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| 02:09.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 10:22.22 | phrearch | hello |
| 10:23.38 | phrearch | i'm trying to setup asterisk 16 for webrtc video conferencing, but got stuck where the res_pjsip_transport_websocket module can't be loaded. The cli doesn't mention what's wrong though. Only: |
| 10:23.39 | phrearch | [Nov 19 11:17:04] ERROR[6705]: loader.c:2249 load_modules: Failed to resolve dependencies for res_pjsip_transport_websocket |
| 10:23.51 | phrearch | any idea what could be wrong? |
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| 10:35.21 | phrearch | hm, probably something wrong with the loaded modules. with autoload, it gets loaded just fine |
| 10:36.40 | wyoung | phrearch: nice |
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| 11:24.41 | wakko | hi, is it possible to bind a pjsip transport to an interface rather than an ip address (because the ip is dynamic) ? |
| 11:25.23 | file | no. |
| 11:25.49 | wakko | so i'll make a script to update pjsip.conf, maybe using an hostname |
| 11:25.53 | wakko | thanks for the info |
| 11:26.06 | wyoung | wakko: hostnames are usually resolved then cached |
| 11:26.37 | wakko | that's why i was thinking about modify the /etc/hosts, and restart the pjsip module :( |
| 11:27.00 | wakko | i don't think this will be cached, but i could be wrong |
| 11:27.59 | wakko | maybe generating the transport block in a seprate file with the correct IP and including it from pjsip.conf might be better then |
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| 11:55.03 | wyoung | just don't use pjsip |
| 11:55.10 | wyoung | SOunds like an easy fix |
| 11:55.51 | wakko | haha |
| 11:57.05 | wyoung | brushes his hands off |
| 11:58.47 | wakko | the thing is, i already implemented the last solution and it works :) |
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| 12:16.28 | wyoung | wakko: woot! |
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| 13:24.19 | Samot | 6:55:01 AM <wyoung> just don't use pjsip <-- You realize this won't be an option in the near future? |
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| 13:54.15 | TECFALL | I have a bunch of Cisco 7961G's with SIP 9-4-2SR3-1S firmware, and now my callFwdALL softkey isn't working. I get the following: NOTICE[2548][C-00006c0d]: chan_sip.c:26672 handle_request_invite: Call from '230' (192.168.101.36:49838) to extension 'x-cisco-serviceuri-cfwdall-235' rejected because extension not found in context 'outbound'. |
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| 13:54.44 | wyoung | nice |
| 13:54.51 | TECFALL | I have never had to used a softkey xml file, or do anything specially within extensions.conf to get this to work on older firmware. Any ideas? |
| 13:55.04 | ttyX | does maxcallbr mean bitrate for all calls through a peer or just one call? |
| 13:55.05 | Samot | Yeah, that doesnt work without the patch. |
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| 13:55.38 | TECFALL | Samot: was that last comment directed towards me? |
| 13:55.42 | Samot | TECFALL: Do you have the Cisco patch? |
| 13:55.47 | Samot | Yes |
| 13:55.50 | TECFALL | Samot: what patch? |
| 13:56.33 | Samot | The Cisco CallManager patch. Those phones are designed for the Cisco UCM |
| 13:57.13 | Samot | So they have headers and info in the sip packet for how thier UCM does things |
| 13:57.38 | TECFALL | Samot: I have been using these phones for 8 years. Is there just something different in the newest firmware? I was recently using 8-4-4 without any issues. |
| 13:58.05 | Samot | Im just telling you how it is. |
| 13:58.38 | TECFALL | Samot: no, i am all ears. So what patch is this that you are referring to? |
| 13:58.40 | Samot | x-cisco-serviceuri-cfwdall-235 <-- Cisco only SIP header. |
| 13:58.53 | Samot | The Call Manager patch for Asterisk. |
| 13:59.17 | Samot | That someone created so these phones could work on Asterisk in the same way they worked in the UCM. |
| 13:59.30 | Samot | Asterisk needs to know what to do with those custom SIP headers and data. |
| 13:59.59 | TECFALL | Samot: that is very interesting... it makes complete sense. but now i don't know how my phones were working for 8 years without it. |
| 14:00.05 | Samot | Shrug. |
| 14:01.24 | TECFALL | Samot, I appreciate your help. I am reading about it now., |
| 14:01.33 | Samot | 8:54:42 AM <wyoung> nice <-- Was this in reply to my question? |
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| 14:27.15 | *** join/#asterisk TECFALL (~dschuett@66.37.233.134) |
| 14:27.39 | TECFALL | So... what is a decent phone for asterisk? Perhaps the best phones for under $100 |
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| 14:34.16 | Slade | cheap cellphones with softphone? :P |
| 14:36.55 | Samot | And what should this phone be able to do? |
