| 00:00.24 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:01.07 | msaraiva | Only thing "special" i can identify on that INVITE is t38, but i might be overlooking something... |
| 00:01.14 | mjordan | nothing about that jumps out at me. My guess is that there is something in the fax image offer that res_pjsip_t38 doesn't like. |
| 00:01.25 | mjordan | do you have the t38 module loaded? |
| 00:04.48 | msaraiva | Let me check |
| 00:08.28 | msaraiva | Never had to use fax on Asterisk before, so i'm not sure if this is correct, but i have res_fax, res_fax_spandsp and res_pjsip_t38 loaded |
| 00:11.48 | mjordan | Hm. |
| 00:12.01 | mjordan | I'd remove your explicit transport usage in the endpoint and see if that magi-fixes it. |
| 00:12.21 | mjordan | actually, your transport option seems invalid |
| 00:12.33 | mjordan | since you didn't include a transport with that name in the config |
| 00:12.49 | mjordan | but, generally, you don't have to specify explicit transports unless you *REALLY* want to force an endpoint onto that transporot |
| 00:12.53 | msaraiva | Sorry, that's not the full config |
| 00:12.56 | mjordan | np |
| 00:12.59 | mjordan | I'd still try removing that. |
| 00:13.09 | msaraiva | Sure, let me try that |
| 00:13.12 | mjordan | Since there's a few IP checks that occur in the SDP handler in T.38. |
| 00:13.38 | mjordan | Otherwise, I'd make a full DEBUG log (instructions on the wiki for that) with logging enabled and make an issue. I can't see why that would get rejected. |
| 00:15.53 | msaraiva | You mean with "core set debug 10" : |
| 00:16.03 | msaraiva | *? |
| 00:18.03 | mjordan | nope |
| 00:18.18 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 00:19.13 | msaraiva | Ah, that. |
| 00:19.16 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 00:19.31 | msaraiva | Yeah, my console channel was setup like that |
| 00:19.47 | msaraiva | I have some logs from that as well, if you'd like to see that |
| 00:21.36 | msaraiva | Paste is updated with the debug info i got |
| 00:25.08 | msaraiva | Removing the transport from the endpoint config didn't magi-fixed it :'( |
| 00:25.08 | mjordan | mind trying a patch? |
| 00:25.16 | msaraiva | Not at all! |
| 00:26.06 | mjordan | it is against 13. Fixed a lot of 488 issues last week. It _may_ apply to 12 |
| 00:26.29 | mjordan | http://pastebin.com/QsrnMXwr |
| 00:26.34 | msaraiva | Humm, let's hope it does |
| 00:27.18 | mjordan | it will spit out more debug information if nothing else :-) |
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| 00:38.23 | msaraiva | http://pastebin.com/SjcVjPyd |
| 00:38.27 | msaraiva | Only this failed to apply |
| 00:38.28 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 00:38.35 | msaraiva | Guess not all is lost :) |
| 00:39.08 | mjordan | you can probably ignore that |
| 01:01.19 | msaraiva | Both modules compiled fine |
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| 01:10.58 | msaraiva | ...but with different compile options |
| 01:10.59 | msaraiva | damn |
| 01:11.04 | msaraiva | Time to look at my docs... |
| 01:20.52 | Apteryx | Hmm. I'm now at studying the wireshark traces of why I can't get TCP and chan_pjsip working with Bria and Jitsi. |
| 01:23.03 | Apteryx | When calling Jitsi at 192.168.0.102, Asterisk tries to establish TCP connection with the port Jitsi used when registering, and Jitsi sends back an ACK. So far so good. |
| 01:23.48 | Apteryx | But following this, Asterisk loops, resending this: 333.435953000192.168.0.10192.168.0.102TCP74[TCP Retransmission] 47535 > 45097 [SYN] Seq=0 Win=29200 Len=0 MSS=1460 SACK_PERM=1 TSval=31373931 TSecr=0 WS=128 |
| 01:24.44 | Apteryx | Which wireshark labels as "TCP retransmission". The call rings until it times out. |
| 01:25.03 | Apteryx | (rings only for the caller, and does not ring for the callee) |
| 01:42.02 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 01:42.26 | Apteryx | Still no joy with simple TCP registration and Asterisk 12/13. A bit frustrating. I love the features of pjsip like multiple endpoint registrations and other such modernities, though! |
| 01:42.49 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
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| 01:59.46 | WIMPy | Well, there's always something... |
| 02:05.20 | ChannelZ | Is there a reason you want TCP specifically? |
| 02:07.22 | Apteryx | ChannelZ: it consumes much less battery on iOS and probably Android also. At least using Bria as a client, because then it does not need to run in the background to receive calls. |
| 02:08.05 | WIMPy | has read about that, but I don't see why it is/has to be that way. |
| 02:08.48 | Apteryx | I don't know also. I just guess Apple came with some smart mechanism regarding TCP connections to save battery. |
| 02:09.45 | WIMPy | I don't see why theprotocoll should make a diference. |
