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| 00:42.26 | vook | I've got two servers connected to bandwidth.com, one an old system that is currently in use, and the other a new server to which we'll be migrating. On the new server, I set up one of our unused numbers on an inbound route and it is working as expected. The question, is adding the other inbound routes dangerous at this point - how will bandwidth.com ultimately determine which server receives the call? |
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| 01:03.42 | [TK]D-Fender | inbound route plays no role in that |
| 01:04.39 | vook | I don't understand that - I was unable to recieve incoming calls via a public number until I created an inbound route and associated it with an extension. |
| 01:04.43 | [TK]D-Fender | Calls go to whichever server registered last with the account related to the DID in question |
| 01:04.52 | vook | o |
| 01:04.53 | vook | ok |
| 01:05.05 | [TK]D-Fender | inbound route PROCESSES the call... once it has actually arrived |
| 01:05.13 | [TK]D-Fender | You asked who gets it |
| 01:05.21 | [TK]D-Fender | the one who registers gets it |
| 01:05.39 | [TK]D-Fender | And if both register then they will be playing tug-of-war over it |
| 01:06.38 | vook | ok, thanks |
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| 01:49.23 | linuxfool | hey everyone |
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| 02:35.36 | DMJC | I've got two sip peers registered but sip show registry lists nothing |
| 02:35.38 | DMJC | is that normal? |
| 02:36.07 | file | sip show registry shows *outbound* SIP registrations |
| 02:36.21 | DMJC | ah thanks |
| 02:36.29 | DMJC | and peers are my inbounds :) |
| 02:37.48 | file | they can reflect information about an inbound registration, yes |
| 02:38.02 | DMJC | what are the free codecs? |
| 02:38.14 | DMJC | I've just realised I'm trying to use ulaw on phones which don't support it |
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| 03:27.27 | Penguin | Phones that don't support ulaw? I've never heard of that. |
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| 04:54.14 | chris349 | On Asterisk 13 is it better to use a SIP trunk or PJSIP trunk? |
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| 05:36.49 | ChannelZ-Wk | define "better" |
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| 05:38.16 | ChannelZ-Wk | PJSIP is the new hotness. But it's also an either/or (well, that's not strictly true I don't think, if you ran chan_sip and chan_pjsip on different ports maybe they coexist together, I've never tried) |
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| 06:44.58 | joako | I have installed the latest AsteriskNOW 64-bit version but it is totally broken in regards to using Digium phones. Is there any particular version that will work well with Digium phones? |
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| 06:55.59 | Penguin | Do you need FreePBX? |
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| 07:31.54 | ChannelZ | Who does! |
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| 07:51.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 13:46.34 | bsdice | Good news I finally figured out why ptime on CSipSimple will not be respected (to decrease packet overhead): You have to pick only ONE codec in CSipSimple and then define the interval server side, e.g. allow=g729:50,speex,g722 |
| 13:47.08 | bsdice | SRTP in UDP will then have 100-byte packets i.e. 50ms of audio per packet |
| 13:48.08 | bsdice | so now I got everything down to duplex 40 packets/sec (20 per direction) and 32kbit/s (16 per direction). Works over GPRS now |
| 13:50.50 | bsdice | too bad codec2 can't be parameterized with regard to ptime, would love to lower overall bitrate per direction below 8kbit/s including IP/UDP/SRTP overhead |
| 13:51.44 | WIMPy | The S&M scene is alive. |
| 13:55.21 | bsdice | Using the Codec2 1200bps mode, I can squeeze about 50 simultaneous phone calls into 64kbit/s ISDN |
| 13:57.03 | bsdice | will sound like going over a Thuraya sat phone though |
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| 14:01.11 | pzn | I have a phone with hardware bugs, that is wrongly registering at sip extension 5430 (stealing other phone extension / I don't have phisical access to it today / its port 80 is locked). tought about a workaround... is there any way in sip.conf to "restrict" which IPs can try to register in sip/5430 account? |
