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| 10:34.35 | Aristide | Hello. Randomly, when I forward to a exten, I get this error : |
| 10:34.35 | Aristide | [Apr 15 12:33:53] WARNING[16709][C-00000005]: app_playback.c:493 playback_exec: Playback failed on SIP/ovh-00000009 for silence/1&custom/blacklistinfo |
| 10:34.46 | Aristide | But sometime its work |
| 10:34.48 | Aristide | Sometime not |
| 10:35.23 | Aristide | I have try to put Progress() before Playback but same problem. And put a Wait(1) |
| 10:36.25 | Aristide | But other commands are runed with success |
| 10:36.35 | Aristide | are run* (sorry my English is not very well ^^) |
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| 10:56.21 | Ellenor | Aristide: I've seen far worse English than yours. Your statement and questions I can at least fully comprehend. |
| 10:59.57 | Aristide | Hm. Maybe its work now |
| 11:00.02 | Aristide | After 4 tests, no problems |
| 11:00.23 | Aristide | Ellenor: I'm making a « blocking number » system with a simple forward to 4000 |
| 11:00.53 | Aristide | Eg : You call me, I don't want to be called by you anymore : Forward â 4000. You get a message about your phone number by a prerecorded message and you cannot calle me after :D |
| 11:01.30 | Ellenor | Aristide: So if you block +1 778 764 0137 and I call you from that #, I'll get a message in français parisien about how Angloids aren't welcome? |
| 11:01.53 | Aristide | xD |
| 11:02.06 | Ellenor | Or will the message be in french and english? |
| 11:02.12 | Aristide | French only |
| 11:02.17 | Aristide | I'm never called from other country |
| 11:02.45 | Aristide | Only new caledonie by my Brother but 1 : He speak french, 2 : Blocking my brother is not in my todo list ^^ |
| 11:02.50 | Aristide | caledonia* |
| 11:03.08 | Ellenor | Is New Caledonia part of the +33 area codeL |
| 11:03.10 | Ellenor | ?* |
| 11:03.36 | Ellenor | Or is it a separate country for telephony sakes? (I know it's part of the French Republic) |
| 11:06.25 | Ellenor | Aristide: I know that what you want to do is possible, but because I haven't wanted to make this system, I'm not able to help you. |
| 11:06.46 | Aristide | Ellenor: My brother has a VoIP phone subscription |
| 11:06.59 | Aristide | Ellenor: But its work very well now :) Thank's for your response |
| 11:08.04 | Ellenor | I'd say pm me your # and I might try it out but you probably don't want to be telephoned by a canadianoid at high noon |
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| 11:08.18 | Ellenor | (when I am meant to be asleep) |
| 11:08.55 | Ellenor | (and will be drunk in English and downright incoherent in French assuming I can speak it at all) |
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| 11:35.05 | DanielYK | @file The IPv6 addresses in the SDP of a pjsip endpoint are with square brackets. Is this issue related to pjsip or asterisk? |
| 11:35.07 | DanielYK | o=- 9416523 9416525 IN IP6 [20xx:e9:xxxx:b200:20c:29ff:xxxx:bf36] |
| 11:35.07 | DanielYK | s=Asterisk |
| 11:35.09 | DanielYK | c=IN IP6 [20xx:e9:xxxx:b200:20c:29ff:xxxx:bf36] |
| 11:36.26 | file | Asterisk is what places things there. |
| 11:37.17 | DanielYK | Okay, I think that is a bug. Is this known or should I open a new issue? |
| 11:37.28 | file | any known issues are filed in the issue tracker |
| 11:38.17 | DanielYK | Or are there any configuration items which could cause this wrong format? |
| 11:38.28 | file | nothing comes to mind |
| 11:38.41 | DanielYK | thanks |
| 11:39.05 | file | what version of Asterisk? |
| 11:39.09 | DanielYK | 16.9.0 |
| 11:39.37 | file | haven't seen such a thing mentioned |
| 11:40.06 | DanielYK | hmm |
| 11:41.05 | file | are you setting an external media address... |
| 11:42.05 | DanielYK | Yes, I use a domain name. But it does this wrong formatting also when hardcoded to the current IPv6 address. |
| 11:42.38 | file | I think it would put brackets in if external address were set for IPv6 |
| 11:42.46 | file | I don't think anyone has really done that |
| 11:43.28 | DanielYK | Yes, that could it be. |
| 11:46.22 | DanielYK | Do you have a hint for me where the code issue could be? I have not checked the code yet. |
| 11:46.58 | file | the application of external_media_address is done in the res_pjsip_sdp_rtp module |
| 11:58.01 | DanielYK | I think ast_sockaddr_stringify_host(&transport_state->external_media_address) returns the IPv6 addresses with brackets. Is there another function which returns without brackets? |
| 11:58.19 | file | probably, I do not know off the top of my head |
| 11:59.04 | file | all that stuff is defined in netsock2.h |
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| 12:17.49 | DanielYK | ast_sockaddr_stringify_addr returns the addresses in the correct format. No changes to IPv4, only IPv6 without brackets. |
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| 12:43.27 | DanielYK | It looks like the issue is in res_pjsip_session. What is the difference? |
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| 12:58.25 | forgotmynick | hello. i'm trying to enable inbound and outbound calls on asterisk for microsoft team users WITHOUT a microsoft calling plan because we can't afford to pay £7 per month per user for 200 users just to connect it to be PBX. does anyone know if there is a module that can connect the two? |
| 12:59.45 | file | DanielYK: res_pjsip_sdp_rtp handles the audio/video SDP and RTP, while res_pjsip_session handles sessions (or calls) and associated stuff |
