| 00:16.22 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
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| 01:34.21 | drmessano | Someone wrote Zaptel patches years ago to simulate some of the phreaking stuff |
| 01:34.58 | drmessano | I guess you would give it a 2600hz tone and get a DISA |
| 01:35.18 | drmessano | Stuff like that |
| 01:41.40 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 01:43.33 | WIMPy | LCR also has C5 support to test your old blue box. |
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| 02:39.29 | drmessano | Oh nice |
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| 05:01.45 | *** join/#asterisk pixelate (~pixelate@139.195.47.169) |
| 05:02.06 | pixelate | hey -- really quick question -- there's a standard name for a telephone number, but i can't at the moment seem to remember what it is |
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| 05:02.21 | pixelate | i want to say 'isdn' 'isi' or something like that, but that doesn't seem right |
| 05:02.36 | pixelate | anybody know what it is? |
| 05:03.24 | WIMPy | E.164 |
| 05:03.34 | pixelate | WIMPy: yeah, but there's a specific name? |
| 05:03.37 | pixelate | that's the standard |
| 05:04.46 | WIMPy | For GSM users it's called MSISDN. |
| 05:05.03 | pixelate | yeah |
| 05:05.09 | pixelate | that's what i was trying to remember |
| 05:05.10 | pixelate | thanks |
| 05:05.36 | WIMPy | But that'd GSM/UMTS specific. |
| 05:06.03 | pixelate | WIMPy: are there other names then for the other networks? or a generic nomenclature for 'phone number' other than 'phone number'? |
| 05:06.34 | pixelate | imsi ? |
| 05:06.39 | WIMPy | IDN |
| 05:06.51 | WIMPy | No, that's not a phone number. |
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| 05:08.06 | pixelate | WIMPy: do you mean x.121 then? |
| 05:08.32 | WIMPy | No. Something different again. |
| 05:08.40 | pixelate | what does 'idn' stand for then? |
| 05:09.07 | pixelate | my apologies, its been a couple years since i was building anything to do with the PSTN |
| 05:09.35 | pixelate | building something really quick, and just wanted to put the right name in for 'telephone' -- as 'tel' would be naive naming... |
| 05:09.38 | pixelate | may as well get the name right |
| 05:09.56 | WIMPy | Idividual Directory Number |
| 05:10.32 | WIMPy | Depending on direction calling/Called Party Number. |
| 05:13.22 | pixelate | thanks |
| 05:14.32 | WIMPy | What's the context? |
| 05:23.41 | snadge | lol.. just doing some further testing with failsterisk 13 ;) |
| 05:23.50 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 05:24.14 | snadge | now if i dial my mobile.. dst shows up in CDR(dst) as s.. but in the record .. its my mobile number |
| 05:35.04 | snadge | so .. clearly the exten => h function in asterisk13.. is horribly broken |
| 05:35.10 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 05:35.32 | snadge | and if thats indeed the case.. then unless there's some other way to do work after a call has hung up.. then we can't use it |
| 05:37.00 | *** join/#asterisk xnaron (~xnaron@S0106b4750e5de3b2.ed.shawcable.net) |
| 05:37.46 | xnaron | moved from asterisk 11 to 13 and video calls no longer work |
| 05:38.05 | xnaron | is there a known issue? |
| 05:39.48 | snadge | well, the guys are probably going to want a bit more info than that |
| 05:40.55 | xnaron | using x-lite to test...was working before the upgrade |
| 05:41.15 | xnaron | I can see 2016-01-24 22:39:34] VERBOSE[16478][C-00000006] netsock2.c: Using SIP VIDEO TOS bits 136 |
| 05:41.15 | xnaron | [2016-01-24 22:39:34] VERBOSE[16478][C-00000006] netsock2.c: Using SIP VIDEO CoS mark 6 |
| 05:41.39 | xnaron | in the logs |
| 05:41.52 | xnaron | checked the codecs are still enabled |
| 05:42.22 | xnaron | I made a call in the reverse direction and I can see distorted video |
| 05:42.35 | xnaron | like a techno color version |
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| 06:00.24 | snadge | im seeing strange cdr handling behaviour from asterisk 13.. but apparently this is to be expected, and by design ;) |
| 06:00.56 | snadge | in some cultures.. 13 is considered a very unlucky number... perhaps it should have been called, asterisk 12b |