| 02:09.46 | Apteryx | Using TCP, the drain is really small. But I can still receive calls. |
| 02:11.19 | Apteryx | WIMPy: I think the Apple services already have to run for notifications and stuff. And this uses TCP. So Bria can just tap into a service that has to run anyway. My wild guess. |
| 02:12.19 | WIMPy | Doesn't make sense to me. You just sleep until you receive anything in both cases. |
| 02:13.20 | Apteryx | Well, something has to run to wake up the sleeping app. and I guess this something is limited to TCP on iOS devices. |
| 02:14.06 | ChannelZ | Leave it to apple for nonsense |
| 02:15.16 | Apteryx | indeed! |
| 02:18.01 | Apteryx | I'm now trying to try my old sip.conf, so I've put a noload => res_pjsip. But my "service asterisk restart" fails at starting Asterisk. And no obvious reason why in /var/log/asterisk/messages |
| 02:19.45 | ChannelZ | rename your configs |
| 02:20.33 | Apteryx | well, sip.conf is there. And I've renamed pjsip.conf to pjsip.conf.bak |
| 02:20.34 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 02:28.18 | Apteryx | no crash when ran from gdb, yet asterisk is not running ^^ |
| 02:33.39 | WIMPy | asterisk -cvvvddd |
| 02:36.45 | Apteryx | ok, thanks. |
| 02:37.12 | Apteryx | Seems I need to put a noload in from of anything pjsip, else it will fail with an undefined symbol error. |
| 02:37.20 | Apteryx | *in front |
| 02:39.31 | Apteryx | can I use a regexp? ;) |
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| 02:55.34 | Apteryx | using the old chan_sip, TCP works wonder. |
| 03:07.11 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 03:07.35 | Apteryx | Is there pass-through support for codecs such as opus in chan_sip? |
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| 03:25.43 | msaraiva | mjordan: always get a core dump with the patched pjsip modules |
| 03:26.58 | msaraiva | Well, only res_pjsip_session causes that, to be precise |
| 03:28.06 | msaraiva | Guess that for now i'll have to go back to chan_sip |
| 03:28.10 | msaraiva | At least for this particular peer |
| 03:54.50 | msaraiva | mjordan: thanks for the help...hope these 488 issues get resolved in the future |
| 03:56.32 | Penguin | apteryx: I can't even use TCP with chan_sip, so you're in a better place than I am. |
| 03:59.56 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 04:00.57 | Apteryx | Penguin: wow, how's that ? Which version ? |
| 04:01.39 | Penguin | 1.8.23.1 |
| 04:01.56 | Penguin | I set the transport to tcp, change the softphone to use tcp... |
| 04:02.51 | Penguin | It will register and it will receive calls... |
| 04:03.11 | Penguin | But all calls from the phone, the phone says 603/Declined. |
| 04:03.13 | Apteryx | Isn't that old |
| 04:03.23 | Penguin | Not that old, no. |
| 04:03.42 | Penguin | Asterisk 1.8.23.1 built by rob @ cpe-e650 on an i686 running Linux on 2013-10-09 17:45:26 UTC |
| 04:03.47 | Penguin | Not even one year old! |
| 04:04.21 | Penguin | So the phone always says 603/Declined to all calls I try to make. |
| 04:04.29 | Penguin | sip debug shows absolutely NOTHING. |
| 04:04.38 | Apteryx | When you place outbound calls from the TCP registered phone, it says 603/Declined? |
| 04:05.01 | Penguin | core verbose shows the call hitting the unauthorized context, which means it must not be sending auth. |
| 04:05.14 | Penguin | That's correct. 603/Declined shows on the screen. |
| 04:05.26 | Apteryx | Is this all happenning on a lan (no nat) ? |
| 04:05.48 | Penguin | Yes, on the LAN, same subnet, no NAT. |
| 04:06.24 | Penguin | Why would sip debug not show anything at all? |
| 04:06.25 | Apteryx | Did you remove all the app-demo stuff from the extensions.conf |
| 04:06.53 | Penguin | This isn't a "new" asterisk. |
| 04:07.02 | Apteryx | ok |
| 04:07.25 | Penguin | Why would sip debug not show anything at all? |
| 04:07.49 | Apteryx | Can you paste the sip debug log somewhere? |
| 04:07.53 | Apteryx | set sip debug on |
| 04:08.01 | Penguin | (2304.28) <Penguin> sip debug shows absolutely NOTHING. |
| 04:08.04 | Apteryx | sip set debug on ;) |
| 04:08.11 | Apteryx | ok... |
| 04:08.38 | WIMPy | I have tried tcp some time. I think it was most probably with 1.8. |
| 04:08.45 | Apteryx | What if you un-register and register again, it should print this at least. |
| 04:08.53 | Apteryx | if not, you're sip debug is broken. |
| 04:09.02 | Apteryx | *your |
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| 04:11.13 | Apteryx | anyway, it's midnight here and working tomorrow. I'll be there again tomorrow evening, come again and I'll try to help you as much as I can. |
| 04:11.23 | Penguin | The register and options packets are getting 200 OK no problem. Calls from the phone don't show any sip debug at all. |
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| 04:12.15 | Apteryx | That is indeed very strange! You should at least see the invite and a trying |