| 13:00.01 | file | forgotmynick: there is no module or functionality built into Asterisk to explicitly facilitate such things |
| 13:00.49 | forgotmynick | i understand but could there be a third party module? i have of course searched on google but cannot find anything and judging by microsofts api, it's probably not going to be possible |
| 13:01.12 | sibiria | doesn't Teams have regular SIP capability for direct dialing of a remote SIP phone or so? |
| 13:01.30 | forgotmynick | paid option |
| 13:01.33 | sibiria | my memory could be off but i could swear i've seen this setup mentioned |
| 13:01.36 | sibiria | oh |
| 13:01.52 | sibiria | shouldn't be surprised, i suppose |
| 13:02.25 | forgotmynick | half the company has been let go while the remaining have agreed to a pay cut so we're trying to shave off costs every where we can |
| 13:02.41 | sibiria | maybe you should just drop Teams entirely then, in favor of something free that works |
| 13:03.14 | sibiria | this may sound odd, but Discord is free to use for businessess, and it's a pretty competent chat + voice solution. its gamer appearance may not appeal, though |
| 13:06.51 | forgotmynick | office 365 subscription is good value for money because you get access to office and teams which everyone is using to work from home minus the factory workers, chat, collaboration, file sharing etc. teams is free to call other team users but need a solution for inbound/outbound. the other issue is directors want physical handsets even though we could save a lot of money by using softphones but that's a whole |
| 13:06.51 | forgotmynick | different problem. thanks anyway, if anyone does have any suggestions please let me know |
| 13:08.35 | Samot | How will any third party solution work if the Microsoft Teams account doesn't have the ability to connect to outside SIP services? |
| 13:08.56 | forgotmynick | i was just about to say on a third party solution there might not be a way to route external calls internally |
| 13:09.10 | Samot | Or vice versa. |
| 13:09.41 | Samot | Unless the PBX has a connection to Microsoft Teams to accept calls from and send calls to, it's not much of a solution. |
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| 13:39.19 | MLC | So I'm trying to enable TLS in PJSIP for the first time. I can connect if the client does not check the server certificate, but when the client does check the server certificate it can't connect. The client says "Certificate name mismatch (503)" but I've double checked that the names match. At the same time asterisk shows me a pjproject error: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000. I'm guessing my server certificate i |
| 13:39.19 | MLC | s in the wrong format. |
| 13:40.24 | sibiria | so correct it? :) |
| 13:40.39 | MLC | not sure how |
| 13:40.40 | sibiria | PEM format, full chain, remember the correct order of entities in the chain |
| 13:46.25 | MLC | I don't want to use a self-signed cert, so I purchased a cheap cert from ssls.com. |
| 13:46.36 | sibiria | why not use let's encrypt? |
| 13:47.03 | sibiria | but whatever you go for, it should still be in PEM format, so start there |
| 13:47.21 | MLC | I'm not familiar with Let's Encrypt, I'll check it out. |
| 13:48.11 | MLC | I tried converting the cert I got to PEM but didn't have any luck |
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| 13:48.22 | sibiria | what were you given by ssls.com? p12? |
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| 13:49.07 | MLC | not sure. I'm not well versed in certificates and such |
| 13:49.28 | sibiria | the file type/ending would be a good indicator |
| 13:49.58 | MLC | they gave me a .crt and a .ca-bundle |
| 13:50.31 | sibiria | then i would think you already have things in PEM format, but perhaps not wrapped-up correctly |
| 13:51.13 | sibiria | if you take a look in those files, are they plain-text and contain a few "---BEGIN CERTIFICATE/PRIVATE KEY ----" sections? |
| 13:51.41 | MLC | yes |
| 13:52.55 | MLC | I did try the .crt file alone and also the concatenated .crt + .bundle that works in nginx |
| 13:54.59 | sibiria | begin with putting the private key in its own file. you may name this "privkey.pem" or something |
| 13:55.18 | MLC | yes. that is already done. |
| 13:55.23 | sibiria | then coalesce the certificates into another file. you may call that "fullchain.pem" |
| 13:55.54 | sekil | forgotmynick: calling plan is one thing...direct routing is other...I think E5 is needed for DR |
| 13:56.09 | MLC | which goes first, the bundle or the cert in that fullchain file? |
| 13:56.27 | sibiria | *most* software is ok with random order, but specifications actually demand an ascending order |
| 13:56.33 | sibiria | first you put your subject, meaning your domain cert |
| 13:56.38 | sibiria | after that you place the issuer of that cert |
| 13:56.43 | sibiria | then repeat |
| 13:56.55 | sibiria | e.g. your cert -> eventual intermediate -> root cert |
| 13:57.04 | MLC | ok, trying that |
| 13:57.32 | sibiria | and just to underline, the key should be alone in its separate file. no cert in there |
| 13:57.37 | sibiria | and ditto for the certs |
| 13:57.47 | MLC | ok |
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| 14:01.17 | sibiria | if you want, you may verify the privkey and the cert to really make sure they match |
| 14:01.23 | sibiria | you can do this with openssl like so: |
| 14:01.40 | sibiria | openssl x509 -noout -modulus -in fullchain.pem | openssl sha1 |
| 14:01.52 | sibiria | openssl rsa ... -in privkey.pem | openssl sha1 |
