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| 00:21.56 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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 |
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| 12:36.42 | LunaLovegood | Sometimes when I send a fax, the destination server detects the CNG and sends a re-invite for T38, but I disabled udptl on my end. If I get the messages "chan_sip.c: Failed to initialize UDPTL, declining image stream", does that mean the call fails, or will the fax still go through in PCMU format? (which is what I want) |
| 12:37.27 | LunaLovegood | I'd normally use PCMU only for all calls, voice or not. |
| 12:40.50 | LunaLovegood | Is there a SIP header or something I can send in my INVITEs and in my responses to indicate I don't want T38 re-invites? |
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| 13:45.47 | Samot | T.38 is not done on the first INVITE |
| 13:46.02 | Samot | It's done when the destination returns it can do or wants T.38 |
| 13:48.21 | Samot | If the other side only wants T.38 you're not going to have much luck. |
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| 13:53.48 | DocScrutinizer05 | ~logs |
| 13:53.48 | infobot | All conversations are logged to http://infobot.rikers.org/%23asterisk/ Lines starting with spaces are not logged. Logs are updated daily. |
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| 13:55.01 | DocScrutinizer05 | ~#asterisk logs |
| 13:55.01 | infobot | i guess #asterisk logs is http://ibot.rikers.org/%23asterisk/ or http://ibot.rikers.org/stats/asterisk.html.gz |
| 13:55.43 | DocScrutinizer05 | ~#asterisk logs is also https://botbot.me/freenode/asterisk/ |
| 13:55.43 | infobot | okay, DocScrutinizer05 |
| 13:58.52 | DocScrutinizer05 | ~logs |
| 13:58.52 | infobot | i heard logs is http://ibot.rikers.org/%23asterisk/ or http://ibot.rikers.org/stats/asterisk.html.gz, or https://botbot.me/freenode/asterisk/ |
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| 15:21.42 | monsterco | Hi - is there any way I can turn on Verbose for a DID number and ignore the rest of inbound calls on other DIDs? |
| 15:23.41 | [TK]D-Fender | no |
| 15:26.11 | monsterco | that's a shame |
| 15:26.16 | monsterco | any creative way you can think of |
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| 15:30.52 | [TK]D-Fender | no |
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| 16:30.19 | luxio | is there a way to put someone on permanent hold so that they're on hold but the line will be free |
| 16:32.03 | rfr__ | <[TK]D-Fender> If I could provide you access to my * machine would you be willing to log in for a fee and fix my problem. Been banging my head for week trying to get * dhadi install to work to no avail. |
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| 16:33.41 | [TK]D-Fender | luxio, your description doesn't add up and we have no details |
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| 16:34.55 | igcewieling | luxio: you can transfer someone to an extension which plays MoH forever. |
| 16:35.27 | igcewieling | "line" can have at least 3 definitions in telecom. |
| 16:58.29 | rfr__ | <[TK]D-Fender> I take it your not interested? If not no big deal. But just wanted to make sure you saw the offer. |
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| 18:04.36 | *** join/#asterisk SpaceInvaders (~SpaceInva@136.57.54.178) |
| 18:05.24 | SpaceInvaders | Can someone please respond with pointers on where I can find a comparison of SPs I can connect to my asterisk pbx? |
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| 18:06.00 | SpaceInvaders | fwiw I have been googling and only the ads seem to answer that question and they are all biased or completely misrepresenting what they offer. |
| 18:06.10 | SpaceInvaders | Thanks! |
| 18:08.55 | [TK]D-Fender | I don't know of any sites that really compare them in ways that are meaningful to you |
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| 18:09.25 | [TK]D-Fender | As for representing what they offer each provider does that on their oown pages quite well 99% of the time |
| 18:09.50 | [TK]D-Fender | for ones we commonly recommend here: voip.ms, vitelity.net, flowroute.com, etc |
| 18:11.44 | igcewieling | vitelity is now owned by Onvoy, which is now owned by Intelliquent, I've been with Vitelity since Oct 2005, but never had a high call volume. |
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| 21:02.25 | dritskalle | (pjsip) is there a way to allow codecs for media point but only if all parties support it? like i want to allow endpoints talking opus, but i dont want to transcode |
