24.00 | igcewieling1 | stephanj: maybe both channels support both codecs, but what ACTUAL codec is selected for each leg of the call? |
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| 15:25.41 | stefan27 | Yeah I'll continue investigation, thanks |
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| 15:26.05 | igcewieling1 | "sip show channels" while the calls are active is a good start. |
| 15:26.10 | [TK]D-Fender | Ther is no "both channels" |
| 15:26.26 | [TK]D-Fender | That is a single channel going through dialplan apps. |
| 15:27.15 | [TK]D-Fender | it has a single codec. * is capable of transcoding whatever format it has to (that it supports at all) and will choose the most economic choice available |
| 15:27.47 | igcewieling1 | [TK]D-Fender: what do you think of the bad file extensions on the filenames? |
| 15:28.01 | igcewieling1 | .al and .ul |
| 15:28.01 | [TK]D-Fender | Mistake |
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| 15:28.16 | [TK]D-Fender | not supported AFAIK |
| 15:28.25 | [TK]D-Fender | still have to name them properly |
| 15:29.00 | igcewieling1 | stefan27: if your files ACTUALLY end in .al and .ul then Asterisk is not using the files. |
| 15:30.50 | stefan27 | sip show channel <mychannel> outputs (alaw|ulaw) |
| 15:30.59 | stefan27 | under "Format" column |
| 15:31.07 | igcewieling1 | how about "sip show channels" |
| 15:31.21 | stefan27 | oh i actually meant sip show channels |
| 15:31.34 | igcewieling1 | pastebin the output |
| 15:31.44 | igcewieling1 | next time type what you mean. |
| 15:32.27 | stefan27 | cannot pastebin the output, it contains pstn numbers of a friend |
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| 15:32.40 | igcewieling1 | I'm sorry, I cannot help you further. |
| 15:33.16 | stefan27 | thanks for the help, Im sorry I couldn't provide a more well-defined problem |
| 15:33.35 | stefan27 | I'll have to dig deeper before asking |
| 15:33.43 | [TK]D-Fender | How much is there to saying what the problem is? |
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| 15:35.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:38.10 | [red] | hypotheical and vauge question out of fustration, is pjsip better than chan_sip? i'm having so many issue with chan_sip just freezing and having to kill -9 asterisk |
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| 15:39.20 | [TK]D-Fender | "better" depends on what you need. |
| 15:39.24 | igcewieling1 | [red]: you have some other problem. |
| 15:39.29 | [TK]D-Fender | And should not affect your issue |
| 15:39.34 | [TK]D-Fender | What version are you running exactly? |
| 15:40.11 | [red] | first 13.20 |
| 15:40.13 | skrusty | anyone know why pjsip would almost totally remove the SDP payload from an invite when sending it on to another desintaiton (i.e. inbound call -> dial (sip uri)) |
| 15:40.14 | [red] | err |
| 15:40.21 | [TK]D-Fender | upgrade first |
| 15:40.21 | [red] | 13.6 |
| 15:40.25 | [red] | then 11.20 |
| 15:40.33 | [red] | now back to my trusty 11.14 |
| 15:40.37 | [TK]D-Fender | What kind of load/environment? |
| 15:40.45 | [red] | high load, 100s of calls an hour |
| 15:40.53 | [red] | CentOS7 on a VPS |
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| 15:43.15 | [red] | it used to work perfectly, however i admit doing some pretty significant config changes and now this insane stability problem |
| 15:43.54 | [red] | running 11.14 appears to be the most stable, but after about 12-24 hours of use, it dies at some point |
| 15:44.33 | WIMPy | First you need to define "dies". |
| 15:44.59 | [red] | asterisk continues to run, however, it completely ignores and is unable to send any sip messages |
| 15:45.12 | [red] | calls that are currently on the system are able to wrap up |
| 15:45.35 | [red] | however, doing a core show channels verbose shows the "active" calls and the duration timer continues to increase |
