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| 01:05.12 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); 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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| 06:25.24 | phormulate | hello, how do I enable srtp for chan motif? |
| 06:28.57 | phormulate | I'm trying to make sure the incoming rtp from google chat are encrypted, and I can't find a thing about it anywhere... successfully decoded the rtp stream as ulaw with wireshark to verify... driving my a bit nuts |
| 06:29.42 | phormulate | google states in the developers section, they support srtp for xmpp media |
| 06:29.55 | phormulate | I just can't find out how to enable it |
| 06:29.59 | phormulate | from the asterisk end |
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| 11:21.40 | cervajs2 | is it possible use tls,wss transport for endpoint with chan_pjsip? |
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| 12:40.37 | rie | hello. im a newbe . Just a quick question. Asterix runs with AIX , so I do not necessarily need SIP, right? |
| 12:41.40 | WIMPy | That obviousely depends on who you want it to talk to. |
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| 12:42.47 | rie | ok. I have some softphones that work with SIP. Now I want make asterisk to send fax |
| 12:46.08 | [TK]D-Fender | rie: "core show applications like fax |
| 12:46.42 | WIMPy | Fax is evil. |
| 12:46.48 | rie | the question is if I can only use iax and no SIP if i want to make internal calls and fax sending or if i have to use SIP for any reasons. |
| 12:47.06 | rie | that makes me hope ;) |
| 12:47.18 | WIMPy | If you want to use SIP phones, you obviousely need SIP. |
| 12:47.42 | WIMPy | If you use IAX (or other types of) phones, you don't need SIP. |
| 12:48.24 | rie | <PROTECTED> |
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| 12:49.13 | WIMPy | If you have something that talks IAX... But that would be T.30 only. |
| 12:49.20 | rie | I already have hylafax running and a registered iaxmodem |
| 12:49.38 | marceloamorim | guys, could I send any variable through iax2? |
| 12:50.01 | WIMPy | marceloamorim: Yes. See function IAXVAR. |
| 12:50.33 | marceloamorim | ty sir |
| 12:50.37 | WIMPy | Unfortunatly IAX seems to be dying. |
| 12:51.03 | rie | me too soon.... |
| 12:51.13 | rie | well, show must go on :) |
| 12:51.22 | rie | is there an alternative? |
| 12:51.40 | WIMPy | Not really. |
| 12:52.02 | WIMPy | Looks like everything that makes SIP look bad must go. |
| 12:52.18 | WIMPy | That would be about ... everything. |
| 12:52.35 | rie | i heard iax is more reliable |
| 12:53.04 | WIMPy | It has several advantages. |
| 12:54.03 | [TK]D-Fender | riei heard iax is more reliable <- it isn't |
| 12:54.32 | WIMPy | I haven't heard of anyone having NAT issues with IAX, yet. |
| 12:54.52 | rie | ok. i have a soft fax that is registered on my aserisk server. then i have a gateway with an old fax also registered. it is correct that the iaxmodem does the "forwarding"? |
| 12:55.02 | [TK]D-Fender | It is simpler as far as handling some mor annoying NAT's that mess with SIP+RTP, but "reliable" isn't an appropriate term |
| 12:55.21 | [TK]D-Fender | rie: "forwarding" is not an appropriate term |
| 12:55.46 | [TK]D-Fender | rie: And I'm not sure where you're going with that question |
| 12:56.11 | WIMPy | I think it is appropriate to call somethign that would work without a potential lot of hazzle, nore reliable. |
| 12:56.27 | rie | [TK]D-Fender, i'm stuck . |
| 12:56.43 | [TK]D-Fender | rie: rephrase your question. |
| 12:58.34 | rie | all devices are in the same network. when i send a fax from one extension to an other how is the aiaxmodem involved? |
| 12:59.14 | [TK]D-Fender | it's involved if you use it |
| 12:59.19 | [TK]D-Fender | nothing "just happens" |
| 12:59.35 | [TK]D-Fender | You get calls from devices that place calls. You send them where you send them. |
| 13:01.32 | rie | if 'iax2 show peers' shows the modem but it is not online, how can i get it on? |
| 13:02.18 | [TK]D-Fender | Make sure to start it |
| 13:03.34 | rie | [TK]D-Fender, errm, how? |
| 13:03.49 | [TK]D-Fender | Read it's instructions |
| 13:04.14 | *** join/#asterisk AlienPenguin (~myRealNam@posta.netresults.it) |
| 13:04.29 | [TK]D-Fender | It has its own webpage with instructions. They tell you how to use it |
| 13:05.07 | AlienPenguin | hi all, i am trying to configure asterisk so that i can have multiple peers with the same name: i.e. 100 but belonging to different realms, is that possible? |
| 13:05.21 | rie | ok. it is running now . thanks. i never tried to start it until now XD |
| 13:06.55 | [TK]D-Fender | AlienPenguin: No. Peername is unique. having differently named peers with the same USERNAME might work if they are on different realms... not sure... |
| 13:07.28 | AlienPenguin | [TK]D-Fender, ok i will try ty |
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| 13:13.14 | AlienPenguin | [TK]D-Fender, it accepts the configuration but the two sip peers cannot register (cannot find matching peers) i named the peers as 100_0 and 100_1 with the same username and different realms |
| 13:13.44 | AlienPenguin | from what i gathered from rfc it should be possible to have the same name on a different realm |
| 13:15.43 | [TK]D-Fender | * isn't a nice full realm supporting SIP proxy. It is a basic B2BUA |
| 13:15.52 | [TK]D-Fender | Don't bet on RFC all the time |
| 13:16.03 | [TK]D-Fender | ESPECIALLY with chan_sip. |
| 13:16.09 | [TK]D-Fender | You might have better luck with PJSIP |
| 13:17.26 | AlienPenguin | pjsip as integrated in * or as a standalone project |
| 13:17.28 | AlienPenguin | ? |
| 13:17.49 | [TK]D-Fender | integrated |