| 21:04.02 | Samot | It's an offer/answer relationship. |
| 21:05.09 | Samot | One side is going to offer the codecs they support, in the order of preference and the other side will answer with the same and then each side determines which is the best match on their end. |
| 21:06.15 | Samot | And if you're using opus the only time you won't be transcoding is between devices. |
| 21:06.43 | Samot | Opus is not a PSTN accepted codec. You'll be transcoding calls to/from the provider |
| 21:08.11 | igcewieling | Eventually I expect the PSTN to support g722 (at least cellphones) |
| 21:08.58 | Samot | It does it other parts of the world. |
| 21:09.10 | Samot | But the NA network is much older. |
| 21:09.15 | Samot | More of a beast. |
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| 21:09.41 | igcewieling | "G.722 has also been widely used by radio broadcasters for sending commentary-grade audio over a single 56 or 64 kbit/s ISDN B-channel (the least significant bit is dropped on 56kb circuits)." |
| 21:10.00 | sibiria | it's an endpoint thing, not a bandwidth thing |
| 21:10.00 | dritskalle | i know that.. but i have two asterisk boxes that trunk between - i want to support calls across opus/opus with MT.. but to PSTN i dont want to.. today all i canget working is that transcoding kick in |
| 21:10.39 | Samot | So make opus the first option in the codecs for the trunks |
| 21:10.43 | Samot | That's easy. |
| 21:10.48 | dritskalle | so a call from a user to pstn i should only offer ulaw - or decline transcoding anyway.. but a call over the inside trunk i like to allow opus |
| 21:10.54 | Samot | If it's a Asterisk to Asterisk peer, just make it the first codec. |
| 21:11.07 | dritskalle | yes - but that invokes transcoding for a pstn call |
| 21:11.10 | Samot | Let's be clear.. |
| 21:11.16 | Samot | You have to transcode the PSTN call |
| 21:11.29 | Samot | If you're going to use Opus internally. |
| 21:11.47 | dritskalle | okay.. i guess that means opus only if there is no MT |
| 21:11.51 | Samot | If you do not offer or accept the codecs the provider supports there is not call. |
| 21:13.27 | dritskalle | i guess the only way to avoid is to not use MT on the local but setup rtp to sbc and direct locally.. |
| 21:13.49 | Samot | What does that mean? |
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| 21:14.53 | dritskalle | <PROTECTED> |
| 21:15.49 | Samot | What type of SBC? |
| 21:15.49 | dritskalle | the original sentence was messed up - only made sense in my head :p point taked though one cant offer MT/sdp optionally based on codec |
| 21:16.04 | dritskalle | whatever really - cisco cube or avaya-something |
| 21:16.22 | Samot | Do they support Opus? |
| 21:16.38 | Samot | Your SBC will need Opus support. |
| 21:16.40 | dritskalle | no |
| 21:16.54 | Samot | You can't be all Opus and make calls to the PSTN |
| 21:16.58 | dritskalle | no i only want opus for all inside calls - but also over my internal trunk |
| 21:17.05 | Samot | Unless you find a provider willing to do the transcoding for you. |
| 21:17.29 | Samot | OK |
| 21:17.43 | Samot | So when you trunk between the two asterisk systems, you set their codecs to Opus. |
| 21:17.58 | dritskalle | that would be nice actually if cisco ucm and avaya whatever did support it, less load on me |
| 21:18.03 | dritskalle | yes |
| 21:18.30 | Samot | How many calls are we talking about here? |
| 21:18.47 | dritskalle | varies - deployments up to 12K users |
| 21:19.00 | dritskalle | max 100 simul calls per box |
| 21:19.14 | Samot | per box? Which box? |
| 21:19.15 | Samot | Asterisk? |
| 21:19.20 | dritskalle | yes |
| 21:19.27 | Samot | So the SBC could be doing 200 |
| 21:19.45 | dritskalle | the sbc would belong to the customer - so it be sized to their needs |
| 21:20.14 | dritskalle | my lab has a 3800 vice gw router |
| 21:20.26 | dritskalle | s/vi/voi/ |
| 21:20.33 | dritskalle | haha |
| 21:21.46 | dritskalle | large customers i wouldnt offer opus tgough - cant take the chance on all that transcoding.. but most are in the <=40 simul calls area and never had any issues |
| 21:22.04 | jamesaxl | HEllo |
| 22:09.30 | *** join/#asterisk infobot (ibot@rikers.org) |
| 22:09.30 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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 |
| 22:34.28 | SpaceInvaders | @[TK]D-Fender & igcewieling - thanks! |
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