| 04:12.20 | WIMPy | Smells like a phone issue. Maybe you explicitely need to set an outboundproxy with ;transport=tcp? |
| 04:12.46 | Apteryx | What kind of phone/softphone is this? |
| 04:13.06 | Penguin | I would have expected to see the INVITE, too. |
| 04:13.07 | Penguin | User-Agent: CSipSimple_vivow-10/r2330 |
| 04:13.14 | Penguin | (on android) |
| 04:13.30 | Penguin | I'll look at the proxy settings. |
| 04:14.04 | Apteryx | Wireshark traces could show you what sip debug is missing |
| 04:14.12 | Penguin | Good idea. |
| 04:14.15 | Apteryx | I've been using this a lot lately. |
| 04:14.55 | Apteryx | talk to you tomorrow. good night/luck :) |
| 04:16.32 | Penguin | wimpy: That fixed it. |
| 04:17.27 | WIMPy | Something that works \o/ |
| 04:17.32 | Penguin | With UDP, the proxy setting was blank. I added the fqdn of the asterisk box to the proxy URI setting and now it makes calls. |
| 04:17.49 | Penguin | Very weird that it worked with UDP when there was no proxy setting. |
| 04:18.42 | Penguin | I think I'm going to clear it and tshark the call to see what it is doing without the proxy URI specified. |
| 04:18.51 | WIMPy | That's what happens if defaults are partially hardcoded and partially copied from other configuration fields. |
| 04:19.25 | WIMPy | Use UDP and not matching a peer as you set transport=tcp in Asterisk? |
| 04:20.01 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 04:20.09 | Penguin | transport is acutally tcp,udp, so both will actually work. |
| 04:20.34 | WIMPy | Ok, in that case my theory fails. |
| 04:21.32 | Penguin | I've always used UDP and always used the general setting of transport=udp. I have never specified the proxy URI value in the phone, and it has always worked. |
| 04:22.51 | Penguin | Today I decided to try out TCP. I set my transport to allow tcp on my peer, forced the transport on the phone to tcp (was set to auto), and re-registered it. |
| 04:23.05 | Penguin | Calls failed with 603 Declined. |
| 04:23.15 | Penguin | No trickery or anything involved. |
| 04:23.42 | Penguin | But taking your suggestion about the proxy, filling in the appropriate value in the proxy URI field... all works now. |
| 04:24.02 | Penguin | tshark shows an INVITE, a 100 Trying, and a 603 Declined. |
| 04:25.10 | Penguin | The registration URI and the proxy URI are now set to the same value and it works. |
| 04:25.54 | WIMPy | Lots of fun stuff... |
| 04:26.01 | Penguin | I wonder if TLS works now, as well. I tried TLS some time ago, and I don't remember what the failure was... but it did not work. |
| 04:26.22 | WIMPy | Yeah, likely the same. |
| 04:26.42 | Penguin | changing to transport=tls,tcp,udp; |
| 04:27.05 | Penguin | Oops. 'tls' is not a valid transport type when tlsenable=no. If no other is specified, the defaults from general will be used. |
| 04:27.52 | WIMPy | Makes sense. |
| 04:28.17 | Penguin | And I didn't make my cert file, so that is another error. |
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| 04:52.44 | Penguin | cpe-e650*CLI> sip reload |
| 04:52.44 | Penguin | SSL certificate ok |
| 04:57.21 | Penguin | I guess the phone doesn't work with TLS. It doesn't even send any packets at all when I press the button to register it. |
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| 05:35.12 | *** join/#asterisk tbsky (~tbsky@dns.annsky.com) |
| 05:39.04 | tbsky | hi! i am facing random dropped call issue with our sip trunk provider. I want to debug the issue. is it enough to open asterisk 'sip set debug peer xxxx' or I need tcpdump the whole sip session? |
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| 05:42.45 | tbsky | or I should open 'core set debug 1' also? |
| 05:44.21 | tbsky | at least I want to find out who hangup the call at first. but at current asterisk log without debugging open, it seems didn't show who hangup the call.. |
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| 05:50.26 | ChannelZ | sip debug is the place to start |
| 05:51.13 | tbsky | ok, thanks ChannelZ! I will start with that. |
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| 08:27.20 | tbsky | oh. it seems the sip trunk didn't drop the call but we did. I saw log that aterisk "pbx.c" hangup, then send sip bye to peer, so that mean we hangup the call right? |
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| 08:56.16 | tbsky | quit |
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| 10:58.07 | [sr] | WIMPy: none of the 2 things worked |
| 10:58.20 | [sr] | WIMPy: even with direct rj45 cable to the shdsl same behavior |
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| 11:25.27 | *** join/#asterisk DelphiWorld (~TheVoicer@openvpn/user/DelphiWorld) |
| 11:25.29 | DelphiWorld | hey |
| 11:25.35 | DelphiWorld | anyone here doing PRN business? |
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| 11:46.29 | phix | DelphiWorld: who? |
