| 00:20.56 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 00:49.58 | *** join/#asterisk stux16777216Away (stux@72.20.50.1) |
| 01:21.53 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:21.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 02:04.29 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 04:11.59 | *** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co) |
| 04:55.36 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
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| 06:18.37 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 07:51.13 | *** join/#asterisk miralin (~Thunderbi@91.237.94.67) |
| 08:19.39 | *** join/#asterisk alpartis (~alpartis@cpe-2606-A000-71D0-5500-1E1B-DFF-FE6B-581.dyn6.twc.com) |
| 08:20.52 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:28.41 | *** join/#asterisk elcontrastador (~textual@206.78.110.8) |
| 08:28.50 | *** join/#asterisk areski (~areski@37.223.2.207) |
| 08:59.18 | elcontrastador | https://gist.github.com/anonymous/65918a0637def89bf15ee6c76dcbc2f0 |
| 08:59.39 | elcontrastador | Attempting to dial cisco router to paging gw |
| 08:59.51 | elcontrastador | sip debug and relevant config above |
| 09:00.26 | elcontrastador | looking for any visible Asterisk side issues |
| 09:10.33 | *** join/#asterisk areski (~areski@37.223.2.207) |
| 09:43.38 | *** join/#asterisk miralin (~Thunderbi@91.237.94.67) |
| 09:48.42 | *** join/#asterisk cwhuang (~cwhuang@210.240.39.201) |
| 09:50.38 | cwhuang | Hello! I have some old asterisk (1.8 or so) experiences before, but I haven't touch it several years. Recently I tried to upgrade to asterisk 15, but all CLI commands I know disappears. WHY? |
| 09:50.52 | cwhuang | no 'ael reload' |
| 09:50.57 | cwhuang | no 'sip debug' |
| 09:51.09 | cwhuang | etc |
| 09:51.53 | cwhuang | I tried to google it, but all docs I saw still describe the old commands |
| 09:52.55 | cwhuang | Is there any doc to explain the commands change of asterisk 15? |
| 10:05.02 | cwhuang | 'sip show peers' also disappeared |
| 10:05.50 | cwhuang | but these still described in the wiki: https://wiki.asterisk.org/wiki/display/AST/Registering+Phones+to+Asterisk |
| 10:06.08 | cwhuang | How can I check sip registration in asterisk 15? |
| 10:08.25 | *** join/#asterisk Rac-on_ (jasper@bambi.rac-on.nl) |
| 10:13.04 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 10:16.36 | *** join/#asterisk jkroon_ (~jkroon@165.16.204.165) |
| 10:35.08 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 10:41.13 | *** join/#asterisk Sprinterfreak_ (~irc@p4FF5ECB2.dip0.t-ipconnect.de) |
| 10:42.44 | elcontrastador | cwhuang: your modules aren't loading....check which modules are loading with 'modules show' or, more specifically, 'modules show like sip' |
| 10:44.39 | Sprinterfreak_ | Hi, does anyone know if it is possible to configure pjsip to a limited range of rtp media ports? Because of the lag of stun I otherwise would have to redirect 15000 udp-ports to my asterisk mashine behind the nat |
| 10:44.57 | elcontrastador | edit your modules.conf and add the line 'autoload = yes' as the first line under [modules]. This will load all modules that are built. It's best practice to have that off and load manually (with load = chan_sip.so, or the like) so you're not running any modules unnecessarily |
| 10:46.03 | Sprinterfreak_ | This ofcourse is nearly impossible because the router allows a maximum of 255 ports redirected in one range |
| 10:47.32 | Sprinterfreak_ | I think the only way to get around this is sticking to the old chan_sip until pjsip supports stun |
| 10:51.10 | Sprinterfreak_ | without stun the only way to receive incoming calls WITH audio would be if asterisk could force my telco to use a limited range of udp ports for rtp |
| 10:54.20 | Sprinterfreak_ | otherwise because the nat router does not allow me to redirect such huge blocks of port-ranges it would be nearly impossible to get incoming calls |
| 10:55.37 | Sprinterfreak_ | my telco sends rtp to any port between 15000 to 30000/udp |