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| 15:45.49 | craigify | I've had channels get into a zombie state. Not sure what else to call them. They would cause asterisk to crawl and nothing fixes it until it restarts. This is a much older asterisk than you are running. |
| 15:46.09 | craigify | and it was usually with IAX, not SIP |
| 15:46.13 | igcewieling1 | I've had similar issues -- turns out DNS was having problems. |
| 15:46.28 | craigify | chan_sip is pretty stable from what I can tell |
| 15:47.10 | craigify | I suppose that doesn't help you much.... |
| 15:47.23 | [red] | igcewieling1: did you see any error messages in the logs that indicated DNS issues? |
| 15:47.57 | igcewieling1 | [red]: send the pastebin link again |
| 15:48.30 | [red] | i didn't generate one yet |
| 15:48.39 | *** part/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 15:48.43 | igcewieling1 | Then how am I supposed to look at the logs? |
| 15:49.01 | [red] | no, i was asking you about your DNS issue |
| 15:49.11 | [red] | how did you figure it out |
| 15:49.14 | igcewieling1 | No, there isn't usually any errors. |
| 15:49.17 | [red] | ok |
| 15:49.22 | [red] | yeah, same situation i'm in |
| 15:49.27 | [red] | no errors at all |
| 15:50.08 | igcewieling1 | try putting entries for all IP addresses in the system into /etc/hosts (correctly formatted) and verify name resolution uses /etc/hosts first. |
| 15:50.16 | [TK]D-Fender | get a good DNS cache.... |
| 15:50.30 | igcewieling1 | I do that and run a local caching nameserver on the box as well |
| 15:50.47 | [red] | interesting |
| 15:50.49 | igcewieling1 | and use dnsmgr |
| 15:51.39 | [red] | yeah, i do have some dns magic i can do |
| 15:51.43 | [red] | it's worth a shot |
| 15:52.13 | [red] | really really odd that if it is a DNS issue, things would just quit working instead of tossing errors |
| 15:52.55 | igcewieling1 | [red]: does any other software complain when it cannot lookup ip addresses of the LOCAL interfaces? Apache does, but I don't know of others at the moment. |
| 15:52.59 | WIMPy | DNS issues can cause all sorts of strange behaviour in Asterisk, especially chan_sip. |
| 15:53.42 | [red] | well the local ip is already in /etc/hosts, so that shouldn't be an issue |
| 15:54.00 | igcewieling1 | I've not gotten complaints for a long time of (local) extension to extension calling not working when the internet goes down. |
| 15:54.00 | [red] | s/local/interface/g |
| 15:54.31 | [red] | :) |
| 15:54.36 | [red] | pets infobot |
| 15:54.57 | igcewieling1 | [red]: all local IPs? |
| 15:55.23 | [red] | no, it's a vps, so it's a "public" ip |
| 15:55.46 | [red] | only one interface |
| 15:56.16 | [red] | the only other thing is i'm using dns names for the sip trunks |
| 15:56.17 | igcewieling1 | All of my hosts have at least 2 interfaces (including localhost), some have 3 or 4 for various things |
| 15:56.22 | igcewieling1 | there you go. |
| 15:56.46 | [red] | yup |
| 15:56.48 | igcewieling1 | run a local caching nameserver and use dnsmgr |
| 15:57.32 | [red] | i will give it a try |
| 15:57.38 | [red] | thanks for saving me once again |
| 16:02.32 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 16:21.51 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 16:41.25 | *** join/#asterisk theron (~theron@2620:10d:c090:200::9:e078) |
| 16:58.38 | *** join/#asterisk qloogkm (sid20721@gateway/web/irccloud.com/x-uzzejqkaezgfzoja) |
| 17:08.00 | *** join/#asterisk rubio (~rubio@181.47.150.170) |
| 17:08.31 | rubio | Hi, anyone can help with ICE conf on asterisk? (i'm trying WebRTC and i can't get audio at all) |
| 17:10.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:11.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:12.44 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 17:14.09 | *** join/#asterisk bluemerlin (~bluemerli@109.236.171.1) |
| 17:14.15 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:21.57 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:25.06 | Kobaz | anyone know the ringtone file size limit for polycom |
