| 00:00.26 | rubio | Hi, i'm having audio issues using asterisk form webRTC, crazy (or not so crazy) thing is that if i plug my client directly to internet (without nat) i have normal audio and rtp debug said "via ICE", but when client is behind a router rtp does not said "via ice" and there is no audio at all. Asterisk has a public ip. Any clues??? (asterisk 11.13.1 on debian 8, clients is SIP.JS on Chrome) |
| 00:16.02 | *** join/#asterisk shudon (~hdon@wsip-24-234-30-238.lv.lv.cox.net) |
| 00:17.17 | shudon | hi all :) we're running freepbx and using isymphony. sorry if this is OT, but does isymphony connect to asterisk or does it get there through freepbx? any idea how i can look up my own authentication credentials? the only authentication credentials i'm sure of are in /etc/asterisk/manager.conf for the admin user but that's for the asterisk manager interface and it's configured to bind to loopback interface only, so that can't be it |
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| 01:10.12 | lvlinux | Phil-Work: ok I uploaded a simplified features.conf and extensions.conf for you to check out http://ruel.io/nway You should be able to tell what's going on by looking at the configs and adapt it to what you need. |
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| 03:44.41 | K0HAX | Question: My SIP provider requires insecure=port,invite and I'm trying to switch to pjsip, when I use type=identify I don't send a SIP Contact: field when I send SIP Trying. |
| 03:44.54 | K0HAX | How do I get insecure=port,invite to work in pjsip? |
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| 04:34.43 | santiago_ITnoob | God night |
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| 09:20.27 | z30 | Hi guys, any one can help me with fax in asterisk? |
| 09:20.55 | WIMPy | Not unless you have a question. |
| 09:21.38 | z30 | i need to change codec for fax |
| 09:21.57 | z30 | my hardware not supported t38 so i must use G711 ulaw |
| 09:22.23 | z30 | and my call is in g729 codec |
| 09:22.56 | z30 | problem is : when i call fax machine and Answer() it with asterisk for fax detection |
| 09:23.19 | WIMPy | Oh, that one. |
| 09:23.42 | z30 | how can i reinvite call with g711 codec when i established call with g729 |
| 09:24.38 | z30 | sorry for too much explanation |
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| 10:32.44 | stevenm | Hey our company uses a crazy hosted PBX which only allows certain vendors of phones to register to their platform. Could I use an extremely simple/basic/stripped-back installation of Asterisk to register to them (with an altered User-Agent so it'll connect) and have my phone connect to Asterisk and have it bridge the two transparently? and is Asterisk the best tool for the job? |
| 10:34.42 | Tim_Toady | you can by setting the parameter useragent in sip.conf |
| 10:35.13 | Tim_Toady | it is a bit of an overkill, maybe a sip proxy would bve more suitable but its not trivial to setup one either |
| 10:35.27 | Tim_Toady | s/bve/be/ |
| 10:35.33 | stevenm | Tim_Toady, do you have a sip proxy in mind? |
| 10:35.57 | Tim_Toady | kamailio, opensips etc |
| 10:36.29 | stevenm | i thought they were more like asterisk |
| 10:36.31 | Tim_Toady | but as I said their setup is far from trivial, require deep knowledge of the sip protocol |
| 10:36.43 | stevenm | ah ok so asterisk is likely simpler? |
| 10:36.50 | Tim_Toady | yes probably |
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| 10:37.25 | stevenm | ok I'm more familiar with asterisk anyway |
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| 13:34.32 | carrar | OK |
| 13:34.48 | carrar | Tokyo Asterisk Users Group meeting completed |
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| 13:40.38 | paraxor | is there an asterisk user's group near DC? |
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| 14:04.13 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:29.44 | s8y | hello, is there anyone who could help with pjsip (I know there are very few people who had a go at it)? |