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| 06:54.42 | snadge | we've been bought out by a larger provider that uses freeswitch.. so im probably not going to bother investing too much effort into fixing our "legacy" asterisk stuff anyway |
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| 08:00.51 | wdoekes | snadge: 12b/14 has been discussed.. and rejected |
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| 08:19.32 | snadge | i dont believe in superstition |
| 08:19.58 | snadge | how on earth im going to fix this cdr stuff though.. i have no idea.. it would be nice if others could reproduce the problem |
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| 08:46.09 | Aamit | <PROTECTED> |
| 08:46.29 | Aamit | I am using multiple Asterisk behind kamailio. |
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| 08:57.49 | snadge | hmm.. okay.. so because a mobile call gets passed to a macro |
| 08:58.00 | snadge | that means overwrite the dst variable |
| 09:05.07 | snadge | exten => _04XXXXXXXX,9,Set(server=${RAND(1,2)}) |
| 09:05.40 | snadge | exten => _04XXXXXXXX,10,Macro(server${server}) |
| 09:06.49 | snadge | macro-server1 and macro-server2 just dial two different servers in opposing order |
| 09:07.02 | snadge | but because they're like |
| 09:07.32 | snadge | exten => s,1,DIAL(SIP/someprovider_nn/61${MACRO_EXTEN:1}) |
| 09:07.51 | snadge | then CDR(dst) gets overwritten with s .. despite the correct value going into the call record.. :| |
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| 09:48.37 | snadge | ok so i replace Macro with gosub.. because macro is deprecated.. then when you call Dial from a gosub.. it doesn't return.. :| |
| 09:54.08 | wdoekes | snadge: a Dial which is answered does not return |
| 09:54.34 | wdoekes | if you want it to return to the dialplan, you should add the "continue-dialplan-after-this-dial-Dial-option" |
| 09:55.37 | snadge | well.. ideally i just want the CDR(dst) value to not change to an "s" when i call a macro |
| 09:55.41 | snadge | this behaviour has changed in asterisk 13 |
| 09:56.16 | snadge | which causes the agi script to screw up.. despite the correct dst being logged into the call record.. thats the fundamental problem.. im just going on a wild goose chase to find a workaround to that |
| 09:59.16 | snadge | gosub seems to work.. but then it wont trigger the hangup context |
| 09:59.52 | wdoekes | the hangup context is triggered where the call is ended, which could be inside the gosub context |
| 09:59.55 | wdoekes | but |
| 10:00.01 | wdoekes | you can use a global hangup handler in 13 |
| 10:01.16 | snadge | cool.. i'll try making the hangup handler global |
| 10:08.07 | snadge | i think this actually ties to my fundamental problem with asterisk 13.. we're using exten => h, and its doing unexplained/different stuff |
| 10:08.41 | snadge | so changing to use the hangup handler instead.. is probably the simple / least intrusive solution |
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| 10:35.56 | snadge | if i just remove the macro/gosub.. the problem goes away |
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| 10:51.17 | snadge | and now its not inserting the custom CDR fields in the cdr table.. asterisk 13 is just an impossibility |
| 10:52.35 | snadge | is there a way to verbose debug cdr? |
| 10:53.15 | wdoekes | core set debug 20 ? |
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| 11:04.24 | snadge | yeah.. need to look at what some of this means.. but it looks like its creating the cdr prior to the variables being set |
| 11:04.35 | kchehab | i cant get the sipcallid for the autocall file unless i set a micro exten => s,1,Dial( ,M(to) ,i am now getting the sipcallid when call is connected ,how can i get the callid if call failes s_BUSY or s_unavailable |
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| 11:48.05 | cargill | hi, does anyone know why asterisk is gone in debian stable? most of the infrastructure says that 11.13 is there but the Packages file disagrees |
| 12:07.17 | wdoekes | gone? I thought jessie shipped with asterisk 13? |