| 17:33.40 | *** join/#asterisk theron (~theron@2620:10d:c090:200::a:7c88) |
| 17:36.13 | [TK]D-Fender | I think I got away with a full song I converted back in 2005 |
| 17:36.21 | [TK]D-Fender | should vary based on available RAM on it |
| 17:39.17 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-viqzhgrvkbvhrnuu) |
| 17:42.01 | igcewieling1 | squirrels: cute, fluffy. evil. |
| 17:42.04 | *** join/#asterisk blooregard (~matt@50.141.78.151) |
| 17:44.09 | *** join/#asterisk bulkorok (~Adium@92.206.230.45) |
| 18:04.10 | Kobaz | i think it's 300kb |
| 18:07.58 | Kobaz | i found some stuff on the polycom forms |
| 18:11.17 | *** join/#asterisk brettnem (~brettnem@cpe-72-177-107-10.austin.res.rr.com) |
| 18:12.30 | brettnem | Hey all, I have a funny issue with Asterisk 11. Call is originatied via AMI via Local/<num>@context. I want to be able to *cancel* the ringing call also via AMI. If I simply use the Local Channel with a hangup, it won't hang the call up until it answers. Am I doing it wrong? |
| 18:14.28 | [TK]D-Fender | how are you trying to do this? |
| 18:14.37 | [TK]D-Fender | What method exactly? |
| 18:17.54 | *** join/#asterisk jasonwert (~jason@75-134-81-98.static.aldl.mi.charter.com) |
| 18:18.57 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:21.03 | brettnem | [TK]D-Fender: making the call with ruby-asterisk via AMI, first channel is Local/<num>@context1 and destination is context + exten.. Right before the dial, I save the ${CHANNEL} (which is local since the dial hasn't completed). sometimes during the call someone might hit a button that sends a manager Hangup command with that Local channel name. It looks like it takes it fine, but it does nothign.. if the call connects it immediate |
| 18:21.16 | brettnem | hangs it up |
| 18:22.52 | brettnem | *sometimes during the ringing... |
| 18:34.20 | *** join/#asterisk Swampy (9f3f46f6@gateway/web/freenode/ip.159.63.70.246) |
| 18:35.56 | *** join/#asterisk jasonwert (~quassel@75-134-81-98.static.aldl.mi.charter.com) |
| 18:37.23 | Swampy | I recently updated from 1.6 to 13.1-cert2 and from a T1 to a SIP trunk provider.... now I'm experiencing a 20 second delay in outbound calls. http://pastebin.com/UQAEEHAQ |
| 18:38.18 | Swampy | I know that's a lot of changes at one time, but it was necessary. Anyone ever see this before? |
| 18:38.28 | [TK]D-Fender | brettnem: Try an AMI Redirect and send it off a cliff |
| 18:39.01 | [TK]D-Fender | is that 20s as of the start of the Dial command itself? |
| 18:39.19 | [TK]D-Fender | [Nov 3 08:50:55] == Using SIP RTP CoS mark 5 |
| 18:39.20 | *** join/#asterisk chadxz (Adium@nat/digium/x-dlijpmgzfxehxfgx) |
| 18:39.20 | [TK]D-Fender | [Nov 3 08:51:15] -- Called SIP/11231231234@207.227.999.19 |
| 18:39.22 | [TK]D-Fender | ah |
| 18:39.31 | [TK]D-Fender | iptables limiteing possibly? |
| 18:39.37 | [TK]D-Fender | check your firewall |
| 18:39.45 | Swampy | Immediately after the Dial command |
| 18:40.02 | [TK]D-Fender | Also you should be using a standard peer VS a raw IP like that |
| 18:40.24 | Swampy | No firewalling... directly connected to an MPLS network with the SIP provider. |
| 18:41.10 | Swampy | Not sure i know what you mean by "standard peer VS a raw IP" |
| 18:43.29 | Swampy | I was trying to use the IP address to rule out any DNS delay. |
| 18:43.44 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-yadsscrsmytebbti) |
| 18:43.53 | [TK]D-Fender | Check iptables.... |
| 18:44.21 | [TK]D-Fender | and enable SIP debug to verify the actual packet timinging |
| 18:44.25 | [TK]D-Fender | "sip set debug on" |
| 18:45.10 | Swampy | [TK]D-Fender: OK... I'll check iptables and get a new call sample with debug on... |
| 18:46.16 | *** join/#asterisk areski (~areski@80.174.128.87.dyn.user.ono.com) |
| 18:58.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:00.18 | igcewieling1 | 20 seconds sounds a lot like a direct media issue. |
