| 00:21.21 | *** join/#asterisk markl (~markl@d54c2c766.access.telenet.be) |
| 00:22.32 | markl | still having problems with stopping rtp streams on asterisk 11.20.0 |
| 00:23.21 | markl | in a call with rtp debug on i suddenly see only "Got RTP", no "Send RTP" |
| 00:23.26 | markl | anyone an idea? |
| 00:32.00 | talntid2 | I don't know. Seems pretty dead here today |
| 00:44.14 | SeRi | so many years of silence |
| 00:50.28 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:50.29 | talntid2 | yeah |
| 00:50.40 | SeRi | :) |
| 00:51.21 | talntid2 | this is so frustrating |
| 00:51.28 | SeRi | :/ |
| 00:51.31 | talntid2 | everything works, except no sound. |
| 00:51.40 | talntid2 | makes me want to bash my head into a woodchipper :) |
| 00:51.46 | WIMPy | You get used to it. |
| 00:51.47 | SeRi | talntid2: rtp issues? |
| 00:52.11 | SeRi | WIMPy: houdy there, Long time no see! |
| 00:52.18 | talntid2 | SeRi, I'm sure it is.. just.. there are no firewalls in the way or anything... I just must have misconfigured transports or something. |
| 00:52.32 | WIMPy | He, I'm still here a lot. |
| 00:52.58 | WIMPy | Or configured NAT even if not needed? |
| 00:52.59 | SeRi | WIMPy: I spent more time with docs and hospitals now... No time fore the fun stuff |
| 00:53.34 | talntid2 | I don't think I have configured NAT, because I don't even think I know how to configure NAT with this PJSip stuff. It's not as straightforward |
| 00:53.39 | WIMPy | Strqange definition of "fun". But might still be better than yours. |
| 00:53.47 | talntid2 | SeRi, that's a bummer :( |
| 00:53.57 | SeRi | talntid2: what WIMPy said |
| 00:54.02 | SeRi | talntid2: mehhh |
| 00:54.12 | SeRi | talntid2: I am sure is a nat issue |
| 00:54.15 | WIMPy | Might be somethign automatic. |
| 00:54.32 | SeRi | WIMPy: correction automagic |
| 00:54.59 | talntid2 | I am sure it is something like that too, just, this new PJSip + realtime stuff, isn't as easy to troubleshoot for me yet, as I don't understand all the moving pieces |
| 00:55.23 | talntid2 | there are no longer things like nat=yes |
| 00:56.01 | SeRi | talntid2: you have to be like me stick to oldskool 1.8 branch... LOL |
| 00:56.34 | talntid2 | haha |
| 00:56.41 | talntid2 | i understand what goes on in there :P |
| 00:56.43 | SeRi | wiats for the fire breathing dragons |
| 00:56.54 | WIMPy | For the moment I still have hopes to be able to stick with 11. |
| 00:57.01 | WIMPy | But you know the thing about hope... |
| 00:57.08 | SeRi | WIMPy: LOL |
| 00:57.09 | talntid2 | no fire breathing dragons, they aren't around.. they are trying to fix pjsip+realtime :P |
| 00:57.26 | talntid2 | or guarding all of the secrets about it |
| 00:58.27 | talntid2 | sucks, because our PBX requires constant additions/removals of extensions and queues and such, so I just write web interfaces for it. I was looking forward to doing it like this |
| 00:58.52 | SeRi | talntid2: LOL |
| 00:59.06 | SeRi | talntid2: sucks to hit a wall |
| 00:59.20 | talntid2 | yep indeed |
| 00:59.22 | talntid2 | oh well |
| 00:59.29 | SeRi | talntid2: I dont even know what you are talking about so I just put my hands up and drink beer. |
| 00:59.39 | talntid2 | that sounds like a better plan ;) |
| 00:59.44 | SeRi | talntid2: sorry I cant be much of help but all this new shit is too much for me... LOL |
| 00:59.55 | talntid2 | basically, cliffnotes |
| 00:59.56 | WIMPy | Oh, someone is being clever here :-) |
| 01:00.09 | SeRi | WIMPy: LOL |
| 01:00.12 | talntid2 | someone decided chan_sip sucked, so they made a new one called pjsip |
| 01:00.27 | talntid2 | so sip.conf goes away, and is replaced by putting things in 3 different places |
| 01:00.36 | talntid2 | additionally, realtime is... |
| 01:00.38 | SeRi | talntid2: just like sytsemd |
| 01:00.39 | WIMPy | No, it's not chan_sip that sucks. SIP sucks. |
| 01:00.50 | SeRi | somebody decided initd sucks |
| 01:00.58 | WIMPy | yikes |
| 01:00.59 | SeRi | what they dont know is that thir mom suck |
| 01:01.12 | talntid2 | all the sip stuff, voicemails, extensions, etc are stored in a database now |
| 01:01.14 | talntid2 | when using realtime |
| 01:01.36 | talntid2 | extensions is optional, and probably left as a flat file |
| 01:01.56 | talntid2 | but queues, sip endpoints, voicemails, etc are in database |
| 01:02.37 | talntid2 | so, enjoy your beer, it's easier to understand! ;) |
| 01:02.53 | SeRi | talntid2: LOL, Well Thanks |
| 01:03.42 | SeRi | talntid2: I just need to catch up is all, I actually like the fact that all is now store in DB, I use the asterisk DB quite allot my self. |
| 01:04.01 | talntid2 | yeah that is why I want to use it too! |
| 01:04.11 | talntid2 | but NoOoOoOoOoooooooooo.... it just wants to be a pain in the ass |
| 01:04.15 | WIMPy | That option is not new at all. |
| 01:04.30 | talntid2 | yeah, that's not new at all. I have built other interfaces with that |
| 01:04.38 | talntid2 | just trying to be a hipster, that's all. |
| 01:13.39 | SeRi | talntid2: :) |
| 01:14.08 | SeRi | WIMPy: How is 11 working out for you? |
| 01:14.20 | SeRi | I need to find the time to move |
| 01:14.22 | SeRi | I am so behind |
| 01:14.33 | talntid2 | i'll move yours, you move mine |
| 01:14.36 | talntid2 | :) |
| 01:14.37 | SeRi | is not like 11 is that new anyways...lol |
| 01:14.46 | SeRi | talntid2: Thats no fun |
| 01:14.47 | SeRi | LOL |
| 01:15.10 | SeRi | I have a new project in my hands and is call open-mesh :/ |
| 01:15.21 | SeRi | oh before I forget, I hate shortel |
| 01:15.22 | *** join/#asterisk dms (~Adium@74.121.217.2) |
| 01:15.39 | talntid2 | doesn't everyone? |
| 01:15.46 | WIMPy | I have a big issue with DTMF on native bridges. Somethign has changed there and it was not for the better. |
| 01:16.05 | SeRi | talntid2: a safe assumption |
| 01:16.26 | SeRi | damn typos |
| 01:16.42 | WIMPy | I don't. I don't know them :-) |
| 01:16.54 | SeRi | WIMPy: lol |
| 01:17.13 | SeRi | anybody here play with open-mesh in the past? |
| 01:17.34 | talntid2 | nopers |
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| 05:04.59 | igcewieling1 | I should have known my "NOT HERE TO MAKE FRIENDS" t-shirt would make people talk to me. |
