| 00:07.08 | AL13N_lappy | hey, is there a way to use realtime external DB with templates?, like you have some values, but since all the rest is the same for all of these, it's based on a template? |
| 00:07.24 | AL13N_lappy | i'm trying to do so with the ldap one, but no luck |
| 00:12.16 | sharpy | does anyone of you know a tutorial/documentation on how to use the asternet library for c#? |
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| 01:14.55 | Kobaz | i broke it |
| 01:15.02 | Kobaz | It's broken... |
| 01:15.25 | Kobaz | > core show channels PJSIP/22860-0000139a (None) Up AppDial((Outgoing Line)) |
| 01:15.33 | Kobaz | channel request hangup PJSIP/22860-0000139a PJSIP/22860-0000139a is not a known channel |
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| 01:26.26 | Kobaz | i'll see if i can reproduce |
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| 09:56.50 | AL13N_lappy | file: hey, i've looked into realtime ext config, wrt pjsip and ldap, is there a way to specify defaults for each endpoint/aor/etc...? or can templates with static values be used with sorcery/wizard/ to complete a section (combined with looked up values and static values?) i tried [ps_endpoints](template-endpoint) in the res_ldap but so far no luck. |
| 09:57.05 | file | no. |
| 09:58.20 | AL13N_lappy | file: i thought so, it seemed like either the full entry comes ext or internal... |
| 09:59.43 | AL13N_lappy | file: i've also looked to find if there was any way to call a function to externally lookup a value (and/or change it) in the config, but it seems like you can't use functions in a config either... |
| 10:00.16 | file | you can't, the only thing would be #exec to execute an external script or application to produce configuration output |
| 10:00.58 | AL13N_lappy | right, but that's on reload, right? |
| 10:01.05 | file | yes. |
| 10:01.36 | AL13N_lappy | it's just seems sad, i mean, there's these powerfull stuff like extconfig/templates/sorcery/wizard/... but no real combining of templates with them |
| 10:01.59 | file | not something anyone has ever written |
| 10:02.57 | AL13N_lappy | myeah, well, i hear you coming, but I've got tons of other stuff to do as well... |
| 10:03.05 | AL13N_lappy | and $dayjo |
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| 11:10.23 | AL13N_lappy | anyone know how to enable textsupport on pjsip? i saw in documentation that sip.conf has textsupport=yes |
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| 12:20.57 | Ravenheart | what like SMS? |
| 12:22.05 | sibiria | if you're referring to sending SIP MESSAGEs then you don't need to do anything special to allow that with pjsip |
| 12:22.05 | AL13N_lappy | yes |
| 12:22.20 | AL13N_lappy | ok, thx |
| 12:23.03 | AL13N_lappy | ok, so the provide would be converting that to messages, so there probably doesn't even need t140? |
| 12:23.20 | sibiria | that's up to the provider |
| 12:23.48 | sibiria | i'm just pointing out that pjsip/asterisk by default will allow MESSAGE requests |
| 12:24.03 | Samot | And SIP MESSAGE != SMS |
| 12:29.11 | AL13N_lappy | ok, thanks, one more stupid question: |
| 12:30.26 | AL13N_lappy | i'm trying out the my trunk, but i get an error message about it; i've looked around, but i found mostly stuff about sip providers which want you to register, i have one that is ipbased, and i put in the identify section for it, but, how does asterisk know that it's up? |
| 12:31.37 | AL13N_lappy | Endpoint 'xxxx': Could not create dialog to invalid URI '012345685' |
| 12:32.17 | AL13N_lappy | this is probably something stupid, but there's not even sip requests going out ; i don't see them with tcpdump |
| 12:33.45 | AL13N_lappy | do i have to put @IP.AD.RE.SS:5060 in the Dial() app? i thought it would use the contact linked to the endpoint? (it has 2 contacts) |
| 12:34.45 | Samot | What does the dial string look like now? |
| 12:35.20 | AL13N_lappy | Dial("PJSIP/201-00000000", "PJSIP/trunk/012334456,20") |