| 19:03.54 | Chainsaw | igcewieling1: Bit short for a session timer, agreed. |
| 19:04.51 | [TK]D-Fender | that's 20s before that ack of the call |
| 19:04.55 | [TK]D-Fender | being PLACED |
| 19:19.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 19:20.18 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 19:24.05 | Swampy | OK... got pulled away for some other support issues... I disabled the firewall all together, and recreated the call with debug on. http://pastebin.com/WnriHdFe |
| 19:25.23 | Swampy | The pause takes place at line 175 and it looks like Asterisk is not even sending the invite until after the 20 seconds. |
| 19:27.08 | [TK]D-Fender | yup |
| 19:27.14 | [TK]D-Fender | iptables --list |
| 19:27.27 | [TK]D-Fender | Time to verify if the stack is clear |
| 19:29.16 | Swampy | http://pastebin.com/HpUJQ9Sv |
| 19:29.59 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:32.42 | Swampy | Looks empty to me... igcewieling1 mentioned a direct media issue... The SIP provider did recommend that we set canreinvite=no is that still valid? |
| 19:32.54 | [TK]D-Fender | not applicable at all |
| 19:33.01 | Swampy | OK.. |
| 19:33.02 | [TK]D-Fender | Something "core-wise" is going on here |
| 19:36.33 | Swampy | [TK]D-Fender: I was originally running 13.6.0 and decided to reload with 13.1-cert2 just in case it was something in the non LTS version. |
| 19:36.50 | Swampy | Happened in both versions. |
| 19:36.58 | [TK]D-Fender | LTS really doesn't mean anything unless you're a paying customer |
| 19:37.05 | [TK]D-Fender | And the 11 branch? |
| 19:37.22 | *** join/#asterisk dms (~dms@162.207.112.26) |
| 19:37.27 | igcewieling1 | Swampy: canreinvite is not a valid option for Asterisk 11. |
| 19:37.43 | file | 13 is an LTS branch. |
| 19:37.56 | Swampy | I didn't try the 11 branch yet... do you recommend that i do? |
| 19:40.04 | igcewieling1 | Swampy: yes. |
| 19:40.36 | Swampy | 11.6-cert11 or 11.20.0 |
| 19:40.51 | igcewieling1 | Swampy: the latest |
| 19:41.20 | [TK]D-Fender | file: Any inspiration on his system showing : |
| 19:41.21 | [TK]D-Fender | [13:39][TK]D-Fender[Nov 3 08:50:55] == Using SIP RTP CoS mark 5 |
| 19:41.23 | [TK]D-Fender | [13:39][TK]D-Fender[Nov 3 08:51:15] -- Called SIP/11231231234@207.227.999.19 |
| 19:41.46 | [TK]D-Fender | file: basically the SIP COS notic first, and 10 seconds before the invite packet is even sent. |
| 19:41.58 | [TK]D-Fender | iptables is empty |
| 19:41.59 | file | that IP address is wrong so it's trying to resolve it into SOMETHING? |
| 19:42.07 | [TK]D-Fender | and he's dialing an IP direct so no DNS |
| 19:42.18 | [TK]D-Fender | acutally... there is that.. |
| 19:42.30 | [TK]D-Fender | is that IP address simply masked, or is it actually wrong? |
| 19:42.36 | [TK]D-Fender | Swampy: ^^^^ |
| 19:42.38 | [TK]D-Fender | ? |
| 19:43.08 | Swampy | The IP was changed to protect the innocent... |
| 19:43.21 | igcewieling1 | Why are you not dialing by peer? |
| 19:43.24 | Swampy | The call eventually goes through on that IP address |
| 19:44.49 | Swampy | Do you mean why do I have the type=friend instead of type=peer? |
| 19:45.08 | igcewieling1 | Swampy: what do you have inside the [] in sip.conf? |
| 19:45.27 | igcewieling1 | I suspect you have the IP address in there. |
| 19:45.41 | brettnem | [TK]D-Fender: I tried your idea with the AMI redirect to nowhere (actually redirecting to an extension that hangs up. Problem is, the redirect (just like the hangup) does nothing until the local channel is optimized out by the SIP channel being answered |
| 19:46.28 | [TK]D-Fender | brettnem: make sure your Channel is /n |
| 19:46.35 | [TK]D-Fender | to prevent optimizing away |
| 19:46.51 | brettnem | [TK]D-Fender: I'll try that.. but I am doing this before the SIP channel answers, so I'm not sure it'd matter? |