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| 14:32.32 | s8y | I am slowly migrating from 1.8 to 13.5 and struggle with pjsip. how do I declare device (e.g. phone-2 in pjsip)? |
| 14:33.03 | newtonr | That is a big move. Let me point you at some examples |
| 14:34.13 | newtonr | Here is the pjsip documentation in general |
| 14:34.14 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 14:34.30 | newtonr | this page has a side by side example of some sip.conf and pjsip.conf configuration |
| 14:34.30 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip |
| 14:35.04 | newtonr | Here are some further configuration examples : https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 14:35.52 | newtonr | s8y, that should get you started! |
| 14:38.03 | newtonr | Oh, in addition the pjsip.conf has examples of device configuration.. https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample |
| 14:38.21 | newtonr | and if you want to see an example in the context of a basic PBX type setup: https://github.com/asterisk/asterisk/tree/master/configs/basic-pbx |
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| 14:38.45 | newtonr | There! More documentation than you'll ever want! :) |
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| 14:43.56 | s8y | newtonr: thanks I already had a go at wiki and sample files. is it just extensions.conf and phsip.conf that I need to configure for simple in pbx calls or is there more to configure? |
| 14:45.53 | newtonr | s8y, that highly depends on what you want to do of course. Yes you could provide a simple setup with only configuring extensions.conf and pjsip.conf |
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| 14:46.21 | newtonr | obviously if you want to use voicemail then you'll need to configure voicemail.conf or if you want to use features provided by the core then you'll want to configure features.conf and so on |
| 14:46.40 | s8y | newtonr: thanks, that's what I did but must be missing something since it doesn't work :-( |
| 14:48.05 | s8y | newtonr: I wanted to try simple: register 2 phones to same pbx and make them to call one another within same pbx (no trunk required) and it doesn't |
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| 14:49.01 | newtonr | s8y, probably want to try this first: https://wiki.asterisk.org/wiki/display/AST/Hello+World |
| 14:49.08 | newtonr | using the pjsip.conf configuration |
| 14:49.54 | newtonr | Then it is a matter of adding a second device configuration (endpoint, aor, auth, etc) and a Dial call to your extensions.conf |
| 14:51.19 | s8y | newtonr: I managed to get play(hello-world) working :-) |
| 14:53.08 | s8y | newtonr: can you show me example of dial(pjsip/) please? |
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| 14:56.44 | newtonr | s8y, If you looked at the documentation I provided to you , you'll find this https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 14:57.03 | newtonr | I do recommend going through the documentation I've linked so far. :) |
| 14:58.33 | s8y | newtonr: where do I declare ${EXTEN} variable? |
| 14:58.35 | igcewieling | Documentation, like marriage, is overrated. |
| 14:59.16 | [TK]D-Fender | s8y: You don't. ${EXTEN} is where you ARE |
| 15:01.16 | newtonr | igcewieling, if your spouse came with documentation things might go easier. That is assuming you read the documentation. |
| 15:03.05 | *** join/#asterisk nix8n82 (~AndChat39@2600:100e:b026:77e1:a58f:3802:7787:63eb) |
| 15:03.43 | igcewieling | resists an "insert tab a into slot b" joke. |
| 15:04.34 | s8y | [TK]D-Fender: so what does following entry means? exten => _6XXX,1,Dial(PJSIP/${EXTEN}) (call where you are?) |
| 15:05.11 | newtonr | s8y, I thought you said you were moving from 1.8 to 13? If you are completely new to Asterisk then there is a lot more documentation you'll want to read through :D |
