| 00:01.33 | Scorp1us | ok finally i got some, but its just 3 lines |
| 00:02.00 | Scorp1us | <PROTECTED> |
| 00:11.42 | [TK]D-Fender | Scorp1us: We haven;'t seen that complete call yet. |
| 00:11.51 | [TK]D-Fender | Scorp1us: And no ANSWER |
| 00:14.10 | Scorp1us | well that's the thing. |
| 00:14.29 | Scorp1us | I can't answer it if it doesn;t call me |
| 00:14.36 | *** part/#asterisk mjordan (~matt@nat/digium/x-nsegdciezllhmjbz) |
| 00:14.50 | Scorp1us | if i call it, it works |
| 00:15.01 | Scorp1us | but calling out is the problem |
| 00:22.50 | charkee | hi guys, do you have any recommendation of a version control system applicable for monitoring changes to asterisk configuration files as well as supporting php/bash scripts |
| 00:23.51 | charkee | basically it's to make sure every changes to the configs are tracked and reverted if necessary |
| 00:24.27 | Scorp1us | they all do |
| 00:24.43 | Scorp1us | as for the configuration management... that's up to you |
| 00:25.01 | Scorp1us | git is fine, subversion, etc. |
| 00:25.53 | WIMPy | Yes. Anything from tar to rsync and from svn to git. |
| 00:27.36 | Scorp1us | [TK]D-Fender: so how do I get an ANSWER if it isn't calling me? |
| 00:28.25 | charkee | yeah I've leaning on using git, have you guys had any first hand experience using git for asterisk config management, any caveats? |
| 00:28.40 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 00:28.46 | [TK]D-Fender | Scorp1us: We still haven't seen a proper call yet |
| 00:29.15 | Scorp1us | I think that's the problem |
| 00:29.23 | Scorp1us | i don't have one, but I should |
| 00:32.23 | Scorp1us | here is all I get: http://pastebin.com/NV3YAceH |
| 00:37.05 | [TK]D-Fender | Scorp1us: "sip show peer callcentric" |
| 00:37.27 | [TK]D-Fender | Scorp1us: That isn't a new pastebin either |
| 00:37.35 | [TK]D-Fender | scor it's just a re-paste |
| 00:38.58 | [TK]D-Fender | Really destroying SIP dialog '4b71f504562072053cd64b093c735849@127.0.0.1' Method: REGISTER - http://pastebin.com/dZQBPw2d #14 |
| 00:39.29 | Scorp1us | i don't know what to show you. that;s what happens when I drop a call file in |
| 00:39.30 | [TK]D-Fender | Really destroying SIP dialog '4b71f504562072053cd64b093c735849@127.0.0.1' Method: REGISTER - http://pastebin.com/NV3YAceH #5 |
| 00:39.47 | [TK]D-Fender | EXACLTY the same SIP transaction #'s? |
| 00:39.53 | Scorp1us | it's not a re-paste, its a fresh one |
| 00:39.56 | [TK]D-Fender | I don't hink that is a NEW call |
| 00:40.10 | Scorp1us | i copy the same call file in for testing |
| 00:40.18 | Scorp1us | same content |
| 00:40.41 | [TK]D-Fender | that is not legit |
| 00:40.47 | Scorp1us | why not? |
| 00:40.51 | [TK]D-Fender | Pastebin what I just requested |
| 00:41.10 | Scorp1us | what did you request? |
| 00:41.20 | [TK]D-Fender | [19:36][TK]D-FenderScorp1us: "sip show peer callcentric" |
| 00:41.32 | Scorp1us | ah missed that |
| 00:41.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.100) |
| 00:42.21 | Scorp1us | http://pastebin.com/dzB4YNwU |
| 00:44.03 | [TK]D-Fender | <PROTECTED> |
| 00:44.16 | [TK]D-Fender | <PROTECTED> |
| 00:44.25 | [TK]D-Fender | it has not resolved an IP to call at all |
| 00:44.31 | [TK]D-Fender | You cannot call out this peer |
| 00:44.36 | Scorp1us | ah |
| 00:44.39 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 00:44.44 | Scorp1us | what file has those settings? |
| 00:44.50 | [TK]D-Fender | ''sip.conf <- |
| 00:44.55 | Scorp1us | k, thaks |
| 00:46.18 | Scorp1us | would you know where in the FreePBX GUI is it? |
| 00:46.37 | [TK]D-Fender | your TRUNK |
| 00:47.12 | Scorp1us | in the peer details box? |
| 00:47.43 | [TK]D-Fender | yes |
| 00:48.02 | Scorp1us | awesome |
| 00:48.14 | Scorp1us | I have no idea why this would have changed |
| 00:50.26 | Scorp1us | bows to [TK]D-Fender |
| 00:50.53 | *** join/#asterisk smirker (~x@CPE-123-211-202-192.lnse4.woo.bigpond.net.au) |
| 00:50.55 | Scorp1us | that's what it was. I guess when they moved my host they also upgraded me to a version that needed that parameter |
| 00:51.09 | [TK]D-Fender | No, |
| 00:51.21 | [TK]D-Fender | *'s requirements for this have never changed |
| 00:51.30 | Scorp1us | huh |
| 00:52.08 | [TK]D-Fender | Same basic settings for over a decade |
| 00:53.32 | Scorp1us | Then its a Schrödenbug. because this worked until November 5th. |
| 00:53.42 | Scorp1us | And not since then until tongiht |
| 00:54.07 | Scorp1us | Anyway, I really appreciate it. If you were local i'd buy you a beer |
| 00:54.19 | Scorp1us | (assumung you are over 21) |
| 01:14.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.104) |
| 01:18.20 | Scorp1us | Anyway, thanks again |
| 01:32.26 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.133) |
| 01:35.38 | *** join/#asterisk unstable (unstable@tor/regular/sid) |
| 01:37.03 | unstable | I live in Manhattan, twice in the last two decades there was a power outage where landlines were the only way to make a call. Cells/Internet went down. Though it's $40 a month for service, and I never use the landline besides emergencies. Is there a better way to do what I"m doing? |
| 01:38.00 | unstable | eg, how can I have a working phone during outages, without paying $40 a month |
| 01:39.03 | [TK]D-Fender | my guess would be UPS <- |
| 01:39.31 | WIMPy | Donate some batteries to the ISPs or PLMN operators. |
| 01:41.20 | outtolunc | openbts on a raspberry pi strapped to a drone? ;) |
| 01:41.35 | paulc | +1 for that option :) |
| 01:41.51 | [TK]D-Fender | WIMPy: Nowhere do I see how "landlines" were the only option... and that carriers went down.... |
| 01:42.09 | [TK]D-Fender | WIMPy: So far all I get is a loose feeling that user sites had no power |
| 01:42.36 | WIMPy | Depends on how you read it I guess. |
| 01:42.46 | [TK]D-Fender | pretty iffy description to start with |
| 01:47.53 | cusco | I guess he meant landline has voltage from elsewhere even though you neighbour has no power |
| 01:48.26 | WIMPy | Yes. That's a pretty big advantage of real phone lines. |
| 01:49.20 | [TK]D-Fender | Doesn't say anything about powering a PBX behind them... |
| 01:49.28 | [TK]D-Fender | which is still an issue |
| 01:49.48 | [TK]D-Fender | "internet down" is still kinda grey. |
| 01:49.55 | WIMPy | If you need it, yes. But you can always connect a phone. |
| 01:50.01 | [TK]D-Fender | no provider was functional during that time? |
| 01:50.26 | WIMPy | And soem PBXs have a physical fail over that connects the first phone to the line. |
| 01:50.29 | [TK]D-Fender | So if consumer has power backup.... noone was working on ANY technology? |
| 01:50.46 | rrittgarn | wonders if unstable is simply trolling #asterisk |
| 01:51.11 | [TK]D-Fender | [20:50]WIMPyAnd soem PBXs have a physical fail over that connects the first phone to the line. <- Cisco/Link SPA-3102 does this already ont eh cheap |
| 01:51.33 | [TK]D-Fender | rrittgarn: Not yet... he only dropped the first bomb... no subsequent strikes ;) |
| 01:51.50 | WIMPy | Not sure I'd call that a PBX, but yeah. |
| 01:52.51 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 02:04.53 | unstable | rrittgarn: no, I'm not trolling. |
| 02:05.02 | unstable | Or at least not intentionally. |
| 02:06.27 | unstable | I'm actually looking for creative alternatives to my CLEC in NY. |
| 02:06.38 | unstable | Minimal price from Verizon for a line with caller ID and call waiting seems to be about $45. |
| 02:07.47 | unstable | What I mean is.. during a power outage, or other emergencies.. the cell network gets flooded and cell phones don't work. The Internet doesn't work. My landlines work. |
| 02:08.19 | unstable | I want to have a contingency plan for the couple times every decade when there is an emergency, so I can still make calls. Though without paying $40 a month. |
| 02:08.51 | unstable | It ends up being like $5k a decade to have a contingency plan that works using traditional phone lines. I'm looking for creative alternatives. |