| 05:05.05 | igcewieling1 | 8-| |
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| 10:48.53 | ak77_ | hello all |
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| 13:17.05 | *** join/#asterisk ModFather (~johnd@athedsl-145228.home.otenet.gr) |
| 13:17.08 | ModFather | hi There |
| 13:19.36 | ModFather | i have setup Asterisk 13, i've got a number from twilio and i setup Sip Trunk |
| 13:19.52 | ModFather | i can get calls from outside, but i cannot make calls to a friend on a US number |
| 13:20.04 | [TK]D-Fender | what happens? |
| 13:20.34 | ModFather | [Oct 14 13:05:26] NOTICE[30713][C-00000000]: chan_sip.c:25660 handle_request_invite: Call from 'johnd' (myiphere:63831) to extension '+171xxxxxxx' rejected because extension not found in context 'LocalSets'. |
| 13:20.45 | [TK]D-Fender | Means exactly what it says |
| 13:20.58 | [TK]D-Fender | you are dialing a number from your phone that you do not have a match for in your dialplan |
| 13:21.18 | [TK]D-Fender | noting in [LocalSets] matches the number you dialed |
| 13:21.21 | [TK]D-Fender | nothing* |
| 13:21.35 | ModFather | yes i purchased a book from amazon regarding asterisk, and i read some blogs on google, and i added on [LocalSets] |
| 13:22.09 | ModFather | exten => +/NXXNXXXXXX,1,Dial(SIP/johnd/${EXTEN}) |
| 13:22.15 | ModFather | exten => same,n,Dial(SIP/twilio0/+${EXTEN}) |
| 13:22.20 | [TK]D-Fender | that / is no good.... |
| 13:22.31 | ModFather | yes i have tried later, after some research |
| 13:22.42 | [TK]D-Fender | <PROTECTED> |
| 13:22.47 | ModFather | i had it _1 before i ad / |
| 13:22.49 | [TK]D-Fender | remove it and fix the rest of the pattern |
| 13:23.15 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:23.17 | ModFather | exten => _1NXXNXXXXXX,1,Dial(SIP/johnd/${EXTEN}) |
| 13:23.29 | [TK]D-Fender | Will not work. |
| 13:23.33 | [TK]D-Fender | does not match what you are dialing |
| 13:23.46 | ModFather | okey 1 sec to read again the section on the dialing match patterns |
| 13:23.59 | [TK]D-Fender | you need to look at what your phone is sending |
| 13:24.25 | ModFather | it calls my friend US number |
| 13:24.58 | [TK]D-Fender | Still doesn't match the number you dialed |
| 13:25.09 | ModFather | +17180000000 |
| 13:25.16 | ModFather | this is the number i am trying to call |
| 13:25.28 | ModFather | its not a real i changed the numbers to 0 |
| 13:25.42 | [TK]D-Fender | And still won't match that |
| 13:25.52 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 13:26.01 | ModFather | alright, i am gonna read again the section on that |
| 13:26.03 | ModFather | and i will retry |
| 13:26.12 | ModFather | many thanks man , i thought it was something else |
| 13:26.49 | [TK]D-Fender | "rejected because extension not found" <- means that you don't have an extension that matches what was dialed. |
| 13:27.44 | ModFather | hmm yes but as i read i thought that _1NXXNXXXXXX matches the number i am dialing |
| 13:27.49 | ModFather | _ for symbol + |
| 13:27.54 | ModFather | 1 the US number |
| 13:28.00 | ModFather | and the rest 718000000 |
| 13:29.09 | [TK]D-Fender | _ is NOT + |
| 13:29.16 | [TK]D-Fender | _ says that what follows is a pattern |
| 13:29.25 | ModFather | sorry yes |
| 13:29.30 | ModFather | N matches to digit from 2-0 |
| 13:29.32 | ModFather | N matches to digit from 2-9 |
| 13:29.40 | ModFather | X matches any digit from 0-9 |
| 13:29.47 | [TK]D-Fender | otherwise those N & X you have in there are taken as LITERAL letters of the alphabet being part of the number being passed |
| 13:30.00 | ModFather | aha yes right |
| 13:30.27 | ModFather | _ = the following is a pattern |
| 13:30.39 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 13:30.47 | ModFather | so in my already config, if i call without + extension |
| 13:30.51 | ModFather | its gonna work? |
| 13:30.56 | ModFather | gonna try |
| 13:31.00 | [TK]D-Fender | So far.... |
| 13:31.12 | [TK]D-Fender | If the patttern matches.. then it matches |
| 13:32.30 | ModFather | exten => _17183XXXXXX,1,Dial(SIP/johnd/${EXTEN}) |
| 13:32.33 | ModFather | okey i am trying this one |
| 13:32.43 | ModFather | i checked and the XXX is exact the other numbers |
| 13:34.19 | ModFather | [TK]D-Fender, you are the best man ;) |
| 13:34.23 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:34.27 | ModFather | now it tries to call ;) |
| 13:34.48 | [TK]D-Fender | Good lesson learned today... |
| 13:35.30 | ModFather | heh yes |
| 13:35.34 | ModFather | i thought i had it correct |
| 13:35.50 | ModFather | without you i would still thinking something else is the wrong there |
| 13:36.08 | ModFather | i did many tries thats why the / on the extension before |
| 13:36.19 | ModFather | now i get an error after 30 seconds |
| 13:36.21 | ModFather | [Oct 14 13:33:38] WARNING[1312]: chan_sip.c:4038 retrans_pkt: Hanging up call 612430fd5dd1bd051f91d9a22f7ote6c@ipofserverhere:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 13:36.40 | [TK]D-Fender | <PROTECTED> |
| 13:36.58 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:37.05 | [TK]D-Fender | That error suggest you've got a networking iswsue, most likely due to improper NAT setup |
| 13:37.33 | ModFather | nat issue from my softphone or servers network? |
| 13:37.47 | ModFather | the server is hosted on RackSpaceCloud ( no firewalls enabled at the moment ) |
| 13:38.03 | ModFather | and my softphone is linphone app on my iphone |
| 13:38.05 | [TK]D-Fender | either |
| 13:38.21 | [TK]D-Fender | "sip set debug on" <- |
| 13:38.25 | [TK]D-Fender | ~pb |
| 13:38.33 | 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. |
| 13:38.37 | [TK]D-Fender | ^^^ |
| 13:39.58 | ModFather | Retransmitting #5 (no NAT) to mysoftphoneip:5060: |
| 13:40.10 | ModFather | you are godlike man.. going to fix that issue |
| 13:40.13 | ak77_ | asterisk/pjsip closes tcp/ip (tls) connection after each REQ/RES... why? |
| 13:41.25 | ModFather | [TK]D-Fender, i think i will need to define on asterisk.conf nat=yes and make my phone lan ip on my router on DMZ setting |