| 14:37.22 | Samot | Limiting your phone to just price can seriously limit what the phone can actually do. |
| 14:37.43 | Samot | So really this should be "I need a phone that can do X, Y, Z and I would like it to be under a $100" |
| 14:38.09 | Samot | Because the answer could end up being either "Here are the phones you can look at" or "There aren't any" |
| 14:39.41 | Slade | grandstream is a reasonably inexpensive brand you can look at.. panasonic has a few as well |
| 14:39.57 | Slade | see Samot's comments on how to make your question a useful engineering question |
| 14:40.15 | Samot | We play this game too much. |
| 14:40.22 | Samot | "Need a phone, any suggestions" |
| 14:40.25 | Samot | "Here's a phone" |
| 14:40.32 | Samot | "Nope, doesn't do X" |
| 14:40.35 | Samot | "Here's this phone" |
| 14:40.40 | Samot | "Nope, doesn't do Y" |
| 14:41.36 | TECFALL | sorry, really nothing too special. Transfer, hold, conference, forward, voicemail, headset if needed, speaker... |
| 14:42.45 | Samot | Yealink T-19E2. |
| 14:43.07 | Samot | 1 line, does all that. Under $100. |
| 14:43.37 | Slade | speaker.. there are certainly times i wish phones didn't come standard with that feature |
| 14:43.58 | [TK]D-Fender | Polycom VVX150 |
| 14:44.42 | Samot | Yeah because Speaker Phone isn't a requested/wanted business phone feature.... |
| 14:45.10 | Slade | Samot, oh it is, it really is.. but its never a requested business feature by the person in the cube next to them :P |
| 14:45.26 | Samot | That has nothing to do with the phone being IP or analog. |
| 14:45.34 | Samot | That has been an issue since before SIP. |
| 14:46.02 | Slade | oh, my tongue in cheek remark wasnt meant to imply otherwise, or to be taken as seriously as you have |
| 14:49.20 | Samot | Well one must realize a comment was a "tongue in cheek" remark. |
| 14:50.58 | Slade | 1 :P isnt enough? will 3 do? :P :P :P |
| 14:51.51 | Samot | TECFALL: So the Yealink T-19P E2 or the Polycom VVX150 are decent options. |
| 14:56.23 | Slade | wonder if he saw |
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| 16:13.13 | wakko1 | pjsip SDP is messed up with more than one transport |
| 16:13.36 | Samot | In what way? |
| 16:14.33 | wakko1 | I have 3 IPs on one server, with 3 transports blocks, on call arrives on one, the c=/o= are using the IP address of another transport |
| 16:14.48 | wakko1 | s/on/one/ |
| 16:15.12 | wakko1 | I have 3 IPs on one server, with 3 transports blocks, one call arrives on one, the c=/o= are using the IP address of another transport |
| 16:15.43 | wakko1 | i have see a blog post about this from 2015 and I am using 15.6.1 |
| 16:17.35 | wakko1 | Samot, i hope this pjsip near future is not that near |
| 16:19.30 | wakko1 | maybe bind= is not enough and I should add localnet= (there is not NAT) ? |
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| 16:24.20 | Samot | Calls arrive from where? |
| 16:26.21 | wakko1 | Calls arrives with INVITE from a phone (192.168.23.3) sent to asterisk (10.0.23.254) - yes there is a route for that - later on asterisk answers back 200 OK with o=- 20019 20021 IN IP4 192.168.199.3 and c=IN IP4 192.168.199.3 (that is a completly different network) |
| 16:27.13 | wakko1 | I do have a udp transport with bind=10.0.23.254:5060 |
| 16:30.01 | wakko1 | and another udp transport with bind=192.168.199.3:5060 |
| 16:30.13 | Samot | And what do you have the actual media ips set to? |
| 16:31.15 | wakko1 | I don't know, it's not set automatically ? |
| 16:31.19 | Samot | No. |
| 16:31.50 | wakko1 | based on the bound address, and the endpoint/transport association, asterisk should know |
| 16:32.00 | Samot | So these are all local? |
| 16:33.05 | wakko1 | not local as accesible throught a switch, but local as accesible throught a tun/tap with no NAT and clean routes |
| 16:33.35 | Samot | Well you need to tell the endpoint what to use for media |
| 16:33.47 | Samot | ;media_address= ; IP address used in SDP for media handling (default: "") |
| 16:34.01 | Samot | ;bind_rtp_to_media_address= ; Bind the RTP session to the media_address. |
| 16:34.22 | Samot | This causes all RTP packets to be sent from the specified address. |
| 16:35.30 | wakko1 | okay! |
| 16:35.37 | Samot | I believe that is what you are looking for and those are [endpoint] settings (along with direct_media, etc) |
| 16:36.49 | wakko1 | i will try this, thanks |
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| 18:00.31 | xhude1 | Any words on STIR/SHAKEN implementation in Asterisk? |
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| 18:12.45 | file | I know of noone working on such a thing |
| 18:14.20 | xhude1 | I think Verizon is close to implementing it soon... |
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| 18:19.16 | [TK]D-Fender | Bond has already spoken with regards to martinis.... |
