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| 05:28.33 | iheartlinux | I put delete=yes in voicemail.conf, but they aren't deleting from voicemailmain. any ideas? |
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| 09:33.27 | clive- | Hi guys. I am still using asterisk version 1.8.23 and considering upgrading. Which version is recommended for stable production use ? |
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| 09:42.25 | file | the wiki has a page, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions which lists versions and when they are supported to |
| 09:42.50 | file | 11 goes security only in a few months, 13 still has 2+ years |
| 09:59.17 | clive- | thanks file |
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| 11:29.56 | file | la la la Wednesday la la la |
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| 12:28.08 | leon_ | Hi :). would you please let me know how to control sdp in asterisk ? when I tried to call to endpoint, the rtp channel is always open video stream with audio. But I dont want to open video stream, and I'm using pjsip channel driver. Can I control this in asterisk with pjsip ? |
| 12:29.31 | leon_ | my asterisk server version is 13.9.1 |
| 12:29.44 | file | how have you configured the endpoint? |
| 12:30.07 | leon_ | I don't know what variable control this. |
| 12:30.23 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:32.46 | leon_ | it's configured in ps_endpoint database table. id, transportm, aors, auth,context, codec setup(disallow, allow), network control(ice_support = no, force_rport=yes) , rewrite_contact=yes, rtp_symmetric=yes. and all the others configured null. |
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| 12:33.09 | file | and what is the configuration? |
| 12:34.00 | leon_ | ??? |
| 12:34.31 | file | what is the configuration for the endpoint? PJSIP will only offer video if you've configured it with video codecs in the allow section |
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| 12:37.41 | leon_ | oh I'm sorry. It works video and audio. can i switch this ? I meant video+audio call and audio only call. |
| 12:38.18 | leon_ | allowed : ulaw,alaw,g722,g729,gsm,speex,opus,h264,h263,h263p,vp8,mpeg4 |
| 12:38.29 | file | you can't control what codecs are used when dialing, except by creating a second endpoint with a limited set of codecs |
| 12:40.00 | leon_ | is it only way to control this ? |
| 12:40.11 | file | actually you may be able to use PJSIP_MEDIA_OFFER - https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_MEDIA_OFFER - but I haven't used it so I can't comment on a working way with it |
| 12:40.39 | leon_ | thank you :) I will check it. |
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| 12:56.53 | stefan27 | in chan_sip you can set dialplan variable SIP_CODEC_OUTBOUND=ulaw just before dialing a sip peer X, in which case it should not offer alaw even if X has both allow ulaw and allow alaw |
| 12:57.05 | stefan27 | PJSIP has something similar? |
| 12:57.16 | file | yes, PJSIP_MEDIA_OFFER |
| 13:00.54 | slima | Is that have any sense? exten => _00ZX.,1,GotoIf($[${REGEX("2433712[0-4][0-9]" $CALLERID(num)})}]?ok:notok) |
| 13:08.16 | [TK]D-Fender | $CALLERID(num)} <----- |
| 13:08.19 | [TK]D-Fender | missin { |
| 13:09.50 | slima | oh, thx a lot! |
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| 13:21.56 | stefan27 | if i recall correctly the behavior of the SIP_CODEC_INBOUND variable was not very intuitive... for example if you receieve an invite from a peer X which in chan_sip.c is configured to have allow:ulaw,vp8 then if you set SIP_CODEC_INBOUND=ulaw,vp8 you might expect nothing to happen |
| 13:22.33 | stefan27 | but if X offers only ulaw and you set SIP_CODEC_INBOUND=ulaw,vp8 then X can not reinvite with vp8 |
| 13:23.33 | stefan27 | so setting SIP_CODEC_INBOUND=<list1> before answering a call from a peer X which has allow=<list2> then X is limited to the intersection of list1 and list2 for the rest of the call |
| 13:25.37 | stefan27 | I meant he is limited to the intersection of list1 and list2 and the actual codecs that were offered in the first invite |
| 13:26.04 | stefan27 | maybe PJSIP_MEDIA_OFFER works differently |
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| 13:29.53 | stefan27 | in short, setting any of the chan_SIP SIP_CODEC* variables prevents reinviting with more codecs |
| 13:30.12 | stefan27 | is that a bug maybe? |
| 13:31.00 | leon_ | can I get the information about endpoint status in dialplan without AGI ? |