| 15:05.38 | newtonr | If you want someone to take you step by step through configuring Asterisk it is going to take a long time. |
| 15:06.00 | [TK]D-Fender | s8y: You are in a PATTERN for that exten |
| 15:06.08 | [TK]D-Fender | that is the NUMBER you dialed that you're execiting there |
| 15:06.21 | [TK]D-Fender | <PROTECTED> |
| 15:06.44 | [TK]D-Fender | EXTEN |
| 15:06.44 | [TK]D-Fender | So whatever number you dialed that got you executing that line... that is ${EXTEN} |
| 15:07.19 | igcewieling | *sigh* eternal september in #asterisk |
| 15:13.56 | s8y | newtonr: I have been using asterisk since 2010 |
| 15:14.51 | s8y | but its not my primary job |
| 15:15.12 | [TK]D-Fender | very curious your not knowing what $EXTEN} is after that kind of time. |
| 15:17.08 | *** join/#asterisk tuxmartin (~martin@89.190.50.140) |
| 15:17.24 | stephanj | newtonr: so you recommend the asterisk book as a good start? |
| 15:18.36 | s8y | I know what exten => is |
| 15:19.47 | newtonr | stephanj, Yeah, the 4th edition. Go through that and reference the wiki. Though you could read straight through the wiki and get a lot out of that too. The book doesn't cover the new stuff like pjsip. |
| 15:19.49 | tuxmartin | Hi, I need to install Asterisk (for testing) that support video calls. I have latest Debian testing and compiled latest Asterisk 13.6.0. In config files I have enabled video support, but in clients video dosen't work. Audio calls works good. On Android I'm using https://play.google.com/store/apps/details?id=com.antisip.vbyantisip and https://play.google.com/store/apps/details?id=org.sipdroid.sipua on pc I am using http://www.microsip.org/ on Windows 7 |
| 15:19.49 | tuxmartin | and Ekiga on Ubuntu 15.10 linux. There are my configs: http://pastebin.com/kjTXD1PJ What I do bad? |
| 15:20.12 | [TK]D-Fender | ${EXTEN} <-- the most important basic variable |
| 15:20.46 | [TK]D-Fender | tuxmartin: Show us your configs |
| 15:21.34 | tuxmartin | [TK]D-Fender, my change in config files are on pastebin http://pastebin.com/kjTXD1PJ do you need all files from /etc/asterisk ? |
| 15:21.58 | [TK]D-Fender | tuxmartin: now a call.. |
| 15:22.53 | tuxmartin | [TK]D-Fender, do you mean asterisk log if I make a call? |
| 15:23.10 | [TK]D-Fender | * CLI w/ SIP DEBUTG |
| 15:23.32 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 15:26.39 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
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| 15:31.15 | tuxmartin | [TK]D-Fender, how can I redirect log output to file if I am in voidp*CLI> ? |
| 15:31.53 | tuxmartin | [TK]D-Fender, I have allow ports 5060 and 10000-11000 tcp and udp. |
| 15:32.02 | [TK]D-Fender | just copy/paste |
| 15:32.05 | *** join/#asterisk DavidHuebner (~hubie@ip-69-178-218-71.far.ideaone.net) |
| 15:32.53 | tuxmartin | [TK]D-Fender, there are a lot of text, I can't get begin of log output. |
| 15:33.05 | [TK]D-Fender | Yes, you can |
| 15:33.33 | tuxmartin | [TK]D-Fender, terminal in linux Mint don't allow me to go too back. |
| 15:34.05 | [TK]D-Fender | get putty or some better client |
| 15:38.16 | igcewieling | tuxmartin: you keep asking questions even total n00bs usually figure out before asking for help. |
| 15:38.34 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
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| 15:40.11 | tuxmartin | [TK]D-Fender, thank you, putty works. There is complete debug: http://pastebin.com/ZSYZMFH1 |
| 15:40.28 | tuxmartin | [TK]D-Fender, I'm trying to make videocall from user 100 to user 101 |
| 15:41.09 | [TK]D-Fender | Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw|alaw|speex|speex16|ilbc|opus)/video=(h263|h263p|mpeg4|h264|vp8)/text=(nothing), combined - (ulaw|h263) |
| 15:41.44 | [TK]D-Fender | answering end = Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) |
| 15:41.49 | [TK]D-Fender | which didn't offer video |
| 15:42.08 | [TK]D-Fender | Go look at your callee |
| 15:42.15 | [TK]D-Fender | they didn't offer it |
| 15:45.32 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-nkxkvclvoyrfikqt) |