| 02:14.39 | [TK]D-Fender | Clarify "the internet doesn't work". |
| 02:14.47 | [TK]D-Fender | NO provider's infrastructure was up? |
| 02:30.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:38.28 | *** join/#asterisk j4jackj (jack@172.218.203.54) |
| 02:39.01 | j4jackj | I'm trying to remake one of the tunes featured in Asterisk |
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| 02:58.31 | j4jackj | hi vlad_starkov, nice to see you... |
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| 04:57.10 | mattsl | I have a question that I feel embarrassed to ask given all the other related things I do know, and it's not actually Asterisk related at all, but I'm sure someone here knows. It just goes to show the ignorance I've managed to retain by never doing any OSP work. At the demarc when you have the nice big green box from the LEC (painted the same color as their crossboxes), what is the technical name for that? i.e. when before it goes to a 6 |
| 05:05.17 | *** join/#asterisk areay (~areay@87.113.217.248) |
| 05:08.14 | ChannelZ | I have no idea what any of that means so you're ahead there |
| 05:10.24 | areay | hi all.. having trouble with sip peers - if i register 2 peers from the same ip i get a username/auth name mismatch when i dial out... i can get around this by setting insecure=invite but the wrong peer appears in the From header of the invite request |
| 05:11.00 | ChannelZ | they should be using different ports.. do they not show up in 'sip show peers' that way? |
| 05:11.44 | ChannelZ | and are they type=peer ? |
| 05:12.00 | areay | ah, i'm using webrtc2sip and they're registered from the same port :( makes sense now |
| 05:12.51 | areay | is there a way to match on username instead of port and ip? |
| 05:13.50 | areay | but yeah they're both type=peer |
| 05:15.59 | ChannelZ | Well you can make them type=user which will only match by name. |
| 05:16.19 | ChannelZ | Although they can't receive calls |
| 05:16.27 | ChannelZ | only make them |
| 05:16.38 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 05:17.32 | ChannelZ | but if they are somehow using the same source port on the same IP that already doesn't work so I assume you're only making calls. |
| 05:18.49 | areay | both can receive incoming calls fine even though asterisk sees them as coming from the same ip/port... webrtc2sip takes care of routing it all |
| 05:18.55 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.110) |
| 05:19.26 | areay | i just get the wrong peer in the From header which means the phone thinks the call is coming from itself. still connects fine tho |
| 05:20.25 | areay | i suppose if the only problem is the bad caller info i could subscribe to the ami event and just have the client display the right username |
| 05:20.39 | ChannelZ | Well I guess you can do type=friend which will give you a user (match by name) and peer (match by ip/port) |
| 05:20.56 | areay | let me try that |
| 05:22.20 | j4jackj | hi vlad_starkov |
| 05:23.22 | areay | nope :( unfortunately still getting the wrong username... it's definitely asterisk changing the username in the invite though |
| 05:24.44 | ChannelZ | What do you mean it's changing it? Are you calling from one to the other or... |
| 05:25.17 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 05:25.43 | ChannelZ | If you turn on SIP debug it should show you the peer it chose. If the From name matches the [peername] that should get matched first if the type=friend or user.. at least I thought |
| 05:27.32 | areay | yeah calling from one to the other: the invite from the first peer is sent from webrtcsip with From:1001,To:1002 but asterisk sends the invite back to webrtc2sip as From:1002,To:1002 |
| 05:35.24 | areay | i had 'fromuser' set in my sip peer definition! removed that and it works :) |
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| 05:40.14 | ChannelZ | details |
| 05:40.17 | ChannelZ | :) |
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| 08:08.36 | smirker | Hi guys, using Asterisk 11.6.0, I am extracting a custom SIP header value using ${SIP_HEADER(X-From-Client)}. This works fine when the call was initiated with an INVITE. However when doing a blind transfer, a REFER is used, and the SIP_HEADER value is blank. I used 'sip set debug on' and looked at the REFER request that was generated, and it correctly contained the X-From-Client custom |
| 08:08.36 | smirker | header. |
| 08:08.47 | smirker | Is there any way to extract a custom header from a blind transfer (ie REFER) ? |
| 08:10.54 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:13.07 | mirela666 | Morning, anyone had experiance with sipml5 (javascript phone) and connecting it to asterisk trough webrtc2sip websocket? |
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| 09:23.40 | michael_work | hey if i do have duplicated variables (after attended transfer checked on 1.8.X) -- someone else example http://lists.digium.com/pipermail/asterisk-users/2010-August/252927.html |
| 09:23.46 | *** join/#asterisk calum_ (~calum_@92.40.248.91.threembb.co.uk) |
| 09:23.47 | michael_work | how do i get both of those? |
| 09:23.59 | michael_work | i mean i want to check in AGI if i have two |
| 09:34.11 | WIMPy | Not, sure, but I think MASTER_CHANNEL should give you wat you want. |
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| 10:08.13 | michael_work | WIMPy, thanks, gonna check |
| 10:08.23 | michael_work | the funny thing that it happens only on incomming |
| 10:18.59 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
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| 10:29.05 | michael_work | WIMPy, MASTER_CHANNEL(value) returns same as value :( |
| 10:29.35 | Psi|4ward | hey there, why is my DEVICE_STATE() = NOT_INUSE even if theres no SipClient registred? |
| 10:30.18 | michael_work | while in dump chan i see it twice and each has different parameter |
| 10:33.21 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 10:42.58 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 10:43.10 | xytis | Hi |
| 10:43.25 | xytis | I am trying to set up realtime sip |
| 10:43.43 | michael_work | ok |
| 10:43.45 | michael_work | and? |
| 10:43.54 | xytis | I reached the point where I may login to asterisk using webrtc clients using realtime data from database |
| 10:44.19 | xytis | yet all calls from realtime user to static user results in 'not acceptable here' |
| 10:44.34 | xytis | Can You provide any hints what have I done wrong? |
| 10:45.34 | michael_work | what do you see in CLI |
| 10:45.45 | michael_work | and do you use encryption |
| 10:45.58 | xytis | encryption is enabled |
| 10:46.12 | xytis | CLI without sip debug on is silent on this matter. |
| 10:47.01 | michael_work | core set verbose 5 |
| 10:47.16 | xytis | silent with verbosity, I meant. |
| 10:47.49 | michael_work | try this : |
| 10:47.51 | michael_work | same => n,Set(CHANNEL(secure_bridge_signaling)=1) |
| 10:47.51 | michael_work | <PROTECTED> |
| 10:47.53 | michael_work | :) |
| 10:48.42 | michael_work | before calling |
| 10:48.45 | michael_work | or opposite |
| 10:48.55 | michael_work | depends if your exten you call is secured or not |
| 10:49.15 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:49.21 | michael_work | as well you can use tcpdump and check in wireshark what's is going on there |
| 10:55.46 | xytis | Ok, I checked above two lines with static user. Pardon my stupidity, but it seems that I use unsecured extensions. |
| 10:56.06 | xytis | Then I called the same extension via dynamic user. |
| 10:56.39 | xytis | I appears that I receive SDP packet with 488 'before' entering dialplan |
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| 11:15.48 | kippi | hey |
| 11:16.36 | kippi | with a .call file is there anyway to make it ring a context? What I want to do is ring phone 1 if that phone isn't answered hunt to another one |
| 11:17.00 | kippi | but I don't want it to ring an extenstion, this is for an alert |