| 11:00.35 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 11:02.15 | Sprinterfreak_ | There is rtp.conf wich in combination with rtp_symmetric = yes lets me force my telco to anser to valid rtp ports during an outgoing call. But this does not solve my problem on incoming calls |
| 11:21.11 | *** join/#asterisk roswell (roswell@roswell.systems) |
| 11:59.18 | *** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com) |
| 12:04.20 | *** join/#asterisk kodomo (~gregor@92.63.174.119) |
| 12:06.39 | kodomo | Hi all - I'm a bit lost with a tcptls issue here - the issue seems common enough, but I didn't find a solution, yet - maybe someone can give me a hint into the right direction... |
| 12:07.32 | roswell | hi. what's the issue? |
| 12:07.54 | kodomo | I'm currently migrating from a crypto-lifetime-ignore patched * 1.6 via a non-patched * 11 to a non-patched * 13 (which includes crypto-lifetime-support, I understand) |
| 12:08.30 | roswell | sounds weird already ) |
| 12:08.47 | kodomo | v1.6 was working fine with TLS on sip ; 11 only between devices not using crypto-lifetime, 13 not at all - outputting the following error over and over again: |
| 12:08.56 | kodomo | [Jan 21 13:07:20] ERROR[20614]: tcptls.c:694 handle_tcptls_connection: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error |
| 12:08.59 | kodomo | [Jan 21 13:07:20] WARNING[20614]: tcptls.c:781 handle_tcptls_connection: FILE * open failed! |
| 12:09.36 | roswell | i'd imagine it has issues reading a file |
| 12:09.49 | kodomo | First thing I thought of was the keys directory setting (checked in both asterisk.conf and sip.conf -they're all right) |
| 12:10.13 | roswell | what about their permissions and ownership> |
| 12:10.34 | kodomo | Next thing I thought of were file permissions - the directory and keys are accessible to the asterisk group (asterisk is running as asterisk.asterisk) - so there shouldn't be a problem either |
| 12:10.49 | kodomo | now I'm a bit puzzled ;) |
| 12:11.28 | roswell | have you tried _temporarily_ setting a+r permission to these? |
| 12:11.54 | kodomo | not yet - will do... |
| 12:14.53 | kodomo | That is... weird... |
| 12:15.11 | roswell | what exactly i wonder |
| 12:15.24 | kodomo | Now I don't get above error anymore, but instead, * reports |
| 12:15.57 | kodomo | [Jan 21 13:14:29] ERROR[21167]: tcptls.c:1046 ast_tcptls_client_start: Unable to connect SIP socket to [my IP]:[some port]: No route to host |
| 12:16.12 | kodomo | mind that it's the IP I'm loggin onto the server from right now... |
| 12:16.44 | roswell | alrighty then, update ownership and revoke a+r permission |
| 12:17.45 | roswell | are you running ipv6 by chance? |
| 12:19.56 | kodomo | ok - re-owned keys to asterisk.asterisk, running * as asterisk and getting the FILE * open failed again... something's definitely amiss. |
| 12:20.26 | kodomo | Moreover, I still get the unable to connect SIP socket - so that's apparently unrelated (must have missed these before) |
| 12:21.23 | roswell | what a mess... the best advice would be to backup everything and reinstall ) |
| 12:21.46 | kodomo | IPv6: your typical system... local link is configured - but it's not used actively |
| 12:22.31 | kodomo | That would be a distro-change, then, as I've just set up this machine :P |
| 12:23.18 | kodomo | Lucky me, I still got the old VM running 1.6 around, though inactive - but it's badly outdated (hence my giant version leap that I'm undertaking) |
| 12:23.30 | roswell | what's 'ps aux | grep asterisk' show? |
| 12:23.44 | roswell | mind usernames |
| 12:23.47 | kodomo | asterisk 21489 2.2 9.2 2002384 94656 ? Ssl 13:18 0:06 /usr/sbin/asterisk -p -U asterisk |
| 12:23.50 | kodomo | asterisk 21490 0.0 0.1 15232 1660 ? S 13:18 0:00 astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 21489 |
| 12:23.53 | kodomo | root 21604 0.0 0.6 78560 6644 pts/1 S+ 13:19 0:00 rasterisk r |
| 12:23.55 | kodomo | it's running as asterisk, all right |
| 12:23.57 | kodomo | root 21894 0.0 0.0 12756 968 pts/5 S+ 13:23 0:00 grep asterisk |