| 11:46.56 | DelphiWorld | phix: prn=premium rate numbers |
| 11:47.40 | phix | ah, 1900 or what ever |
| 11:47.51 | phix | (depending on your country) |
| 11:48.17 | DelphiWorld | ph8: do you? |
| 11:48.54 | DelphiWorld | phix: do you? |
| 11:48.58 | DelphiWorld | sory ph8 |
| 11:52.00 | phix | ph8 heh |
| 11:52.09 | phix | What do you want to know about them? |
| 11:52.21 | DelphiWorld | phix: looking for a partner;) |
| 11:52.53 | phix | I see |
| 11:54.26 | phix | well time for bed, nn |
| 11:54.42 | DelphiWorld | see ya phix |
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| 12:08.36 | fixx | guys, i am struggling a bit with auto call monitoring. i have the sox application installed, created /var/spool/asterisk/monitor with the correct pemissions and i use the following on the extension: |
| 12:08.46 | fixx | exten => 5001,1,Dial(SIP/5001, 10) same => n,Monitor(wav,${EXTEN},m) |
| 12:08.55 | fixx | same => n,Monitor(wav,${EXTEN},m) |
| 12:09.35 | fixx | but it doesn't create any wav file |
| 12:09.37 | fixx | Spawn extension (starbright, 5001, 1) exited non-zero on 'SIP/trunk-old-pbx-00000013' |
| 12:09.53 | fixx | that exited non-zero does mean that there is an error right? |
| 12:10.26 | fixx | is there anything else that i need? |
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| 12:15.17 | fixx | ponders rather trying MixMonitor |
| 12:18.04 | fixx | seems as though the result is the same |
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| 12:22.49 | fixx | ah ok, so i have to start the recording before ringing the extension *sigh* |
| 12:42.40 | mjordan | Dial is a blocking application. It doesn't return until the bridge established between the two parties ends. |
| 12:42.49 | mjordan | (really, everything in dialplan is synchronous) |
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| 12:55.24 | Apteryx | Does this: "switching from simple_bridge technology to native_rtp" mean that it's now using direct_media ? |
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| 13:01.04 | Apteryx | Answering myself: from Wireshark, seems like it's still using the bridge. |
| 13:02.53 | Apteryx | Second question: I thought that with registration, I didn't need to open RTP ports for SIP. Is this true? From Wireshark trace again, I see my RTP packets leaving NAT, and nothing coming in. The address in the SDP c= field looks OK. |
| 13:03.08 | Apteryx | *but nothing coming in. |
| 13:11.28 | mjordan | Apteryx: simple_bridge means the media is fully decoded. native_rtp means the RTP stack is bridging the two channels. That can be either a packet to packet bridge - where packets are sent to Asterisk but not fully decoded - or a direct media bridge, where media is re-INVITEd between the participants |
| 13:11.42 | mjordan | Apteryx: Registration != RTP. |
| 13:11.55 | mjordan | Registration merely tells Asterisk the location of a SIP UA. |
| 13:12.05 | mjordan | It has nothing to do with media. |
| 13:12.13 | Katty | file: YOU |
| 13:12.17 | Katty | file: are now a year old. |
| 13:12.21 | Katty | file: HAPPY BIRTHDAY!!!!!!!!!!!! |
| 13:12.26 | file | Katty, hi |
| 13:12.45 | Katty | file: hi! |
| 13:15.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:15.41 | Apteryx | mjordan: OK, but being registered, it can "test" RTP trafic and see if it's going through or not? Is this not what the "Probation passed" messages are about? |
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| 13:17.00 | Apteryx | And it also means, by using signaling, it can sends its RTP listening port to the the NATed party, which, just sending back RTP at this port, will create a NAT rule? |
| 13:17.16 | Apteryx | *PAT |
| 13:18.09 | Apteryx | That's what I understood by "it's not required to port forward RTP ports on your router if you are registering your devices" |
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| 13:43.09 | mjordan | no, that has nothing to do with registration. |
| 13:43.42 | mjordan | Again, registration does not test the media path. It is simply the act of a UA informing another UA about an address it can be contacted at. It is a signaling mechanism. |
| 13:43.53 | mjordan | The probation messages you are seeing are related to the strictrtp setting in rtp.conf. |
| 13:44.14 | mjordan | Asterisk 'locks' onto a source of media for a particular RTP stream. That prevents another media source from 'hijacking' the RTP stream. |
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| 13:44.29 | mjordan | Apteryx: And I'm not sure where you read that. |
| 13:44.35 | mjordan | Apteryx: But that sounds very wrong. |
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| 14:18.23 | Katty | pokes drmessano |
| 14:20.37 | carrar | pokes at the keyboard |
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| 14:22.47 | [TK]D-Fender | pokes the hokey does... |
| 14:22.51 | [TK]D-Fender | </yoda> |