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| 13:14.16 | *** join/#asterisk matthew-moretalk (~matthew@88.96.27.150) |
| 13:14.30 | matthew-moretalk | hey im having an issue with asterisk where its taking a long time to accept initial invites and respond with a 100 trying. any ideas? its not a latency issue and its not a firewall issue. |
| 13:14.47 | matthew-moretalk | we are using asterisk realtime |
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| 13:16.43 | Greenlight | DNS issue is my bet |
| 13:17.08 | Greenlight | BUt if you're using reamtime i could be slow db i suppose |
| 13:17.22 | matthew-moretalk | currently just using straight ip addresses at the moment no dns involved |
| 13:17.41 | Greenlight | Doesn't it do a reverse lookup ? |
| 13:17.42 | matthew-moretalk | anyone know if you can debug realtime requests from asterisk? |
| 13:18.17 | matthew-moretalk | why would it need to do a reverse lookup greenlight? |
| 13:18.32 | Greenlight | I've just got a feeling that it does |
| 13:18.36 | Greenlight | Could well be wrong |
| 13:20.56 | Greenlight | Try enabling debug and full verbosity and see if you can tell where the delay is |
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| 13:22.30 | matthew-moretalk | ill give that a try thanks. |
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| 13:50.56 | WIMPy | michael_work: Oh, so there are two variables with the same name on one channel? |
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| 13:53.33 | michael_work | WIMPy, yes i found the place in code that actually leads to it |
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| 13:53.40 | michael_work | try to found workaround |
| 13:53.55 | michael_work | the problem that incmoing and outgoing calls are acting different |
| 13:54.10 | WIMPy | Ok, that must be a bug then. |
| 13:54.37 | michael_work | are you related to asterisk team btw? |
| 13:55.39 | WIMPy | No |
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| 15:17.53 | lnb | tech guy at office for mikrotic said the rb2011 will get rid of the sip breaking.. yay |
| 15:18.16 | Penguin | Way to go, sales department! |
| 15:18.21 | lnb | for sure |
| 15:18.40 | lnb | i only used that device for one day and it has a ton of features with its OS |
| 15:20.17 | *** part/#asterisk fullstop (~fullstop@64-121-16-14.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
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| 15:24.12 | lnb | http://routerboard.com/RB2011UiAS-RM |
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| 16:28.57 | polysics | hi! so, I was wondering, I am trying to make an Asterisk server "pretend" it is a UA |
| 16:29.06 | polysics | since we need to drive and sink some load |
| 16:29.27 | polysics | I configured a register line on the "sink" box to register to the "load" box |
| 16:30.15 | polysics | is it a good thing to just "wait" forever? |
| 16:30.24 | polysics | or maybe it's better to wait just a long time |
| 16:33.03 | Penguin | Asterisk doesn't need to pretend it's a UA. |
| 16:33.52 | pabelanger | ! |
| 16:34.15 | pabelanger | If you are load testing, you want to use SIPp |
| 16:34.34 | Penguin | Based on what you said you want to accomplish, I have no idea what you're doing. |
| 16:34.41 | polysics | so, rationale is, we are not load testing the SIP stack |
| 16:34.46 | polysics | but the application logic |
| 16:34.56 | Penguin | You can register asterisk to another asterisk. |
| 16:35.08 | Penguin | You can make calls between them. |
| 16:35.12 | polysics | I have agents waiting in a confbridge, and I dial out to customers, then join them to the confbridge if the customer answers |
| 16:35.28 | polysics | the dialing and additional logic is done through Adhearsion and Redis |
| 16:35.42 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:35.46 | polysics | my reasoning was, since I am not driving the load through "dialing a number" |
| 16:36.01 | polysics | I might as well have another Asterisk act as a transparent client |
| 16:36.14 | Penguin | Go for it. |
| 16:36.21 | Penguin | That's exactly what asterisk does. |
| 16:36.25 | polysics | posing as both agents and customers, and essentially playing tt-monkeys all day. |
| 16:36.27 | Penguin | ~b2bua |
| 16:36.27 | infobot | methinks b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 16:36.43 | polysics | or lots-of-monkeys if I can find it. |
| 16:36.45 | polysics | BUT |
| 16:36.49 | polysics | this is my current setup |
| 16:36.50 | polysics | https://gist.github.com/polysics/56d0220a2e7fb4671d58 |
| 16:37.08 | polysics | and when I originate, call goes through the dialplan and falls through, call do not stay up |
| 16:37.18 | polysics | the goal is for that call to last for the whole 3600 seconds |
| 16:37.31 | polysics | can someone spot something that is wrong ,please? |
| 16:37.49 | pabelanger | polysics, show us the issue, get up log files |
| 16:38.04 | pabelanger | everything will be logged and tell you what the issue is |
| 16:39.09 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 16:39.40 | Penguin | polysics: The problem seems to be that you are not using MusicOnHold() like you should be. |
| 16:39.50 | Penguin | same => n,Set(CHANNEL(musicclass)=default); |
| 16:39.53 | Penguin | same => n,MusicOnHold(); |
| 16:40.00 | Penguin | It will sit there until you end it. |
| 16:40.58 | Penguin | You could set a timeout value prior to running MusicOnHold() if the call won't end on its own. |
| 16:41.01 | polysics | pabelanger: load side: https://gist.github.com/polysics/d4d782e320c2ad2ad541 - sink side: https://gist.github.com/polysics/5e8e94d47099b1cfed0a |
| 16:41.41 | polysics | Penguin: the idea was to have timeout as a "safety feature" :) |
| 16:42.01 | pabelanger | polysics, https://gist.github.com/polysics/d4d782e320c2ad2ad541#file-gistfile1-txt-L301 |
| 16:42.06 | pabelanger | it is hanging up |
| 16:42.07 | Penguin | Set the absolute timeout to 3600 if that's what you need. |
| 16:42.33 | pabelanger | since the prompt is only like 4 seconds long |
| 16:42.55 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:43.02 | pabelanger | polysics, go look at the Asterisk testsuite |
| 16:43.08 | pabelanger | there are many examples on how to do this |
| 16:43.29 | polysics | so the originate finishes executing on its own, basically? |
| 16:43.45 | [TK]D-Fender | originate SIP/301 application Wait |
| 16:43.49 | [TK]D-Fender | I see no timeout there |
| 16:44.04 | polysics | oh, I got the wrong command from history then, sorry |
| 16:44.27 | [TK]D-Fender | <PROTECTED> |
| 16:44.31 | polysics | still no cigar, though |
| 16:44.37 | polysics | Wait changes nothing |
| 16:44.45 | [TK]D-Fender | Which is what we see |
| 16:44.55 | Penguin | When that file is done playing, guess what happens. |
| 16:45.13 | [TK]D-Fender | <PROTECTED> |
| 16:45.23 | [TK]D-Fender | That is a fixed playback called against the originated channel |
| 16:45.33 | pabelanger | Right |
| 16:45.37 | [TK]D-Fender | Indeed a stupid setup |
| 16:45.47 | [TK]D-Fender | that is not "indefinite on both sides" |
| 16:45.51 | Penguin | When the file is finished playing, the dialplan moves on to the next step. If there is no next step, the call ends. |
| 16:45.52 | [TK]D-Fender | or "fixed long" |
| 16:46.13 | [TK]D-Fender | not "if there is no next step", he's using "application"... that is always and only a single step. |
| 16:46.29 | [TK]D-Fender | this is outside the concept of dialplan & priorities\ |
| 16:47.13 | Penguin | I couldn't determine that from the command he showed. originate SIP/301 application Wait didn't really have anything to do with what the verbose output showed. |
| 16:47.53 | polysics | I got the wrong command from history. Re-pasting the proper output |
| 16:48.01 | [TK]D-Fender | Penguin: Correct, but the output showed that he used "application" thus no "steps exist, just a single call to Playback which runs out an d*poof* |
| 16:48.29 | polysics | load side with Wait: https://gist.github.com/polysics/d4d782e320c2ad2ad541 |