| 12:08.39 | wdoekes | apparently not, but according to my jessie box here, it would install 1:11.13.1~dfsg-2+b1 |
| 12:09.25 | cargill | wdoekes: does it still say that after an apt-get update? |
| 12:10.19 | wdoekes | yup.. although my apt cache url may be a couple of hours old |
| 12:13.15 | wdoekes | http://ftp.nl.debian.org/debian/dists/Debian8.3/main/binary-amd64/ <-- has it |
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| 12:24.54 | kchehab | any hint |
| 12:27.32 | snadge | i wonder if i need to update phpagi for asterisk 13 ? |
| 12:32.43 | cargill | that's weird, I just installed a jessie host and can't get aptitude to see asterisk as an available package |
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| 12:35.25 | snadge | Section: comm |
| 12:35.25 | snadge | Priority: optional |
| 12:35.25 | snadge | Filename: pool/main/a/asterisk/asterisk_11.13.1~dfsg-2+b1_amd64.deb |
| 12:36.53 | snadge | you have main and contrib enabled? |
| 12:37.34 | cargill | ah, found it, I didn't actually install jessie, but stretch, duh |
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| 13:02.05 | mccar | Hi there! Is anybody may to help with GOIP8? |
| 13:06.41 | mccar | I'm using VoIP call forwarding to asterisk, and everything goes well, except I see random CID on my VoIP phone (all calls forwards to one extension). I have no idea why, maybe because all peers have the same IP. I need a way to identify on which number in GOIP I have an incoming call. |
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| 13:16.42 | *** join/#asterisk vlt (~nobody@lvps87-230-93-209.dedicated.hosteurope.de) |
| 13:17.19 | vlt | Hello. Is there an easy way to solve a "No RTP ports remaining. Can't setup media stream for this call." problem w/o interrupting current calls? |
| 13:18.28 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 13:19.45 | mccar | Did you try extend port range and reload SIP module? |
| 13:19.52 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:21.44 | vlt | Iâll try that, thanks. |
| 13:23.42 | vlt | mccar: Which conf file do I have to edit? |
| 13:25.09 | vlt | founf rtp.conf |
| 13:25.11 | vlt | *found |
| 13:26.44 | mccar | rtp.conf |
| 13:26.59 | vlt | `sip reload` wasnât enough but `reload` did it. Thanks! |
| 13:27.40 | mccar | You're welcome) |
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| 14:06.33 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:09.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:09.50 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:14.50 | [TK]D-Fender | mccar: Peer matches by IP, and the first hit is what will get used. Unless your gateway is sending a unique username you'll have to find it in a header or something. |
| 14:14.55 | [TK]D-Fender | mccar: Go look at the actual calls |
| 14:19.54 | mccar | In INVITE from GOIP to asterisk i see FROM:79992343322 TO:527 |
| 14:20.46 | mccar | from - extension on asterisk where GOIP is registered and TO - number to call on asterisk |
| 14:21.41 | mccar | But in INVITE from asterisk to peer 527 I see FROM:(random extension like 79992343322) TO:527 |
| 14:22.46 | mccar | So first INVITE is correct, but when asterisk calls to local 527 number CID is random... |
| 14:23.35 | [TK]D-Fender | PASTEBIN it |
| 14:27.12 | mccar | How to save it in convenient form? I use wireshark... |
| 14:30.06 | mccar | Found. Just a minute |
| 14:31.48 | [TK]D-Fender | asterisk SIP debug, nothing else |
| 14:31.53 | mccar | http://pastebin.com/eVVNiXDB |
| 14:33.20 | [TK]D-Fender | The complete call... |
| 14:33.23 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 14:38.27 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 14:40.25 | stefan27 | VoiceMail() according to 'core show application VoiceMail' should exit if it receives a '0' dtmf, but it doesn't for me. Is there perhaps a switch in voicemail.conf which turns this feature off/on that I cannot find? |
| 14:41.16 | Chainsaw | stefan27: And DTMF is being obeyed for other voicemail commands? |
| 14:42.04 | stefan27 | not sure, I only use the VoiceMail(14,su) command |