| 13:31.08 | file | you can't add codecs to an answer, you can only respond with what they offer or a subset |
| 13:31.25 | file | it's not an Asterisk limitation - it's SDP itself |
| 13:33.25 | stefan27 | I'm not talking about adding codecs to the answer sent to the offerer X, I'm talking about asterisk rejecting codecs in the second offer coming from X |
| 13:34.08 | file | oh, yeah, I'm not surprised |
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| 14:22.49 | jeffspeff | looking at https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial is the ip or hostname required for the client certificate? just wondering about the use case of phones having a dynamic public IP. |
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| 15:10.33 | polysics | hello! is there a way to list installed modules an versions, please? |
| 15:11.14 | polysics | I am trying to figure out what exactly is installed on an Asterisk box I have |
| 15:11.33 | Samot | module show |
| 15:11.38 | Samot | from the asterisk cli |
| 15:12.39 | polysics | that does not have versions, though |
| 15:14.15 | [TK]D-Fender | Modules don't have independent version #'s |
| 15:15.20 | [TK]D-Fender | If you installed it you should have the source or packages handy you did it from |
| 15:15.28 | [TK]D-Fender | outside of that * should show you on connect tto CLI |
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| 15:19.51 | polysics | [TK]D-Fender: the issue here is that the source folder contains 3 versions of the same thing |
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| 15:20.13 | polysics | I tried to reinstall using a few combinations but did not get anything working |
| 15:24.51 | [TK]D-Fender | At this point then you have whatever you just built since you've done it so recently. You should know the last one you did. |
| 15:25.07 | [TK]D-Fender | "few combinations" really tells us nothing of course |
| 15:25.22 | [TK]D-Fender | And no details on why this would be a combination effort in the first place. |
| 15:25.32 | [TK]D-Fender | Make sure your recompiles are CLEAN |
| 15:25.45 | polysics | I am installing Asterisk + UniMRCP + UniMRCP plugin |
| 15:25.59 | polysics | has anyone done that recently? |
| 15:26.36 | polysics | most combinations of versions i have tried result in some compile error, while installing everything using the "all in one" package UniMRCP provides results in a segfaulting asterisk |
| 15:26.40 | [TK]D-Fender | Never heard of it |
| 15:26.46 | [TK]D-Fender | Get * working by itself first |
| 15:26.52 | polysics | it's a TTS/ASR interface |
| 15:27.07 | polysics | Asterisk itself compiles and works, it has never not worked for me :D |
| 15:27.08 | [TK]D-Fender | If you have a problem with some all-in-one installer they offer, then you'd need to ttake it up with them |
| 15:27.48 | polysics | I guess my real problem is that I am not sure what the problem is :) |
| 15:29.21 | [TK]D-Fender | Time to find a clue |
| 15:29.38 | [TK]D-Fender | You don't know what packages you're even working with and have no details for us. |
| 15:29.45 | [TK]D-Fender | And lay off the doobie :p |
| 15:34.19 | qakhan | hi all, i have few exts which are being used in 1 office over NAT. i am trying to restrict them dont configure except 1 IP address. |
| 15:34.36 | qakhan | here is my config of ext http://pastebin.com/Xhuy31A4 |
| 15:35.20 | qakhan | i am getting Peer '42xx' is trying to register, but not configured as host=dynamic |
| 15:35.23 | qakhan | on cli |
| 15:35.49 | scv | the error message is rather clear |
| 15:35.58 | WIMPy | If you specify a host, there's no need to register. |
| 15:36.00 | polysics | right, time to step back and see what's wrong in the current install |
| 15:36.04 | WIMPy | See permit/deny |
| 15:36.08 | qakhan | Registration from '<sip:42xx@x.x.x.x>' failed for '110.9.x.x:1177' - Peer is not supposed to register |
| 15:36.24 | Samot | qkhan: Because you have a host specified. |
| 15:36.39 | Samot | Change your host back to host=dynamic |
| 15:37.12 | polysics | first thing first, my current * installation fills /tmp with core dumps |
| 15:37.21 | qakhan | Samot i want to restrict sip registeration with 1 IP address |
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| 15:37.48 | qakhan | its works if i use host=dynamic |
| 15:38.21 | polysics | and the logs do no seem to tell me why |
| 15:38.39 | WIMPy | But the core dumps will. |
| 15:38.42 | Samot | qakhan: Yes, because that's how it is supposed to work. |
| 15:39.10 | Samot | WIMPy pointed you in the right direction. permit/deny settings. |