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| 14:30.03 | Katty | hugs carrar |
| 14:30.10 | Katty | carrar: how was your weekend? |
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| 14:31.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:42.01 | Penguin | apteryx: The reason you don't need to forward ports to your phone is much like the reason you don't need to forward ports to your web browser to surf the interweb. |
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| 14:43.29 | file | sorta, kinda, not quite |
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| 14:44.17 | Penguin | Your NAT device knows about RELATED and ESTABLISHED packets. That's all I'm saying. |
| 14:44.35 | WIMPy | Hopefully. |
| 14:44.46 | Penguin | It most cases, it does. |
| 14:45.10 | Penguin | There are cases where it doesn't know and doesn't care, but that's for a different day. |
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| 14:55.49 | jeev | has anyone ipsec interconnected to verizon? |
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| 15:11.16 | MadHatter42 | hello everyone |
| 15:11.22 | MadHatter42 | i was trying to use the chanspy option on freepbx |
| 15:12.40 | [TK]D-Fender | keep it in #freepbx . we will assist you there |
| 15:13.30 | MadHatter42 | thnx |
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| 15:35.14 | carrar | hugs Katty |
| 15:35.20 | carrar | Katty: it was great! |
| 15:38.31 | Katty | that's good! :> |
| 15:41.12 | voip | carrar: you've been MIA for quite some time. |
| 15:41.46 | carrar | I've still been around here |
| 15:41.48 | voip | gasmasters |
| 15:41.54 | carrar | Just not as vocal lately |
| 15:42.00 | voip | afk on the eris |
| 15:42.17 | voip | werd i figured you were over the pond |
| 15:42.22 | carrar | life gets crazy busy sometimes |
| 15:42.25 | voip | we're trying to plan a pond hop now |
| 15:42.34 | voip | you aren't kidding. how's the family? |
| 15:42.43 | voip | hope all is going well for you old timer |
| 15:42.44 | voip | :> |
| 15:42.56 | carrar | doing good, just dropped the child at kindergarten |
| 15:43.07 | carrar | house shopping |
| 15:43.17 | carrar | Time to move out of tent city |
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| 15:46.06 | voip | i thought you lived in the ritz / bellevue |
| 15:46.09 | voip | tent city? |
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| 16:13.32 | Katty | carrar: good luck house shopping!!! |
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| 16:19.30 | *** join/#asterisk Pegasus_RPG (~Icedove@pool-141-155-60-246.nycmny.fios.verizon.net) |
| 16:20.02 | Pegasus_RPG | Hello. Does anyone here run their * behind pfSense? I'm having trouble getting audio though I can make and receive calls. |
| 16:20.18 | Pegasus_RPG | And it worked fine behind my old Cisco PIX firewalls |
| 16:20.36 | Pegasus_RPG | I suspect the magic was the PIX's sip fixup functionality |
| 16:20.55 | Pegasus_RPG | which I understand changes the IP addresses to the translated ones in the paackets on the fly |
| 16:21.19 | WIMPy | You don't want that. It usally does more harm than good. |
| 16:22.04 | Pegasus_RPG | WIMPy: okay. In any case, I'm on pfSense boxes now and have done nothing other than the usual 1:1 NAT (I have a block of external IPs) and allowing SIP and RTP ports |
| 16:22.59 | Pegasus_RPG | SO what am I missing? |
| 16:23.48 | Pegasus_RPG | I have externhost set in my sip.conf to a FQDN of the machine |
| 16:25.03 | Pegasus_RPG | which resolves to the correct address depending on whether you query from inside the LAN or on the Internet |
| 16:27.17 | [TK]D-Fender | Which is pointless |
| 16:27.37 | [TK]D-Fender | it needs to resolve to the PUBLI IP on the * box itself |
| 16:28.30 | [TK]D-Fender | As for audio you also need to ensure your forwarding RTP, have PORT RANDOMIZATION disabled, and no other ALG in play |
| 16:29.31 | Pegasus_RPG | meaning doing 'dig pbx.example.com' on the * should give its public address? |
| 16:29.42 | [TK]D-Fender | on the * box itself |
| 16:31.13 | Pegasus_RPG | ahh, port randomization is likely the main problem as it's default on pfSense. Fixing... |
| 16:31.56 | Pegasus_RPG | Just for SIP or for RTP too? |
| 16:32.17 | [TK]D-Fender | kill it |
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| 17:05.10 | pjensen00 | I'm doing originates via ARI and I can get it to ring my SIP endpoint no problem. However, I'm trying to add a predial handler to it and am failing. |
| 17:05.31 | pjensen00 | I add the "b(context^exten^priority)" portion and I keep getting 500's back |
| 17:06.08 | pjensen00 | Is it supposed to be in the 'appArgs' query parameter? I tried putting it there but didn't have much luck. |
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| 17:19.47 | pjensen00 | looks like asterisk can't read the parenthesis. |