| 14:42.27 | Chainsaw | stefan27: I'm wondering if it's not just complete inability to receive DTMF. |
| 14:44.09 | [TK]D-Fender | 0 will exit .... if it has somewhere to exit TO |
| 14:44.39 | [TK]D-Fender | Did you also tell it it was allowed to do this in the VM box definition line? |
| 14:50.01 | stefan27 | I use realtime mysql and I can't find a column in the db that looks like it's related to whether or not it accepts dtmf exits, I did try putting a exten => 'o',1,Hangup within same context as the VoiceMail call. |
| 14:50.26 | *** join/#asterisk mjordan (mjordan@nat/digium/x-wrnsqfsurhnejihi) |
| 14:50.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:51.04 | [TK]D-Fender | There is a very clear conf option for whether a specific box is allowed tto use that or not |
| 14:51.06 | stefan27 | the console logger does show that DTMF '0' + '0 end' is received on the sip channel that executes VoiceMail |
| 14:51.11 | mccar | [TK]D-Fender: Full log here http://pastebin.com/SH7KRiUG |
| 14:51.47 | *** join/#asterisk captain118 (~Adium@132.sub-70-193-65.myvzw.com) |
| 14:52.20 | [TK]D-Fender | mccar: So we see a "to:" in there, that looks like the username you gave it |
| 14:53.54 | mccar | [TK]D-Fender: "to:" is a destination peer, not a user name |
| 14:54.09 | mccar | [TK]D-Fender: username in "from:" |
| 14:54.45 | [TK]D-Fender | mccar: that from: should be the callerid from the incoming call |
| 14:56.30 | mccar | [TK]D-Fender: GOIP has 8 port for SIM and each SIM has a connection to asterisk through username/password |
| 14:57.06 | [TK]D-Fender | Then you should be setting these up as type=user |
| 14:57.27 | *** join/#asterisk anonymouz666 (bd192dc9@gateway/web/freenode/ip.189.25.45.201) |
| 14:57.31 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-jrshzdurfkpmtrrx) |
| 14:57.45 | [TK]D-Fender | since type=peer is matched only on IP |
| 14:57.59 | [TK]D-Fender | And the first one it lands on is all you're going to get |
| 14:58.05 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 15:00.50 | mccar | [TK]D-Fender: I've changed and now asterisk response that password is wrong! |
| 15:01.46 | [TK]D-Fender | Show us the call, the config on *, and on the device itself |
| 15:02.45 | mccar | I gave you a call... |
| 15:03.15 | [TK]D-Fender | a new one that FAILS |
| 15:03.25 | [TK]D-Fender | And the configs from both sides |
| 15:05.39 | anonymouz666 | when you look closer to pjsip, it's very hard to think about chan_sip again. |
| 15:06.10 | mccar | [TK]D-Fender: How can I show you a call when I get SIP/2.0 403 Forbidden? |
| 15:06.23 | [TK]D-Fender | PASTEBIN <- |
| 15:06.29 | mccar | Sure |
| 15:07.10 | mccar | [TK]D-Fender: http://pastebin.com/fHuPHzvM |
| 15:07.55 | [TK]D-Fender | You should set your gateway to NOT register, and just set the IP's in each |
| 15:09.39 | mccar | I didnt get you... |
| 15:09.58 | mccar | gateway is a GOIP? |
| 15:11.49 | mccar | [TK]D-Fender: peer config http://pastebin.com/WLns0vY4 |
| 15:13.07 | [TK]D-Fender | mccargateway is a GOIP? <- yew |
| 15:13.09 | [TK]D-Fender | yes |
| 15:13.22 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 15:16.41 | *** join/#asterisk kharwell (kharwell@nat/digium/x-knxmetqzpygtwqji) |
| 15:19.27 | mccar | [TK]D-Fender: GOIP settings http://pastebin.com/48t935b9 |
| 15:19.54 | [TK]D-Fender | Registrar server 192.168.100.250 |
| 15:19.59 | [TK]D-Fender | don't tell it to register |
| 15:20.06 | [TK]D-Fender | And then show us a call |
| 15:22.13 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-xmftdtppwbzdczeo) |
| 15:24.24 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 15:25.08 | catphish | if you specify multiple destinations to Dial(), does it create a channel for each destination, then bridge the one that answers, destroying the others? |
| 15:26.35 | xnaron | I upgraded from asterisk 11 to 13 and am having an issue with video calls. Previously working with C-lite softphone and h.263/h.263+ codec. With asterisk 13 calls establish but the video does not come through or comes through garbled. Is there any known issues with asterisk 13 and h.263/+ or x-lite soft phone? |
| 15:26.37 | mccar | [TK]D-Fender: There is no way to prevent registration. I tried to clear password fields and registrar. |