| 16:48.30 | Penguin | I guess I don't see a problem. Everything looks like it did what it was supposed to do. |
| 16:49.33 | *** part/#asterisk matthew-moretalk (~matthew@88.96.27.150) |
| 16:49.37 | polysics | sink side just says: https://gist.github.com/polysics/5e8e94d47099b1cfed0a |
| 16:49.37 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:49.58 | [TK]D-Fender | polysics: devbox*CLI> originate SIP/301 application Wait <-- what part of "you aren't telling it HOW LONG" are you missing? |
| 16:50.18 | [TK]D-Fender | polysics: -- Launching Wait() on SIP/301-0000000d |
| 16:50.22 | polysics | doesn't default to "forever"? |
| 16:50.24 | Penguin | Try supplying a time to wait. |
| 16:50.42 | [TK]D-Fender | nothing is nothing |
| 16:51.48 | polysics | oh, giving it a time works, let's see if it tears down properly |
| 16:52.01 | [TK]D-Fender | and people aren't reading instructions and making very bad assumptions |
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| 17:13.58 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:18.17 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.5) |
| 17:18.40 | ruben231 | hi guys is it possible to have fax feature on an asterisk server 1.4 |
| 17:18.45 | XandriX | i take it its possible to make an incoming call ring all 3 available phones at the same time and who ever awnsers first awnsers ? |
| 17:19.00 | [TK]D-Fender | ruben231: what "fax feature"? |
| 17:19.13 | [TK]D-Fender | XandriX: Correct/. Dial all 3 |
| 17:19.22 | [TK]D-Fender | XandriX: "core show application dial" |
| 17:19.24 | Chainsaw | ruben231: Did you want to send outbound faxes, or receive faxes and turn them into e-mails? |
| 17:20.50 | ruben231 | Chainsaw: all those features you mention..is it possible |
| 17:21.02 | ruben231 | basically a fax function.. |
| 17:21.04 | XandriX | [TK]D-Fender: thx :) and o/ Penguin :P |
| 17:21.30 | [TK]D-Fender | [12:21]ruben231basically a fax function.. <- there are MULTIPLE "fax functions". Give us a SPECIFIC list of exactly what you want to do |
| 17:22.15 | ruben231 | ok: outbound fax- received fax and turn into emails |
| 17:23.52 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:25.44 | [TK]D-Fender | ruben231: Outbound fax from what? |
| 17:27.26 | XandriX | oh man thats easier than i thought to do ( my ring all 3 phones thing) |
| 17:29.32 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 17:30.08 | ruben231 | [TK]D-Fender: from our asterisk server |
| 17:30.27 | [TK]D-Fender | ruthat is not a "how". There is no technology of origin there. |
| 17:30.39 | [TK]D-Fender | ruben231: that is not a "how". There is no technology of origin there. |
| 17:31.28 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 17:33.09 | ruben231 | [TK]D-Fender: oh ok sorry, how does fax function on asterisk somehow |
| 17:33.34 | [TK]D-Fender | ruben231: You seem to have a serious conceptual problem wirth the idea of where something starts, and where it ends |
| 17:35.34 | XandriX | [TK]D-Fender: would this make sense ? exten => s,4,Dial(SIP/5001&SIP/5002&SIP/5003&SIP/5004&SIP/5005,40) |
| 17:35.47 | [TK]D-Fender | XandriX: Sure, why not... |
| 17:36.01 | XandriX | [TK]D-Fender: i was asking cuz i suck at syntax and wanted to make sure :) |
| 17:36.08 | [TK]D-Fender | XandriX: Try & see. |
| 17:36.40 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:36.43 | Penguin | I'm not a fan of your device naming convention, but the syntax is right. |
| 17:37.19 | Penguin | And I'm not a fan of your numbered priority, either. |
| 17:37.29 | [TK]D-Fender | I am :) |
| 17:37.43 | Penguin | Some of us can count, some can't. |
| 17:38.01 | Penguin | For those who can't, there is priority n. |
| 17:38.12 | [TK]D-Fender | If you fail at hard-numbered priorities then you don't deserve to admin a PBX :) |
| 17:38.26 | [TK]D-Fender | built-in IQ test |
| 17:38.45 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:39.07 | Penguin | I prefer n because it makes changes in lines of dialplan much easier. You never have to renumber if you don't use numbers in the first place. |
| 17:39.49 | *** part/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 17:39.58 | XandriX | Penguin: its just an example i pasted my numbering is a bit different in my actual config and i know the n trick thanks to you |
| 17:40.17 | ruben231 | [TK]D-Fender: can asterisk send faxs..? |
| 17:40.24 | Penguin | I also like same => n, |
| 17:40.43 | XandriX | :) |
| 17:41.16 | [TK]D-Fender | [12:40]ruben231[TK]D-Fender: can asterisk send faxs..? <- what is the point of origin? What is the trigger? |
| 17:42.22 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
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| 17:57.40 | XandriX | i need to better setup my firewall cant externaly connect to my sip |
| 17:58.49 | *** join/#asterisk fonewiz (~fonewiz@cpe-173-174-255-129.satx.res.rr.com) |
| 17:59.45 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 18:06.59 | fonewiz | Hello, long time user of Asterisk. Currently using 1.8.x. I have default=US set in my indications.conf but need to play Ringing in other country styles depending on the DID the call comes in on. Thanks in advance for any help. Seems like it should be straight forward but I am having a hard time find the info I need. I am currently playing US style ring indicator in my dial plan using "Ringing". |
| 18:16.32 | *** join/#asterisk human39 (~human39@c-71-236-110-155.hsd1.pa.comcast.net) |
| 18:17.56 | Penguin | fonewiz: Why are you providing the ringing sound for calls coming in? That's not normal. |
| 18:18.11 | Penguin | The call has to be answered before you can provide inband ringing sound. |
| 18:18.29 | human39 | hi all. I'm looking for tips on allowing remote call center staff. I assume that softphones would be the way to go. |
| 18:20.21 | fonewiz | Penguin: Normally I do an Answer then play a background sound telling the caller what to do next. This causes the call to be answered immediately and the calling party does here any ringing sound, just an immediate answer then my background sound. Some of our customers say they don't like that, for them we just do a "Ringing" then Wait(5) then move on to play the background sound and that makes them happy. |
| 18:20.40 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 18:21.24 | fonewiz | Penguin: Issue now is that we are taking on some customers outside the US and they are hearing the US style Ring indicator and that's obviously odd to them. So, trying to change this on a per call or per DID basis. |
| 18:21.55 | Max_E | human39, yes if you dont have Hardphones to use, yes softphones is the option |
| 18:22.05 | fonewiz | I think changing my Ringing to something like Playtones(something) with solve my issue but?? |
| 18:22.34 | fonewiz | Maybe Playtones(400+450/400,0/200,400+450/400,0/2000) to indicate ringing in the UK for example? |
| 18:23.03 | human39 | Max_E: I assume SIP is the only way to go. |
| 18:24.09 | [TK]D-Fender | fonewiz: Set(CHANNEL(tonezone)=XX) |
| 18:24.30 | fonewiz | Perfect |
| 18:24.31 | fonewiz | Thanks! |
| 18:24.38 | fonewiz | I will give that a try |
| 18:24.43 | fonewiz | That will work with 1.8x |
| 18:24.46 | fonewiz | ? |
| 18:25.47 | [TK]D-Fender | yes |
| 18:26.11 | fonewiz | The Playtones(400+450/400,0/200,400+450/400,0/2000) seems to work also but I like the solution you provided much better because I don't have to define each tone |
| 18:26.55 | Max_E | human39, or IAX2 |
| 18:27.42 | human39 | IAX2 works better with NAT.. right? |
| 18:27.54 | [TK]D-Fender | [13:18]human39hi all. I'm looking for tips on allowing remote call center staff. I assume that softphones would be the way to go. <- Softphones is the el-cheapo way to go |
| 18:30.31 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 18:31.02 | Katty | hello my asterisk does not work at all how to fix plz?? is urgent thx. |
| 18:32.12 | *** join/#asterisk [SySteM] (~antoine@ip-202.net-89-2-251.rev.numericable.fr) |
