| 01:50.09 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 01:50.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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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| 05:40.29 | persiever | Hello, Is there a way in asterisk to send Sip video (h263 & alaw) in h223 protocol? Can we add functionality to stream video as h223 data |
| 05:43.26 | persiever | Actually I have got 3g Ip connectivity from Telecom provider and after SDP session Ok. The media server of Telecom open a H223 Channels in Clearmode codec. So I was just asking can we develop a functionality that will send video+audio data in h223 stream |
| 05:47.46 | persiever | join |
| 05:47.58 | [TK]D-Fender | * doesn't do H.223 |
| 05:48.24 | [TK]D-Fender | There is also no "SIP video". SIP negotiates RTP |
| 05:48.31 | [TK]D-Fender | RTP is the transport |
| 05:48.36 | [TK]D-Fender | for the media |
| 05:52.36 | persiever | Ok my scenario is that I am sending Clearmode codec in SDP for sip invite (Please note I have kept clearmode in Passthrough only as suggested by Asterisk and followed steps on how to support clearmode in asterisk). The SIP server replies with Clearmode codec and start the RTP session. In the RTP session I can see Payload Type= Clearmode/8000 and H.223 ITU-T protocol. I know asterisk cannot process such RTP stream. |
| 05:53.07 | persiever | So I am in search of Solution where I can send the RTP stream in same way the Channels opened by Remote SIP server |
| 05:54.55 | MelMalik | persiever: Nobody can help you |
| 05:55.51 | persiever | @MelMalik: ok |
| 05:57.26 | persiever | As h.223 is 3g video and sip is ip video. There must be a gateway which will convert 3g video to ip video and vice versa |
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| 06:06.58 | persiever | I have seen chan_ooh323. what it does? How I can use in send h323 protocol |
| 06:10.30 | MelMalik | 223 != 323 |
| 06:10.42 | MelMalik | 323 is microsoft videoconferencing |
| 06:10.56 | [TK]D-Fender | no. |
| 06:11.08 | [TK]D-Fender | H.323 is a standard separate for any specific company |
| 06:11.09 | MelMalik | [TK]D-Fender: to which |
| 06:11.17 | MelMalik | [TK]D-Fender: oh, i was just going by what i know it frmo |
| 06:11.30 | [TK]D-Fender | your statment is incorrect. |
| 06:12.00 | MelMalik | the takeaway you took from it is incorrect |
| 06:12.12 | MelMalik | the statement itself is simply based on what i know about its usage |
| 06:12.23 | [TK]D-Fender | You put the words implying it's "theirs" |
| 06:13.08 | [TK]D-Fender | Microsoft makes all sorts of software, as do others. They may speak all sorts of protocols |
| 06:13.24 | MelMalik | and again, the point stands that 323 has not lip nor ass to share with 223 |
| 06:13.38 | MelMalik | and now i remember why i didn't fucking stay here TKD |
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| 06:23.36 | persiever | h.323 is 3g video protocol and many Telecom operator has adapted this protocol as 3g324m protocol. |
| 06:24.17 | [TK]D-Fender | https://www.voip-info.org/asterisk-h324m/ |
| 06:24.32 | [TK]D-Fender | Was something I found a few minutes ago for 3rd party odd bits |
| 06:24.46 | [TK]D-Fender | no idea if there is a functional modern adaptation that can do anything useful or not |
| 06:25.04 | MelMalik | persiever: no? |
| 06:25.15 | MelMalik | persiever: have you had your adderall today? |
| 06:25.32 | MelMalik | persiever: from where do you hail |
| 06:27.32 | persiever | <[TK]D-Fender>: I found the above link https://www.voip-info.org/asterisk-h324m/ interesting. I have installed asterisk 16 on server. How to configure h324 now? |
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| 06:28.00 | [TK]D-Fender | Go drill the author's resources for this |
| 06:28.01 | [TK]D-Fender | . |
| 06:28.09 | [TK]D-Fender | it was never part of main-line * |
| 06:28.10 | persiever | As I can't find any proper documentation for this |
| 06:28.52 | MelMalik | https://www.news.com.au/travel/travel-updates/incidents/scottish-excage-fighter-suffering-recurring-flashbacks-after-taxi-driver-bashing-in-melbourne/news-story/28f4d95c3333d009b2f68739085067d7 |
| 06:29.10 | MelMalik | that's as relevant as what persiever is saying so... |
| 06:37.55 | persiever | MelMalik: may be you are right at your place. I need to know if the code for h324 is available on latest asterisk release i.e, 16 . |
| 06:40.24 | [TK]D-Fender | It's not part of * itself |
| 06:40.30 | [TK]D-Fender | And the only stuff I see is a decade old |
| 06:40.39 | [TK]D-Fender | So I doubt it's compatible as-is |
| 06:40.57 | [TK]D-Fender | and no idea what it might take to make usable |