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| 17:31.27 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:31.27 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.0 (2014/09/24), 1.8.31.0 (2014/09/24); Standard: Asterisk 12.6.0 (2014/09/24); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 17:38.19 | mjordan | pjensen00: er |
| 17:38.34 | mjordan | why would a pre-dial handler, which is a specific construct to Dial, work with ARI? |
| 17:38.53 | mjordan | Consider this: do you think the same thing would work with the Originate AMI action? |
| 17:40.29 | Pegasus_RPG | Okay, I've tried disabling port randomization but it doesn't have any effect. To clarify: when a call comes in, they hear the VM greeting and can leave a message. If I call in from the outside, I too can check the VM. |
| 17:41.06 | Pegasus_RPG | But if I try to answer a call from a softphone or a call forwarded to my cell phone, I hear nothing (and presume the other party doesn't either.) |
| 17:41.19 | [TK]D-Fender | Presume = bad |
| 17:41.22 | Pegasus_RPG | If I call in to the voice mail from a soft phone, I hear nothing |
| 17:41.26 | [TK]D-Fender | and you haven't shown us anything |
| 17:41.41 | Pegasus_RPG | and neither does it (dialing digits don't register.) |
| 17:41.46 | Pegasus_RPG | What would you like to see? |
| 17:42.07 | [TK]D-Fender | debug for calls, firewall settings, * core settings, etc |
| 17:42.47 | Pegasus_RPG | I just verified and the other party doesn't hear me either if I take a forwarded call |
| 17:42.55 | Pegasus_RPG | collecting information... |
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| 18:33.47 | pbxman | hello I need some advice on developing IVRs, please |
| 18:34.42 | pbxman | What are the best ways to develop them in Asterisk? Php, Java, LUA? |
| 18:35.29 | [TK]D-Fender | ~best |
| 18:35.30 | infobot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 18:35.37 | [TK]D-Fender | There is no such thing as "best" |
| 18:35.45 | [TK]D-Fender | and "IVR's" is ratehr vague |
| 18:35.54 | [TK]D-Fender | rather* |
| 18:37.37 | mjordan | the answer, of course, is "Flock of Seagulls". |
| 18:37.42 | ovoshlook | hello! I find a vay how to do transfer through REST interface. |
| 18:38.04 | mjordan | ovoshlook: blind or attended? |
| 18:38.05 | pbxman | well I'm migrating an IVR app from VXML to something else. I'm thinking about using Java or PHP maybe LUA. It must be easy, dynamic, not hard to maintain, etc. I was wondering what is the first option for people to develop a complex IVR now with Asterisk |
| 18:38.31 | [TK]D-Fender | pbxman: "complex" is rather subjective |
| 18:38.33 | mjordan | I think language choice is a personal decision. |
| 18:38.48 | [TK]D-Fender | ^ as well |
| 18:39.04 | mjordan | Whether or not you do it in the dialplan or through AGI (or some other interface) seems to be the question. And that's very subjective still, as it depends on the requirements of your system. |
| 18:40.25 | ovoshlook | @mjordan does not matter. For example blind at first time |
| 18:40.29 | mjordan | ovoshlook: generally, ARI doesn't provide transfers itself. Channel drivers, of course, can still perform transfers, in which case ARI will emit events that tell you what just happened to your channel. |
| 18:40.32 | pbxman | mjordan, how would you do it, if you have to call Web Services all the time, use databases, Mysql + Oracle and so on. |
| 18:40.46 | mjordan | A blind transfer, in Asterisk, is rather trivial. You just release a channel to the dialplan using continue |
| 18:41.45 | [TK]D-Fender | pbxman: then AGI in a language of your choice |
| 18:41.56 | mjordan | With an attended transfer, you can simply move a channel between the bridges. You control the bridges and the channels, so you can remove the transferer from their bridge, start MoH on the destination, make a new bridge, and perform an originate to the transfer target. |
| 18:42.13 | mjordan | pbxman: what [TK]D-Fender said (who will have a better opinion on this question than me) |
| 18:42.33 | pbxman | [TK]D-Fender, I think of using a FastAgi + Java it's my best option here, the Asterisk Java APi is quite good, the problem is that I would like to be able to modify the code without having to restarting the application |
| 18:42.39 | mjordan | in fact, the concept of an 'attended' transfer is up to you to define. |
| 18:42.52 | mjordan | you could have an attended transfer always be a 3-way party. |
| 18:43.03 | mjordan | let everyone talk it out, then have the transferer drop out when they want. |
| 18:43.05 | *** join/#asterisk hecatae (~philip@host-78-151-178-31.as13285.net) |
| 18:43.38 | mjordan | you could make it so that the transferer could talk to both parties at once while the target and destination can't talk to each other until the transfer is completed (although that would require some interesting usage of bridges/channels) |