| 14:04.02 | sibiria | you can also verify that the cert aligns with the CA's cert: openssl verify -verbose -CAfile fullchain.pem fullchain.pem |
| 14:05.29 | MLC | the first 2 give matching results. I assume this is good? |
| 14:06.06 | MLC | and the last one reports OK |
| 14:06.26 | sibiria | yes, the private key and the cert yields the same checksum, so they match |
| 14:06.32 | sibiria | and OK is... OK :) |
| 14:06.44 | sibiria | so your certificate and private key files should be in order |
| 14:07.57 | MLC | so with fullchain.pem containing cert then bundle I get the same symptoms. There are 2 certs in the bundle. Maybe they are in the wrong order? |
| 14:08.11 | sibiria | the correct order is what i stated earlier |
| 14:08.37 | sibiria | first your cert, and then the issuer of that cert, and then the issuer of THAT cert etc. |
| 14:08.50 | MLC | I'll break the bundle up so that I can look at each part |
| 14:09.11 | sibiria | if your certs have attributes in plain text, they will indicate what's what |
| 14:09.20 | MLC | they don't |
| 14:09.30 | sibiria | aka "bag attributes" |
| 14:10.02 | sibiria | ok, you may take a look at the separate certs then: openssl x509 -in some_cert_file -text -noout |
| 14:10.48 | sibiria | just curious, what client software is it you're using that doesn't like the cert? |
| 14:11.06 | MLC | Bria on iPhone, also got similar symptoms from a Digium phone |
| 14:11.19 | sibiria | i think i've only once ever run into software demanding the correct chain order of a cert |
| 14:12.00 | MLC | I think it may not be the client but asterisk. Still seeing this: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000 |
| 14:12.05 | MLC | in asterisk |
| 14:13.36 | sibiria | i presume the priv key isn't password-encrypted? :P |
| 14:13.49 | MLC | it is not |
| 14:13.49 | sibiria | and that it's readable by asterisk (file ownership/permissions) |
| 14:15.38 | MLC | permissions look ok |
| 14:16.46 | MLC | The first cert in the bundle has |
| 14:16.47 | MLC | <PROTECTED> |
| 14:16.47 | MLC | <PROTECTED> |
| 14:16.47 | MLC | <PROTECTED> |
| 14:16.55 | MLC | And the 2nd |
| 14:16.55 | MLC | Authority Information Access: |
| 14:16.55 | MLC | <PROTECTED> |
| 14:17.03 | MLC | I think that means they are in the correct order? |
| 14:17.20 | sibiria | not sure from just that. the subject and issuer are the two details that are important |
| 14:18.38 | MLC | First: |
| 14:18.38 | MLC | Issuer: C = US, ST = New Jersey, L = Jersey City, O = The USERTRUST Network, CN = USERTrust RSA Certification Authority |
| 14:18.49 | sibiria | subject: you, issuer: subca1/ca1 -> subject: subca1/ca1, issuer: big corp ca -> subject: big corp ca |
| 14:18.52 | MLC | Subject: C = GB, ST = Greater Manchester, L = Salford, O = Sectigo Limited, CN = Sectigo RSA Domain Validation Secure Server CA |
| 14:20.05 | *** join/#asterisk lyr (~lyr@51.15.165.159) |
| 14:20.10 | lyr | Hi there |
| 14:20.26 | MLC | got it. I think they are in the correct sequence |
| 14:20.33 | lyr | I'm trying to get prometheus metrics configured |
| 14:20.33 | sibiria | sectigo is an intermedia |
| 14:20.40 | sibiria | intermediate |
| 14:20.54 | MLC | the 2nd is USERTrust RSA Certification Authority |
| 14:20.58 | sibiria | their cert is signed by USERtrust |
| 14:21.10 | lyr | I'm not used to asterisk conf, I dropped a prmetheus.conf file in /etc/asterisk, restarted asterisk, nothing in the log, where's the /metrics ? |
| 14:21.36 | sibiria | a simple way to identify a root CA is to see if its issuer is the same as the subject |
| 14:21.50 | sibiria | which you should see on the USERTrust cert |
| 14:22.00 | sibiria | so that's the one you place last in the file |
| 14:22.33 | sibiria | and the one issued by USERtrust (subject sectigo) goes before |
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| 14:22.44 | sibiria | and in the top your subject, which will have sectigo as issuer |
| 14:23.59 | MLC | I have it that way, except that the issuer != subject on the final one |
| 14:24.59 | sibiria | so the last one's subject is USERTrust RSA CA or something, but issued by another CA? |
| 14:25.15 | MLC | yes |
| 14:25.15 | MLC | Issuer: C = SE, O = AddTrust AB, OU = AddTrust External TTP Network, CN = AddTrust External CA Root |
| 14:25.16 | MLC | Subject: C = US, ST = New Jersey, L = Jersey City, O = The USERTRUST Network, CN = USERTrust RSA Certification Authority |
| 14:26.21 | sibiria | i guess your digium phone doesn't have that addtrust caroot available |
| 14:26.55 | sibiria | iirc that's one of comodo's root cas |
| 14:27.06 | MLC | Sounds right. It is a Comodo cert |
| 14:27.28 | MLC | I'll try a let's encrypt cert and see what happens. |
| 14:28.18 | sibiria | sounds like what you got from ssls.com is a cross-signed product and the clients don't want to acknowledge it |
| 14:30.42 | MLC | Have you used Let's Encrypt for asterisk? It seems to be geared toward web sites. |
| 14:31.49 | sibiria | i have. certs are transparent in this regard, there's no such thing as web cert, telephony cert etc. |
| 14:32.57 | sibiria | but no matter where you get the cert from, in the end your clients must still have a root bundle that connects the whole chain in order to verify trust |
| 14:33.12 | MLC | makes sense |
| 14:33.17 | sibiria | if you have a client with a really outdated certstore then you may run into problems no matter where you get the cert |
| 14:35.07 | MLC | Any thoughts on this error in asterisk? SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000 |