| 12:24.11 | roswell | hm. i'm actually out of ideas |
| 12:24.45 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-180-17-122.ph.ph.cox.net) |
| 12:24.54 | roswell | maybe running strace would help to find out the cause, but i'm far from my mood helping you with that |
| 12:24.54 | kodomo | roswell: thx for trying, though - at least we've established that it's a (weird) permission problem. |
| 12:25.08 | kodomo | Do you have ideas w.r.t. the route issue? |
| 12:25.37 | kodomo | It's weird as well - getting a no route to host message being logged on via ssh from exactly that IP |
| 12:25.39 | roswell | try checking 'iptables -L -n' output |
| 12:26.12 | roswell | perharps it's some weird setup |
| 12:26.45 | kodomo | roswell: oh - looking at this mess, I'd say it most definitely must be! XD |
| 12:27.22 | Sprinterfreak_ | kodomo: to the permission issue. Does any folder in the tree at least have execute permissions to asterisk? |
| 12:27.58 | kodomo | all seems in order - SIP, SIPS, and the dedicated RTP port are permitted explicitely in my INPUT chain, OUTPUT has POLICY ACCEPT, all right... |
| 12:28.45 | roswell | also mind Sprinterfreak_ 's remark, and check whether ICMP is allowed aswell |
| 12:29.52 | kodomo | Sprinterfreak_: everyone is allowed access (x flag) and read access (r flag) to the directories in the path to the keys |
| 12:30.58 | kodomo | the keys directory itself belongs to root.asterisk (the idea was that asterisk needs to read the keys, but not to change them), but that shouldn't be an issue, as it's still world-xr. |
| 12:32.12 | roswell | a bad idea actually. am unsure what os you've got running, but in debian /etc/ssl/private is only urwxgrx , whilst asterisk shares ssl-cert group assigned to the private directory |
| 12:33.55 | kodomo | It's a Devuan jessie (with asterisk + dependencies updated to ascii) |
| 12:34.24 | roswell | what's 'id asterisk' output? |
| 12:34.26 | kodomo | mind that while the directory is world accessible and readible, the files themselves are not |
| 12:36.25 | kodomo | roswell: ok... my world view snapped back - I just tried the chmod 644 on all key files again, restarted asterisk and got the FILE * open error again... |
| 12:36.33 | roswell | i'm unsure about devuan (i know it's debian w/o systemd) but perharps they've introduced some apparmor or selinux. try also checking these |
| 12:36.43 | kodomo | sad, because it's a step back, but good, because it's consistent with my world view |
| 12:36.57 | kodomo | uid=104(asterisk) gid=108(asterisk) groups=108(asterisk),20(dialout),29(audio) |
| 12:38.33 | kodomo | hm - this seems so off that I'm seriously considering scrapping the system :-| |
| 12:39.28 | tzafrir | kodomo, what's ascii? Based on Stretch? |
| 12:39.36 | roswell | and what's show 'ls -ld /etc/ssl/private' ? |
| 12:40.28 | kodomo | maybe I'll give a proper ascii install another chance before moving to another distro - but the weirdness in this makes me think that maybe the dependencies are not strictly respected upon mixing jessie with ascii (that seem like a probable enough reason to me) |
| 12:40.49 | kodomo | ascii is whatever comes after jessie (there they have the same name) |
| 12:40.53 | kodomo | I'll check |
| 12:41.08 | roswell | most likely it's the reason ) |
| 12:41.19 | kodomo | roswell: drwx------ 2 root root 4096 Nov 2 15:06 /etc/ssl/private/ |
| 12:41.31 | roswell | hence the issue |
| 12:41.40 | tzafrir | kodomo, I don't know about the case of devuan specifically, but mixing two releases is something that has to be done carefully. |
| 12:42.06 | tzafrir | Specifically in Debian Stretch (9), Debian switched to OpenSSL 1.1 by default. |
| 12:42.32 | tzafrir | IIRC the Asterisk package is built vs. openssl 1.1 |
| 12:42.36 | kodomo | tzafrir: I did it in the old days, so I'm aware... but I was also fighting with dependency resolution upon trying between jessie and ascii this time... |
| 12:42.55 | tzafrir | But all too many libraries are still linked with openssl 1.0.2 |