| 13:41.33 | ak77_ | registration makes two connections, as first RES is 401, then new connection is maed with REGISTER and basic authentication |
| 13:41.55 | [TK]D-Fender | ModFather: You normally never need to DMZ or forward clients |
| 13:42.40 | ModFather | my router using NAT already so i guess i need to open ports ? |
| 13:44.00 | [TK]D-Fender | no |
| 13:44.09 | [TK]D-Fender | just let * know what its situation is |
| 13:44.29 | [TK]D-Fender | which should be "nat=yes", "qualify=yes", "directmedia=no" <- last one is critical. |
| 13:53.09 | ModFather | [TK]D-Fender, http://pastebin.ca/YhYtQGOS |
| 13:54.25 | ModFather | [TK]D-Fender, i should add those on asterisk.conf and not on any file sip.conf or extensions.conf right? |
| 13:54.26 | ModFather | nat=yes", "qualify=yes", "directmedia=no" |
| 13:55.31 | *** join/#asterisk cmendes0101 (~cmendes01@172.56.27.180) |
| 13:59.50 | [TK]D-Fender | no |
| 13:59.55 | [TK]D-Fender | that is not for asterisk.conf |
| 13:59.59 | [TK]D-Fender | that is sip.conf |
| 14:00.25 | [TK]D-Fender | SIP stuff clearly does not belong anywhere else except sip.conf (or pjsip.conf if that's the channel driver you're using) |
| 14:01.42 | ModFather | aha i understand now, i am using only sip channel not pjsip, so i am gonna use those on the [twilio-trunk] and on my phone [1000] |
| 14:03.04 | [TK]D-Fender | no, those are just for your PHONE |
| 14:03.09 | [TK]D-Fender | twilio is NOT behind NAT. |
| 14:03.14 | [TK]D-Fender | They should specifically be nat=no |
| 14:03.18 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 14:03.40 | ModFather | yes i added only to my phone section |
| 14:03.41 | ModFather | nat=yes is deprecated, use nat=force_rport,comedia instead |
| 14:03.55 | ModFather | so i guess i should change it to nat=force_rport,comedia |
| 14:04.14 | *** join/#asterisk kharwell (kharwell@nat/digium/x-vlybfmppuwlseyvn) |
| 14:05.40 | ModFather | okey now it doesn't dialing and i get: WARNING[5041][C-00000003]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 14:06.03 | *** join/#asterisk defsdoor (~andy@cpc37-sutt4-2-0-cust84.19-1.cable.virginm.net) |
| 14:06.24 | [TK]D-Fender | The device being called is considered non-contactable |
| 14:06.45 | [TK]D-Fender | if you changed twilio's settings to include "qualify" and they reject it then they will be considered non-contactable |
| 14:06.56 | [TK]D-Fender | You should not have those there probably |
| 14:07.06 | *** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 14:07.11 | ModFather | yes i just added it |
| 14:07.21 | ModFather | and now i cannot call the same number i was trying |
| 14:07.45 | ModFather | the number i am calling is valid, is my friends mobile, so i guess i should set qualify to no |
| 14:08.47 | ModFather | now i see its using NAT |
| 14:09.01 | ModFather | but i still get WARNING[4773]: chan_sip.c:4009 retrans_pkt: Retransmission timeout reached on transmission |
| 14:09.04 | ModFather | like before |
| 14:13.31 | ModFather | [TK]D-Fender, hehe ... i just installed another softphone on my pc Zopier, and on my iphone i have Linode , when i am calling my friends US mobile i get incoming call to my softphone from me |
| 14:14.41 | ModFather | i think i know what is |
| 14:19.15 | ModFather | SIP/2.0 603 Declined |
| 14:21.04 | [TK]D-Fender | [09:38]infobotA "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. |
| 14:21.05 | [TK]D-Fender | [09:38][TK]D-Fender^^^ |
| 14:23.54 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 14:24.12 | ModFather | [TK]D-Fender, http://pastebin.ca/Wz8yFbYb my extension.conf pass: 12344 |
| 14:24.52 | [TK]D-Fender | 118dialplan has nothing to do with this |
| 14:26.04 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-svcjtirdbzxguzrx) |
| 14:26.27 | ak77_ | anyone has an idea why tcp/ip connection is not preserved between sip each req/res ? (asterisk trunk, pjsip, outbound calls with registration) |
| 14:27.36 | ModFather | [TK]D-Fender, my sip.conf http://pastebin.ca/QWwcJbbU pass:12344 |
| 14:27.43 | ModFather | and now i am gonna send you the debug |
| 14:28.37 | [TK]D-Fender | ModFather: I told you to do "nat=no" for your twilio peers |
| 14:31.57 | ModFather | [TK]D-Fender, sorry, i just added nat=no to twilio-trunk |
| 14:32.30 | ModFather | do i have to add nat=no to every twilio ip? |
| 14:33.07 | ak77_ | i am testing SIP calls with asterisk and with pjsua cli app... there are few differences (don't know if that is the reason tcp connection is closed after each req/res) like SIP url in REGISTER and SIP url in Route header are in each other places (compared to pjsua cli). |
| 14:33.10 | ModFather | on my sip.conf i had twilio-trunk inside a template, so i guess the rest peers will get that nat=no |
| 14:34.33 | WIMPy | ak77_: I don't know much about PJ, but if the connection was persistant, there wouldn't be a point in registereing, would there? |
| 14:35.00 | ak77_ | WIMPy: it's TLS and ITSP requires registration |
| 14:37.14 | ak77_ | WIMPy: ok, maybe I can leave out registration, but problem remains that on INVITE request I got ringing status but because connection closed, I never got notified that other party answered |
| 14:40.15 | igcewieling | ak77_: sounds like classic NAT issues. |
| 14:40.33 | igcewieling | does it work if you use UDP? |
| 14:40.52 | WIMPy | No, sounds like something's seriousely broken. |
| 14:41.59 | ModFather | [TK]D-Fender, http://pastebin.ca/AUfjMLUD debug info pass: 12344 |
| 14:42.37 | ak77_ | igcewieling: with another ITSP/udp it works yes. |
| 14:43.01 | *** join/#asterisk cmendes0101 (~cmendes01@172.56.27.79) |
| 14:43.59 | ak77_ | igcewieling: and... it works with pjsua cli app |
| 14:44.18 | igcewieling | I can't help with pjsip. |
| 14:44.40 | igcewieling | I don't plan on using it until Asterisk 15. |
| 14:46.42 | *** join/#asterisk dms (~Adium@74.121.220.2) |
| 14:47.39 | ak77_ | igcewieling: yes. i maybe made a mistake for using it too early. |
| 14:49.06 | igcewieling | ak77_: I very much dislike angry customers calling so tend to run not-the-newest-code. |
| 14:49.36 | igcewieling | the main advantage of chan_sip, in my opinion, is almost all Asterisk docs on SIP and online assume chan_sip. |