| 06:40.58 | [TK]D-Fender | https://www.google.ca/search?q=Asterisk+H.324 |
| 06:41.45 | persiever | yes, that's why I am not able to find any support on this. I have googled as well and stopped at above link provided by you |
| 06:43.13 | persiever | The support for h.324 is end now. I think so.. But I recently receive a Request from a client heil from India. As he was saying the network support 3gvideo call. |
| 06:48.08 | persiever | i have found this link http://lists.digium.com/pipermail/asterisk-dev/2005-June/013114.html that best describe my question and state |
| 06:48.46 | [TK]D-Fender | That's 13 years ago and pretty wworthless now.... |
| 06:51.37 | persiever | Yes, it may be worthless as No one will use in future. And also I know where its worthy. that's why i was searching for the solution. Anyway you have better experience in Asterisk and its capabilities, so I will have to look for another solutions for this |
| 06:54.23 | [TK]D-Fender | The evidence points to development on this to have stopped about a decade ago. Go track down the author if you want to dicuss what it might take to bring this into being usable with current releases or you'll need to look elsewhere fo solutions to this situation you have found yourself having to deal with |
| 06:55.42 | persiever | ok any suggestion where I can get discus on this |
| 06:56.05 | [TK]D-Fender | As I said several times, the author of that 3rd party module |
| 06:56.20 | persiever | thanks |
| 06:56.21 | [TK]D-Fender | There is no "place" for this. The development cycle appears to be dead |
| 06:56.35 | [TK]D-Fender | So you'll need to hunt down the people the hard way and contact them directly |
| 06:59.27 | *** join/#asterisk jackmor (566c2961@gateway/web/freenode/ip.86.108.41.97) |
| 07:00.50 | jackmor | Hello Guys ! I have asterisk installed with chan mobile connected to my iphones via bluetooth ! |
| 07:01.21 | csavinovich | nice! |
| 07:01.25 | jackmor | I need to send uri for example to call emails via facetime ? any possible way to send a call or to open a link via asterisk through mobile |
| 07:02.52 | jackmor | I have been searching on net for the past year ! the problem when i send the call to iphone with originate command the letters are changed to numbers then dialed again ? but if open a link with facetime-audio://email it will send the call ? any possible way to ask asterisk to open a link via bluetooth |
| 07:12.11 | [TK]D-Fender | And bed-time.... |
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| 09:47.07 | persiever | Hi, Is there any possibilities that we can send audio and video in single port |
| 09:48.52 | file | over... SIP? |
| 09:49.31 | persiever | what is the best possible way to integrate Asterisk with 3g324m protocol. As 3g324m protocol open 1 media port during session setup. But sip uses individual port for audio, video and sip call control |
| 09:50.10 | file | I have no idea then. |
| 09:50.25 | persiever | ok |
| 09:51.22 | persiever | what is the best possible way to support Clearmode/8000 codec in asterisk. I need to make it passthrough only for SDP session negotiation |
| 09:52.07 | file | you write the code to do so. |
| 09:52.19 | file | there is no guide or anything, following an existing one would be the easiest option. |
| 09:54.04 | persiever | ok |
| 09:54.18 | MelMalik | looks like alex bkash hired a white monkey |
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| 14:11.20 | igcewieling | Us: "it doesn't work" isn't a trouble report. Customer: It doesn't work! |
| 14:12.13 | file | those are fun |
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| 17:08.51 | Timvde | Hi. I'm looking to set up some hardware (arduino/pi/...) to do network testing. I'd like to cover data on 4G/3G/2G, as well as voice and SMS (both sending and receiving). Only at the last moment, I noticed that the module I wanted to order does not support voice calls (https://sixfab.com/product/quectel-ec25-mini-pcle-4glte-module/). Do you guys have any recommendations for me? |
| 17:13.43 | [TK]D-Fender | a cell-phone using chan_mobile, or something supported by chan_dongle (get googling) |
| 17:24.02 | Timvde | Thanks for the keywords :) |
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| 20:34.52 | igcewieling | With chan_sip the command "sip show channels" output included the codec for each channel. Is there something similar for pjsip? |
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| 21:14.12 | Samot | pjsip show channelstats |
| 21:14.47 | Samot | https://www.irccloud.com/pastebin/VBa2P2wZ/ |
| 21:14.56 | Samot | You end up with that |
| 21:33.27 | igcewieling | Samot: thanks |
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