| 15:39.19 | qakhan | but how can i restrict the sip registration with 1 ip |
| 15:39.28 | wyoung | WIMPy is the best |
| 15:39.43 | WIMPy | In what? |
| 15:39.58 | wyoung | At being as awesome as [TK]D-Fender |
| 15:40.18 | polysics | WIMPy: gdb seems to be quite cryptic though |
| 15:40.37 | WIMPy | To answer myself: I guess in guessing. I just hoped that were the right names :-) |
| 15:40.43 | polysics | https://gist.github.com/polysics/7de2cb6c06c5d1a11a8fcc3483fb4e2a |
| 15:41.01 | WIMPy | polysics: Depends on how much debuggung you enabled when compiling. |
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| 15:47.28 | polysics | waht it looks like is that safe_asterisk is what is segfaulting |
| 15:47.41 | polysics | if I run * from the CLI it works and does not core dump |
| 15:47.45 | polysics | whatever that implies |
| 15:48.56 | WIMPy | That is just a shellscript. |
| 15:50.41 | qakhan | any thought on my question |
| 15:52.35 | Samot | You want the phones to only register to one IP? |
| 15:52.42 | qakhan | yes |
| 15:52.46 | Samot | Uhm. |
| 15:52.52 | Samot | proxy |
| 15:53.06 | Samot | SIP Domain/SIP Proxy/whatever they call it in the device. |
| 15:53.24 | Samot | You tell it to register to a IP or FQDN |
| 15:53.35 | qakhan | i want to allow only 1 IP address which is remote office router IP address |
| 15:54.44 | WIMPy | Do it! We told you how. |
| 16:08.30 | polysics | WIMPy: rather, the safe_asterisk script works if called manually. At this point, it does look like the service/etc.d script is somehow breaking * |
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| 16:18.22 | leon_ | somebody let me know how to use dialplan function named PJSIP_MEDIA_OFFER(media) in dialplan. I can't find what is the media, and how can i use it on dialplan. |
| 16:22.36 | scv | 12:12 < rm> you still win |
| 16:22.38 | scv | er |
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| 16:44.06 | charlot | Hi |
| 16:44.17 | charlot | I have a question could someone help me? |
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| 16:54.06 | WIMPy | Not if you keep it to yourself. |
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| 17:19.03 | [TK]D-Fender | <qakhan> i want to allow only 1 IP address which is remote office router IP address <- so do it |
| 17:19.49 | drmessano | 11:53:37 <qakhan> i want to allow only 1 IP address which is remote office router IP address <-- Sounds like a good idea |
| 17:20.05 | drmessano | got here late, isn't sure who to pass the dead horse to next |
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| 17:28.30 | tikun | I have a SIP trunk that has three phone numbers associated to it, the main number lets say is 555-1515 and the other is 555-5500, the last is an 800 number. So the SIP trunk account is setup w/ the auth name being 555-1515. Now, I'm trying to take all calls made to 555-5500 I have created an inbound route with the DID 555-5500, created an IVR and then a ring group. |
| 17:28.45 | tikun | but when you call 555-5500 it sends it to 555-1515 no matter what. |
| 17:30.50 | shido6 | whats the cli tell you, Asterisk is very chatty |
| 17:31.24 | shido6 | Looking for X in context Y is what I'm looking for - you could also look through the sip debug to find out whats being sent and watch the debug to find out how its being matched |
| 17:31.39 | shido6 | use pastebin, for posting configs and debug |
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| 19:46.23 | Janko | Hello, I've got a lot of NOTICEs from probably malicious IPs in my logs, like http://pastebin.com/7XQ1A3Hs |
| 19:46.32 | Janko | Is it something to be concerned about? |
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| 19:55.18 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 19:55.59 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 19:56.27 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 20:14.12 | [TK]D-Fender | Janko, You're allowing un-authed calls in the first place. And they may eventually land on something that can dial out, etc. |
| 20:14.19 | [TK]D-Fender | and they are wasting your resources even as they fail |
| 20:17.04 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:32.26 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
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| 20:36.36 | Janko | [TK]D-Fender: Thanks, I've disallowed allowguest and installed fail2ban, and it looks better |
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| 21:32.16 | K0HAX | Is there a way for me to send a SIP 603 when hanging up on an inbound caller? The call is coming in on a PJSIP trunk. |