| 14:39.24 | sibiria | no idea, it's a bit undescript |
| 14:39.34 | sibiria | old pjsip/chan_sip? |
| 14:40.11 | MLC | I am 16.6.2 which is a few points behind current |
| 14:41.34 | sibiria | are you giving asterisk a ca list file/path or letting it use the system defaults? |
| 14:41.52 | sibiria | maybe it's just a simple pjsip config problem after all |
| 14:42.35 | MLC | I'm only specifying the cert_file and priv_key |
| 14:43.15 | sibiria | priv_key_file* you mean |
| 14:43.34 | MLC | duh, no |
| 14:43.48 | MLC | smacks his head and will fix that |
| 14:43.49 | sibiria | explodes |
| 14:44.12 | sibiria | at least your cert chain and key is now in order............ |
| 14:44.29 | MLC | no, I have it correct. priv_key_file |
| 14:45.17 | sibiria | for my let's encrypt setup i only point pjsip to the cert and key as well. it should suffice |
| 14:46.06 | MLC | I'll figure out the ACME / certbot thing for Let's Encrypt and try that |
| 14:49.02 | jjrh | Anyone here ever played around with a android based handset? (like the grandstream android line, or yealink ) |
| 14:49.08 | igcewieling | When a cert expires (90 days for certbot, I think) Does Asterisk re-read the cert file when it expires or does it need to be told to reload the new cert |
| 14:49.25 | jjrh | Not sure if there are any other companies shipping android deskphones handsets |
| 14:49.53 | sibiria | igcewieling: i think a minimum of reloading pjsip is needed |
| 14:50.10 | sibiria | and yes, it's 90 days for Let's Encrypt's certs |
| 14:50.31 | igcewieling | https://www.ui.com/unifi-voip/uvp/ is basically an android phone with a handset |
| 14:50.58 | jjrh | nifty |
| 14:51.16 | jjrh | not a big fan of having no physical buttons on that model |
| 14:52.54 | igcewieling | *nod* If I wanted a SIP WiFi phone, there are hundreds of models around to choose from. I don't think sticking one in a cradel counts as anything but a disappointment. |
| 14:54.07 | igcewieling | I stick with Polycom because we have at least 1,000 Polycom phones and wrote our own EPM-like program almost 10 years ago to handle setup. |
| 14:54.56 | jjrh | I'm more interested in writing some software for these devices. If they run android it opens up a lot of possibilities for using non sip voice and video |
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| 14:56.15 | igcewieling | Never underestimate the stupidity of users. |
| 14:56.54 | jjrh | in this case i'm mostly interested in something for myself or a couple clueful users |
| 14:59.44 | jjrh | the gxv3350 is actually a somewhat reasonable price - $300USD the yealink stuff is nice but really expensive |
| 15:04.22 | drmessano | I wanted a phone that I could run existing Android apps on, specifically Pushover. |
| 15:07.14 | *** join/#asterisk thansen (~thansen@192.74.130.86) |
| 15:11.22 | MLC | sibiria: getting the same results with a Let's Encrypt key. I'll try upgrading asterisk to current. |
| 15:21.34 | *** join/#asterisk matrix1233 (~matrix123@80.215.34.196) |
| 15:22.35 | *** join/#asterisk miralin (~Thunderbi@host-46-45-217-207.stavropol.ru) |
| 15:24.29 | *** join/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 15:24.47 | CRCinAU | Hi all, so mucking around with 17.x.x, it gives a warning to move to pjsip |
| 15:25.01 | igcewieling | YAY! |
| 15:25.05 | CRCinAU | does pjsip handle multiple registrations against the same hostname now? |
| 15:25.10 | CRCinAU | ie two accounts on the same sip server? |
| 15:25.25 | CRCinAU | in the past when I've tried to switch, its never been able to route calls properly |
| 15:26.57 | igcewieling | Asterisk has never had that problem. PJSIP allows you to register multiple devices to the same "account". Asterisk has always allowed multiple devices to register from the same IP to different accounts, otherwise NAT would never work with Asterisk. |
| 15:27.19 | CRCinAU | shrugs |
| 15:27.33 | CRCinAU | I know it has had that problem - but I haven't tried again in a number of years. |
| 15:27.58 | CRCinAU | I have 2 upstream providers, both have two sip accounts on each server - ie different username / password |
| 15:28.08 | *** join/#asterisk miralin1 (~Thunderbi@85.115.248.115) |
| 15:28.36 | CRCinAU | calls from the upstream to either of accounts on the same provider would always drop into one of the lines |
| 15:28.45 | CRCinAU | ie you could never get an incoming call on the second account |
| 15:29.36 | CRCinAU | from memory, it was something to do with pjsip using the hostname to match incoming calls |
| 15:29.37 | igcewieling | An example: https://pastebin.com/PRTfzD5x |
| 15:30.45 | CRCinAU | is that the same for the registrations? |
| 15:30.56 | CRCinAU | whatever the translation of 'sip show registry' is? |
| 15:31.00 | igcewieling | those are the registrations of the phones to the server. |
| 15:31.13 | *** join/#asterisk _abc_ (~usre@unaffiliated/ccbbaa) |
| 15:31.26 | igcewieling | sip show registery shows remote devices you are registered TO. It does not show devices registered to the server. |
| 15:31.31 | _abc_ | Hi. Is there any documentation on * concerning SIP MESSAGE handling? |
| 15:31.34 | CRCinAU | yeah - I'm talking about Asterisk (with pjsip) registering to an upstream SIP server |
| 15:32.12 | CRCinAU | handsets registering to Asterisk from the same IP isn't an issue |
| 15:32.20 | igcewieling | I don't think I've set that sort of thing up in 10 years. |
| 15:32.37 | CRCinAU | its kinda coming back to me now..... |