| 15:27.28 | [TK]D-Fender | Then change to type=friend. |
| 15:28.17 | stefan27 | xnaron which minor 13.? version of asterisk? |
| 15:31.01 | xnaron | Asterisk 13.7.0 |
| 15:33.23 | mccar | [TK]D-Fender: http://pastebin.com/WqTdcSnk |
| 15:34.14 | [TK]D-Fender | To: <sip:102@192.168.100.250> |
| 15:34.16 | mccar | [TK]D-Fender: Nothing changed, I see on my phone all peers name randomly. 3 calls - 3 different CID |
| 15:34.19 | stefan27 | okay then I don't know. There were a few video issues that were resolved just before 13.7.0; and I never used asterisk11 myself and I have only tried clients that use vp8 or h264 and they work for me |
| 15:34.21 | [TK]D-Fender | Where is it getting that 102 from? |
| 15:35.25 | mccar | [TK]D-Fender: Here is it INVITE sip:102@192.168.2.51:5068 SIP/2.0 |
| 15:35.31 | [TK]D-Fender | I see that. |
| 15:35.36 | [TK]D-Fender | I asked where the 10 comes from |
| 15:36.27 | mccar | Its my local number... I want GOIP to forward mobile calls to my local number 102.. |
| 15:36.39 | xnaron | are there any free soft phone clients that use h.264? x-lite does not |
| 15:37.14 | mccar | Maybe I didnt get your question... |
| 15:37.56 | stefan27 | I dont think so, but Bria wasnt so expensive |
| 15:38.47 | stefan27 | you could try linphone (vp8) ? |
| 15:39.53 | [TK]D-Fender | Looking for 102 in from-internal (domain 192.168.100.250) |
| 15:39.57 | [TK]D-Fender | It's going to 102..... |
| 15:40.50 | mccar | [TK]D-Fender: Its right. |
| 15:41.14 | mccar | [TK]D-Fender: Oh, sorry... I changed destination from 527 to 102.. |
| 15:41.59 | mccar | [TK]D-Fender: Previously you see 527 in debug |
| 15:43.33 | [TK]D-Fender | I don't see this |
| 15:43.41 | [TK]D-Fender | I also don't see why it should be either number |
| 15:43.49 | [TK]D-Fender | I was shown no settings for any of that |
| 15:48.18 | mccar | [TK]D-Fender: Sorry, Its another one param in settings from another section of GOIP: "Forwarding to VoIP number" |
| 15:52.19 | [TK]D-Fender | Are you using the same secret on all ports in your * peers? |
| 15:52.34 | mccar | Yes, one for all |
| 15:53.17 | [TK]D-Fender | well * can't tell the difference between your lines |
| 15:53.46 | [TK]D-Fender | And I can't go through this one tiny piece at a time |
| 15:53.57 | [TK]D-Fender | maybe someone will |
| 15:54.49 | mccar | I see |
| 15:54.52 | mccar | Thank you |
| 15:58.52 | mccar | [TK]D-Fender: One more question, please |
| 16:00.02 | mccar | [TK]D-Fender: May I make a 8 new peers and forward calls from these peers to my local one and set new CID? |
| 16:01.03 | [TK]D-Fender | You don't normally get tto set CID on cell networks |
| 16:02.07 | *** join/#asterisk hfb (~hfb@96.247.48.232) |
| 16:02.55 | WIMPy | Considering the compressed format of the signalling, I doubt that it's even possible. |
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| 16:33.12 | danielyk | Hey there, I have a question to PJSIP. Is there someone can help me? |
| 16:33.19 | [TK]D-Fender | Depends on the question |
| 16:33.24 | [TK]D-Fender | Never ask to ask. Just ask it. |
| 16:33.35 | danielyk | Okay, sorry. this is my first time. |
| 16:35.14 | danielyk | I have an Asterisk 13 with can_sip running. My ISP gives me for a reconnection a new IP address. Now I want to migrate to PJSIP. PJSIP works very well, until the IP address changes. I can't register my trunks any more. If I shutdown Asterisk and wait for example 30 Minutes (PJSIP Expiration: 480 sec), Asterisk is able again register successfully. |
| 16:36.51 | [TK]D-Fender | Should look to force a higher reg frequency |
| 16:37.31 | danielyk | 480 sec works with chan_sip without problems. |
| 16:37.46 | danielyk | res_pjsip_outbound_registration.c: Temporal response '500' received from 'sip:tel.t-online.de:5060' on registration attempt to 'sip:phonenumber@tel.t-online.de:5060', retrying in '60' res_pjsip_outbound_registration.c: Fatal response '403' received from 'sip:tel.t-online.de:5060' on registration attempt to 'sip:phonenumber@tel.t-online.de:5060', stopping outbound registration |
| 16:38.28 | danielyk | Those are the answers after the IP address change. |