| 15:45.45 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-mvfhxqemovtyhjry) |
| 15:45.46 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:58.09 | tuxmartin | [TK]D-Fender, Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(h263)/text=(nothing), combined - (ulaw|h263) |
| 15:58.18 | tuxmartin | [TK]D-Fender, Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) |
| 15:58.44 | tuxmartin | [TK]D-Fender, both side support h263 video codec and ulaw audio codec |
| 16:00.36 | igcewieling | tuxmartin: keep in mind Asterisk cannot transcode video |
| 16:01.20 | [TK]D-Fender | The answering side didn't accept the video |
| 16:01.35 | [TK]D-Fender | line 454+ |
| 16:01.53 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net) |
| 16:01.57 | [TK]D-Fender | Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) |
| 16:02.03 | [TK]D-Fender | Peer doesn't provide video |
| 16:02.05 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 16:04.19 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 16:05.15 | tuxmartin | [TK]D-Fender, I modify DTFM setting (I read in in manual), but it still do not work. Please look at to my new debug output: http://pastebin.com/zXYZLnWf |
| 16:06.36 | [TK]D-Fender | Retransmitting #2 (NAT) to 89.190.50.140:39059: |
| 16:06.41 | [TK]D-Fender | Contact: <sip:101@192.168.1.7:5060> |
| 16:06.47 | [TK]D-Fender | Your NAT settigns are also all screwed up |
| 16:06.56 | [TK]D-Fender | you're missing the basics for * to work from behind NAT |
| 16:07.12 | igcewieling | [TK]D-Fender: you never get tired of setting up systems for people? |
| 16:07.24 | [TK]D-Fender | daily |
| 16:07.32 | [TK]D-Fender | And then another day comes and I get tired again |
| 16:07.38 | igcewieling | then stop. Life is a lot better. |
| 16:07.55 | igcewieling | I stopped years ago. |
| 16:10.30 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 16:11.31 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 16:13.15 | tuxmartin | [TK]D-Fender, please look at my next debug output: http://pastebin.com/qHQXTTPA |
| 16:13.54 | *** join/#asterisk ldiamond (~ldiamond@unaffiliated/ldiamond) |
| 16:13.57 | [TK]D-Fender | Reliably Transmitting (NAT) to 89.190.50.140:5060: |
| 16:14.02 | [TK]D-Fender | Contact: <sip:100@192.168.1.7:5060> |
| 16:14.19 | [TK]D-Fender | Still a fail |
| 16:14.25 | tuxmartin | [TK]D-Fender, how can I disable internal address? |
| 16:14.31 | ldiamond | I want to use a laptop with a phone jack (modem) to plug a normal phone to make calls through a sip provider. Any idea how I can do that? |
| 16:14.32 | [TK]D-Fender | set you localnets properly |
| 16:14.36 | [TK]D-Fender | set your wan IP properly |
| 16:14.41 | [TK]D-Fender | prevent reinvites. |
| 16:14.49 | [TK]D-Fender | all the usual things required since... pretty much forever |
| 16:26.29 | igcewieling | [TK]D-Fender: since at least 2001 perhaps earlier. |
| 16:31.01 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 16:37.08 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 16:43.15 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:57.32 | ldiamond | If I understand correctly, to plus a phone into my computer's modem, it has to support FXO. How do I check that? |
| 16:57.49 | [TK]D-Fender | It doesn't |
| 16:58.02 | igcewieling | ldiamond: the one modem where you could do that with has not been made in 15 years. |
| 16:58.07 | [TK]D-Fender | Asterisk can't use generic junk modems |
| 16:58.45 | [TK]D-Fender | And the only one that did you could not plug a phone into and use |
| 16:59.13 | igcewieling | [TK]D-Fender: good point. that modem only supported phone LINES, not phones. |
| 17:03.54 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:04.26 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:06.58 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:16.59 | ldiamond | [TK]D-Fender: igcewieling: so what are the alternatives? I need to buy another card? |
| 17:17.18 | [TK]D-Fender | yes |