| 15:32.58 | CRCinAU | it was how pjsip handled the INCOMING call from the upstream provider.... it matched the first hostname match, then associated the calls with that |
| 15:33.08 | igcewieling | All of my servers have public static IPs so I've not had to register one server to another server in many years |
| 15:33.09 | CRCinAU | although it may have been the wrong account that it matched first |
| 15:33.23 | igcewieling | CRCinAU: you tell pjsip the match order. |
| 15:33.36 | igcewieling | can't do that with chan_sip. |
| 15:33.49 | CRCinAU | in chan_sip, I register each endpoint with FNN/FNN |
| 15:33.55 | CRCinAU | (fnn = full national number) |
| 15:34.13 | CRCinAU | which is how it tells the difference - but that didn't work with pjsip - not sure if that's changed now |
| 15:34.50 | igcewieling | endpoint_identifier_order=username,ip |
| 15:35.22 | CRCinAU | however, pjsip was rather new back then.... so yeah - I'm assuming a bit has changed |
| 15:37.30 | drmessano | I don't recall that being an issue with PJSIP |
| 15:38.12 | drmessano | Bur it certainly works fine |
| 15:38.23 | *** join/#asterisk miltux (~miltux@94-225-27-17.access.telenet.be) |
| 15:38.36 | seanbright | _abc_: what is your question |
| 15:39.14 | CRCinAU | yeah - on another note, it doesn't look like the sip_to_pjsip.py handles multiple upstream accounts on the same server :) |
| 15:39.33 | CRCinAU | it creates a [blah] with username / username / password / password :) |
| 15:39.40 | CRCinAU | at least the version in 16.9.0 anyway |
| 15:40.14 | _abc_ | seanbright: whether SIP MESSAGE settings or treatment are mentioned anywhere in the * docs. I can't find anything because all SIP messages are SIP messages. |
| 15:40.20 | _abc_ | aka requests |
| 15:41.04 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_MessageSend |
| 15:41.12 | _abc_ | seanbright: in particular SIP MESSAGE (chat) from SIP client to SIP client both on the same * proxy server and no reinvite, i.e. data flow through the *. |
| 15:42.04 | seanbright | asterisk isn't a proxy, but that isn't relevant |
| 15:42.16 | seanbright | take a look at MessageSend as igcewieling mentioned |
| 15:42.21 | _abc_ | It's a SIP server among other things, in chan sip terms |
| 15:42.29 | _abc_ | I missed that, looking. |
| 15:45.09 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@gateway/tor-sasl/ai9zo5ap) |
| 15:47.56 | Samot | A MESSAGE request is not the same as an INVITE request. |
| 15:48.55 | _abc_ | The page mentions pjsip enhanced MESSAGE support since 13.22 . Does anyone know if 13.14 had any message support properly working on SIP (not pjsip)? This is what's on this machine, default distribution (stretch) * version, will have to connect to another for a higher version. |
| 15:50.39 | sibiria | you can send MESSAGEs with chan_sip, too, since ages |
| 15:50.39 | seanbright | can you type out the word 'asterisk' please? |
| 15:52.53 | Samot | OK if you're trying to send a MESSAGE from a device to Asterisk, the first thing you need to do is tell the Chan_SIP peer that it can accept out of call messages and where to process them. |
| 15:53.14 | Samot | That is covered in the sip.conf samples and in the Wiki for chan_sip. |
| 15:59.34 | _abc_ | Does cmp2k work nicely in modern android phones? https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone |
| 16:03.19 | MLC | sibiria: FYI upgrading to 16.9.0 did not solve that TLS issue |
| 16:05.00 | sibiria | MLC: sorry, i have no other advice, but i would at least again verify that the full path and cert/key file really really is readable by the user/group asterisk runs under |
| 16:16.09 | *** join/#asterisk meek424 (~stefan@host62-7-189-109.range62-7.btcentralplus.com) |
| 16:17.26 | meek424 | hi there, anyone can please help me to install the sRTP module on an existing asterisk server |
| 16:17.59 | meek424 | just need some high level instructions, maybe an article |
| 16:22.30 | *** join/#asterisk MLC (~MLC@63.249.40.11) |
| 16:22.42 | Samot | Have you tried loading the module? |
| 16:23.02 | jjrh | drmessano: yeah just having a android phone on the desk would open up a ton of nice possibilities. Using stuff like skype, discord, etc |
| 16:23.21 | _abc_ | Is there a way to issue a message using MESSAGE(body) and MessageSend(to) from the cli without editing the dialplan? In place macro or interpretation? |
| 16:23.35 | meek424 | Samot: nope |
| 16:23.50 | Samot | meek424: Then you should. |
| 16:24.08 | meek424 | Samot: hehe, will do |
| 16:24.20 | meek424 | Samot: thx |
| 16:24.34 | igcewieling | _abc_: no. You could use something which generates SIP messages like sipp or you could create call files to trigger a call. |
| 16:24.49 | _abc_ | Also, is there a way to batch it? using pipe | asterisk -rx perhaps? |
| 16:24.56 | igcewieling | no. |
| 16:24.57 | Samot | Batch what? |
| 16:24.59 | _abc_ | ok |
| 16:25.04 | Samot | You're being very random. |
| 16:25.13 | _abc_ | Samot: originate a MESSAGE using a shell command. |
| 16:25.19 | jjrh | use sipsak |
| 16:25.36 | igcewieling | You can't run random applications from the Asterisk CLI. Asterisk -rx is considered "the CLI" in this case. |
| 16:25.49 | _abc_ | Yes for cli, understood about cannot run. |
| 16:25.54 | jjrh | That is if you're trying to do SIMPLE messaging. |
| 16:26.06 | _abc_ | quips cisco has tclsh in the clish path on exe cli... |
| 16:26.26 | _abc_ | jjrh: simple messaging is perfect for now. Looking, thanks. |
| 16:26.38 | meek424 | Samot: is not here, so i guess i have to install it from source somehow (ls /usr/lib/asterisk/modules/ | grep srtp) |