| 12:36.06 | AL13N_lappy | (this is from the logs) |
| 12:36.29 | Samot | So yeah, in that format you need a proper SIP URI |
| 12:37.31 | AL13N_lappy | Samot: how can i do that, if there's 2 ip addresses i need to use? |
| 12:37.50 | Samot | At the same time? |
| 12:37.59 | AL13N_lappy | isn't the contacts for that to complete that? |
| 12:38.47 | AL13N_lappy | Samot: so, you're saying i need to manually fix the extension so that it would try one and in case of failure try the second? |
| 12:39.40 | AL13N_lappy | or is rewrite_contact meant for this? |
| 12:40.17 | AL13N_lappy | (also, i'll probably need to set the outgoing callerid somewhere, should i do that in the dialplan too?) |
| 12:40.43 | Samot | Well your dial string ignores AORs/contacts. |
| 12:40.50 | AL13N_lappy | Samot: i figured with 2 contacts it would autocomplete and pick one of both randomly |
| 12:41.09 | AL13N_lappy | Samot: ah, how should i write it? |
| 12:41.15 | Samot | No, that's not how it works. |
| 12:42.45 | AL13N_lappy | ok, so, how can i fix the dial string , to use the AoR/contacts? |
| 12:45.51 | Samot | https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels |
| 12:48.37 | AL13N_lappy | yeah, i've seen that |
| 12:49.25 | AL13N_lappy | ah, you mean the @mytrunk |
| 12:49.27 | AL13N_lappy | i'll give that a try |
| 12:50.49 | AL13N_lappy | "Endpoint: 'edpnet': Unable to create request with auth. No auth credentials for realm(s) 'edpnet.net' in challenge." |
| 12:51.04 | AL13N_lappy | argh, i have no auth section for this, because there is no registration... |
| 12:51.44 | AL13N_lappy | ooh, maybe that means the provider is rejecting me... |
| 12:52.00 | AL13N_lappy | maybe because my callerid isn't set yet |
| 12:52.25 | AL13N_lappy | yes! |
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| 13:17.40 | AL13N_lappy | Samot: ok, halfway there: right now i have an INVITE that goes to trunk, but due to NAT it's still wrong: |
| 13:17.40 | AL13N_lappy | From: "AL13N" <sip:0234542465@10.X.Y.Z> |
| 13:18.06 | AL13N_lappy | i think my public ip address should be put there, but i did put the ip in the transport for this trunk endpoint |
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| 13:56.19 | AbortRetryFail | I've got Asterisk registered to another pbx via SIP. Anyone know how I can tell Asterisk to tell the other PBX to transfer the call? |
| 13:56.56 | AbortRetryFail | I've tried setting REDIRECTING() like the wiki says, but from what I've been reading that seems to just change the caller ID appearance. |
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| 15:43.48 | AL13N_lappy | ok, this is weird, so now, i have 2 linphones connected... A can call B, but B cannot call A, server says busy... but in cli "pjsip show endpoints" shows it as available....? |
| 15:43.54 | AL13N_lappy | quite weird |
| 15:44.18 | AL13N_lappy | the server literally says busy... |
| 15:49.36 | AL13N_lappy | <PROTECTED> |
| 15:49.36 | AL13N_lappy | [Oct 22 14:56:55] VERBOSE[885][C-00000003] app_dial.c: Called PJSIP/201/sip:201@10.238.9.151:5060 |
| 15:49.37 | AL13N_lappy | [Oct 22 14:56:55] VERBOSE[885][C-00000003] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) |
| 16:08.45 | AL13N_lappy | ok, i needed more debug logs |
| 16:10.20 | AL13N_lappy | basically B has a newer linphone version than A; and A is rejecting SIP from server with "488 not acceptable here" though i checked the codecs... |
| 16:11.04 | AL13N_lappy | futhermore A is actually also used with a different (very old) asterisk server at the same time, which works just fine |
| 16:11.29 | AL13N_lappy | so, iow: the server here is maybe using/forceing something that older linphone is not compatible with...? |
| 16:16.32 | AL13N_lappy | also, isn't it weird that there's a bridge set up between them when both were using opus? |
| 16:17.05 | AL13N_lappy | i'm wondering if i should enable AVP on the older linphone... though i can't think that would be the reason it breaks... |