| 19:47.06 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 19:47.16 | Swampy | Here is my SIP config http://pastebin.com/ekDwfFK1 |
| 19:47.17 | [TK]D-Fender | Not sure what the 'running" difference is pre-brdge, but it's a thing to try |
| 19:48.48 | igcewieling1 | Swampy: replace 207.227.999.19 with 9006 on your Dial line and it should work. |
| 19:49.13 | igcewieling1 | otherwise you have no peer/friend/user to Dial |
| 19:49.31 | igcewieling1 | Swampy: you might consider reading the Asterisk book |
| 19:49.58 | brettnem | [TK]D-Fender: even with /n it does the same thing. redirect happens after the new channel is answered, but not when the command is issued |
| 19:50.14 | brettnem | it's like the local channel's ability to respond to ami actions is paused until the call is ANSWERED |
| 19:50.15 | [TK]D-Fender | brettnem: Hrm.... |
| 19:51.16 | Swampy | My dial string is as follows: exten => _91NXXNXXXXXX,n,Dial(SIP/${EXTEN:1}@207.227.999.19,300) |
| 19:51.25 | brettnem | [TK]D-Fender: so I noticed this.. when I perform the initial dial to Local/<num>@context, and that context issues a dial, on the CLI it says that the call is proceeding on SIP/<actual_chan_nam> |
| 19:51.36 | brettnem | like *proceeding* when it's ringing. |
| 19:51.44 | brettnem | now if I could capture THAT channel, I might be able to hang that one up |
| 19:51.48 | [TK]D-Fender | well that is another channel.... |
| 19:51.48 | brettnem | but I can't find a way to capture it |
| 19:52.01 | [TK]D-Fender | are you trying the redirect on the SIP channel, or on the LOCAL? |
| 19:52.13 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 19:52.14 | Swampy | Are you saying to add the port as Dial(SIP/${EXTEN:1}@207.227.999.19:9006,300) |
| 19:52.26 | brettnem | [TK]D-Fender: on the local, because I don't KNOW the SIP channel until it answers (thanks to a Dial "M" macro) |
| 19:53.00 | igcewieling1 | Dial(SIP/${EXTEN:1}@9006,300) |
| 19:53.16 | igcewieling1 | Though I don't know how you are going to do a call if you only have one device setup. |
| 19:53.19 | [TK]D-Fender | You could use a pre-hook to get it |
| 19:53.47 | igcewieling1 | Swampy: Don't dial by IP unless you have a very good reason. |
| 19:54.37 | igcewieling1 | Swampy: Also stop using srvlookup |
| 19:57.20 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 19:57.25 | Swampy | igcewieling1: Can you point me to an example. The dial string listed above actually uses variables, I just converted it to the value to show you what I was sending. I'm not sure how else to day without sending it to a host... I must be missing something. |
| 19:59.04 | Swampy | igcewieling1: Can you please explain what you mean by "replace 207.227.999.19 with 9006 on your Dial line" |
| 20:00.50 | [TK]D-Fender | make a sip.conf section for it |
| 20:00.55 | [TK]D-Fender | just like any other |
| 20:00.58 | [TK]D-Fender | with ty=peer |
| 20:01.04 | [TK]D-Fender | type=peer |
| 20:01.06 | [TK]D-Fender | rather |
| 20:03.36 | igcewieling1 | Swampy: what [TK]D-Fender said. Have you considered reading the Asterisk book? |
| 20:04.07 | igcewieling1 | You have a fundamental misunderstanding of how to setup Asterisk with SIP. |
| 20:07.41 | Swampy | <PROTECTED> |
| 20:08.38 | igcewieling1 | Swampy: stuff setup wrong often works until it doesn't. I never setup any system without all the correct peers and I *NEVER* dial by IP. |
| 20:09.19 | igcewieling1 | That's just asking for all sorts fo complicated tough to diagnose issues. We've spent all this time and are no closer to solving the issue. |
| 20:09.21 | Swampy | igcewieling1: OK.. .point taken... I'll go RTFM for proper SIP setup. |
| 20:09.42 | igcewieling1 | anyway, I have another squirrel to deliver. |
| 20:09.45 | igcewieling1 | bbiab |
| 20:15.42 | *** join/#asterisk mzbotr (~mzbotr@196-102-35-72.mtaonline.net) |
| 20:20.55 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
| 20:24.31 | *** join/#asterisk eofster (~eofster@ip5b4063a6.dynamic.kabel-deutschland.de) |