| 16:38.48 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/pjsip.conf.sample?revision=433338&view=markup |
| 16:38.49 | [TK]D-Fender | 882 |
| 16:39.28 | [TK]D-Fender | And make sure your enternal IP is getting re-evaluated frequently. |
| 16:40.57 | danielyk | I have not set any external IP in Asterisk |
| 16:41.14 | [TK]D-Fender | is your server NAT'd? |
| 16:41.20 | danielyk | yes |
| 16:41.35 | [TK]D-Fender | Then you have skipped a critical step |
| 16:41.43 | [TK]D-Fender | * needs to know what your WAN IP is. |
| 16:42.53 | danielyk | Okay, my IP Address changes if my router re-connect to the ISP. How can I set automatically the WAN IP? |
| 16:43.24 | danielyk | With sipgate it works without WAN IP without any problems... |
| 16:44.06 | [TK]D-Fender | They are probably ignoring the IP you are telling them to contact you at. |
| 16:44.16 | [TK]D-Fender | Providers shouldn't be doing that |
| 16:45.02 | [TK]D-Fender | There are clear settings in the Transport for what subnets are considered local, and what to use as, or how to find, your WAN IP. |
| 16:45.10 | danielyk | hmm, i don't know. It works with my provider without problems till the ip address changes... I would say, they ignore it too... |
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| 16:45.46 | [TK]D-Fender | Do the job right |
| 16:45.56 | danielyk | okay, yes |
| 16:46.05 | danielyk | How should I update the IP Address? |
| 16:46.27 | danielyk | With chan_sip it was so easy... |
| 16:52.59 | danielyk | I have with that not so much experience... |
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| 16:59.26 | danielyk | Would that be useful (adapted to PJSIP)? http://lists.digium.com/pipermail/asterisk-users/2012-February/270057.html |
| 17:03.29 | [TK]D-Fender | reinventing the wheel |
| 17:03.45 | [TK]D-Fender | There was already 'externrefresh" for chan_sip |
| 17:03.50 | [TK]D-Fender | this looks pointless |
| 17:04.04 | [TK]D-Fender | And the equivalent exists in chan_pjsip |
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| 17:08.32 | danielyk | I can't find something equivalent for chan_pjsip |
| 17:10.26 | danielyk | I think pjsip can't work with external domain names (e.g. dyndns), so there can't be an externrefresh equivalent? |
| 17:11.23 | danielyk | I can only set an external ip, so I must update this IP-Address with an script. Or have I got that wrong? |
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| 17:14.08 | [TK]D-Fender | hrm... |
| 17:14.14 | [TK]D-Fender | not seeing it in the sample myself... |
| 17:16.18 | danielyk | I must update with an script external_media_address and external_signaling_address |
| 17:16.49 | [TK]D-Fender | I would doubt that. |
| 17:17.09 | [TK]D-Fender | Downloading the lastest stabe to see if the samples are better than the more dated SVN I've been referencing |
| 17:17.45 | danielyk | I will do that, one moment. |
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| 17:21.55 | danielyk | I can only set an domain, but I think that is not what I'm looking for... |
| 17:22.26 | [TK]D-Fender | It's half of the picture |
| 17:22.36 | [TK]D-Fender | the other is a refresh rate to see if it changes |
| 17:24.53 | danielyk | Joshua Colp: "PJSIP does not currently have functionality to automatically update the public address (externhost / externrefresh)" |
| 17:26.02 | [TK]D-Fender | ... |
| 17:26.10 | [TK]D-Fender | OMFG |
| 17:26.47 | danielyk | I think I must write an script... |
| 17:26.49 | [TK]D-Fender | file: WTF? |
| 17:27.01 | [TK]D-Fender | :| |
| 17:27.07 | file | it doesn't have those options |
| 17:27.34 | [TK]D-Fender | That looks pretty bad.... |
| 17:27.52 | danielyk | Yeah right... |
| 17:27.54 | [TK]D-Fender | What's the suggested workaround? |
| 17:28.13 | file | I don't have one presently |
| 17:29.29 | [TK]D-Fender | danielyk: Looking like a modded script is in order (at best), or switching to chan_sip until something usable arrives |
| 17:29.48 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:29.51 | danielyk | If I update my IP in pjsip.transport.conf, does an "core reload" sufficient? |