| 17:17.32 | [TK]D-Fender | When what you have isn't usable... you have to get something else. |
| 17:17.48 | [TK]D-Fender | So to be clear .. you want to plug just a PHONE in and use the PHONE with Asterisk, right? |
| 17:17.53 | ldiamond | Basically, the situation is that I have a medical device at home which needs to send data back to the hospital using a phone line. The interface is a phone jack. All I have is a cell phone |
| 17:17.56 | igcewieling | ldiamond: I think your best option is to spend $2/month and get a SIP number. |
| 17:18.11 | ldiamond | igcewieling: I have a sip number and a sip provider |
| 17:18.19 | igcewieling | ldiamond: Asterisk does NOT support modems |
| 17:18.22 | ldiamond | Just no way to plug the device unless I buy the cisco thing |
| 17:18.50 | igcewieling | ldiamond: even if you buy the cisco thing it won't work if you are using a modem to send the data back. |
| 17:19.19 | igcewieling | if the medical device sends data using only DTMF (not modem) then it should work with an ATA |
| 17:19.21 | ldiamond | SIP won't work with modems? |
| 17:19.25 | igcewieling | ldiamond: correct. |
| 17:19.31 | igcewieling | modems send non-voice signals. |
| 17:19.38 | ldiamond | Yea I don't really know what the device uses. |
| 17:19.49 | igcewieling | ldiamond: then you don't know enough. |
| 17:20.11 | ldiamond | ugh, 37$/month for a phone line. |
| 17:20.39 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.39.238) |
| 17:21.29 | igcewieling | ldiamond: multiply by 12, see if the total is less than upgrading to a version of the medical device to support cellular |
| 17:21.55 | ldiamond | I don't own the device, it's the hospital that lends it. |
| 17:22.01 | ldiamond | It's for 1-2 months too |
| 17:22.53 | igcewieling | Are you in the USA? |
| 17:22.57 | ldiamond | Canada |
| 17:23.54 | igcewieling | ah, a country with a civilized health care system. If I had this issue (unless I'd die without it) I'd hand the device back to them and say it won't work with your phone line. |
| 17:27.24 | ldiamond | That's all they have available |
| 17:38.11 | *** join/#asterisk jpastore (~jpastore@50-200-51-210-static.hfc.comcastbusiness.net) |
| 17:38.27 | jpastore | hi is it appropriate to ask a vicidial question in here? |
| 17:39.44 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:41.22 | [TK]D-Fender | maybe barely... |
| 17:41.51 | [TK]D-Fender | finishes burying the body of the last guy.... |
| 17:45.09 | cmendes0101 | haha |
| 17:50.13 | *** join/#asterisk scinawa (~scinawa@client-8-21.eduroam.oxuni.org.uk) |
| 17:50.59 | scinawa | hi guys! :) |
| 17:54.45 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 17:54.52 | Dovid | hellp all. ling time no time |
| 18:02.35 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 18:03.05 | scinawa | What's the right way of setting the timeout of a call from a callfile? I spawn the call from a local channel and then I connect to a context+extension. Using Set(Timeout(absolute)=${duration}) results in a crash each time I spawn a call. |
| 18:06.07 | jpastore | ha. well...then I'll keep googling... |
| 18:12.10 | igcewieling | scinawa: Using WaitTime is the documented way. 0004f235a510-phone1.cfg |
| 18:12.17 | igcewieling | ok, that's not the right link |
| 18:12.26 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 18:13.27 | scinawa | but that's not for hanging up the call after xx seconds of duration, right? |
| 18:13.37 | scinawa | that's for hanging up after some seconds before answer |
| 18:14.17 | igcewieling | ah, good point. What version of Asterisk? |
| 18:14.56 | scinawa | 11.13.1~dfsg-2+b1 currently running on Debian-82-jessie-64-minimal |
| 18:17.03 | scinawa | :v |
| 18:18.26 | igcewieling | Asterisk should not lockup when setting a timeout |
| 18:18.53 | scinawa | that's for sure :D |
| 18:23.04 | igcewieling | try upgrading |
| 18:23.18 | scinawa | kernel: [4954888.638913] asterisk[9420]: segfault at 72c763f1bb6b ip 00007f2d2ea41c8a sp 00007f2ce1364058 error 4 in libc-2.19.so |