| 14:01.25 | WIMPy | That must be without IP. |
| 14:02.05 | WIMPy | pzn: Yes |
| 14:02.16 | bsdice | pzn in sip.conf of peer write deny=0.0.0.0/0.0.0.0 + permit=192.168.1.0/255.255.255.0 to allow only that subnet |
| 14:03.57 | pzn | bsdice, worked as expected! problem solved for the weekend :-) tks! |
| 14:04.06 | bsdice | np cheers |
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| 14:21.14 | bsdice | 450 bits/sec does not sound quite like me anymore http://www.rowetel.com/downloads/codec2/robust4/ve9qrp_450.wav |
| 14:22.21 | bsdice | so I'll settle for 1200 bps or a little more after all. Raytheon has 300 bps voice going for Blackhawk voice comm that supposedly will not make you sound like Stephen Hawking |
| 14:29.49 | WIMPy | You should use a combination of speech recognition and TTS. That way yoy could get away with far less. |
| 14:30.38 | bsdice | I am not smart enough to implement that |
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| 15:27.51 | goseeped | have you experience with this : SRTP unprotect failed with: authentication failure 10 |
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| 15:44.19 | bsdice | goseeped not much, maybe this https://issues.asterisk.org/jira/browse/ASTERISK-22911 |
| 15:45.43 | bsdice | also https://issues.asterisk.org/jira/browse/ASTERISK-16298 |
| 15:54.29 | goseeped | thanks bsdice but its wear because this is happening on asterisk 13 pjsip sip stack ! with webrtc peers, maybe a regression ? |
| 15:57.34 | bsdice | are you using sipml5 by chance? |
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| 15:58.01 | bsdice | I don't use webrtc, sounds like this https://code.google.com/p/sipml5/issues/detail?id=44 |
| 16:00.34 | bsdice | maybe AES_CM_128_HMAC_SHA1_32 <-> AES_CM_128_HMAC_SHA1_80 compatibility problem |
| 16:00.52 | bsdice | if that's not it, open a ticket... |
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| 17:31.14 | bsdice | I just found this http://www.essentialpatentblog.com/wp-content/uploads/sites/234/2013/02/AIM-IP-v-Digium.pdf and according to http://en.wikipedia.org/wiki/Term_of_patent_in_the_United_States this patent should expire on August 24th 2016 |
| 17:31.52 | bsdice | were there any press releases by Digium over this complaint? |
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| 19:03.39 | krapper | need some assistance on getting a call to proceed with t.38 |
| 19:04.13 | krapper | 11.6-cert7 |
| 19:08.03 | WIMPy | ~ask |
| 19:08.03 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 20:44.01 | DMJC | I've got two snom320 handsets I'm trying to have call each other. I've got them both appearing in the sip peers list |
| 20:44.10 | DMJC | but the handsets are still saying they are not registererd |
| 20:44.34 | [TK]D-Fender | "sip show peer X" |
| 20:44.54 | [TK]D-Fender | show us each, and the call attempt. And while you're there, reboot one and watch it re-register |
| 20:44.59 | [TK]D-Fender | "sip set debug on" |
| 20:45.01 | [TK]D-Fender | ~pb |
| 20:45.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:45.02 | [TK]D-Fender | ^^^ |
| 20:45.48 | DMJC | thanks |
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| 21:03.25 | DMJC | http://pastebin.ca/2902859 |
| 21:03.38 | DMJC | pretty sure it's to do with the 404 errors on the debug output |
| 21:05.47 | DMJC | http://pastebin.ca/2902860 |
| 21:06.05 | DMJC | second link is sip show peer phonename |
| 21:09.23 | [TK]D-Fender | those are options (qualify" requests used as a keep-alive and to checkif they are still responding. |
| 21:09.51 | [TK]D-Fender | I don't see the reboot & reregister attempt, nor a call. |
| 21:10.05 | [TK]D-Fender | get the rest. Also, pastebin your peer's making only the secret. |
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| 23:42.20 | krapper | looking for some assistance in understanding t.38... my SIP provider supports it and it appears on inbound calls, Asterisk is detecting... on outbound calls, do I need to set something in my dial strings in to indicate to SIP provider that particular call will require t.38? |
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