| 17:30.37 | file | transports are marked as being unable to reload due to PJSIP restrictions |
| 17:31.11 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 17:31.25 | file | as chan_pjsip does not have the functionality you need, then you'll need to fallback on chan_sip |
| 17:32.53 | [TK]D-Fender | file: How big a challenge do you believe it should be for it to be added? |
| 17:33.07 | file | shouldn't be THAT bad |
| 17:35.00 | danielyk | I use chan_sip already, but my ISP is signaling calls from 217.0.0.0/13. The implementation of this wide range ip subnet is with pjsip so easy... |
| 17:35.46 | [TK]D-Fender | danielyk: Agreed, PJSIP's modularization is a great idea I proposed before I knew of it as a library (or much else) |
| 17:37.05 | danielyk | Right. I had to create a separate peer for any incoming IP address ... |
| 17:38.35 | danielyk | And that's really ugly ... |
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| 17:44.35 | anonymouz666 | file: pjsip could acct like a PUA? |
| 17:44.46 | file | I don't understand the question |
| 17:45.00 | anonymouz666 | the sip stack sending the subscribe's |
| 17:45.07 | anonymouz666 | like an user agent |
| 17:45.43 | file | the functionality is there to add modules to do so... yes |
| 17:47.02 | anonymouz666 | the goal is to integrate the presence state with skype for business through SIP |
| 17:48.07 | anonymouz666 | I don't know if Skype for Business can send PUBLISH to us, but if we can subscribe, we could reach the same result |
| 17:48.17 | anonymouz666 | that is what I expect but I might be wrong |
| 17:48.34 | file | you'd have to write code. |
| 17:49.46 | anonymouz666 | to solve specifically what part? |
| 17:50.24 | file | there is no module to do the outbound SUBSCRIBE or to handle the incoming NOTIFY for it |
| 17:50.32 | file | same for PUBLISH of other types |
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| 17:57.13 | thiagoc | I'm receiving messages from "Message/ast_msg_queue" on CLI, how can I disable it? |
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| 18:04.28 | anonymouz666 | file: what about outbound-publication and inbound publications types? |
| 18:04.42 | file | the only supported one is an Asterisk specific JSON based one |
| 18:08.36 | anonymouz666 | so that answers redirect me to the sip proxy solution |
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| 18:44.24 | anonymouz666 | "AWS launches tool that provides SSL and TLS certificates for free" |
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| 19:45.00 | josefig | someone has used sendmycall as a provider ? |
| 19:50.22 | [TK]D-Fender | Look like another fly-by-night little company |
| 19:50.31 | [TK]D-Fender | they are not a name I've ever heard mentioned in here |
| 19:54.43 | josefig | [TK]D-Fender: I need help to accept incoming calls from them but I've looked for information regarding that, i setup something but when I try to call the DID and wait for the incoming call i dont see nothing and on the provider's logs say Request Timeout |
| 19:55.19 | [TK]D-Fender | If you don't see packets.. then there are no packets getting that far |
| 19:55.31 | [TK]D-Fender | Is this with SIP DEBUG enabled at * CLI? |
| 19:55.40 | josefig | yes |
| 19:55.49 | josefig | sip set debug on |
| 19:56.09 | [TK]D-Fender | And using chan_sip? |
| 19:56.42 | josefig | yes, i have a provider to make my outbound calls and is working perfectly just not the DID yet |
| 19:57.00 | [TK]D-Fender | Are you supposed to register to them? |
| 19:57.07 | [TK]D-Fender | go look att that first if so |
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| 20:01.06 | josefig | [TK]D-Fender: no, I needed to create the SIP URI at their PBX side in order to accept at our side the incoming calls, how it should be the sip uri ? DID@IP-from-our-server ? |
| 20:02.22 | [TK]D-Fender | yes |
| 20:02.34 | [TK]D-Fender | And you'll need forwarding/firewalls open for this |
| 20:04.07 | josefig | [TK]D-Fender: I already opened the port and the incoming IPs |
| 20:04.30 | [TK]D-Fender | Then you should see traffic art the border (if you have a decent router) |
| 20:04.34 | [TK]D-Fender | at* |