| 18:24.18 | igcewieling | try upgrading libc 8-) |
| 18:25.06 | scinawa | wtf indeed |
| 18:25.37 | igcewieling | I think 11.15 might be the latest version in that branch, but it is easy enough to check. |
| 18:26.40 | *** join/#asterisk areski (~areski@80.174.128.113.dyn.user.ono.com) |
| 18:36.09 | *** join/#asterisk blooregard (~matt@50.141.76.9) |
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| 18:45.22 | blooregard | is there any reason to use IAX2 over SIP when trunking between Asterisk boxes? |
| 18:46.27 | WIMPy | Less bandwidth, less trouble. |
| 18:47.07 | blooregard | got it |
| 18:50.23 | igcewieling | less bandwidth, lots more hassle. |
| 19:12.30 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 19:13.37 | blooregard | igcewieling, why do you say more hassle? |
| 19:14.39 | WIMPy | He meant to say "mor features". |
| 19:14.56 | WIMPy | But the mediaonly transfer is obviousely broken. |
| 19:15.33 | igcewieling | blooregard: now you have to debug two protocols, not just one. |
| 19:15.46 | *** join/#asterisk bulkorok (~Adium@92.206.230.45) |
| 19:15.59 | igcewieling | also so few use IAX that you'll have trouble finding anyone to help with it when you have issues. |
| 19:16.18 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 19:16.44 | igcewieling | There is little reason not to use SIP only. |
| 19:17.15 | blooregard | I was just curious. |
| 19:18.02 | blooregard | I haven't used IAX in years. I just remember it works over NAT a lot easier than SIP |
| 19:18.17 | igcewieling | blooregard: SIP has caught up for most setups. |
| 19:18.45 | igcewieling | I manage 200 - 300 endpoints, about 20% are NAT'd, none use IAX. |
| 19:19.31 | igcewieling | if you REALLY need to avoid NAT, then install a tunnel between the two sites so you don't need NAT |
| 19:19.45 | *** join/#asterisk bulkorok (~Adium@92.206.230.45) |
| 19:21.15 | igcewieling | heh, a few more than 300 endpoints. 450 endpoints online |
| 19:21.42 | igcewieling | Around 175 are NAT'd |
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| 19:33.08 | WIMPy | Yes. Many more people like SIP because they are familiar with debugging it. Whay that is, is left for the reader to decide :-) |
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| 20:07.15 | Nugget | I hate NAT so much. |
| 20:09.32 | stephanj | and NAT hates you. |
| 20:09.50 | *** join/#asterisk jjrh (~jjrh@2607:f0b0:1:6e12:6890:8222:5321:2c0e) |
| 20:14.00 | [TK]D-Fender | not as bad as ... telnet |
| 20:14.09 | [TK]D-Fender | :/ |
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| 20:50.33 | epinky | how to change multicastrtp ttl? |
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| 20:57.40 | *** join/#asterisk dorphalsig (be939928@gateway/web/freenode/ip.190.147.153.40) |
| 20:57.56 | dorphalsig | Hello |
| 20:58.18 | dorphalsig | anybody lives? |
| 20:59.38 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 20:59.44 | pjensen00 | Depends. |
| 21:03.32 | [TK]D-Fender | Everyone dies (over an extended enough period of time) |
| 21:03.38 | [TK]D-Fender | #existentialism |
| 21:04.00 | *** join/#asterisk hecatae (~hecatae@host-89-240-2-225.static.as13285.net) |
| 21:11.32 | *** join/#asterisk dorphalsig_ (be939928@gateway/web/freenode/ip.190.147.153.40) |
| 21:11.44 | dorphalsig_ | So yeah, I just discovered mortality myself |
| 21:12.28 | dorphalsig_ | anyway, I want to send some custom headers when a queue member rings |
| 21:13.01 | dorphalsig_ | So far I've thought of using local channels (one for each member of the queue), but it sounds hackish |
| 21:13.13 | dorphalsig_ | does anybody have a better idea? |
| 21:14.26 | [TK]D-Fender | that's the most explicit way |
| 21:14.31 | [TK]D-Fender | or set it before they hit the queue |
| 21:15.07 | dorphalsig_ | can I do that? |
| 21:15.20 | [TK]D-Fender | Can't you? |
| 21:15.32 | dorphalsig_ | I'm not sure :P |
| 21:15.52 | igcewieling | dorphalsig_: have you read the output of "core show application queue" ? |