| 12:43.34 | kodomo | roswell: you think it needs to access this directory? There's nothing in there... I keep the * keys under /etc/asterisk/keys |
| 12:43.44 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:43.58 | roswell | does himself a facepalm |
| 12:43.59 | tzafrir | If it's called "private" then: no |
| 12:45.21 | kodomo | roswell: ok... ? Up to now I could keep my keys where I wanted and it worked all right... so please explain :) |
| 12:45.30 | tzafrir | kodomo, you mentioned strace. Strace defaults to showing you the main thread. |
| 12:45.42 | tzafrir | Try -f to see all threads? |
| 12:45.53 | roswell | tzafrir, -ff actually |
| 12:46.03 | kodomo | tzafrir: roswell mentioned strace, I did not, so far... |
| 12:46.04 | tzafrir | ah, ok |
| 12:46.33 | roswell | kodomo, check all permissions on that directory |
| 12:47.20 | roswell | ls -ld /etc/asterisk/keys |
| 12:47.24 | kodomo | roswell: I did (report is above) - you figured it a bad idea to give access and read permissions to the world (to the directory only, I'd like to point out) |
| 12:47.37 | kodomo | root@newselena:/etc/asterisk/keys# ls -ld /etc/asterisk/keys |
| 12:47.37 | kodomo | drwxr-xr-x 2 root asterisk 4096 Dec 17 20:50 /etc/asterisk/keys |
| 12:48.19 | roswell | alrighty. ls -l /etc/asterisk/keys |
| 12:48.34 | roswell | don't forget to paint out valuable data |
| 12:51.12 | kodomo | roswell: mind that it's the test with a+r so all files are like this: |
| 12:51.20 | kodomo | -rw-r--r-- 1 asterisk asterisk 1220 Jan 14 19:19 asterisk.crt |
| 12:51.36 | roswell | looks fine to me |
| 12:51.38 | kodomo | do you want the entire file list? |
| 12:51.58 | kodomo | roswell: indeed - before the test it was -rw-r----- |
| 12:53.13 | kodomo | sorry - gotta jump to catch a train - will be bak online in about 40 minutes ; thank you very much for spending time on this! |
| 12:54.32 | roswell | yvw mate |
| 12:58.12 | Sprinterfreak_ | So for my initial pjsip issue i gave up now. I don't see any way to get pjsip working behind nat with my provider |
| 13:45.35 | kodomo | ... ok - looking at strace doesn't help me much, either - the thread throwing the error message does some futex business before throwing the error - the next open() above it is for /var/tmp - no entry trying to open something in *keys* |
| 13:46.16 | kodomo | sad, but I'm getting the impression that reinstalling the system with another release or distro will be the easier way out |
| 13:46.31 | roswell | sure ) |
| 13:46.53 | roswell | just don't plug any else repositories along |
| 13:47.22 | kodomo | It's annoying, though - I usually like knowing what's amiss :p ;) |
| 13:48.18 | kodomo | lesson 1) do backups, lesson 2) really! do backups, lesson 3) if you can't help it, keep the old system around until the new one works ;) |
| 13:53.36 | roswell | lesson 3) hire someone who reads this manthra for you |
| 13:53.51 | kodomo | *g* |
| 13:54.10 | roswell | lesson 4) then fire him and hire someone who really does backups |
| 13:54.37 | kodomo | *lol* ack! |
| 14:00.05 | roswell | Sprinterfreak_, perharps ur provider may be plainly blocking RTP or SIP traffic |
| 14:02.19 | roswell | because 'it does not work' doesn't give any clue |
| 14:02.58 | Sprinterfreak_ | roswell: No it dosn't. SIP goes through. If a call comes in my provider invites me and told me our rtp-stream goes to <externip>:17567 |
| 14:03.04 | *** join/#asterisk jkroon (~jkroon@165.16.204.165) |
| 14:04.09 | Sprinterfreak_ | Problem is this is different on any call. the ports audio gets sent to range between 15000 and 30000 |
| 14:05.33 | Sprinterfreak_ | Since my asterisk sits behind nat and my router doesn't let me forward such huge ranges of ports I never get the incoming rtp stream |
| 14:06.57 | Sprinterfreak_ | The only way to get this done seems to be using stun, wich of course is not implemented in pjsip yet |
| 14:07.03 | roswell | narrow rtp port range in /etc/asterisk/rtp.conf |
| 14:07.20 | roswell | with rtpstart and rtpend |