| 16:26.53 | jjrh | to send the messages look at sipsak. |
| 16:26.59 | igcewieling | meek424: just run menuconfig and select the module to build |
| 16:27.02 | Samot | meek424: How did you try to load the module? |
| 16:27.02 | _abc_ | Is voip-info.org considered obsolete? It is very good sometimes but information is seriously out of date on many pages. |
| 16:27.08 | seanbright | yes |
| 16:27.08 | igcewieling | this isn't frickin rocket science. |
| 16:27.22 | seanbright | voip-info is hot garbage |
| 16:27.38 | jjrh | seanbright: what's the alternative? |
| 16:27.41 | _abc_ | Like 15 years out of date to quote a real example. |
| 16:27.51 | igcewieling | wiki.asterisk.org |
| 16:28.06 | seanbright | the wiki, this channel, the forums, etc. |
| 16:28.17 | _abc_ | hot garbage it may be but there are good ideas sometimes in it |
| 16:28.19 | seanbright | stop giving voip-info the add revenue and hopefully they go away |
| 16:28.28 | seanbright | ad* |
| 16:28.37 | _abc_ | is accept_outofcall_message=yes relevant for SIP MESSAGE working well now? |
| 16:28.45 | _abc_ | seanbright: I have adblock on |
| 16:29.01 | jjrh | wiki doesn't have the same number of examples as voip-info unfortunately. |
| 16:29.29 | seanbright | examples of what? |
| 16:29.36 | seanbright | disinformation? |
| 16:29.38 | seanbright | we'll get on that |
| 16:29.40 | igcewieling | jjrh: voip-info can be useful for example/samples, but never consider it an authoritative source |
| 16:29.42 | _abc_ | rephrased: a) check date relevance on voip-info b) exert due diligence editing ideas which are 15 years old |
| 16:30.00 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 16:30.08 | *** join/#asterisk irrgit (~ch33se@parajsa.chat) |
| 16:30.33 | jjrh | seanbright: rough examples of stuff - like igcewieling, you have to use your judgement, look at the wiki documentation to determine how wrong it is |
| 16:30.51 | seanbright | i respectfully disagree |
| 16:31.15 | igcewieling | the sample configs are also useful. |
| 16:32.27 | jjrh | Yes. |
| 16:32.29 | Samot | So using Chan_SIP with MessageSend() and accepting MESSAGEs has been documented on the Internet since Asterisk 10 when this was introduced. |
| 16:32.47 | Samot | The current issue we see these days are people trying to use those examples with PJSIP. |
| 16:35.16 | Samot | MESSAGEs are accepted on a peer/endpoint by default. Both Chan_SIP and PJSIP. If the message context is not specified on the peer/endpoint then the default context= is used. |
| 16:35.37 | Samot | This is documented in the sample files. |
| 16:35.52 | jjrh | I'm mostly saying the wiki could use some more examples - it's why people end up on voip-info. |
| 16:36.22 | Samot | Well |
| 16:36.32 | Samot | Then that means either Digium or the community has to do that |
| 16:36.36 | seanbright | examples of what? |
| 16:36.39 | Samot | ^^^^ |
| 16:36.46 | Samot | That is a 100% valid question. |
| 16:36.53 | _abc_ | Is there perhaps a debug extension for asterisk to play with various Set() and application function calls on the cli without reloading config files all the time? |
| 16:37.18 | seanbright | jjrh: examples of what? |
| 16:37.41 | _abc_ | I was able to do some diaplan set global ... -- is that a way to at least set vars on cli at runtime? |
| 16:38.22 | Samot | No. |
| 16:38.33 | jjrh | mostly of application usage. https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Application_Transfer for example doesn't provide anything. While I realise it's obvious to most, it doesn't hurt to give a simple one line example. |
| 16:38.36 | Samot | _abc_: If you want to do MESSAGEs you need to do this in the dialplan. Period. |
| 16:39.02 | jjrh | I'm not complaining, I get someone has to do this. I'm only pointing out /why/ people may go to voip-info |
| 16:39.19 | _abc_ | Is there a reason for not having testing/interpreter mode commands implemeted on the cli? Samot it is not for MESSAGEs it is for general debugging of expressions and such. |
| 16:39.52 | Samot | exten => _X.,n,Transfer(PJSIP/100) |
| 16:39.57 | Samot | There's your example. |
| 16:40.01 | jjrh | exactly. |
| 16:40.20 | Samot | So which part of that is consistent? |
| 16:40.28 | seanbright | the 'Arguments' documention is pretty sparse on that page |
| 16:40.30 | Samot | Transfer([Tech/destination]) <- That part |
| 16:40.31 | seanbright | and that is being generous |
| 16:40.33 | meek424 | igcewieling: I've issued "make menuconfig" but i cant select res_srtp.so it has in front [XXX] what does that mean please? |
| 16:41.05 | igcewieling | that means something isn't loaded. It should tell you. |
| 16:41.13 | Samot | 12:27:01 PM <Samot> meek424: How did you try to load the module? |
| 16:41.14 | sibiria | it means you don't have the necessary development libraries installed |
| 16:41.16 | jjrh | Samot: for people learning asterisk, that one line is exactly what they might need. |
| 16:41.20 | igcewieling | usually it means the required libraties are not installed. |
| 16:41.30 | Samot | jjrh: That's basic dialplan |
| 16:41.49 | jjrh | Exactly, but most of the stuff on voip-info is basic stuff. |
| 16:42.00 | Samot | Then contribute. |
| 16:42.06 | Samot | There is the Wiki, there is the forum. |
| 16:42.10 | Samot | Someone has to write that crap up |
| 16:42.14 | sibiria | meek424: what OS are you building on? |
| 16:42.21 | Samot | Which means someone has to do it as well to prove it works. |
| 16:42.22 | jjrh | look i'm only pointing out /why/ voip-info continues to exist. |