| 21:15.56 | dorphalsig_ | I mean, I thought I had to know which channel to send it to |
| 21:16.26 | *** join/#asterisk stkoch (~stefan@p5B357D06.dip0.t-ipconnect.de) |
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| 21:17.29 | dorphalsig_ | igcewieling: yeah, with the macro thingy? But doesnt that get triggered when the call connects (the agent answers)? I need to send an auto answer header |
| 21:18.05 | stkoch | hi, I have two SIP providers (sipgate and tonline) and want to forward calls from sipgate to tonline - with that I get the CHANUNAVAIL error, see: http://paste.debian.net/hidden/1a1e135c/ |
| 21:18.56 | stkoch | directmedia=no |
| 21:20.35 | stkoch | It seems that asterisk wants to connect both SIP connections directly. But it should do it internal itself... |
| 21:21.22 | [TK]D-Fender | So far nothing seems like anything |
| 21:21.29 | [TK]D-Fender | We aren't looking at SIP debug |
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| 21:30.34 | [TK]D-Fender | packs up to head home |
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| 21:51.59 | stkoch | this is the sip.conf: http://paste.debian.net/hidden/9ff5871f/ |
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| 21:53.03 | stkoch | this is the extensions.conf: http://paste.debian.net/hidden/dfd927e8/ |
| 21:53.57 | stkoch | Why comes the CHANUNAVAIL error at call forwarding from sipgate to tonline ? |
| 21:54.33 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:54.48 | robmal | Can you call out via tonline via callfile/manager/cli/smoke signals/enchanted pidgeons? |
| 21:57.33 | *** join/#asterisk chadxz (Adium@nat/digium/x-sypnqchqbmkytnsq) |
| 21:58.26 | stkoch | robmal: I can call outside. I can also forward from tonline to tonline... |
| 21:59.23 | robmal | Is the number format correct when forwarding? |
| 22:00.40 | stkoch | robmal: here is a sip debug: http://paste.debian.net/hidden/ee69e90e/ |
| 22:01.02 | robmal | Ok, now show me a call which does connect via tonline. |
| 22:01.35 | stkoch | The sip debug is incoming sipgate forwarded to tonline |
| 22:03.18 | robmal | "488 Gewünschter Rufaufbau wird vom Ziel nicht unterstützt." - i think it means they don't like forwards, Stefan. You might have to proxy that rtp stream. |
| 22:04.33 | *** join/#asterisk talntid (~talntid@mail.llcagent.com) |
| 22:04.34 | talntid | http://pastebin.com/5bJq5Etu |
| 22:04.55 | talntid | can someone help assist and help me figure out why line 48-49 happens? |
| 22:05.18 | talntid | basically sometimes when i call out, once the other party answers, the call drops - and that is the sip debug. other times, it works fine. |
| 22:06.08 | stkoch | robmal: yes I want to proxy that, how to do? |
| 22:07.08 | robmal | talntid: From: "Name" <sip:blahblah@blah> etc |
| 22:07.54 | robmal | stkoch: Kamailio? |
| 22:08.18 | talntid | robink, I don't see what you are pointing out? |
| 22:08.27 | robmal | From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29 |
| 22:08.33 | robmal | Shouldn't look like this. |
| 22:08.41 | talntid | oh, what should it look like? |
| 22:09.16 | robmal | It should look like RFC specifies it too look like. |
| 22:09.18 | robmal | ~book |
| 22:09.23 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:09.24 | talntid | roger that |
| 22:11.38 | stkoch | robmal: is it possibe to use only asterisk? |
| 22:12.06 | stkoch | The principe would be: |
| 22:12.22 | stkoch | -incoming call (sipgate) |
| 22:12.31 | stkoch | -outgoing call (tonline) |
| 22:12.53 | stkoch | connect these with asterisk internally? |
| 22:12.55 | robmal | Some mix of canreinvite and nat might work. |
| 22:13.20 | stkoch | you see in sip.conf, nat=yes, directmedia=no |
| 22:14.09 | stkoch | I have read that directmedia replaces canreinvite? |
| 22:18.00 | robmal | Looks like so. |
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| 23:07.50 | stkoch | robmal: thanks |
| 23:08.53 | robmal | Always a pleasure. |
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