| 14:07.23 | Sprinterfreak_ | This only affect outgoing calls. |
| 14:07.37 | file | it affects both. |
| 14:07.44 | file | it controls the range from which we locally allocate RTP pors |
| 14:07.52 | file | you can look at the SIP signaling to see this |
| 14:08.48 | Sprinterfreak_ | file: sadly my sip provider doesn't follow this |
| 14:09.17 | Sprinterfreak_ | It is narrowed indeed. But it doesn't take effect |
| 14:09.33 | file | then that's not the fault of Asterisk, it could even be your NAT setup establishing a port mapping outside of the range |
| 14:09.39 | Sprinterfreak_ | Only on outgoing calls with symetric_rtp set to yes |
| 14:09.47 | roswell | Sprinterfreak_, uhm... have you tried to tcpdump traffic on all ends you have access to? |
| 14:11.05 | Sprinterfreak_ | Yes. This way I figured out my provider doesn't recognise the narrowed range of rtp ports set in my rtp.conf. |
| 14:11.44 | roswell | switch to another one. or perharps use a vpn |
| 14:12.30 | Sprinterfreak_ | At the moment I only see the option to switch back to the working but insecure chan_sip |
| 14:12.50 | file | what option are you using in chan_sip that "makes it work"? |
| 14:12.57 | Sprinterfreak_ | stun |
| 14:13.28 | file | stun is not an option. |
| 14:14.52 | Sprinterfreak_ | ok, its called outboundproxy |
| 14:15.08 | file | that's not STUN, that sets a SIP outbound proxy |
| 14:15.24 | file | which can also be done in chan_pjsip using the outbound_proxy option |
| 14:16.45 | file | that influences the routing of the SIP messages though, not the RTP |
| 14:17.38 | Sprinterfreak_ | :q |
| 14:17.52 | Sprinterfreak_ | uses vim |
| 14:18.03 | roswell | pervert |
| 14:18.05 | roswell | :D |
| 14:18.15 | roswell | sorry ) |
| 14:21.23 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 14:29.26 | Sprinterfreak_ | still debugging |
| 14:30.00 | *** join/#asterisk DanB (~DanB@clt-195.192.206.104.ip-anschluss.net) |
| 14:34.00 | Samot | Uhm. |
| 14:34.29 | Samot | Sprinterfreak_: So this is a case of having one way audio on incoming calls from your provider? |
| 14:35.03 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:36.16 | Samot | Because, reading the scroll back so far leads to me believe this is a 100% NAT issue. |
| 14:36.37 | Samot | Also the "outbound proxy" isn't something that is used when incoming traffic is sent to the system. |
| 14:37.06 | Sprinterfreak_ | Samot: Currently testing this. |
| 14:37.20 | Samot | But that is the issue? |
| 14:37.28 | Samot | One way audio from the provider on inbound calls? |
| 14:37.49 | Sprinterfreak_ | Samot: As far as I can remember there was no audio at all and the call dropped after a timeout |
| 14:38.03 | Samot | So the call drops after what, 30 some seconds? |
| 14:38.14 | Sprinterfreak_ | Yes |
| 14:38.21 | Samot | Only for inbound? |
| 14:38.49 | Sprinterfreak_ | Only for inbound calls, yes |
| 14:38.54 | Samot | OK. |
| 14:39.05 | Samot | So what kind of router do you have there in front of the PBX? |
| 14:39.56 | Sprinterfreak_ | Its a Fritz!Box. Not the best choice for that |
| 14:40.00 | Samot | OK. |
| 14:40.08 | Samot | The issue is your Fritz!Box. |
| 14:40.12 | Samot | This is 100% networking. |
| 14:40.26 | Samot | Not Chan_SIP or Chan_PJSIP. |
| 14:40.37 | Samot | This is that those Fritz!Box's are POSes. |
| 14:41.22 | Sprinterfreak_ | Samot: Agree that those Boxes have way too much intelligence |
| 14:41.23 | Samot | Because for the most part a Fritz!Box is an ATA/VoIP device as well. |
| 14:41.27 | Samot | It's an all in one box. |
| 14:41.37 | Samot | It's mean for when an ISP is giving you both data and voice. |
| 14:41.59 | Samot | So you have to deal with the Fritz!Box and it's voip stuff. |
| 14:43.39 | Sprinterfreak_ | But my FritzBox has no SIP configured. I simply forwarded 5060, actually 5061 (used with pjsip) and my rtp port range wich is configured in rtp.conf to my asterisk box |
| 14:44.59 | Samot | And what's the RTP port range you have? |
| 14:45.19 | Samot | Because said earlier that it doesn't support a wide range for port forwarding |