| 18:32.17 | [SySteM] | Hello anyone got an idea |
| 18:32.21 | [SySteM] | when a call is make |
| 18:32.32 | [TK]D-Fender | Katty: Mew. |
| 18:32.36 | Katty | hugs [TK]D-Fender |
| 18:32.46 | Katty | [TK]D-Fender: how're you dear? |
| 18:32.49 | [SySteM] | work 2or3 sec |
| 18:32.55 | [SySteM] | and after |
| 18:33.06 | [SySteM] | seems to have some buffring/codec problem i dont know |
| 18:33.12 | human39 | [TK]D-Fender: cheapo is okay, if it works. I can get phones/ATA's if needed. I don't have much experience with SIP through a NAT, I just heard that it can be a headache |
| 18:33.17 | [SySteM] | but only for caller, not for called |
| 18:33.21 | [TK]D-Fender | [SySteM]: Clarify the "I don't know". |
| 18:33.28 | [TK]D-Fender | [SySteM]: Idea : SHOW US the call |
| 18:33.34 | [TK]D-Fender | ~pb |
| 18:33.34 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:33.36 | [TK]D-Fender | ^^^^^ |
| 18:33.42 | [SySteM] | the sip log debug ? |
| 18:34.12 | [TK]D-Fender | [SySteM]: * CLI with "sip set debug on" |
| 18:34.19 | fonewiz | human39: SIP issues won't really be any different with Softphones vs ATA's or Hard Phones |
| 18:35.12 | [SySteM] | yes but how i can take only log |
| 18:35.17 | [SySteM] | about my two phones |
| 18:35.27 | [SySteM] | cause there is lots of thing which i can show |
| 18:35.39 | human39 | fonewiz: true.. would IAX2 be better in this situation? |
| 18:36.05 | fonewiz | human39: What is your situation? |
| 18:36.21 | fonewiz | human39: Personally, I stick with SIP but others like IAX |
| 18:36.55 | human39 | Managment wants senior sales (and some call center) employees to be able to work from home. |
| 18:36.56 | [SySteM] | i got log |
| 18:36.58 | fonewiz | human39: Is your Asterisk box behind a NAT with the end points out on the Interenet? |
| 18:37.01 | Max_E | [SySteM], https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:38.37 | human39 | it's on a private network.. but I'll have it on a public facing IP by the time we deplay our solution. |
| 18:39.26 | [SySteM] | http://pastebin.com/WUqXkZzM |
| 18:40.30 | fonewiz | human39: If your Asterisk box is NOT behind a NAT but your remote employees are, then you should be fine with SIP. This is the situation for me and ALL users for years.. Just about every end user is behind a NAT using soft phones or ATAs or Hard phones, all using SIP excluesively |
| 18:40.57 | fonewiz | human39: I think where you start to run into trouble is when you have the end points AND the Asterisk box all behind different NATS |
| 18:41.07 | [TK]D-Fender | incorrect |
| 18:41.20 | fonewiz | Um, who me? |
| 18:41.34 | [TK]D-Fender | yes |
| 18:41.41 | fonewiz | OK, been doing it for years |
| 18:41.51 | fonewiz | But I also do things to keep the NAT port open |
| 18:41.55 | [TK]D-Fender | having * behind a NAT typically changes nothing |
| 18:42.12 | fonewiz | OK, never had done that |
| 18:42.15 | [TK]D-Fender | You FORWARD SIP & RTP to *. That is all. |
| 18:42.31 | [TK]D-Fender | [SySteM]: that is not a COMPLETE call. |
| 18:42.37 | fonewiz | OK, cool so no worries for him either way.. |
| 18:42.46 | [SySteM] | do you want the invite ? |
| 18:42.48 | [TK]D-Fender | Nothing to assume so far |
| 18:43.02 | [TK]D-Fender | [13:42][TK]D-Fender[SySteM]: that is not a COMPLETE call. <------- |
| 18:44.19 | [SySteM] | is that can help http://pastebin.com/j96DiFkC |
| 18:44.27 | [SySteM] | cause its very difficult to keep just only this call |
| 18:44.39 | [TK]D-Fender | [SySteM]: that is useless. |
| 18:44.57 | [TK]D-Fender | [SySteM]: One last chance before I give up on this. |
| 18:45.14 | [SySteM] | Tell me what information do you search |
| 18:45.26 | Max_E | ups less karma points |
| 18:46.03 | Max_E | [SySteM], he is looking for the complete debug of the call fdrom initaila invite to the BYE(destroying dialog) |
| 18:46.10 | Max_E | all |
| 18:46.31 | *** join/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 18:46.45 | j-fish | Any Recommendations for Softphone app for pc ? |
| 18:47.52 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 18:48.04 | Max_E | Blink, Linphone, Jitsi, zoiper, xlite |
| 18:48.23 | j-fish | Max_E: your favorite ? |
| 18:48.38 | Max_E | for windows Blink, for Linux Linphone |
| 18:48.39 | j-fish | Xlite is a bit heavy ,slow |
| 18:48.50 | j-fish | Perfect thank you i will try it now |
| 18:49.21 | bitfury | I recently started using Acrobits softphone |
| 18:49.24 | bitfury | for ios and android |
| 18:49.26 | bitfury | works great |
| 18:49.54 | bitfury | their push notifications feature is nifty |
| 18:50.15 | j-fish | they have it for pc as well? |
| 18:50.43 | Max_E | [SySteM], run this: asterisk -rnvvvvvvd | teee sipdebug.txt, then enter in the cli and run sip set debug on, make the call when you hangup, wait 2-3seconds and exit the cli. In that path you will find a txt file called sipdebug.txt pastebin that and advide [TK]D-Fender about the called number or device used |
| 18:50.53 | [SySteM] | i log in a file |
| 18:50.56 | bitfury | j-fish: I don't think so |
| 18:50.59 | [SySteM] | cause i cant get all log in just my console |
| 18:51.04 | fonewiz | j-fish: Zoiper works well, it's cross platform and they have a free version |
| 18:51.48 | fonewiz | j-fish: I don't love the user interface but it's got a lot of options, cross platform and you can pay them to brand it etc.. |
| 18:53.19 | j-fish | fonewiz: Thanks i will try that as well. |
| 18:53.23 | fonewiz | I am looking for a soft phone that will be easy to answer and stay out of the way. I already have popups coming up with things the agent needs to click, the soft phone is just another thing they need to click to answer. Using all hard phones right now but this is a concern as I move forward |
| 18:54.28 | fonewiz | Guess I need something like a tray or task bar popup or growl notification that the agent can click on but isn't in the middle of the screen causing confusion between answering this call on the soft phone and reading the popup with call information etc.. |
| 18:55.53 | [SySteM] | i cant post on pastebin cause more 500 kilobytes |
| 18:56.59 | [SySteM] | during call there is lots of [2013-12-11 19:50:21] DEBUG[27204] audiohook.c: Write factory 0x95101b4 was pretty quick last time, waiting for them. |
| 18:58.04 | Max_E | ok disable the debug from cli, only stay with verbose |
| 18:58.47 | [SySteM] | http://privatepaste.com/5775a5f67e |
| 18:58.52 | [SySteM] | here |
| 18:58.53 | [SySteM] | :D |
| 19:00.42 | [SySteM] | call to 2005 |
| 19:02.00 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 19:04.32 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 19:07.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:08.01 | [SySteM] | :/ |
| 19:08.12 | *** part/#asterisk fonewiz (~fonewiz@cpe-173-174-255-129.satx.res.rr.com) |
| 19:09.12 | Penguin | Is there a free SIP soft phone with xmpp support? |
| 19:09.14 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 19:16.20 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 19:17.51 | XandriX | Penguin: i would not know but i have a question for ya a while ago on irc in this channel did u use the nick p3nguin aswell ? |
| 19:18.00 | Penguin | Yes. |
| 19:19.54 | Katty | chuckf: ping |
| 19:23.32 | Max_E | [SySteM], after trying to sanitize your log, i cant see something obvious, im not an sip expert so surely there are a lot of things that i could miss, but after the last ACK the log show a BYE, but no nat issues or retransmission. Maybe I mess some headers since I only get thelog based on the initial CALL-ID |
| 19:24.18 | Penguin | xandrix: Why do you ask? |
| 19:24.41 | XandriX | saw that you might have had a similar issue to me with pfsense a while ago |
| 19:26.29 | Katty | [TK]D-Fender: fender bender do you want to exchange xmas cards this year? |
| 19:26.58 | Max_E | [SySteM], and the call seesm to use alaw so not sure why you have codecs issues |