| 14:50.59 | stefan27 | I get [2015-10-14 16:39:54] WARNING[19082] http.c: HTTP session count exceeded 100 sessions at a time where I expect the number of sessions to be 2 (netstat -tupan | grep asterisk shows 2 sessions) . How can I debug this? |
| 14:51.13 | stefan27 | (asterisk 13.5.0) |
| 14:51.35 | ModFather | [TK]D-Fender, i did nat=no to all twilio peers, nothing changed i still see the same debug logs |
| 14:57.57 | [TK]D-Fender | ModFather: "core set verbose 10" |
| 14:58.00 | [TK]D-Fender | place another call |
| 14:58.28 | ModFather | [TK]D-Fender, ok |
| 14:59.28 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ftmyfnfnnxjbrymm) |
| 14:59.44 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-qegejzgodmjggtii) |
| 15:01.41 | *** join/#asterisk vandyk (~vandyk@191.187.218.127) |
| 15:03.45 | ModFather | [TK]D-Fender, done.. http://pastebin.ca/XLS-hPnl pass: 12344 |
| 15:05.18 | [TK]D-Fender | -- Auto fallthrough, channel 'SIP/johnd-00000009' status is 'UNKNOWN' |
| 15:05.23 | [TK]D-Fender | you ran out of dialplan to execute |
| 15:05.30 | [TK]D-Fender | it did a Set, and then ended |
| 15:05.37 | [TK]D-Fender | -- Executing [17183000000@LocalSets:1] Set("SIP/johnd-00000009", "CALLERID(all)="+18883306354" <+18883306354>") in new stack |
| 15:05.50 | [TK]D-Fender | You have no 2nd step to continue on to doing |
| 15:06.51 | ModFather | looking on my extensions.conf [TK]D-Fender right now 1 sec |
| 15:07.21 | ModFather | exten => same,n,Dial(SIP/twilio0/+${EXTEN}) |
| 15:07.27 | ModFather | isn't that the 2nd step? |
| 15:07.43 | [TK]D-Fender | PB the newst version |
| 15:07.50 | [TK]D-Fender | and no, that is BASD |
| 15:08.13 | *** join/#asterisk K0HAX (michael@shellhost.home.englehorn.com) |
| 15:08.14 | [TK]D-Fender | That is not how "same" works |
| 15:08.28 | ModFather | now i am really confused :( |
| 15:08.49 | [TK]D-Fender | read the syntax for using "same" again |
| 15:09.16 | ModFather | is that what i need ? the same syntax? or am i missing something else after "set" |
| 15:09.34 | [TK]D-Fender | ... |
| 15:09.49 | [TK]D-Fender | ModFatherexten => same,n,Dial(SIP/twilio0/+${EXTEN}) <- this line is WRONG |
| 15:10.00 | [TK]D-Fender | Why did you put the word "same" there? |
| 15:10.08 | [TK]D-Fender | This is not proper syntax |
| 15:10.32 | [TK]D-Fender | Go re-read the page you got that idea from and see how it is supposed to be done |
| 15:10.41 | ModFather | sorry for asking i am not very sure, i started from 0, all i did is what i have read on asterisk book and on the documentations ( without that means that those were wrong ) i am wrong i did it wrong |
| 15:12.11 | igcewieling | ModFather: you read the docs wrong. Your same has invalid syntax. |
| 15:15.09 | ModFather | yes |
| 15:15.28 | ModFather | i know guys i just now realized that "same" should be replaced with _17173XXXX |
| 15:15.50 | ModFather | the number i wrote is example pattern not the real one |
| 15:16.03 | igcewieling | scream* change it to: same => n,Dial(SIP/twilio0/+${EXTEN}) |
| 15:16.15 | igcewieling | now go back and read the damn docs again. |
| 15:17.46 | *** join/#asterisk overyander (~jeff.clay@12.49.160.131) |
| 15:19.34 | ModFather | igcewieling, i did the change you mention |
| 15:19.37 | ModFather | [Oct 14 15:19:10] NOTICE[13211]: pbx.c:5017 pbx_extension_helper: Cannot find extension 'n' in context '' |
| 15:19.37 | ModFather | [Oct 14 15:19:10] WARNING[13211]: pbx_config.c:1783 pbx_load_config: Invalid priority/label 'Dial' at line 8 of extensions.conf |
| 15:19.43 | [TK]D-Fender | Show us |
| 15:19.48 | [TK]D-Fender | the new config... |
| 15:19.59 | igcewieling | ModFather: I'm not helping you further. The proper syntax is documented. |
| 15:21.11 | ModFather | igcewieling, sorry man for the disturb, can you send me a link if you had it handy for to follow it? |
| 15:21.26 | igcewieling | ~book |
| 15:21.26 | 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 |
| 15:21.34 | ak77_ | igcewieling: i change to using res_sip. :) it doesn't disconnect tcp/ip (tls) when two req/res are sent during registration. |
| 15:22.14 | igcewieling | ak77_ now you have years worth of documentation on the internet you can read. |
| 15:22.15 | ModFather | this is the book 'ive got |
| 15:22.21 | *** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 15:22.50 | [TK]D-Fender | [11:19][TK]D-FenderShow us |
| 15:22.51 | [TK]D-Fender | [11:19][TK]D-Fenderthe new config... |
| 15:23.57 | ModFather | [TK]D-Fender, its still the same, i just removed the same before ,n,Dial .. but i supposed is not that a fix.. really sorry, i am going again into the book and try find the correct syntax.. really sorry for anoying you guys.. i appreciate so far [TK]D-Fender |
| 15:24.15 | [TK]D-Fender | What part of "show" are you having trouble with? |
| 15:24.28 | [TK]D-Fender | If you'd like to get it fixed .... SHOW US so we can point out the mistake |
| 15:24.47 | [TK]D-Fender | Should have taken all of 10 seconds. |
| 15:26.15 | ModFather | [TK]D-Fender, http://pastebin.ca/3196576 |
| 15:26.57 | [TK]D-Fender | I don't see the line he gave you there... |
| 15:27.24 | [TK]D-Fender | This is failure to take the 100% complete solution he handed you. |
| 15:27.39 | [TK]D-Fender | [11:16]igcewielingscream* change it to: same => n,Dial(SIP/twilio0/+${EXTEN}) |
| 15:28.40 | ModFather | you are totaly true. sorry for that i am little bit stressed |
| 15:31.09 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:33.12 | Phil-Work | I'm doing "channel originate SIP/number@peer-name" from the CLI... how do I set the Caller ID that this used? |
| 15:33.49 | [TK]D-Fender | You can't via that method |
| 15:34.10 | [TK]D-Fender | you could if you dial a Local channel, set it first and continue to a Dial() |
| 15:37.38 | Phil-Work | thought that might be the case, thanks |
| 15:40.37 | Phil-Work | [TK]D-Fender, what's the syntax to originate to a local channel? |
| 15:40.55 | [TK]D-Fender | Local/extension@context/n |
| 15:41.38 | Phil-Work | that works, thanks |
| 15:42.03 | [TK]D-Fender | You're welcome |
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| 15:48.12 | ModFather | [TK]D-Fender, sorry for late response, i went out for cigarets |