| 18:43.50 | mjordan | ARI turns Asterisk into tinker toys: you get to build what you want out of it |
| 18:44.48 | mjordan | pbxman: your requirement would be difficult for any scripting language. |
| 18:45.18 | mjordan | pbxman: any scripting language generally loads things into memory for the interpreter to execute. Modifying the file and expecting the interpreter to re-parse the source is generally not how things work. |
| 18:45.32 | pbxman | mjordan, I'm also concerned about performance, we are getting about 30K calls a day |
| 18:45.33 | mjordan | pbxman: if you need that level of flexibility, you should store the 'mutable' parts in a database or some other storage |
| 18:46.30 | pbxman | i'm also thinking about creating my own JSP - Java layer based on the current java API but that could take me a while |
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| 18:47.51 | pbxman | mjordan, yes, I have something like that in mind. Perhaps with an web application was backoffice for the IVR in which users could make changes live |
| 18:48.11 | pbxman | *was = as |
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| 19:00.06 | pjensen00 | mjordan: Hrm I guess you're right. I'm just trying to figure out how to add sip headers when I originate a call through ARI. Was hoping to use the predialer but I'll have to try something else. |
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| 19:45.44 | mjordan | pjensen00: you can do it using the body variable parameters |
| 19:45.57 | mjordan | pjensen00: those are evaluated and added to the channel prior to performing the outbound INVITE request |
| 19:46.19 | pjensen00 | Man, I really am trying to make this way too hard on myself. Thanks sir! |
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| 20:27.51 | MadHatter42 | any asterisks gurus around |
| 20:27.56 | MadHatter42 | <PROTECTED> |
| 20:28.09 | MadHatter42 | i'm getting a bit desperate >_< |
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| 20:45.59 | newtonr | MadHatter42, I see retransmissions with a timeout |
| 20:46.17 | newtonr | Follow call-ID 14f0958d2c4d2c992030ec4127af35e4 through |
| 20:46.24 | MadHatter42 | i've disabled all firewalls and stuff |
| 20:47.02 | newtonr | then run a packet capture on your machine to see if any responses to that call are getting back to the machine |
| 20:47.20 | MadHatter42 | i have two numbers from the same provider |
| 20:47.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:47.37 | newtonr | if you don't see anything there, and you are completely exposed, then either you are telling the far end the wrong address to send stuff to, or the problem exists outside your box and network |
| 20:47.38 | MadHatter42 | http://pastebin.com/ecnF70HB full trace |
| 20:47.57 | MadHatter42 | i've got two numbers |
| 20:48.02 | newtonr | I Just looked at that. |
| 20:48.16 | MadHatter42 | one inbound toll free, and one for dialing |
| 20:48.30 | MadHatter42 | the inbound works, i can receive calls |
| 20:48.40 | newtonr | Google packet captures and wireshark. :D |
| 20:49.01 | MadHatter42 | i have more experience with tcpdump |
| 20:49.08 | MadHatter42 | what am I looking for ? |
| 20:49.13 | newtonr | Then compare the IP addresses used in both the SIP traffic of your working calls with your failing ones. |
| 20:49.54 | [TK]D-Fender | MadHatter42: they aren't answering |
| 20:50.19 | MadHatter42 | but i called with my mobile phone and it worked fine |
| 20:50.26 | [TK]D-Fender | means nothing |
| 20:50.40 | [TK]D-Fender | they are not answering your call out from your server |
| 20:51.05 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
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| 20:51.46 | MadHatter42 | what does that exactly means ? |
| 20:52.10 | MadHatter42 | its a provider block or receiver block ? |
| 20:52.15 | [TK]D-Fender | what part of "they are not answering" is unclear? |
| 20:52.40 | [TK]D-Fender | You are originating a call to your provider. They are NOT responding whatsoever |
| 20:52.41 | *** join/#asterisk spectr3 (~spectr3@unaffiliated/spectr3) |
| 20:52.56 | MadHatter42 | but this happens with every number |
| 20:53.02 | newtonr | MadHatter42, If your network isn't blocking the response then I'd call your SIP provider and have them tell you what they see from their side. |
| 20:53.04 | MadHatter42 | numbers that I know they work |
| 20:53.07 | [TK]D-Fender | ... |
| 20:53.14 | [TK]D-Fender | this is not a problem with the NUMEBR you are dialing. |
| 20:53.18 | [TK]D-Fender | If your PROVIDER |
| 20:53.27 | [TK]D-Fender | it's |
| 20:53.33 | MadHatter42 | i use didforsale |
| 20:53.38 | [TK]D-Fender | I zsee that |
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| 20:56.06 | MadHatter42 | 209.216.15.70 +12392054843 486729483_41556 (nothing) No Tx: BYE didforsale |
| 20:56.30 | MadHatter42 | i tried some combinations of the config they gave me |