| 14:46.15 | Sprinterfreak_ | 20000-20099 |
| 14:46.52 | Samot | And that's what you have in your rtp.conf? |
| 14:46.58 | Sprinterfreak_ | right |
| 14:52.42 | Sprinterfreak_ | The curious thing about that is, it currently works |
| 14:53.11 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 14:53.36 | Sprinterfreak_ | I just don't know why |
| 14:54.00 | Sprinterfreak_ | Maybe because Wireshark is running |
| 14:54.48 | Sprinterfreak_ | Will test this in an hour again an see if it still works |
| 15:43.07 | *** join/#asterisk kkocaerkek (~kkocaerke@95.8.207.19) |
| 15:55.35 | *** join/#asterisk Typhon (~Typhon@dslb-084-056-191-139.084.056.pools.vodafone-ip.de) |
| 16:04.06 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 16:09.22 | Sprinterfreak_ | Samot: Strange. Sometimes RTP data get in on a different port as subscribed in the SDP. Then asterisk drops the RTP data |
| 16:10.05 | Samot | I'm not sure what you mean by that. |
| 16:10.14 | Samot | Subscriber by who in what SDP? |
| 16:10.21 | Samot | subscribed* |
| 16:17.51 | Sprinterfreak_ | oh no. I see. Maybe the rtp stream comes from a different ip then the signaling |
| 16:20.21 | Samot | Generally that is the case with providers. |
| 16:20.31 | Samot | The media is coming from another source |
| 16:52.27 | znoteer_ | I've got this problem with iax2 for a few days now. I don't receive incoming calls. When someone calls, I can see the packets arriving at the network interface with tcpdump. But, nothing appears on the asterisk console and my phones don't ring. The only weird thing that I've noticed is running "iax2 show registry" returns n.m.74.179 for both host and perceived addresses. It should show my public |
| 16:52.33 | znoteer_ | IP as the perceived address. Does anyone understand what is going on here? |
| 16:58.52 | Samot | So is this an IAX trunk and an IAX peer for the phone? |
| 16:59.49 | *** join/#asterisk alpartis (~alpartis@cpe-2606-A000-71D0-5500-1E1B-DFF-FE6B-581.dyn6.twc.com) |
| 16:59.52 | Samot | Because if this inbound call is from your carrier and supposed to hit Asterisk then dial the phone, the issue is with the incoming call not getting to Asterisk to actually dial the phone. |
| 16:59.55 | Samot | Not the phone. |
| 17:04.22 | znoteer_ | yes, that is what I figure too. The question is why are these incoming packets not showing up in Asterisk console. I've made no recent changes to this system. The only thing that I see that is different from when it did function is that weird thing with my perceived address when I run "iax2 show registry". What could cause it to show my provider's IP instead of mine? |
| 17:04.40 | Samot | OK |
| 17:04.51 | Samot | So this is an IAX trunk you're having a problem with? |
| 17:04.58 | znoteer_ | yes |
| 17:05.10 | Samot | And you have no issues make calls out of it? |
| 17:05.27 | znoteer_ | outgoing calls and internal calls work fine |
| 17:05.34 | Samot | Internal calls wouldn't use it. |
| 17:05.57 | Samot | Show an incoming call from the provider.. |
| 17:06.05 | Samot | With the Asterisk debug running.. |
| 17:06.25 | znoteer_ | OK |
| 17:29.19 | znoteer_ | rejected. No such extension : https://paste.debian.net/hidden/6ded1599/ |
| 17:35.31 | Samot | Uhm. |
| 17:35.37 | Samot | This is not at all what you described. |
| 17:35.44 | Samot | So the call IS making it into Asterisk. |
| 17:35.52 | Samot | What is the context that you have for this trunk? |
| 17:35.57 | Samot | Do that extension exist? |
| 17:36.25 | znoteer_ | Sorry, before putting debug on I didn't see anything coming in |
| 17:36.51 | znoteer_ | yes, it exists. This was all working fine until a few days ago |
| 17:37.14 | Samot | Well that's a pretty generic IAX error. |
| 17:37.21 | Samot | Generally means it can't find a match in the context. |
| 17:37.28 | Samot | Or a context at all. |
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| 17:43.26 | znoteer_ | Is this likely to be related to the fact that my percieved address is wrong? https://paste.debian.net/hidden/7a8580bf/ My public IP is c.d.150.230 |