| 20:27.26 | *** join/#asterisk theron (~theron@2620:10d:c091:200::a:72b6) |
| 20:28.54 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
| 20:33.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:33.27 | *** join/#asterisk arcticfox (~arcticfox@95-31-33-189.broadband.corbina.ru) |
| 20:33.52 | arcticfox | 43770 ///0r1d |
| 20:35.44 | arcticfox | Hello, could someone help with chan_dongle and /run/lock issue with non-root asterisk process? |
| 20:39.07 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 20:39.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:41.58 | WIMPy | What's the issue? |
| 20:49.13 | arcticfox | asterisk couldn't create /run/lock device locks (/run/lock/LCK..ttyUSB1, /run/lock/ttyUSB2) |
| 20:50.18 | arcticfox | in case that station uses webRTC all services runs own users |
| 20:52.58 | WIMPy | Do the lock already exist when it tries them? |
| 20:53.53 | arcticfox | no, lock dir is just empty... only know locks |
| 20:53.57 | arcticfox | knows |
| 20:54.03 | arcticfox | dmnt: known |
| 20:55.49 | WIMPy | Can't you put them somewhere where Asterisk has permissions? |
| 20:56.50 | igcewieling1 | often /var/run or /var/run/asterisk |
| 20:57.26 | arcticfox | it's idea, how to change lock dir? and how to keep locks system-wide? |
| 20:58.18 | arcticfox | I'm about changing lock dir from /var/run (/run/lock) to /var/run/asterisk (/run/lock/asterisk) |
| 20:58.31 | igcewieling1 | I've never looked at chan_dongle, seemed like a huge amount of work with little advantage. |
| 20:58.41 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 20:59.26 | arcticfox | yeah, very big amount of work, but no other access to PSTN... |
| 20:59.34 | arcticfox | only GSM |
| 21:02.26 | igcewieling1 | I suspect a real GSM gateway, a GSM card telephony card, or SIP is what most people use for connecting to the PSTN |
| 21:04.46 | arcticfox | usb dongles i think, GSM-SIP gateway - more time to double transcoding, and it's notebook and then PCI cards also not available. So digium(c) hardware is useless for me :( |
| 21:05.00 | igcewieling1 | you will double transcode anyway. |
| 21:05.22 | arcticfox | sure? gsm->gsm bridging |
| 21:05.53 | WIMPy | Who wants to use the GSM CODEC supported by Asterisk or VOIP phones? |
| 21:06.39 | igcewieling1 | arcticfox: they might be the same "codec" but they are not as far as Asterisk is concerned. |
| 21:07.03 | igcewieling1 | WIMPy: he isn't using voip phones. |
| 21:07.39 | WIMPy | Not? |
| 21:07.55 | igcewieling1 | WIMPy: correct. |
| 21:08.10 | WIMPy | What else do you connect to a laptop? |
| 21:08.29 | igcewieling1 | WIMPy: cell phone using chan_dongle |
| 21:08.40 | arcticfox | only usb hub with few usb dongles |
| 21:08.46 | WIMPy | And the other end? |
| 21:08.54 | arcticfox | other end of the world |
| 21:09.22 | WIMPy | You are using a laptop to bridge one call from GSM to another via GSM? |
| 21:10.20 | arcticfox | mostly - yes, detailed - very sensitive dialplan for a roaming and local CellOPs |
| 21:10.53 | WIMPy | What's the point of doing that? |
| 21:11.16 | igcewieling1 | WIMPy: so he can use the cellular network for calls. |
| 21:11.55 | arcticfox | have you in mind zero cost roaming for whole country in one cellop, and VERY expensive call to others? |
| 21:12.00 | arcticfox | ÑеалÑÐ½Ð°Ñ Ð·Ð°Ñада... |
| 21:12.10 | arcticfox | missed window |
| 21:12.51 | WIMPy | Would make sense if you're involved somehow. But using GSM surely isn't the obvious choice for bridging 3rd parties. |
| 21:15.04 | arcticfox | 0.003 usd/min and 0.2 USD in one city make sense. Obvious - it's for self. |
| 21:15.47 | arcticfox | Some town with a very bad infrastructure. One fixed phone, but very good cell covering. Sad but true. |
| 21:16.47 | brettnem | [TK]D-Fender: what's a pre-hook? |
| 21:17.01 | WIMPy | So you're calling from one GSM MT to another via some provider to reach your laptop? |
| 21:17.22 | [TK]D-Fender | brettnem: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers |
| 21:17.26 | arcticfox | no laptop totally offline, no 3G/HSDPA also |
| 21:17.35 | arcticfox | no data, only voice |
| 21:18.01 | WIMPy | Oh, it was a RPi, not a laptop, right? |
| 21:18.06 | arcticfox | palnned to add LTE dongle for a data. but it's also costs |
| 21:18.20 | arcticfox | nope - laptop - Asus EEE 1015PE |
| 21:19.29 | WIMPy | You just said no laptop. |
| 21:19.36 | arcticfox | seved me well in clean VoIP solutions, and currently planned to work as home gateway on fazenda(village house, dacha) |
| 21:19.49 | arcticfox | missed semicolod |
| 21:19.50 | arcticfox | missed semicolon |
| 21:20.00 | arcticfox | => no, laptop |
| 21:20.13 | WIMPy | So it IS voip to gsm? |
| 21:20.50 | arcticfox | nope :) clean GSM, voip is in plans, but not earlier than some comes to us with LTE |
| 21:21.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:23.03 | WIMPy | I'm not getting it. What is the complete indended call flow? |
| 21:24.09 | arcticfox | call flows GSM->GSM |
| 21:24.20 | arcticfox | GSM in, other GSM out |
| 21:24.44 | WIMPy | And where does the call come from and where does it go? |
| 21:25.08 | arcticfox | come to 1 GSM and out for another (CO) |
| 21:25.17 | arcticfox | (CO = Capt. Obv.) |
| 21:25.44 | WIMPy | Where does it come from to 1st GSM and where doe it go to from 2nd GSM? |
| 21:26.03 | WIMPy | Are you trying to implement a hosted PBX for others via GSM dongles? |
| 21:26.58 | arcticfox | omg... when will you stop about 3rd partys |
| 21:27.29 | arcticfox | for hosted PBX u need at least internet at place of colo |
| 21:27.40 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 21:28.01 | WIMPy | Well, I have no idea what you're up to. |
| 21:28.02 | arcticfox | there is no internet, no fixed line, no 3G/LTE internet, just only classic GSM + GPRS |
| 21:28.14 | arcticfox | does it needed? |
| 21:28.27 | igcewieling1 | arcticfox: something like GSM Mobile Phone -> chan_dongle -> Asterisk -> chan_dongle -> GMS Mobile phone ? |
| 21:28.47 | arcticfox | yeah, as i understend - exatly |
| 21:28.57 | igcewieling1 | what a horrible idea. |
| 21:29.03 | WIMPy | I already asked that. |
| 21:29.16 | arcticfox | i know, just for excuses it's Huawei E173 usb dongles |
| 21:29.22 | WIMPy | Ok, I wrongly assumed the dongles were connected to a laptop. |
| 21:29.36 | arcticfox | sorry, now i'm confused. |
| 21:29.46 | WIMPy | too |
| 21:31.20 | arcticfox | Total schema: 1 Laptop, have 3 USB donles (Huawei E173, with unlocked voice). One of them used to recieve and send calls to my cell-phone, other 2 - city and roaming egress line |
| 21:32.01 | igcewieling1 | have you tried chan_mobile? |
| 21:32.16 | arcticfox | isn't chan_mobile for bluetooth? |
| 21:32.25 | WIMPy | So the dongles ARE connected to a laptop. |
| 21:32.36 | arcticfox | yeah, they placed in usb sockets |
| 21:32.41 | WIMPy | So you use it to make two mobile calls instead of one? |
| 21:32.58 | arcticfox | nope ;) just for more suitable billing |
| 21:33.29 | igcewieling1 | he means "cheaper calls" |
| 21:33.41 | arcticfox | exactly |
| 21:33.46 | WIMPy | wonders how two calls can be cheaper than one. |
| 21:34.10 | rrittgarn | instead of using the h exten, is there a way to pop a hangup handler on to every single call no matter where it originates in dialplan? or do i have to get creative on how I dial anything to pop a hangup handler onto the channel? |
| 21:34.16 | *** join/#asterisk Trioxin (~Trioxin@2602:306:3ad9:d460:78a0:326e:ebdc:b1ad) |
| 21:34.16 | arcticfox | one incoming (free) and one in same-net-roaming (free) |
| 21:34.54 | arcticfox | it's cheaper than one incoming in roaming (0,2-1 USD/min) |
| 21:35.28 | arcticfox | and much cheaper than one egress (about 1USD in roaming) |
| 21:36.06 | Trioxin | what would be minimum server specs for a KVM VPS @ http://ramnode.com/vps.php ? Something like 12 extensions and at the most maybe half as many channels in use at any given time? Also call recording, IVR, departments and whatnot which shouldn't add any overhead beyond disk space. |