| 19:27.15 | Katty | drmessano: i'm going to go looking for a tardis one sometime this week or next ^_^ once the snow melts. |
| 19:27.19 | Katty | drmessano: it will be an interesting hunt |
| 19:29.29 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 19:33.42 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 19:33.52 | Kobaz | mm |
| 19:33.54 | Kobaz | so |
| 19:34.04 | Kobaz | anyone know how do you do https provisioning with polycom? |
| 19:34.12 | Kobaz | do you need to stick in the https:// to the server url? |
| 19:34.24 | Kobaz | we've got https selected, but the phone is connecting to port 80 |
| 19:34.32 | [SySteM] | Max_E, thanks |
| 19:34.36 | Katty | Kobaz: do you want an xmas card this year Kobaz? |
| 19:34.41 | Kobaz | sure |
| 19:34.46 | Katty | send me your address |
| 19:35.16 | Kobaz | are you going to come visit too? |
| 19:35.29 | Kobaz | we have excellent christmas parties |
| 19:37.00 | Katty | no probably not. |
| 19:37.02 | Katty | bit snow up there :< |
| 19:37.05 | Katty | snowy. |
| 19:37.34 | rrittgarn | Kobaz: Yes, and make sure you have your intermediary certs set up correctly too |
| 19:37.46 | rrittgarn | same thing works with Aastra and Cisco too iirc |
| 19:37.50 | Kobaz | does it like self signed? |
| 19:42.01 | Penguin | xandrix: Nah, I don't use pfsense. |
| 19:42.13 | Penguin | xandrix: Never have. |
| 19:42.28 | Katty | Penguin did you want an xmas card this year |
| 19:42.35 | Penguin | xandrix: I did think about it a long time ago, though. |
| 19:42.41 | Penguin | katty: Sure! |
| 19:42.47 | Katty | Penguin: send me your address |
| 19:44.12 | XandriX | Penguin: oh my bad the person you wer helping seemed to be using pfsense |
| 19:46.08 | Penguin | xandrix: That's very possible. pfSense seems to have some problems playing nicely with asterisk at times. |
| 19:48.42 | XandriX | yeah well wich is logical inside the network it works fine |
| 19:48.48 | XandriX | wan wise no comments :P |
| 19:53.40 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 20:01.34 | XandriX | yay i got the sip client to register externaly but now i have encounterd another problem hehe |
| 20:02.27 | XandriX | http://pastie.org/private/1nrxy4mmqlq960buascelg what does this mean ? |
| 20:02.38 | Penguin | xandrix: I think it is due to pf's implementation of NAT. I use pf and have set up NAT on it, and it works fine for things like forwarding traffic to a web server. |
| 20:05.38 | *** join/#asterisk mirela666 (~mirko.bra@95.180.35.162) |
| 20:06.02 | XandriX | Penguin: oh |
| 20:06.13 | XandriX | i thought my issue was do to something else in regards to that error message |
| 20:07.41 | XandriX | Penguin: is that what the error i pasted in the pastie link regarding to nat or something i didint setup ? |
| 20:13.26 | XandriX | it might still be my firewall seems to mean my asterisk box cant send out a packet |
| 20:13.39 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 20:15.26 | XandriX | hmm now it seems to be working but i get no sound |
| 20:19.59 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:24.51 | XandriX | im getting closer |
| 20:25.09 | XandriX | now calls pop in and my errors for outbound packets stopped but if i call a local extension it "works" but i hear nothing |
| 20:25.12 | pigpen | so, anybody using realtime? I an extension in realtime (psql) that pages a list of about 43 sip phones. Seems it limits target (list of extensions) to about 120 char. |
| 20:25.32 | pigpen | If I put the list into static file, it works fine. |
| 20:25.51 | pigpen | anybody know how to lengthen the "window" of realtime? |
| 20:26.10 | Penguin | xandrix: I didn't look yet. I was just saying I think BSD's implementation of NAT is why asterisk has trouble sometimes. |
| 20:27.10 | XandriX | oh yeah |
| 20:27.16 | XandriX | in my case i forgot to add the outbound for it |
| 20:27.28 | XandriX | but now im wondering if i need a stun service |
| 20:28.59 | Penguin | Did you configure your asterisk to work in your NAT conditions? |
| 20:29.12 | XandriX | err i set this extension to nat=yes |
| 20:29.56 | Penguin | Wrong. |
| 20:30.03 | Penguin | Extensions don't have NAT settings. |
| 20:30.19 | XandriX | i meant sip.conf |
| 20:30.31 | XandriX | the "extension / sip peer |
| 20:30.40 | Penguin | Extensions aren't found in sip.conf. |
| 20:31.21 | XandriX | ok |
| 20:31.32 | XandriX | sorry for bein an idiot it just how i saw it |
| 20:31.53 | XandriX | my setup is odd and my peer number is the same as its related extension number |
| 20:32.03 | Penguin | That's not ideal. |
| 20:32.30 | XandriX | that will be added to my notes on cleaning up my configs once im done experimenting and getting the features i want to work to work :P |
| 20:33.35 | Penguin | If you're going to fix it, you should fix it immediately. It just precipitates the notion that the device is the same as the extension. |
| 20:34.23 | Max_E | why people still replying on the Windows CDR toipic on the mailist now im going to mark that as spam |
| 20:34.25 | Penguin | When you follow a good naming convention, there is a clear disassociation between the two. |
| 20:35.32 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 20:35.44 | XandriX | duely noted |
| 20:35.50 | XandriX | i shall change this |
| 20:35.58 | XandriX | wont be long only have 2 peers |
| 20:36.05 | Penguin | Where is your asterisk located? |
| 20:36.17 | XandriX | at home actualy on the pfsense box |
| 20:36.49 | XandriX | or did you mean geographicaly speaking ? |
| 20:37.09 | atan | I'm having trouble with a Cisco phone leaving lines "ringing" or just leaving lines on the screen once the call has ended. Not even sure where to start on this one but I suspect it has something to do with the phone config. Anyone here use the 99xx phones? |
| 20:37.20 | Penguin | Nah, I meant the relation between the firewall/NAT and the asterisk box. |
| 20:38.15 | atan | What blows my mind is if I watch the sip traffic I see SIP/2.0 487 Request Cancelled & ACK |
| 20:39.07 | XandriX | Penguin: aaaaah |
| 20:39.15 | Penguin | xandrix: Will you have phones on the LAN as well as remote locations? |
| 20:39.22 | XandriX | yes |
| 20:39.34 | pigpen | so, regards to my realtime issue, i found pbx/pbx_realtime.c, line 60: #define EXT_DATA_SIZE 256 |
| 20:39.36 | XandriX | but only 1 remote location |
| 20:39.57 | XandriX | for now being my cell phone that unfortunatly has a dynamic adress |
| 20:39.58 | pigpen | I wonder if this is what was delimiting my command. |
| 20:39.59 | Penguin | xandrix: You'll have to configure it the same if you have one or 100. |
| 20:40.09 | XandriX | i figured as much |
| 20:40.15 | XandriX | shuts up and reads |
| 20:41.28 | XandriX | Penguin: i take it there are a fiew things i need to configure for this to work properly |
| 20:41.29 | Penguin | xandrix: I don't see the entire picture, but I'm fairly sure the problem will be caused by pfsense. That is if you have properly configured each of the settings for asterisk to work behind NAT. |
| 20:42.03 | XandriX | my sip phone inside my lan can recieve and make calls with audio fine my cisco 7940 on lan can recieve and send calls find with audio |
| 20:42.06 | Penguin | ~sipnat |
| 20:42.06 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 20:42.19 | XandriX | my remote cell phone cannot get audio but it can send and recieve calls |
| 20:42.31 | Penguin | For asterisk to work behind NAT, the typical settings in the [general] section are nat=yes, directmedia=no, externhost or externaddr, and your localnet. |
| 20:42.48 | Penguin | No audio? That's a NAT problem. |
| 20:43.07 | XandriX | Penguin: can externhost be externhos=noipdomain.no-ip.org ? |
| 20:43.21 | Penguin | Sure. |
| 20:44.00 | Penguin | externhost is usually used with a dynamic IP address because you use a dynamic DNS name. If you had a static address, you would just use externaddr. |
| 20:45.41 | XandriX | aaah |
| 20:45.59 | Penguin | Actually, I didn't state that well... |
| 20:46.25 | XandriX | hmmm i hear a faint noise |