| 21:47.58 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 21:51.02 | *** join/#asterisk acidfu_ (~acidfoo@198-48-218-90.cpe.pppoe.ca) |
| 21:53.47 | SpaceInvaders | I have a question about what I'm reading in The Definitive Guide. It uses the following example http://pastebin.com/HjCpTGcz |
| 21:54.08 | *** join/#asterisk billum (~Bill@106.216.172.116) |
| 21:54.12 | SpaceInvaders | My question: Why is the 3rd line called the 2nd priority? |
| 21:54.33 | SpaceInvaders | The reason I'm asking is I thought the 3rd line would be the 3rd priority as they are numbered in priority 1,2,3 |
| 21:55.26 | SpaceInvaders | I thought exten => 321,1,NoOp() was in the format ext,priority,command such that the next line the n would be incremented (and thus, 2) |
| 21:55.48 | SpaceInvaders | Is the 1st n always the same priority as the above line? |
| 21:56.10 | WIMPy | NO, N IS THE PREVIOU LINE +1 |
| 21:56.15 | WIMPy | oops |
| 21:56.31 | WIMPy | Where do you see anything different? |
| 21:56.51 | SpaceInvaders | http://pastebin.com/HjCpTGcz I copied from the guide |
| 21:57.12 | SpaceInvaders | which also says - "In the second priority, we assign the value of 3 to the variable named COUNT ." |
| 21:57.26 | SpaceInvaders | which is why I asked |
| 21:57.50 | WIMPy | That's obviousely wrong then. |
| 21:57.57 | SpaceInvaders | Thanks! |
| 21:58.03 | SpaceInvaders | that means I understand |
| 21:59.45 | billum | Hello kind people. I wondered if I could please ask for some help. I am facing a consistent problem forwarding DID numbers to a SIP extension. I have tried 3 different DID providers (Twilio, CallCentric, Zadarma) and both sip and pjsip channels. The calls go through fine but for some reason they spawn multiple dialplans in parallel. The command line shows them proceeding in parallel, |
| 21:59.46 | billum | and you can hear them all at once, talking over each other. Does this ring a bell for anyone? I can share more details. Thank you! |
| 22:06.16 | billum | In other words, Asterisk is constantly spawning new extensions, as if the DID provider does not understand that prior SIP invitations were accepted/answered. |
| 22:07.09 | robmal | o_O |
| 22:07.17 | billum | (Though I am not actually "answering" the call - I want to let it ring. Is that a problem?) |
| 22:12.42 | rmudgett | That sounds like a NAT issue where the responses are not received. |
| 22:13.51 | rmudgett | Or sent to the wrong place. |
| 22:14.00 | billum | Thank you. But hmm, I don't have any NAT. I am trying to understand "re-invites" in sip.conf |
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| 22:25.51 | billum | You're right, Asterisk's SIP responses are not getting received by the DID host. |
| 22:47.02 | *** part/#asterisk kharwell (kharwell@nat/digium/x-qznudeuebjhmzbxf) |
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| 22:55.37 | *** join/#asterisk Micc_ (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 22:55.39 | wyoung | dang |
| 22:57.38 | Micc_ | What's the best way to add a jitter buffer to a sip channel that is created as a result of a dial? Should I use option b to do a gosub before the dial but with the channel then I call JITTERBUFFER(fixed)=200? |
| 22:58.44 | Micc_ | Or would I use option U to execute via gosub for the called channel before connecting to the calling channel? |
| 22:58.52 | Micc_ | or is there some other way? |
| 22:59.50 | Micc_ | Or if I set the JITTERBUFFER(FIXED)=200 before I do the dial, will it apply to those channels? |
| 22:59.55 | Micc_ | the new ones I mean. |
| 23:00.30 | wyoung | *shrugs* |
| 23:05.00 | rmudgett | JITTERBUFFER works on the channel that executed the function. I would use the Dial b option to add it to the outgoing channel before dialing. Using the U option after the call is answered will delay connecting the parties to run the dialplan to add the jitter buffer. |
| 23:06.48 | Micc_ | ok, thanks |
| 23:15.19 | billum | I am still facing the problem of parallel spawns of the dialplan. Your note about NAT made me doubt my hosting environment, so I moved it. Now the same problem is there with a VPS on both DigitalOcean and Vultr, and also with multiple OS's, sip vs. pjsip, and multiple DID providers. Any other tips what I could explore? |
| 23:17.34 | WIMPy | How do you "let it ring"? |
| 23:18.42 | billum | with the Ringing() command |
| 23:19.16 | billum | My whole dialplan is Ringing(), Wait(1), Hangup(). The problem is same with or without the wait. |
| 23:20.34 | WIMPy | Maybe you've just found some of those annoying providers that retry endlessly when a call isn't answered. |