| 20:57.04 | MadHatter42 | now it rings but i cant hear anything |
| 21:00.04 | [TK]D-Fender | Reliably Transmitting (NAT) to 209.216.2.212:5060: <- for starters your provider is NOT behind NAT... this should be "nat=no". |
| 21:00.40 | MadHatter42 | ok just a sec |
| 21:17.35 | Penguin | If I don't use TLS but do use TCP, asterisk complains that I am not using TLS. Is that normal? |
| 21:18.31 | WIMPy | doesn't remember such a message. |
| 21:20.39 | Penguin | WARNING[23329]: chan_sip.c:15444 in register_verify: peer 'D8B377ABCDEF' HAS NOT USED (OR SWITCHED TO) TLS in favor of 'TCP' (but this was allowed in sip.conf)! |
| 21:21.09 | WIMPy | Are you using the wrong port? |
| 21:21.26 | Penguin | I wouldn't think so. |
| 21:21.42 | Penguin | I can't imagine it would work if I used the wrong port. |
| 21:22.38 | WIMPy | I have no clue if it starts encrypted communication right away or if it uses somethin like starttls. |
| 21:23.13 | Penguin | Since I am not using TLS, it will never start encrypted communications. Therefore the warning message. |
| 21:23.40 | WIMPy | Well, if you did, obviousely. |
| 21:24.06 | Penguin | I want to know if the warning is normal when using TCP but not using TLS. |
| 21:24.21 | WIMPy | If it works the later way, you might be able to not use it even on the port that's meant to use it. |
| 21:24.21 | Penguin | And if it is normal, can I remove it? I don't need to see that message -- I know I'm not using TLS. |
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| 21:58.27 | pjensen00 | mjordan: I've been trying to find out why I haven't been able to set variables in the originate REST request and I think it's because the body variable "variables" isn't in the right encoding. I've tried encoding it in JSON to work around that and a packet capture shows the following text |
| 21:58.28 | pjensen00 | http://pastebin.com/Wjhx7h7Q |
| 21:58.43 | pjensen00 | I can't tell what's wrong with it |
| 21:58.56 | pjensen00 | just trying to set the caller ID to Frankenstein at the moment |
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| 22:13.51 | pjensen00 | If anyone has an example of the content body for the REST request with the CALLERID(name) variable set I would greatly appreciate it. Currently trying to figure out why app/extension/endpoint etc work but my "variables" won't |
| 22:16.22 | mjordan | pjensen00: they are JSON, but they are only specified as body parameters, not query parameters |
| 22:16.25 | mjordan | looks for an example |
| 22:20.43 | pjensen00 | danke |
| 22:22.58 | mjordan | it figures that the test that covers this use case doesn't log out the request |
| 22:23.00 | mjordan | sighs |
| 22:23.07 | mjordan | let me see if I can't get it to dump it out in a nice pretty format |
| 22:23.17 | mjordan | but if you'd like some example python code using the requests library: |
| 22:23.33 | mjordan | http://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/rest_api/channels/originate_with_vars/originate_with_vars.py |
| 22:24.57 | pjensen00 | I shall check this out |
| 22:25.48 | pjensen00 | yeah that example is what I'm trying to do, but my lib is doing it's own converstion and mashing the data I think. |
| 22:28.56 | mjordan | I'd expect an equivalent CURL command to look something like this: |
| 22:28.57 | mjordan | curl -H "Content-Type: application/json" -d '{"variables": {"CALLERID(name)": "Alice"} }' http://localhost:8080/channels?app=haa&extension=100&context=default&endpoint=PJSIP%2Fdefault_outbound_endpoint%2Fsip%3Ac4p5b1%40cc.voicegw.varvidate.com |
| 22:29.04 | mjordan | note that the variables aren't part of the query |
| 22:29.12 | mjordan | they're passed in as a body parameter to the post |
| 22:29.51 | pjensen00 | Thanks, I'll keep hacking away until I figure it out. |
| 22:29.56 | mjordan | right now, you're passing them in as query parameters... so. Yeah. Your underlying library shouldn't do that. |
| 22:30.09 | mjordan | which library are you using? |
| 22:30.19 | pjensen00 | http://search.cpan.org/dist/libwww-perl/lib/LWP/UserAgent.pm |
| 22:30.25 | mjordan | ah |
| 22:30.33 | pjensen00 | The issue seems that when it encodes the body, it only encodes one level deep |
| 22:30.58 | pjensen00 | So the container of 'variables' gets left to birds |
| 22:31.04 | mjordan | Hm. |
| 22:31.23 | mjordan | http://search.cpan.org/dist/libwww-perl/lib/LWP/UserAgent.pm |
| 22:31.24 | mjordan | er |
| 22:31.30 | mjordan | http://stackoverflow.com/questions/4199266/how-can-i-make-a-json-post-request-with-lwp |
| 22:31.33 | mjordan | that looks relevant |
| 22:32.02 | pjensen00 | hrm, I will give that a shot thanks |
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| 23:37.58 | Apteryx | Good evening. |
| 23:38.07 | Apteryx | Penguin: still no joy with TCP? |
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