| 15:49.11 | ModFather | [TK]D-Fender, its fixed, i apologize for the disturb i made, big thanks to you man, and also i would like to thanks igcewieling for send me the exact command to make it work. now i can call outside |
| 15:49.31 | ModFather | but without you i wouldnt made it |
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| 16:04.55 | *** join/#asterisk ksuvakin (~ksuvakin@static-dsl-141.213-160-188.telecom.sk) |
| 16:05.00 | ksuvakin | Hi folks! |
| 16:06.05 | ksuvakin | How to configure an asterisk (13.5.0/PJSIP 2.4.5) to answer 200/OK to OPTIONS request? My SIP provider using it as keep-alive.. |
| 16:06.53 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:09.03 | [TK]D-Fender | Look at the debug as to what it is looking for. |
| 16:09.08 | [TK]D-Fender | SIP debug + verbose tells you |
| 16:10.25 | ksuvakin | [TK]D-Fender: For nothing - the same functionality i can reach when i set qualify_frequency=30 in SIP trunk aor |
| 16:11.23 | [TK]D-Fender | Not entirely |
| 16:11.29 | [TK]D-Fender | there are other purposes served |
| 16:11.36 | ksuvakin | [TK]D-Fender: Basically it could complain qualify need - aor knows its latency and the fact that remote point is alive |
| 16:12.14 | ksuvakin | [TK]D-Fender: Hmmm, i receive this request every 20s from provider |
| 16:12.56 | [TK]D-Fender | That would normally be considered quite excessive |
| 16:13.05 | ksuvakin | [TK]D-Fender: My Asterisk answer with 403/Unauthorized, but providet tell me that the response should be 200/OK |
| 16:13.27 | [TK]D-Fender | And most don't send anything. Yours seems over the top. I'd probably ask them to stop if it isn't required |
| 16:13.53 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-mqclowpuovrtzeez) |
| 16:14.08 | ksuvakin | [TK]D-Fender: Understand, i will talk about it with provider. Thank you for help |
| 16:17.51 | ModFather | [TK]D-Fender, is the best! |
| 16:19.00 | [TK]D-Fender | I am not the hero #asterisk deserves, I am the hero #asterisk needs. </darkircvet> |
| 16:19.36 | ModFather | i believe in heroes i can see.. |
| 16:20.11 | ModFather | i will join this channel every day to spam 1 line every single day, [TK]D-Fender is the best, until i got g-line |
| 16:20.21 | [TK]D-Fender | Please don't |
| 16:20.55 | [TK]D-Fender | You're welcome for the time I spent kicking you in the right direction. Your acknowledging having learned a few things is thanks enough. |
| 16:21.29 | ModFather | your time were appreciated and just thanks isnt enough from my side |
| 16:22.00 | ModFather | i buy a book, i read 40+ blogs , and you just fixed it :) |
| 16:22.19 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 16:23.00 | [TK]D-Fender | Technically igcweiling ghanded you the answer. I pointed out what was incorrect hoping you'd re-read the paragraph that explained where you were going with that in hopes you'd find it yourself first |
| 16:23.09 | [TK]D-Fender | Then he handed it to you... and you didn't follow it. |
| 16:23.15 | [TK]D-Fender | So not quitre where I was going with things. |
| 16:23.26 | [TK]D-Fender | but hopefully you understand what it was that was given to you |
| 16:27.31 | *** join/#asterisk bluemerlin (~bluemerli@109.236.171.1) |
| 16:31.38 | ModFather | [TK]D-Fender, i went very stressed with all this, thats why i missunderstood his guide, but without you i wouldnt get that far so the fix of igcweiling fix that |
| 16:31.52 | ModFather | you helped me debug the nat issues and some other stuff i had wrong |
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| 17:48.53 | ModFather | [TK]D-Fender, do you know any sites i can see the queue's ? i would like when someone call the twilio number, to ring on all available VoIPS |
| 17:50.47 | [TK]D-Fender | Then dial multiple devices |
| 17:50.50 | [TK]D-Fender | or setup a queue |
| 17:50.58 | [TK]D-Fender | "core show application dial" |
| 17:51.12 | [TK]D-Fender | "core show application queue" |
| 17:51.49 | [TK]D-Fender | Queue's are more for the point of continuing to hunt for longer periods of time, scaling in new devices as time goes on, playhing music while they wait, etc... |
| 17:51.53 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 17:52.04 | [TK]D-Fender | If you just want to ring all... then you're best off just Dialing mutliple at once |
| 17:52.45 | ModFather | i want to setup a queue |
| 17:53.03 | ModFather | i think its more advance this to be setup |
| 17:55.30 | ModFather | hmm now may main issue is that i cannot call another voip which is registered |
| 17:55.57 | ModFather | i am 100 and the other voip is 101, seems i need to create new extension for internal calls |
| 17:58.23 | ModFather | fixed |
| 17:59.06 | ModFather | but i've got NOTICE[8457]: res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination |
| 17:59.33 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:04.15 | [TK]D-Fender | IPv4 vs IPv6 mismatch in the contact and media |
| 18:05.20 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 18:05.58 | ModFather | those settings are inside asterisk.conf |
| 18:06.00 | ModFather | ? |
| 18:06.12 | ModFather | i am using only ipv4 i dont want to use ipv6 |
| 18:09.05 | [TK]D-Fender | ModFatherthose settings are inside asterisk.conf <- NO. There is virtually no reason to be touching this file |
| 18:09.14 | [TK]D-Fender | again, this is SIP configs |
| 18:09.25 | [TK]D-Fender | Asterisk talks to a PILE of different techs. |
| 18:09.27 | ModFather | aha i understand so i need to force sip to use ipv4 |
| 18:09.34 | [TK]D-Fender | Each is configured in their own file |
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| 19:13.55 | jPadawan | Hello everybody |
| 19:15.23 | jPadawan | When i access FreePBX 2.11, it says that my asterisk service is not running, but when i ran "asterisk service status", "service asterisk restart" it seems to be ok. |
| 19:15.42 | WIMPy | #freepbx |
| 19:15.52 | jPadawan | thanks! WIMPy |