| 16:17.06 | Ravenheart | maybe run it with a high verbosity level |
| 16:17.10 | Ravenheart | or debug mode |
| 16:17.50 | AL13N_lappy | oh, you mean run the linphone with debug? |
| 16:17.58 | AL13N_lappy | that's a good idea! |
| 16:24.40 | AL13N_lappy | Ravenheart: i got this from the linphone rejecting it: https://pastebin.com/VCvexpU8 |
| 16:25.17 | Ravenheart | no no i mean asterisk |
| 16:25.31 | Ravenheart | you can login to the console (asterisk -r) |
| 16:25.42 | Ravenheart | and then there was a command to enter it in debug mode |
| 16:25.48 | Ravenheart | where the logs are very detailed |
| 16:26.03 | AL13N_lappy | oh, yeah, i got the debug from the asterisk, it said the linphone at A was rejecting with 488 |
| 16:26.05 | seanbright | codec mismatch |
| 16:26.09 | AL13N_lappy | well |
| 16:26.11 | AL13N_lappy | Declining mline 0, no corresponding stream in local capabilities description. |
| 16:26.17 | AL13N_lappy | but both are opus |
| 16:26.33 | AL13N_lappy | i think the linphone crapped out because of the text capability request |
| 16:27.05 | file | opus is a codec, not a media stream type |
| 16:27.11 | AL13N_lappy | this older linphone doesn't seem to have text capabilities anywhere |
| 16:27.29 | AL13N_lappy | maybe it rejects because of that |
| 16:28.00 | AL13N_lappy | file: is "text" a media stream type? |
| 16:28.12 | file | it can be, but it's not supported in PJSIP |
| 16:28.20 | AL13N_lappy | oh |
| 16:28.57 | AL13N_lappy | too bad, i guess i should turn it off on the server, maybe that'll fix the issues |
| 16:29.01 | file | you haven't provided the actual SIP INVITE that was sent, or how the endpoint is configured, so can't really say anything else |
| 16:32.03 | AL13N_lappy | file: sorry, this is the data of the SIP INVITE that the linphone receives from the server: https://pastebin.com/cMd8dFbw |
| 16:32.17 | file | that's a WebRTC SDP |
| 16:32.52 | file | with DTLS-SRTP, ICE, opus/speex/alaw/ulaw/gsm, bundle, and rtcp-mux |
| 16:33.44 | AL13N_lappy | i did enable webrtc support, since i was planning on using it from a firefox |
| 16:33.54 | AL13N_lappy | should i disable webrtc? |
| 16:34.18 | AL13N_lappy | or maybe only enable it for a specific client |
| 16:34.37 | file | WebRTC SDP is generally incompatible with normal SIP clients. |
| 16:34.42 | AL13N_lappy | ic |
| 16:34.46 | AL13N_lappy | i'll try it |
| 16:36.04 | AL13N_lappy | file: that cleared it right up... |
| 16:36.06 | AL13N_lappy | ok, thanks! |
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| 16:44.27 | AL13N_lappy | the older linphone also can't understand the video codec sent by the newer one, but that's likely this older linphone's issue (and that there's no video recoding of course) |
| 16:45.56 | AL13N_lappy | ok, so i just need to get the sip trunk working now and do the incoming and outgoing calls test, and it'll be good... (apparently my sip trunk was never set up properly on provider side, since i wasn't using that one yet) |
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| 19:23.51 | Ravenheart | so for a very small office pbx |
| 19:24.08 | Ravenheart | one would only need a few extensions, prolly A queue |
| 19:24.15 | Ravenheart | and inbound/outbound dialing |
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| 21:28.46 | Kobaz | https://community.asterisk.org/t/configure-multiple-transport-for-per-endpoint/70989/6 |
| 21:29.32 | Kobaz | So..... apparently you can leave transport blank on an endpoint?? This doesn't appear to work |
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| 23:16.37 | Samot | Yes you can. |
| 23:16.54 | Samot | Not setting a transport will load the first available transport that matches the request. |
| 23:17.13 | Samot | So if it's a TLS based request, it will look for the first TLS transport. |
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