| 23:21.12 | billum | Hmm, any suggestions for better DID providers? I see identical behavior from Twilio, CallCentric, and Zadarma |
| 23:21.42 | WIMPy | Where does the call come from? |
| 23:21.57 | billum | Same behavior if I initiate from a mobile phone or from Skype |
| 23:21.58 | WIMPy | It doesn't have to be the last provider in the chain. |
| 23:22.18 | billum | I didn't know that though, thank you |
| 23:22.47 | billum | Could an international source make it worse? |
| 23:23.25 | WIMPy | Sure. Anything can. |
| 23:23.28 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:25.11 | billum | I've seen consistent behavior from three origin countries, but none of them are where the DID number is hosted |
| 23:27.25 | WIMPy | So the provider you're getting the calls from isn't the one where the number is hosted? |
| 23:29.22 | billum | Correct. Calls originate from India and Myanmar, number hosted by DID provider in US, Asterisk running on VPS by different host in the US |
| 23:34.27 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 23:38.12 | billum | Just replicated with a third host as well. So have now tried (Twilio, CallCentric, Zadarma) x (Linode, Vultr, DigitalOcean) x (Ubuntu, Fedora, Debian, CentOS) x (Asterisk 13, 11) x (sip, pjsip) and I still see the problem. |
| 23:39.13 | shido6 | ok |
| 23:39.26 | shido6 | scrolls back - |
| 23:39.32 | shido6 | whats happening? |
| 23:39.45 | shido6 | parallels spawns of the dial plan - what is that? |
| 23:39.57 | shido6 | parallel spawns of the dial plan - what is that? |
| 23:40.40 | shido6 | show the dial plan in pastebin.com, please - billum |
| 23:41.12 | shido6 | and the debug , stop it after a few bad events. |
| 23:42.57 | justdave | anyone use Polycom phones and happen to know what config I have to tweak to make them send a proxy hostname in the INVITE request when picking a call to return from the missed calls list? |
| 23:43.55 | billum | thank you shido6. coming right up... |
| 23:44.01 | shido6 | okie dokie |
| 23:44.06 | WIMPy | billum: It about what the calls have in common, not what's different. |
| 23:44.08 | justdave | if I dial a number manually, the packet trace shows "INVITE sip:1234@my.asterisk.server;user=phone". If I hit the Dial button on a call list entry for the same number it does "INVITE sip:1234" |
| 23:44.31 | justdave | everything after the @ is missing, and Asterisk apparently doesn't like that |
| 23:44.41 | shido6 | what? |
| 23:44.43 | justdave | sends a 404 back to the phone when it does that |
| 23:44.46 | Samot | Yeah. |
| 23:44.47 | shido6 | it actually makes it to asterisk? |
| 23:44.54 | justdave | it actually makes it to asterisk |
| 23:44.58 | Samot | Because ive never have a polycom do that. |
| 23:45.16 | Samot | Have the INVITE in the call history. |
| 23:45.22 | shido6 | ok well you can write for the SNAFU or fix the phone... sounds like you want to fix it in the phone |
| 23:45.23 | shido6 | :) |
| 23:45.55 | shido6 | is it hitting the default context? |
| 23:45.59 | justdave | We just upgraded the firmware recently on the phones, it's got 4.1.1 on them now |
| 23:46.03 | Samot | If its making it to Asterisk then the polycom is using the proxy for the call |
| 23:46.24 | justdave | yeah, the polycom is sending it to the defined place, it's just not including it in the headers |
| 23:46.31 | Samot | Otherwise it wouldnt know where to send the call |
| 23:46.36 | shido6 | ok - what about + |
| 23:46.38 | Samot | Show a call |
| 23:46.41 | shido6 | do they have +'s |
| 23:47.02 | justdave | nope. they do show up with a sip: in front of the callerid number in the missed/received call lists now |
| 23:47.25 | justdave | it didn't do that before we upgraded (they previously had 3.3.1 on them) |
| 23:47.32 | shido6 | looks at justdave and gestures toward Samot - [ show the cal ] |
| 23:47.35 | Samot | Show a call. |
| 23:51.40 | Samot | No pms |
| 23:51.58 | Samot | Put it in the channel |
| 23:52.44 | justdave | so I need to sanitize it first then. what part if it's actually important to you? |
| 23:53.07 | Samot | All of it, |
| 23:53.32 | Samot | We need to see what is happening and when people "sanitize" debugs they generally cut something out. |
| 23:56.15 | justdave | https://it.pastebin.mozilla.org/8887425 <- there's the one that doesn't work, that's when I hit the Dial button on the number in the call list |
| 23:56.21 | justdave | https://it.pastebin.mozilla.org/8887427 <- that's the one that does work, that's when I just dial the number on the keypad and hit Send |