| 16:42.35 | sibiria | you need openssl dev package, libsrtp and libsrtp's dev package |
| 16:43.03 | Samot | jjrh: Please show me where everything post Asterisk 12 is in there? |
| 16:43.12 | Samot | CDR updates, CEL updates, PJSIP |
| 16:43.15 | jjrh | It's most likely not. |
| 16:43.19 | Samot | All the new stuff that has happened.. |
| 16:43.21 | jjrh | it's not a good source. |
| 16:43.22 | seanbright | ok |
| 16:43.28 | seanbright | get off jjrh's ass |
| 16:43.35 | Samot | Hah. |
| 16:43.47 | seanbright | everyone has made their respective points |
| 16:43.48 | seanbright | let's move on |
| 16:43.50 | jjrh | I'm really not defending voip-info. |
| 16:43.57 | Samot | 12:27:01 PM <Samot> meek424: How did you try to load the module? |
| 16:44.05 | Samot | I've just been waiting for that. |
| 16:44.46 | jjrh | But yeah little off topic so sorry about that :) |
| 16:44.55 | meek424 | Samot: sorry, I've done this --> I've issued "make menuconfig" but i cant select res_srtp.so it has in front [XXX] |
| 16:45.12 | meek424 | sibiria: Debian 10 minimal |
| 16:45.17 | Samot | So you didn't do the basic: module load res_srtp.so |
| 16:46.02 | sibiria | meek424: then the packages you need are: libssl-dev, libsrtp2-1 and libsrtp2-dev. you may need to run configure again to make sure everything is picked up correctly |
| 16:46.14 | meek424 | Samot: the whole think is missing, --> [Apr 15 17:45:44] ERROR[1191]: loader.c:281 module_load_error: Error loading module 'res_srtp.so': /usr/lib/asterisk/modules/res_srtp.so: cannot open shared object file: No such file or directory |
| 16:46.45 | meek424 | sibiria: cool, i will install them, brb |
| 16:52.25 | _abc_ | What do you suggest to people running debian whose asterisk in-distribution version is usually hopelessly out of date? Like mine? Revert to out of distrubution source builds? |
| 16:52.42 | sibiria | build your own |
| 16:53.11 | _abc_ | pines for the simple life in the 1990s with slackware and build everything from source after overnight download on modem line. |
| 16:53.59 | sibiria | iirc debian buster's package repo has asterisk 16 finally |
| 16:54.20 | sibiria | albeit an earlier v16 |
| 16:54.55 | joepublic | 16.2.1 in debian buster, vs. current 16.9(?) |
| 16:55.18 | meek424 | sibiria: "res_srtp [*]" is what I'm looking for right? |
| 16:55.22 | joepublic | not fresh, but not hopelessly out of date either |
| 16:55.52 | sibiria | meek424: yes. i'm uncertain if pjsip has some separate module for TLS, though i don't think it does |
| 16:56.34 | sibiria | and i believe all of it should be selected by default anyway |
| 16:56.38 | meek424 | I'll be using sip only, not touching pjsip |
| 16:56.44 | meek424 | yes it was already selected |
| 16:56.59 | sibiria | you should be touching pjsip. no touching chan_sip |
| 16:58.02 | meek424 | hang on, I thought is either one or another chan_pjsip or chan_sip |
| 16:58.22 | sibiria | pjsip is your friend. pjsip is the beginning and the end. mother and father. the alpha and the omega, the all and everything |
| 16:59.13 | sibiria | chan_sip has imploded and any associating with it may or may not lead to headache, shortness of breath, facial rashes and even sudden death |
| 16:59.19 | drmessano | I wish people would stop calling it "sip" and "pjsip" ... It's chan_sip and chan_pjsip, which are both providing "SIP". chan_sip is old and should not be used |
| 16:59.45 | sibiria | meek424: you can use chan_sip and pjsip next to eachother, if absolutely necessary |
| 17:00.24 | meek424 | is see |
| 17:00.44 | jjrh | pjsip library is pretty handy too. |
| 17:00.47 | meek424 | so am Im right to completely move of sip.conf to pjsip.conf? |
| 17:00.54 | sibiria | you are |
| 17:01.00 | drmessano | ditch everything chan_sip |
| 17:01.05 | drmessano | rm -rf it |
| 17:01.08 | jjrh | chan_sip will eventually be deprecated won't it? |
| 17:01.20 | meek424 | cool |
| 17:01.21 | sibiria | it's a bit unintuitive to get going with pjsip but the configuration wizard helps a lot |
| 17:01.22 | drmessano | It already is |
| 17:01.55 | _abc_ | Content-Type: application/im-iscomposing+xml in SIP MESSAGE requests is supported since what version of 13.x please? If at all? |
| 17:02.23 | drmessano | chan_sip is getting no new features and is communtiy-supported at this point |
| 17:02.30 | drmessano | So it's dead |
| 17:04.02 | jjrh | ah I knew it wasn't getting new features but didn't know it was community supported. |
| 17:04.49 | meek424 | is it possible to reply to multiple users at once? |
| 17:05.08 | _abc_ | pjsip and sip are mutually exclusive, right? Can't run both at the same time? |
| 17:05.34 | jjrh | according to sibiria you can. |
| 17:05.50 | _abc_ | Hmm yes both are on here |
| 17:05.50 | Samot | Yes, you can. |
| 17:05.57 | sibiria | you can, but they'll have to listen on separate ports of course |
| 17:07.46 | *** join/#asterisk FH_thecat (~FH_thecat@75.11.25.212.ftth.as8758.net) |
| 17:08.46 | drmessano | Running chan_sip and chan_pjsip at the same time is like driving cross country on 3 good tires and the undersized donut |
| 17:08.52 | drmessano | Just commit |
| 17:11.23 | jjrh | Yeah i'm not sure why one would bother pjsip can do everything chan_sip can now I believe? |
| 17:11.40 | sibiria | and then some |
| 17:12.21 | MLC | Does anyone know how to tell the Digium phone not to verify the server certificate? |
| 17:12.21 | meek424 | shot-out to sibiria: Samot: igcewieling: for helping me achieve my first tls/srtp call |
| 17:12.34 | _abc_ | is the content type mentioned above in MESSAGES supported in pjsip in 13.14 ? |