| 21:36.34 | WIMPy | rrittgarn: With hangup handlers you control what's happening at the beginning of a call, with h extensions it depends on where you end up in the dialplan. |
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| 21:42.01 | *** part/#asterisk arcticfox (~arcticfox@95-31-33-189.broadband.corbina.ru) |
| 21:42.28 | rrittgarn | couple days ago i was turned on to hangup handlers as a replacement for the h extension because setting cdr values in the h exten isn't working in 13 - so trying to transition my dialplan over to hangup handlers instead - hoping that will fix it. Otherwise more creativity will be required |
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| 21:59.32 | igcewieling1 | rrittgarn: Paste your current line for setting the hangup handler. |
| 22:02.13 | lvlinux | Trioxin: |
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| 22:18.44 | rrittgarn | @igcewieling1, I'm not currently sure where to put it dialplan wise to make it attach to all calls. I have different entry points into dialplan based on what was dialed from local extensions and incoming. The only 'easy' one is the outbound calls via a SIP trunk because i have those go through the same macro |
| 22:30.09 | igcewieling1 | rrittgarn: 1) setting a hangup handler is, in theory, the same as setting a dialplan variable. 2) it is possible to have variables specified in the peer entry in sip.conf to automatically set a dialplan variable anytime the device makes a call. |
| 22:30.45 | igcewieling1 | see "setvar" in the sip.conf.sample file in the asterisk source tree |
| 22:30.59 | igcewieling1 | I do not know if this will work, but you might consider trying it. |
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| 22:42.18 | rrittgarn | I will thanks igcewieling |
| 22:42.43 | rrittgarn | unrelated question, in 13, does res_parking play nice with realtime? |
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| 23:50.38 | Trioxin | lvlinux, ? |
| 23:50.54 | lvlinux | Trioxin: what? |
| 23:51.03 | Trioxin | you mentioned me |
| 23:52.00 | lvlinux | oh, iddn't actually mean to send that, but what I would have said is a 1 core 256MB VPS should work great and really be overkill. You may be able to get by with 128MB. |
| 23:52.19 | wyoung | Hi, does asterisk support RFC 3326? In particular, so my Snom SIP phones don't log calls as missed if another handset handled the call |
| 23:52.42 | wyoung | Or do I still need to apply a patch to Asterisk from the snom website? |
| 23:53.20 | wyoung | [TK]D-Fender: Good morning |
| 23:54.00 | [TK]D-Fender | "core show application dial" <--------------- |
| 23:54.36 | wyoung | [TK]D-Fender: Was that directed to me? |
| 23:54.50 | [TK]D-Fender | yes |
| 23:56.50 | Trioxin | lvlinux, okay thx, I would have wound up doing 512 or a gig with multiple cores. I just want the connections to remain good. Then there's the issue of what exactly to install with all these different flavors of asterisk. I have a list somewhere. Just don't want to spend days setting it up. I used to use trixbox for my initial install but that's pretty much long gone. |
| 23:57.47 | wyoung | [TK]D-Fender: ok thank you, I will have a read through although which bit in particular should I be reading? |
| 23:57.58 | wyoung | [TK]D-Fender: the u flag? |
| 23:58.14 | [TK]D-Fender | the one that says .... exactly what you asked for |
| 23:59.16 | lvlinux | Trioxin: I just ran HTOP on my 256MB VPS with Debian 7 and Asterisk and it says it's using 106MB RAM. |
| 23:59.47 | wyoung | hehe ok [TK]D-Fender :) I hope I am running the version of Asterisk that supports that feature |
| 23:59.55 | lvlinux | Trioxin: That's including mysql server for CDR logging and Nginx for a web interface to CDR info :-) |
| 23:59.58 | [TK]D-Fender | Is it from the alst decade? |