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| 20:17.46 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 20:32.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:38.39 | *** join/#asterisk talntid2 (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 20:38.43 | *** join/#asterisk ed_ (~zed_@82.179.199.146.dyn.plus.net) |
| 20:40.01 | ed_ | hello, can anyone point me to some tutorial for res_pjsip please? |
| 20:40.57 | ed_ | I have looked into wiki and got as far as "hello world" but that's about it |
| 20:41.17 | talntid2 | good luck, ed_, hah |
| 20:41.51 | talntid2 | but what are you trying to do? |
| 20:42.10 | WIMPy | Achieve world domination! |
| 20:43.05 | ed_ | talntid2: for now do simple call within pbx and maybe configure trunk and do 2 way call between 2 pbxes |
| 20:43.47 | ed_ | telntid2: nothing complicated, just struggle to translate chan_sip to pjsip |
| 20:44.45 | robmal | There was this thing called wiki on the asterisk page, did they remove it? |
| 20:44.48 | talntid2 | ed_, I am having similar issues, and have been in here for days searching, nobody seems to know much about it |
| 20:45.14 | talntid2 | ed_, although I can point you toward a page that does what you are looking for |
| 20:45.36 | talntid2 | https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 20:45.44 | talntid2 | that one uses realtime database though, which you may not want |
| 20:46.08 | ed_ | robmal: I already looked into wiki, but looks like I am too dumb |
| 20:47.07 | ed_ | talntid2: thanks, I wanted to start simple with baby steps, |
| 20:48.42 | ed_ | telntid2: doing multi master database sync is probably overkill for me this moment in time. I need to learn basics before diving deep |
| 20:49.28 | robmal | But... there're like examples there, you can copy them, you know, to test. |
| 20:49.48 | robmal | I think my cat could do that. |
| 20:50.31 | ed_ | tobmal: any idea why there would be multiple [sections] with same phone No as section name? |
| 20:51.01 | robmal | Sure. |
| 20:51.13 | robmal | Each serves a different purpose. |
| 20:52.00 | robmal | It looks like you know how to read, copy the examples and try changing shit. |
| 20:52.10 | robmal | If it stops working, try to figure out why. |
| 20:52.14 | robmal | Wiki will help. |
| 20:52.26 | ed_ | robmal: I learnt asterisk on 1.8 there was nothing like it there |
| 20:52.44 | robmal | I know. |
| 20:53.00 | robmal | Any reason you need pjsip over chan_sip? |
| 20:53.56 | ed_ | robmal: thought pjsip is preferred method to talk to end points, will need to learn it at some point so can do it now. |
| 20:54.14 | robmal | So learn. |
| 20:54.52 | ed_ | robmal: not trying to avoid the subject. just need some hints where to start with it |
| 20:55.15 | robmal | On the internet. |
| 20:55.39 | robmal | Wiki is a good starting point. |
| 20:56.55 | ed_ | robmal: tried wiki and doesn't feel any smarter :-( |
| 20:57.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:57.26 | ed_ | robmal: have you looked into pjsip yourself? |
| 20:57.41 | robmal | Who's forcing you to do this? Blink twice if you're safe. |
| 20:57.53 | robmal | Sure i have, hated it. |
| 20:57.57 | ed_ | robmal: what did you do? |
| 20:58.11 | robmal | I copied the examples and figured shit out along the way. |
| 20:59.18 | ed_ | robmal: I generated sample files, looked into res_pjsip.conf and can't say I understood much. |
| 20:59.56 | robmal | I don't know how to help. If you want me to do it - sure. Paypal is ok? |
| 21:02.00 | ed_ | robmal: though I will try to get my head around it first but in case I fail what is your rate? |
| 21:02.30 | robmal | Depends what do you want me to do ;-) |
| 21:02.43 | *** part/#asterisk kharwell (kharwell@nat/digium/x-vlybfmppuwlseyvn) |
| 21:03.05 | *** join/#asterisk kharwell (kharwell@nat/digium/x-oagmtzcmrwwbuszk) |
| 21:03.46 | ed_ | robmal: tutorial for pjsip :-) |
| 21:04.23 | robmal | I can send you a printed copy of asterisk wiki for $99,99 |
| 21:04.35 | robmal | Limited time offer. |
| 21:04.54 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:05.04 | ed_ | robmal: nice try |
| 21:05.31 | robmal | I can change the fonts and colors for an extra $9,99 |
| 21:06.35 | ed_ | robmal: so now its going to be colour copy? |
| 21:07.05 | robmal | Sure! |
| 21:08.43 | ed_ | robmal: do you actually use pjsip or you stuck with chan_sip for now? |
| 21:09.28 | robmal | I tried, didn't like it, went back. Except for multiple registrations for the same extension i don't see any benefits. |
| 21:10.21 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 21:10.48 | ed_ | robmal: was it too complicated or not reliable/underperform? |
| 21:11.00 | WIMPy | is still unsure if that's actually a good thing. |
| 21:12.40 | robmal | Nothing is too complicated if you mess with it long enough. I didn't see any downfalls in reliability or performance, cause hey, it's SIP, it just makes ends meet, no effect in rtp stream, so nobody will notice. |
| 21:14.56 | ed_ | robmal: it has been preferred method 2dn version in the row while chan_sip is on extended support. so pjsip is likely to stay |
| 21:15.13 | robmal | I have no trouble with that. |
| 21:16.18 | ed_ | robmal: ok lets start with the basics. why there are multiple [sections] for single phone No? |
| 21:16.24 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 21:16.46 | robmal | I'm too lazy to copy and paste the wiki here. |
| 21:18.46 | ed_ | robmal: ok, would you use multiple sections for single phone No in your configuration? |
| 21:19.10 | robmal | Because they forced me to. |
| 21:23.42 | *** join/#asterisk Cyphor (~Cyphor@69.9.239.74) |
| 21:23.44 | ed_ | robmal: is there just wiki or are there any blogs covering pjsip? |
| 21:24.01 | robmal | I'm pretty sure there are. |
| 21:24.54 | ed_ | can anyone point me to some blog explaining pjsip (preferably in simple terms) pretty please? |
| 21:25.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:25.21 | robmal | There's a company called google. They do that search shit quite well. |
| 21:26.26 | robmal | Seriously, what are you trying to achieve here? |
| 21:28.29 | ed_ | robmal: need to learn pjsip in baby steps. e.g. establish a call between 2 extensions on same pbx (nothing complicated really) |