| 20:46.41 | Penguin | externhost is typically set to your a dynamic DNS name because you have a dynamic IP address. |
| 20:46.57 | Penguin | Still not much better, but you get the idea. |
| 20:47.35 | Penguin | If you have a static address and a DNS name with the appropriate record, you could still use externhost... but that isn't the usual use of it. |
| 20:48.28 | XandriX | so i added externhost and nat=yes and directmedia=no to my general settingand well now even if i am peerd after reloading sip i cant call my sip number anymore it just doesnt ring hehe |
| 20:48.33 | XandriX | thinks he broke something |
| 20:49.33 | Penguin | To figure out why, sip set debug on. |
| 20:50.27 | XandriX | i did figure out why my connection at work dropped everyones phone dc'd so outgoing calls werent working from here for a second lol |
| 20:50.38 | XandriX | still no sound with those settings tho :S |
| 20:50.59 | Penguin | Did you open the RTP port range on the firewall? |
| 20:51.34 | XandriX | si |
| 20:51.44 | XandriX | 10000 to 32000 |
| 20:51.58 | XandriX | in udp |
| 20:52.10 | XandriX | im thinking my sip client on my cellphone is set to the wrong codec |
| 20:52.34 | Penguin | The sip debug will show that. Waiting. |
| 20:54.29 | XandriX | sorry the sip debug will say that my phone is using a weird codec or ? |
| 20:54.45 | *** join/#asterisk jpoz (~jpoz@c-50-139-46-49.hsd1.or.comcast.net) |
| 20:55.52 | Penguin | Less talking, more debugging and pasting. |
| 20:59.06 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 20:59.16 | *** join/#asterisk jsjc (~Adium@154.Red-83-32-91.dynamicIP.rima-tde.net) |
| 20:59.29 | XandriX | Penguin: hold on about a billion lines pass by when i do that and is proving slightly tricky to paste |
| 21:02.46 | Penguin | I need to see the entire call. It will be a couple hundred lines. |
| 21:03.23 | XandriX | oh |
| 21:03.28 | XandriX | oh crap haha sorry about that |
| 21:04.08 | XandriX | so im guessing i should asterisk -r >> logfile.txt and make the call ? |
| 21:04.16 | Adylas | Hello, I m having some hard time with a SIP channel with a carrier. After 30 minutes, a call would get disconnected. This seems to be because our asterisk is setup with a timer of 1800 sec and does not "requiere" "timer" in the SIP neg. Any idea ? |
| 21:06.40 | Penguin | xandrix: I connect to the asterisk CLI as usual, sip set debug on, make the call, sip set debug off, scroll up, copy to the bottom, paste in the pastebin. |
| 21:12.07 | Penguin | That paste is truncated. |
| 21:12.13 | Penguin | Stuff to the right side is missing. |
| 21:12.24 | Penguin | wait |
| 21:12.32 | Penguin | The pastebin is just junk. |
| 21:12.47 | Penguin | It's there, but they expect me to scroll sideways. |
| 21:12.51 | Penguin | Try pastebin.com next time. |
| 21:12.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.88) |
| 21:13.48 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 21:18.18 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 21:19.37 | XandriX | Penguin: i appolagize for that :S |
| 21:22.14 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 21:22.29 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 21:24.00 | pabelanger | Adylas, disable session timers |
| 21:24.03 | pabelanger | sip.conf |
| 21:25.39 | XandriX | Penguin: could i perhaps still have a nat issue ? |
| 21:53.47 | *** part/#asterisk unstable (unstable@tor/regular/sid) |
| 22:06.13 | *** join/#asterisk u0m3_ (~u0m3@92.80.102.164) |
| 22:11.15 | Penguin | xandrix: I wasn't able to follow the call in your paste. But if you have an audio problem, I'd blame nat. |
| 22:13.15 | Penguin | xandrix: What calls are having audio problems? All calls from all phones to all extensions? Only calls from the ITSP to asterisk? Only calls from your SIP phone to asterisk? I need to understand what the problem is. |
| 22:15.01 | phix | Penguin: Mornin' |
| 22:15.19 | Penguin | You live in the future. |
| 22:15.27 | Penguin | 1615 here. |
| 22:16.13 | Kobaz | okay so |
| 22:16.18 | Kobaz | still having problems provisioning polycom with https |
| 22:16.21 | Kobaz | anyone have that going? |
| 22:20.29 | Kobaz | i have https://... as my server url |
| 22:20.30 | Kobaz | and server type is set to http(s) |
| 22:20.30 | Kobaz | but it keeps connecting on regular hp, and it's ignoring the login/pass |
| 22:39.15 | *** join/#asterisk Deeewayne (~dwayne@173-164-56-74-colorado.hfc.comcastbusiness.net) |
| 22:39.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:40.14 | Kobaz | okay so |
| 22:40.24 | Kobaz | even telling the polycom to do https://foobar:443 |
| 22:40.32 | Kobaz | it's still using plain http, but on the https port |
| 22:40.34 | Kobaz | SDJKHSKLDHJLSKDF |
| 22:40.58 | phix | hmmm, ok s,1,NoOp(some message) Is still being executed when s/100,1,.. matches and is executed |
| 22:41.12 | phix | does this mean s,2, will be called too? |
| 22:41.36 | phix | or is s/100,n,.. now overiding it? |
| 22:41.54 | phix | Kobaz: That's some mighty fine keyboard smashing there |
| 22:42.31 | Kobaz | thanks |
| 22:45.16 | Penguin | phix: Did you test the dialplan I gave you? What I gave you worked 100%, so if it didn't work for you, you have a bug or you copied and pasted it incorrectly. |
| 22:50.08 | Kobaz | so no one does https provisioning? |
| 22:52.41 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 22:52.56 | phix | Penguin: I have been reading up on http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 22:53.40 | Penguin | You really should have tested what I provided for you. |
| 22:55.12 | phix | Penguin: It answers my question, unless things have changed since that was written. Basically s,1,... is always called, you can then specify the next priority s/100,n,... which overrides s,n,.. (priority 2) but as soon as a priority is reached that there isn't a s/100 for it continues onto the next priority (without the /100) |
| 22:55.17 | phix | kind of not what I want |
| 22:55.39 | phix | as I want to put in a fallback which I dont want to run if the callerid matchd |
| 22:57.30 | Penguin | If you have priorities 1, 2, and 3, and only priority 2 has a callerid match... priority 3 will still execute when the callerid matched at 2. The only thing the callerid match does is prevent that particular line from executing when the callerid does NOT match. |
| 22:58.31 | Penguin | If I understood EXACTLY what you want (your end goal), I could probably devise a solution. |
| 22:59.27 | phix | ok, well what I want to do is create a dialout macro, I want certain SIP phones to use particular SIP trunks and set Callerid if required by the external SIP provider. |
| 22:59.59 | phix | I have 5 SIP trunks, the 5th one has 10 numbers associated with it which I need to set the callerid(number) to use |
| 23:00.34 | Penguin | Okay, that can be done with context and setvar. |
| 23:01.52 | phix | does that mean my dialout macro will contain alot of gotoif's? |
| 23:02.29 | Penguin | That's one possibility. |
| 23:02.43 | phix | what is the cleanest / easiest to read and follow way to do it :) |
| 23:02.56 | phix | which doesn't involve writting out an essay? |
| 23:03.39 | Penguin | I still don't fully understand all angles. Can you lay out a few examples? |
| 23:04.15 | *** join/#asterisk apb1963 (~quassel@174.134.232.228) |
| 23:05.33 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 23:05.56 | Penguin | So far you've only painted the frame. I need to see the whole picture. |
| 23:08.28 | phix | ok, I have an office what contains 12 SIP phones, I have 5 SIP accounts (from a VoIP provider) which has 14 phone numbers (that are accessible via landline). I want to setup an external phone number to each phone when they dial and external / landline extension. |
| 23:09.08 | phix | s/w/t/ |
| 23:11.43 | phix | so in other words the SIP phone uses the correct SIP trunk that the external phone number is associated with |
| 23:11.43 | [sr] | hi WIMPy |
| 23:11.43 | Penguin | You want to "assign" callerid numbers to the phones. Right? |