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| 17:45.21 | Samot | I honestly don't know. |
| 17:45.55 | Samot | I rarely used IAX when it was actually solving issues 12 years ago. Since it the issues it was fixing really don't exist anymore... |
| 17:46.13 | Samot | And that it has very, very, very limited and documented support. |
| 17:46.43 | znoteer_ | so it's a lame duck? |
| 17:50.02 | drmessano | Not necessarily |
| 17:50.24 | drmessano | IAX2 is to your provider? |
| 17:50.30 | znoteer_ | yes |
| 17:52.24 | drmessano | Ok so my thoughts. (1) IAX2 on a LAN is still a good solution between Asterisk boxes. (2) IAX2 between owner sites using Asterisk is still a good solution. (3) IAX2 with a provider means they are using Asterisk, which is absolutely terrible. |
| 17:52.51 | Samot | Hey now. |
| 17:53.18 | Samot | I mean, I would never use IAX or offer it as an option for that. |
| 17:53.21 | znoteer_ | why is 3 terrible? |
| 17:53.27 | drmessano | You have all your customers hanging their calls off Asterisk boxes? |
| 17:53.37 | drmessano | Asterisk is not a proxy |
| 17:53.40 | Samot | Calls go through Asterisk. |
| 17:53.53 | Samot | My customers hang off proxies. |
| 17:54.01 | drmessano | Thatâs fine |
| 17:55.22 | drmessano | I donât trust a provider that offers IAX because likely they are Asterisk boxes in a hall closet |
| 17:55.27 | znoteer_ | I chose IAX2 because it only requires a single network port, so I have to diddle less with firewall rules, and because it can trunk multiple calls to save network overhead |
| 17:55.39 | drmessano | SIP is one rule |
| 17:55.50 | drmessano | You donât need firewall rules for RTP |
| 17:56.01 | drmessano | Itâs an associated protocol to most firewalls |
| 17:57.02 | drmessano | Youâre a client to your ISP so you actually donât need any firewall rules if youâre allowing all outgoing |
| 17:57.07 | drmessano | ITSP |
| 17:57.51 | drmessano | Donât waste your time on IAX2 |
| 17:58.06 | drmessano | Nobody else is, even Digium |
| 17:58.21 | znoteer_ | so it is a lame duck, then |
| 17:59.08 | Samot | IAX hasn't been Digium supoprt since, what? |
| 17:59.09 | drmessano | Points 1 and 2 above are still valid |
| 17:59.13 | Samot | I know it's been years. |
| 17:59.59 | znoteer_ | OK, I guess I'll ditch it. It seemed like a good thing in the O'Reilly book |
| 18:00.10 | drmessano | Actually with PJSIP taking over IAX2 between internal boxes has even more of a place. An IAX2 peer is just a couple of lines |
| 18:00.15 | drmessano | And it works well |
| 18:00.23 | drmessano | But donât use IAX2 to your provider |
| 18:00.45 | drmessano | Itâs not worth the lack of community support or provider support |
| 18:02.08 | drmessano | Who is the provider? |
| 18:03.59 | znoteer_ | voipms |
| 18:06.17 | drmessano | Hah. Ok. Yeah skip the IAX2 |
| 18:06.39 | drmessano | Youâre not the first one with VoIP.ms to set up IAX2. |
| 18:07.18 | drmessano | Their implementation doesnât work as well as one would think and the IAX2 support isnât so great. |
| 18:09.48 | Samot | I think that's part of Zopier's issues too. |
| 18:10.16 | Samot | Any softphone that I can download in 2018 and it has IAX2 support, I pretty much dismiss. |
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| 18:58.27 | drmessano | lol |
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| 21:17.23 | hdon | hi all :) when asterisk wants to register to a sip server, it should show up in "sip show registry" right? even if the registration hasn't succeeded? |
| 21:21.35 | hdon | oh i see. i actually need a register line in my sip.conf |
| 21:40.47 | [TK]D-Fender | Yes, telling * to register is kinda important |
| 21:42.57 | hdon | :) |
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| 23:52.08 | Hsilamot | Hello there, anyone knows by any chance if there's an easy way to install Asterisk/DAHDI in a FreeBSD system? |
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