| 13:17.55 | [TK]D-Fender | Read up... |
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| 13:45.51 | cervajs2 | anyone with working webrtc with chan_pjsip? |
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| 13:58.47 | monsterco | Hi - anyone has experience with goautodial? I am wondering where I can set the trunk? |
| 13:59.10 | [TK]D-Fender | Not supported here |
| 14:00.02 | monsterco | Thanks - Do you know if there is a channel for it? |
| 14:00.14 | [TK]D-Fender | No. Their webpage should tell you. |
| 14:00.27 | [TK]D-Fender | I doubt the have enough users to consider IRC worthwhile |
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| 15:23.45 | Dick-Tracy | Good Morning #asterisk - I've been digging for a way to get the called number for a session in the h context. Running stock 11. Is this possable without setting it as a goblal or tossing it in a local? |
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| 15:50.09 | dan_j | Hi, What softphones would you recommend? |
| 15:55.01 | [TK]D-Fender | none |
| 15:55.03 | [TK]D-Fender | All suck |
| 15:55.08 | [TK]D-Fender | some only slightly less than others |
| 15:55.30 | [TK]D-Fender | Bria seems popular |
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| 16:28.38 | dan_j | Yeh, thats what I'm finding. Thanks for the confirmation. |
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| 17:25.44 | mutilator | is the only way to do a call back queue to mod teh Queue application directly? |
| 17:25.54 | mutilator | i cant think og any snazzy way to hold a spot otherwise |
| 17:28.44 | WIMPy | call back? |
| 17:29.57 | rrittgarn | i use call files |
| 17:30.03 | rrittgarn | to do a CBQ |
| 17:30.25 | rrittgarn | i also have two separate queues, but could probably do it in one |
| 17:30.57 | rrittgarn | first queue times out, gives them the option for the CBQ, if they opt in, generate a call file to join the queue in their place, then they hang up, otherwise if they don't want a call back, they join the queue right then instead |
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| 17:31.28 | rrittgarn | On the Call Backs i also modify the caller ID for the agents so they can see its a call back and aren't confused by ringing |
| 17:31.45 | mutilator | hm |
| 17:31.54 | mutilator | so how does that preserve their spot in the queue? |
| 17:33.39 | mutilator | someoen calls the queue, "you're position 29, press 1 to hold your place and receive a callback..." |
| 17:35.49 | nny | Any old schoolers here? I am working with an Ast 1.2 client (we are upgrading to Ast 11 later this year) to move a server install. The new location test server has a connection to a mysql database for realtime. I have the client installed on CentOS7 (I know I know) and running, using the configs from the old server. I have the database config the s |
| 17:35.49 | nny | ame (and it is loaded as needed) yet asterisk doesn't seem to be picking up realtime entries for voicemail, sip or iax. There isn't a lot of commands in 1.2 from console, maybe someone knows one that can define the reason (log shows config loaded, but not much else in full). I have used realtime show status http://pastebin.com/YStfT9vx |
| 17:36.23 | nny | I have tested the database with asterisk credentials and was able to get a listing of users from the proper database, etc etc |
| 17:37.50 | nny | As this is 1.2 there is a distinct lack of feedback for diagnostic. Although challenging I am finding myself out of resources for troubleshooting |
| 17:43.57 | rrittgarn | @mutilator it adds the call file to the queue at the same time they would have entered it anyway. In a high volume situation you might have a caller bump a couple spots, but nothing too terrible |
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| 17:46.35 | mutilator | hmm |
| 17:46.39 | nny | starting to think the configs aren't setup for using realtime right, event hough they work on the old server somehow |
| 17:46.41 | mutilator | oh |
| 17:46.47 | mutilator | you just add the .call to the queue |
| 17:47.04 | mutilator | which would jump to the end of the queue |
| 17:47.06 | mutilator | i see |
| 17:48.24 | mutilator | wonder if you could make the call cut in line.. |
| 17:51.34 | rrittgarn | @mutilator I think you could if you wanted to. There's a variable for that you can set i believe. |
| 17:51.58 | WIMPy | A parameter to Queue. |
| 17:54.27 | nny | ok extconfig has definitions for the different families, so that's fine |
| 17:54.32 | mutilator | alrite, i'll dig into this a bit then |
| 17:54.37 | mutilator | thanks guys |
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| 17:59.44 | nny | let me reorganize my question, so it appears in 1.2 there was no caching of realtime meaning it didn't show up in SIP SHOW PEERS until it was being used. This explains a lot. I can test this by adding an entry and testing it with a sip client. Is there some way to know this is what the situationi s in 1.2? |
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| 18:09.18 | file | nny, that's the situation in each version with chan_sip realtime |
| 18:23.04 | nny | got it ty |
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| 19:39.28 | phormulate | any idea how to enable srtp for chan_motif? |
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| 22:06.49 | px1mp | hello, I've noticed I cannot register with sip provider using register |
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| 22:08.39 | px1mp | i've tried several providers and watched the communication with tcpdump, but it seems, no communication with any of the servers is happenin |
| 22:09.23 | px1mp | any idea how to troubleshoot ? |
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