| 18:22.52 | igcewieling | Verizon? Implement something less than 10 years old? never gonna happen. |
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| 18:25.06 | Samot | STIR/SHAKEN is not something that Asterisk will be geared for. |
| 18:25.17 | Samot | It's a Carrier level thing. |
| 18:26.08 | Samot | I use Bandwidth, when Bandwidth gets a call from ATT the STIR/SHAKEN process is going to happen between Bandwidth and ATT. |
| 18:29.59 | trmg | N00b question here...what is this STIR/SHAKEN thing? |
| 18:30.11 | trmg | ...asking for a friend. |
| 18:30.17 | Samot | It's new. |
| 18:30.28 | Samot | So new it's not even implemented, it was approved in May. |
| 18:30.44 | trmg | Oh is this related to the FCC thing about robocalls? |
| 18:30.49 | trmg | Well maybe not directly related. |
| 18:30.57 | file | it is a method by which certificates are used to attest that the calling identity in a call is actually from the owner of said number |
| 18:31.01 | Samot | When PSTN Carrier A sends calls they will be "signed" |
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| 18:31.21 | Samot | So Carrier B can validate the presented call information from the carrier, caller, etc. |
| 18:31.30 | Samot | Reputation checks, etc. |
| 18:32.41 | trmg | Ah, gotchya |
| 18:32.41 | Samot | And since this takes place at a pretty high level of the PSTN, Asterisk isn't going to really be the choose platform to make it happen. |
| 18:32.52 | trmg | Right |
| 18:33.04 | trmg | It's basically carrier to carrier. |
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| 18:33.52 | trmg | It's something my trunk provider will need to sort out. :-D |
| 18:35.07 | trmg | I wonder what affect sending anonymous CID will have with STIR/SHAKEN? |
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| 18:36.32 | trmg | I'm reading this right now: https://transnexus.com/whitepapers/understanding-stir-shaken/ |
| 18:38.26 | Samot | It's was never truly anonymous. |
| 18:38.36 | trmg | That is true. |
| 18:38.55 | trmg | And in that case you're not advertising false CID. |
| 18:38.55 | Samot | If order for you to make a call through my network, I need to know who you are... |
| 18:38.59 | trmg | You're just not advertising any. |
| 18:39.09 | trmg | Right. |
| 18:39.14 | Samot | Now just because I hide your CID presentation when the call goes out... |
| 18:39.22 | Samot | Or tell the other carrier "Private" |
| 18:39.36 | Samot | without actually hiding it per se... |
| 18:39.43 | trmg | nods |
| 18:40.23 | trmg | At a provider level it's not hidden. It's just not displayed on the destination endpoint device. |
| 18:40.32 | trmg | "privacy bit" so to speak. |
| 18:41.00 | Samot | In the PSTN/Telephony world, yes. |
| 18:41.37 | trmg | On the SIP side you have IPs I suppose. |
| 18:41.54 | trmg | A simple query with the originating provider would yield you the subscriber information. |
| 18:42.08 | trmg | "Who placed call x from IP y at this time..." |
| 18:46.03 | trmg | Perhaps some other unique ID in SIP headers (I'm not initmately familiar with these quite yet). |
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| 19:28.43 | jrun | has asterisk been thought maildir? |
| 19:28.58 | jrun | also, what does Cyrus mean here: https://wiki.asterisk.org/wiki/display/AST/IMAP+Server+Implementations |
| 19:29.29 | jrun | UW IMAP been tested against it or it could be used as arg of --with-imap= ? |
| 19:29.56 | [TK]D-Fender | That page is really old |
| 19:30.06 | [TK]D-Fender | Maybe you should start by actually describing what you're doing. |
| 19:30.16 | [TK]D-Fender | And if you have an actual problem or not |
| 19:30.54 | jrun | we are looking into if we can use an external *sync* process to sync local maildir with and imap server. |
| 19:31.11 | jrun | our voicemails are on nfs and that's been the source of lots of problems. |
| 19:33.31 | [TK]D-Fender | Then that's a question for the IMAP software support channel |
| 19:33.53 | jrun | oh, didn't know there was one :) |
| 19:34.11 | jrun | would that be asterisk-dev ? |
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| 19:41.04 | drmessano | jrun: You don't SYNC your VM store to IMAP, it actually stores it there |
| 19:41.49 | jrun | drmessano: no local storage? |
| 19:42.00 | jrun | what if imap server is down? |
| 19:42.31 | drmessano | What if you're using Realtime and your SQL server goes down? |
| 19:42.32 | jrun | just quick grep shows codes refering to mbox |
| 19:42.46 | drmessano | Don't let it go down |
| 19:43.01 | drmessano | or don't use IMAP |
| 19:43.50 | jrun | hmm, what's the local storage format? |
| 19:44.11 | drmessano | Whatever you tell it to be in voicemail.conf |
| 19:50.46 | [TK]D-Fender | jrun> hmm, what's the local storage format? <- files |
| 19:51.04 | [TK]D-Fender | (if using the sample config as your initial basis) |
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