| 23:11.43 | phix | Penguin: yes, except for the 4 special cases which have their own SIP account with the external provider |
| 23:11.45 | phix | so 4 of the phones have their own account, the rest share the same trunk but need their callerid to be set |
| 23:12.14 | phix | the SIP phones all have SIP accounts on the asterisk server which sets their caller id, which is what I filter on |
| 23:13.33 | phix | atm everything is working, but I need to include a fallback, that is, if a phone hasn't been assigned a special phone number it will use the generic number that most calls come in as (and which rings all phones / goes to the queue) |
| 23:14.18 | phix | and to make things even better, the last SIP trunk has 10 phone numbers allocated to it but can only have 4 concurrent channels :) |
| 23:14.52 | phix | (which can be changed at a later date, but atm 4 concurrent channels is enough) |
| 23:15.32 | *** join/#asterisk JustinAiken (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 23:15.45 | JustinAiken | hello, |
| 23:15.54 | JustinAiken | i'm trying to do a bridge over AMI as described here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Bridge |
| 23:16.09 | *** join/#asterisk apb1963 (~quassel@174.134.232.228) |
| 23:16.12 | JustinAiken | but getting back "No such command 'Bridge'" on Asterisk 11 |
| 23:16.23 | phix | Penguin: DOes that explain it all yet? |
| 23:16.37 | JustinAiken | I can't find much other information on bridges via AMI - anyone have a good resource to look at? |
| 23:18.01 | mjordan | JustinAiken: there's not much to say about that AMI command. It takes two channels and puts them in a new bridge. That's about it. |
| 23:18.15 | mjordan | The action is in Asterisk 11. It's provided by features.c. |
| 23:18.21 | mjordan | So it's actually in the core of Asterisk. |
| 23:18.40 | mjordan | If it isn't available, that means it didn't register - which probably means you have an error in your features loading. Do you have a valid features.conf? |
| 23:19.11 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 23:24.54 | XandriX | Penguin: all local phones by that i mean phones hooked up to the same lan as the asterisk box work perfectly everything works my external sip phone however (the phone i use when im on the road) does not get audio it recieves and can make calls fine tho and sorry for the late response just got back home from work |
| 23:26.58 | JustinAiken | Matt, my features seem to be loaded okay - the dialplan Bridge() command works, okay, and 'core show application Bridge' gives me useful instructions - just doing the AMI command Bridge gives me no such command |
| 23:28.17 | JustinAiken | https://gist.github.com/d8b0220a52d7ae383e55 |
| 23:32.03 | outtolunc | JustinAiken: its action: bridge .... |
| 23:32.29 | outtolunc | not, action: command command: bridge ... |
| 23:37.17 | JustinAiken | lol |
| 23:37.21 | JustinAiken | thanks outtolunc |
| 23:37.33 | JustinAiken | now bridgin' away |
| 23:37.50 | Penguin | phix: A macro isn't required to do what you're trying to do. SetVar in each peer in sip.conf with an "external CallerID" to use. Here's an example of how I do it: in the peer entry, SetVar=externalCID=3149691077; in the dialplan, Set(CALLERID(num)=${IF($[${EXISTS(${externalCID})}]?${externalCID}:3056512345)}) |
| 23:38.28 | Penguin | phix: This says if the externalCID variable is set for the peer, set the CALLERID(num) to the value of it. If it is not set, set the CALLERID(num) to 3056512345 instead. |
| 23:38.46 | newtonr | JustinAiken, you can do "manager show command Bridge" |
| 23:38.48 | Penguin | phix: This allows for each phone to have its own external caller id number, and a fallback of a default number. |
| 23:39.02 | Penguin | phix: One single line of dialplan, not a macro. |
| 23:40.57 | Penguin | xandrix: Not even one-way audio? |
| 23:41.00 | phix | Penguin: And how do you handle the special cases? |
| 23:41.29 | phix | Penguin: do I then create a dialout macro and filter on the callerid? |
| 23:41.36 | phix | so it uses the correct SIP trunk |
| 23:41.39 | XandriX | Penguin: not even one way audio :( |
| 23:41.59 | Penguin | phix: What special cases? So far I haven't seen any special cases or requirements. |
| 23:42.00 | XandriX | i am currently cleaning up my configs |
| 23:42.51 | Penguin | phix: You can set another variable if you want to do per-device peer dialing. Or you can use another context for those other phones. |
| 23:43.08 | Penguin | There's no need for a macro for anything you've mentioned. |
| 23:44.31 | phix | no need for a macro? well how do I dial out then? |
| 23:44.41 | Penguin | Are you kidding me? Dial() |
| 23:44.56 | phix | yes but I need to filter out the internal and external nmbers |
| 23:44.59 | phix | numbers* |
| 23:45.05 | outtolunc | press 0 and ask the operator to connect you? ;) |
| 23:45.16 | Max_E | +100^ |
| 23:45.34 | Penguin | That's what extensions already do. Unless your internal phones have the same 10-digit phone numbers as PSTN phone numbers, you're trying to complicate things. |
| 23:45.44 | Penguin | There is no "filtering." |
| 23:45.48 | Penguin | It's just pure dial plan. |
| 23:46.11 | Penguin | If I had 10 phones that all needed to dial out through one particular peer, I'd probably assign a special outbound context with extensions containing Dial()s using that particular peer. |
| 23:46.26 | phix | Penguin: yes but I have multiple lines in my external dial plan, so doesn't it mae sense to use a macro so I dont have to repeat stuff to figure out which line to give to dial? |
| 23:46.28 | Penguin | That's exactly what contexts are for. |
| 23:46.53 | Penguin | Macro? No. |
| 23:47.18 | phix | Penguin: some numbers I want to block, other I want to play a pre-recorded message before dialing, then after that I need to decide which line to use based on the phone making the call |
| 23:47.22 | Penguin | If a subroutine /is/ needed, GoSub() will help you. But so far, I haven't seen a need for a subroutine. |
| 23:47.40 | mjordan | Penguin: +10 for GoSubWhile we're |
| 23:47.42 | mjordan | er |
| 23:47.52 | mjordan | Ignore the While we're part of that. Just GoSub. |
| 23:47.54 | mjordan | Silly Enter key. |
| 23:48.14 | Penguin | I'm not understanding the fascination with macro. |
| 23:48.33 | outtolunc | doesn't use macros either |
| 23:49.15 | Penguin | I only have a couple macros hanging off of Dial(), but that's because I cannot make the gosub from a Dial() work. |
| 23:49.16 | phix | Penguin: macros allow arguments like a function? |
| 23:49.44 | phix | contexts dont? |
| 23:49.48 | Penguin | Who cares? You haven't said a single thing that requires a macro. |
| 23:50.07 | phix | Penguin: figuring out what to give Dial() |
| 23:50.21 | Penguin | If you insist on using all these worthless macros, I wish you good luck. |
| 23:50.28 | phix | Penguin: plus I may or may not need to setup a callerid |
| 23:50.39 | phix | Penguin: well how else would I achieve that? |
| 23:50.48 | phix | without repeating myself |
| 23:50.53 | Penguin | If you NEED a subroutine, use GoSub(). But so far, you haven't shown me anything that requires a subroutine. |
| 23:51.06 | *** part/#asterisk mjordan (~matt@nat/digium/x-pxhqmhhbuwcvpjmg) |
| 23:51.20 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 23:51.31 | phix | Penguin: I need to set the callerid or may not need to, I then need to tell dial() what trunk to use for external calls depending on the callerid |
| 23:51.46 | Penguin | What you've told me your requirements are can all be achieved inline. |
| 23:52.34 | outtolunc | ${IF(${ISNULL(... is your friend ;) |
| 23:52.43 | Penguin | ISNULL and EXISTS |
| 23:53.28 | Penguin | Depending on which of those you use determines which side of the : you put the target. |
| 23:54.15 | Penguin | But seriously, if you can accomplish exactly what you're after using inline dialplan, why would you complicate it using a subroutine? |