| 21:29.53 | robmal | You've been here for an hour. I'm pretty sure you could've done that if you weren't chatting here. |
| 21:31.16 | ed_ | robmal: I hoped for eureka moment, and hint like "have a look there, they explained it pretty well" |
| 21:31.39 | robmal | 22:48:32 < robmal> There was this thing called wiki on the asterisk page, did they remove it? |
| 21:32.00 | robmal | have a look there, they explained it pretty well |
| 21:36.19 | *** join/#asterisk u0m3 (~u0m3@92.80.81.240) |
| 21:37.45 | ed_ | robmal: I already looked into wiki and got nowhere with it, as I said before way too dumb |
| 21:38.52 | robmal | So what are you expecting? You want to learn and say you're too dumb to do it. I don't think there's a cure for that. |
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| 21:41.46 | ed_ | robmal: I expected some help not insults. |
| 21:42.49 | robmal | I tried, but all you say is you don't understand how it works. |
| 21:43.03 | talntid2 | [Oct 14 21:40:59] ERROR[16328]: res_pjsip.c:2432 sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 'transport-udp-nat' for endpoint '102' |
| 21:43.18 | robmal | Now i'm trying to figure out why are you here. |
| 21:43.29 | talntid2 | anyone know what causes this? pjsip.conf has http://pastebin.com/eWkTK7W3 |
| 21:44.15 | robmal | talntid2: Paste ext 102 config as well. |
| 21:44.54 | talntid2 | its in realtime config, so here's the database line |
| 21:44.55 | talntid2 | http://puu.sh/kKrN0/343a7b37ba.png |
| 21:45.03 | talntid2 | everything else is null, except for the settings you can see in there |
| 21:45.07 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:46.32 | ModFather | get_defaults: Unable to find a valid server address or name. |
| 21:46.39 | ModFather | does anyone knows what cause this error? |
| 21:47.11 | ModFather | seems it doesnt get a valid server address or server name, but i dont see somewhere i can set a value for this |
| 21:47.56 | ed_ | modFather: are you refereeing to the server as its name or IP? |
| 21:48.19 | ModFather | either |
| 21:48.22 | ModFather | <PROTECTED> |
| 21:48.28 | ModFather | i get this on every core reload |
| 21:48.45 | talntid2 | that is in the phone provisioning config |
| 21:49.00 | talntid2 | phoneprov.conf |
| 21:49.11 | talntid2 | if you don't use auto provision, you can just disable the module |
| 21:50.01 | ModFather | ah yes i just saw it, really thanks |
| 21:50.25 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 21:51.29 | robmal | talntid2: I'll need sorcery.conf too |
| 21:51.38 | talntid2 | will do, thanks |
| 21:52.09 | talntid2 | sorecery.conf http://pastebin.com/ZxG14sVx |
| 21:55.10 | talntid2 | ohhh.. |
| 21:55.31 | talntid2 | because I have transport-udp and transport-udp-nat on the same bind addr, i suspect thats the problem |
| 21:55.38 | *** join/#asterisk Deeewayne (~Deeewayne@2600:1005:b169:c427:c8f1:c6c7:6e4c:b4e) |
| 21:55.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:56.05 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-sdengegirlxehsqz) |
| 21:56.05 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:57.36 | ModFather | talntid2, i have everything ( ; ) on sorcery.conf |
| 21:57.40 | *** join/#asterisk cmendes0101 (~cmendes01@198.199.206.2) |
| 21:57.40 | ModFather | but i still get this: sorcery.c:1376 sorcery_object_load: Type 'transport' is not reloadable, maintaining previous values |
| 21:57.57 | robmal | talntid2: A dumb question, did you reload res_pjsip module once you've set up this transport? |
| 21:58.42 | robmal | Or, in other words, does pjsip show transports list it? |
| 22:01.43 | ModFather | in my case i am not using pjsip |
| 22:02.42 | robmal | ModFather: Unload the module, add it as noload to modules.conf |
| 22:03.10 | ModFather | robmal, i am not sure for the name i should use on the module.conf as noload |
| 22:03.35 | robmal | res_pjsip |
| 22:03.40 | ModFather | noload res_pjsip ? |
| 22:03.47 | ModFather | without any extensions or something? |
| 22:03.49 | robmal | There are examples there. |
| 22:03.58 | ModFather | yes i saw them, they are with extension |
| 22:04.13 | talntid2 | yes, robmal, i have even done full asterisk restart |
| 22:05.05 | robmal | ModFather: That's a new one, modules.conf doesn't give a crap about extensions. |
| 22:05.40 | ModFather | i was meaning extension of the module .so for example noload res_pjsip.so |
| 22:06.02 | ModFather | because i've tried noload res_pjsip already |
| 22:06.04 | robmal | Ye, should be noload => res_pjsip.so |
| 22:06.12 | ModFather | thanks man ;) |
| 22:06.54 | robmal | talntid2: Maybe you're right about the bindaddr, did you try setting it to specific ip addresses? |
| 22:07.32 | ModFather | i thought if i disable the res_pjsip i wouldnt get again this error: sorcery.c:1376 sorcery_object_load: Type 'transport' is not reloadable, maintaining previous values |
| 22:07.33 | ModFather | :( |
| 22:08.10 | robmal | module show like pjsip and noload all of them. |
| 22:12.01 | ModFather | oh thats many in there |
| 22:12.02 | ModFather | hehe |
| 22:12.15 | robmal | Like 2? ;-) |
| 22:12.20 | ModFather | no man :) |
| 22:12.23 | ModFather | its like 50 |
| 22:12.34 | ModFather | 50 modules loaded |
| 22:12.59 | robmal | Oh, ye, they made everything a separate module. Just disable chan_pjsip, should be enough. |
| 22:13.21 | ModFather | http://pastebin.ca/3196997 |
| 22:13.55 | ModFather | hehe thanks robma1 , may i ask you something related to pjsip and sip ? |
| 22:14.13 | robmal | Sure. |
| 22:15.03 | ModFather | for what reasons someone will choose pjsip instead of sip? i see on many blogs talking about asterisk, they suggest pjsip instead of sip without explain the main reason for that |
| 22:17.12 | robmal | Because chan_sip got bulky over the years, nobody wants to maintain it and it'll soon be obsolete. So, like Henry Ford said 'if i asked people what they wanted, they'd ask for a faster horse', asterisk is moving to the next best thing, pjsip ;-) |
| 22:19.13 | talntid2 | i got it to work, robmal. thanks! |
| 22:19.23 | ModFather | robmal, sounds very solid |