| 20:04.50 | josefig | but about the SIP URI how is a valid format? DID@IP should be a normal right? |
| 20:04.56 | josefig | Ah cool, support is coming online to help us |
| 20:05.00 | josefig | their support |
| 20:06.02 | [TK]D-Fender | generally enough. depends if they expect anything extra on their side like a prefix "sip:" or a port on the end, etc. That is something they should have documented |
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| 20:32.59 | danielyk | I have a new problem (chan_sip). After the first re-invite (comming from the provider), all incoming and outgoing RTP is interrupted. But the phone call still exist, every 15 minutes comes an re-invite. I only cannot hear or say something. |
| 20:33.39 | *** part/#asterisk cargill (~ondra@host86-182-2-170.range86-182.btcentralplus.com) |
| 20:34.06 | danielyk | I have already "session-timers: refuse", because at the beginning all calls ended after 15/30 minutes... |
| 20:35.19 | [TK]D-Fender | It's advised to not allow reinvites |
| 20:36.31 | danielyk | What does it mean exactly? |
| 20:37.13 | [TK]D-Fender | Don't let your endpoints attempt to rebridge. |
| 20:37.18 | [TK]D-Fender | Always force it through the PBX |
| 20:39.56 | danielyk | session-timers: refuse is for the trunk, not for the endpoints. All endpoints are connected via chan_pjsip. |
| 20:45.05 | danielyk | I don't think that this is an session-timers problem if the RTP Stream is lost... |
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| 20:57.13 | josefig | how can I see the invites from the voip provider? |
| 21:03.42 | danielyk | sip set debug on |
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| 21:22.41 | danielyk | @[TK]D-Fender: I have "session-timers: refused" removed - the problem is still there. |
| 21:22.52 | [TK]D-Fender | Not what I was saying |
| 21:23.38 | danielyk | [TK]D-Fender: What I have to do? |
| 21:30.10 | danielyk | [TK]D-Fender: Do you mean "session-timers = originate"? |
| 21:30.24 | [TK]D-Fender | I didn't not mention session timers at all. |
| 21:30.38 | [TK]D-Fender | I said not to allow reinvites at all and force everything through the server |
| 21:34.20 | danielyk | [TK]D-Fender: I think we have understood wrong here. Canreinvite is set to no. I meant with re-invites the re-invites because of elapsed session-timers... |
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| 21:52.50 | danielyk | @file: Have you got an idea, why my RTP Stream is lost after 15 minutes? Problem came up today... |
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| 22:00.46 | file | not really |
| 22:01.30 | danielyk | @file: Maybe it's an problem by Telekom Germany... |
| 22:02.54 | wdoekes | danielyk: if both the incoming *and* outgoing rtp is interrupted, then there must be something in the sdp |
| 22:03.36 | danielyk | wdoekes: Yes, both are affected |
| 22:04.14 | wdoekes | asterisk won't stop sending just because it stops receiving, so the session-reinvite triggers some kind of "stop" |
| 22:04.51 | wdoekes | *gone*, batt. low, and it's time anyway. good luck |
| 22:05.28 | danielyk | Asterisk is sending, but the callee hear nothing... |
| 22:06.00 | danielyk | Thanks, bye |
| 22:06.06 | WIMPy | Have you read through the whole SIP dialogs? |
| 22:06.42 | danielyk | Yes, they show normal... |
| 22:08.31 | danielyk | Immediately after the provider sends ACK for the re-invite I can not hear anything. |
| 22:09.02 | WIMPy | Did you also check the RTP streams? |
| 22:10.04 | WIMPy | once had a user with a FritzBox where the calls would disconnect after 5, 10 or 15 minutes. |
| 22:10.36 | WIMPy | Funnily enough someone else with the same model and even the same firmware never had that issue. |
| 22:13.01 | danielyk | I see in the SDPs, that Telekom's RTP is at port 64544 and my own at 18296. I think all is okay... |
| 22:14.03 | danielyk | I had the problem with disconnection too, here helped "session-timers=refuse"... |
| 22:16.19 | danielyk | And asterisk does also sent RTP packets to telekom: Sent RTP packet to 217.0.5.69:64544 (type 08, seq 046953, ts 11553280, len 000160) |
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