| 22:19.35 | talntid2 | it was a combo of.... having two transports binding to the same thing, and.... not having rtp_symetric and force_rport enabled |
| 22:21.15 | robmal | talntid2: Thanks for the feedback, i'm the one who should be making notes ;-) |
| 22:22.47 | ModFather | robmal, i have my asterisk working just fine with sip if i change to pjsip and copy exact the content of sip.conf to pjsip.conf do i need to do extra configuration adjusts to make it work over pjsip? |
| 22:24.26 | robmal | There's a migration tool in the wiki, works great for 9 out of 10 users. You usually are the 10th, because 9 are asterisk employees ;-) |
| 22:25.13 | ModFather | hehehehehe |
| 22:29.30 | *** join/#asterisk ricksebak (~jwr@c-73-52-251-249.hsd1.pa.comcast.net) |
| 22:33.52 | ricksebak | does anyone know of a way to get monit to detect (and preferably fix) this condition: "chan_sip.c: Peer 'didlogic' is now UNREACHABLE!" |
| 22:34.16 | robmal | AMI? |
| 22:36.08 | ricksebak | robmal: i am using the AMI and would be fine with using that interface to fix this condition, but i have no clue how to do that (via bash or asterisk or anything else) |
| 22:38.57 | robmal | You're looking for PeerStatus |
| 22:44.53 | ricksebak | robmal: is there an asterisk command that will cause it to reconnect? |
| 22:45.07 | ricksebak | within the asterisk shell i mean, not in the AMI |
| 22:47.17 | robmal | sip reload? Depends if your asterisk is the peer initiating registration to a sip provider or you'd like i.e. the phone to re-register. |
| 22:49.39 | ricksebak | the unreachable condition is between my server and my sip trunk. so "sip reload" will cause it to attempt to reconnect? |
| 22:50.40 | robmal | Sure, but that's an ugly hack and you should resolve the problem in some other way. |
| 22:52.25 | WIMPy | There is no such thing as a connect. |
| 22:52.56 | WIMPy | If the peer goes unreachable, that's what it means. I.e. there is some networking issue. |
| 22:52.58 | robmal | WIMPy: Please don't go [TK]D-Fender now ;-) |
| 22:55.46 | WIMPy | I certainly won't ask for sip debug. that's noth what I want to read. |
| 22:56.00 | robmal | ;-) Thanks. |
| 22:56.32 | ModFather | robmal, maybe to you sounds easy but for me seems a huge issue, i purchased a number from twilio and with sip trunk setup on my asterisk, i can call outside, and get call to a specific extension. i want when someone calls our number, to ring on all available extensions, i have adjust queue config for that, but propably i am missing something to my sip.conf |
| 22:57.40 | ModFather | http://pastebin.ca/3197026 |
| 22:58.12 | robmal | Queues are boring, Dial(SIP/john&SIP/alice&SIP/paul) |
| 22:58.21 | ModFather | lol hehe thanks mate |
| 22:58.25 | *** join/#asterisk theron (~theron@173-28-183-190.client.mchsi.com) |
| 23:00.40 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:00.50 | ModFather | robmal, WARNING[30187][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 23:00.52 | *** part/#asterisk kharwell (kharwell@nat/digium/x-oagmtzcmrwwbuszk) |
| 23:01.52 | robmal | Meh, i'm not used to unregistered extenions. Show me your queue.conf |
| 23:07.29 | ModFather | robmal, http://pastebin.ca/3197033 |
| 23:10.01 | robmal | member => SIP/100 etc is enough. Now make exten => 1234,1,Queue(general) |
| 23:12.19 | ModFather | <PROTECTED> |
| 23:12.24 | ModFather | i think i miss a step |
| 23:13.34 | robmal | Meh, these pastebins disappear before i have a chance to look at them again ;-) There should be another [something] |
| 23:14.37 | ModFather | my bad for this, let me reupload it |
| 23:14.58 | ModFather | http://pastebin.ca/3197038 |
| 23:15.54 | ModFather | and inside my extensions.conf : exten => 5555,1,Queue(general) |
| 23:17.46 | robmal | Maybe general is (or should be imho) a reserved name, add [whatever] before strategy and correct the dialplan. |
| 23:19.21 | ModFather | i changed it to: general-queue |
| 23:19.25 | ModFather | and updated extensions.conf |
| 23:24.47 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 23:25.04 | ModFather | robmal, ;) you had right man, it was that, now no errors, but when i call my external number i am not hearing anything and it doesnt ring, with verbose 10 i see that: http://pastebin.ca/3197042 |
| 23:28.39 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 23:28.50 | robmal | Silent MOH default maybe? Try setting Queue(general-queue,r) |
| 23:30.55 | ModFather | ah robmal now i changed the SIP/100 to names like SIP/john and it worked it rings, but from the phone i am calling its silent |
| 23:30.58 | ModFather | going to test the ,r |
| 23:33.15 | ModFather | still nothing, |
| 23:33.43 | ModFather | from the phone i am calling the number i dont hear ringing or something, but when the member accept the call, he can hear me |
| 23:33.54 | ModFather | but i cannot, not even the tone on ringing |
| 23:34.11 | ModFather | the queue seems working like a charm except that issue |
| 23:35.27 | *** join/#asterisk cmendes0101 (~cmendes01@198.199.206.2) |
| 23:36.14 | robmal | K, so maybe that's a problem with the rtp stream? |
| 23:36.34 | robmal | Some nat along the way? |
| 23:37.08 | ModFather | hmm, if i do calls internal it works, if i call the member alex we can talk just fine |
| 23:37.27 | ModFather | also if i remove queue and set the incoming to arrive to my softphone, it works too |
| 23:39.06 | robmal | Damn, i'll have to [TK]D-Fender myself. |
| 23:39.18 | robmal | Paste the sip debug for a call that has one way audio. |
| 23:41.35 | robmal | Oh, pcap would be even better. |
| 23:47.57 | ModFather | robmal, http://pastebin.ca/3197058 |
| 23:48.27 | ModFather | i changed the incoming call to arrive to my softphone, i've answered the call and i was able to speak and hear just fine |
| 23:48.44 | ModFather | once i enable queue i cannot hear , but the caller can hear me just fine |
| 23:49.08 | ModFather | its kinda weird |
| 23:50.39 | ModFather | i will enable also queue again and give you debug logs too after a call that i cannot hear |
| 23:52.00 | robmal | pcap would be better, i'm a friend of wireshark ;-) |
| 23:54.44 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