| 17:13.02 | _abc_ | MLC: self generated / signed? |
| 17:13.13 | sibiria | MLC: maybe the digium phone can have its certstore updated instead? |
| 17:13.13 | MLC | no |
| 17:13.37 | sibiria | feed it new/updated CAs |
| 17:13.48 | MLC | sibiria: there is mention of that https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phones+and+Secure+Calling but no indication of how to do so |
| 17:14.50 | igcewieling | MLC: have you made any progress at all in the past few days? |
| 17:14.58 | MLC | no |
| 17:15.23 | MLC | well, I can get Bria on iPhone to connect when it doesn't verify the cert |
| 17:18.21 | sibiria | seems odd it can't verify even the Let's Encrypt cert. it should be using iOS' native CA bundles, and those are up-to-date |
| 17:20.44 | _abc_ | This is driving me batty. 2 android phones using SIP MESSAGE to talk to each other in linphone 4.0.1 through an asterisk sip chan on same lan/wifi. Same version on both? One sends empty messages to the other, the other sends nothing to the 1st. |
| 17:20.45 | MLC | Ahh, breakthough. |
| 17:21.16 | _abc_ | And asterisk sip debug messages show 200 codes in both directions. And the correct message bodies. |
| 17:21.17 | MLC | I was still using the IP address on the proxy statement which directed it to port 5061. When I changed that to the domain name on the cert the Bria can connect. |
| 17:23.16 | sibiria | yes that's paramount for verification. a host name (or SNI) is required |
| 17:23.52 | MLC | yup. I kept changing the main "domain" field and forgot about the proxy field. It's hidden in the Advanced section. |
| 17:25.40 | _abc_ | Apart from the ancient version of asterisk, do you see a problem with the dialplan script at [astsms]? http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html -- it fails succesfully here. |
| 17:27.07 | _abc_ | Meaning, it sends messages to the wrong recipient (== sender) among other things. |
| 17:28.57 | _abc_ | Set(ACTUALTO=${CUT(MESSAGE(to),@,1)}) would cut a header like: To: <sip:4002@192.168.0.101:48817;app-id=929724111839;pn-type=firebase;pn-tok=dap_tLRiViQ:APA91bFXVS-CUwR52Sqcser7BBzCB9Hp-zZCpUb_6XGrGPIrZtwzsUk0RaRHmTQZsGdJB-Osj-VTiL6FivunZ0I-OHEkgfK6oQ-epp0EZ0_twVR5C-hXHBA3hxPJ1TI42xrtRs_M-E1t;pn-silent=1;transport=udp> to just sip:4002 right? |
| 17:34.11 | igcewieling | Why don't you try it and see? |
| 17:34.32 | _abc_ | Something is very wrong with this. I see the SIP MESSAGE headers with From and To set to the SAME address in the debug log. These are asterisk printed debug messages. It already does this. I think I need to parse the Via header instead. Because this is not directmedia. |
| 17:34.42 | _abc_ | * not reinvite. |
| 17:37.10 | Samot | Why would you expect a re-INVITE on a MESSAGE transaction? |
| 17:37.17 | Samot | They aren't the same thing. |
| 17:37.38 | _abc_ | not a reinvite, no |
| 17:38.31 | _abc_ | Why would From and To be set to the same peer?! |
| 17:38.42 | _abc_ | is tired, evening here |
| 17:42.14 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:45.42 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-hcayvwtfvfayskcm) |
| 17:46.49 | _abc_ | Has anyone got a link online to a debug/dumped SIP MESSAGE request? I just can't follow the debug log now. |
| 17:47.04 | _abc_ | google failed to find one when I searched |
| 17:47.59 | _abc_ | How would one parse the IP address from "Via: SIP/2.0/UDP 192.168.0.110:5060;branch=z9hG4bK7e6e8ed1" using CUT()? 2 CUT()s nested? |
| 17:48.23 | seanbright | you would use something that is not asterisk |
| 17:48.30 | _abc_ | ? |
| 17:48.32 | seanbright | ? |
| 17:48.40 | _abc_ | use something that is not asterisk?! |
| 17:48.48 | seanbright | sorry... i meant * |
| 17:49.05 | _abc_ | Uhh. Rephrase please? |
| 17:49.09 | seanbright | parsing a SIP header? i am not sure that CUT (or multiple CUTs) will do it |
| 17:49.14 | seanbright | reliably |
| 17:49.36 | _abc_ | Did you see the diaplan script in the link I pasted above at [astsms ? |
| 17:50.04 | seanbright | no |
| 17:51.33 | Samot | Why would you even need to touch the via header? |
| 17:51.37 | igcewieling | I imagine if you strip off Via: SIP/2.0/UDP using the ${VARIABLE:PREFIXLEN} method |
| 17:52.12 | _abc_ | The sms goes out as-if from asterisk instead of from sender as is now. |
| 17:52.19 | igcewieling | Then take the result and use CUT with : delimiter and get the IP. The whole thing is a dumb idea. |
| 17:52.25 | _abc_ | This causes the recipient device to not display it. |
| 17:52.34 | MLC | sibiria: I got the Digium phone working also, the problem there was similar. |
| 17:52.45 | _abc_ | I agree igcewieling but I do not see how to persuade the script to report the real sender. |
| 17:53.08 | igcewieling | But the "real sender" is not the real sender. the real sender is Asterisk. |
| 17:53.22 | igcewieling | If you were using a SIP PROXY, then the real sender is the actual sender. |
| 17:53.26 | _abc_ | Well yes but I mean the real real sender. The initiator of the SIP MESSAGE |
| 17:53.42 | igcewieling | But Asterisk isnt a SIP proxy, it is a B2BUA, which means it makes calls |
| 17:54.04 | Samot | _abc_: How about you show your dialplan? |
| 17:54.38 | _abc_ | Samot: did you see the link above? 3rd time I use the script at [astsms] from within that. Only change is _X. instead of _X at all lines. |
| 17:54.38 | drmessano | _abc_: I never got that dialplan to work when 11 was new |
| 17:54.51 | MLC | Unfortunately, it seems that the mDNS that DPMA uses for the phones to configure can only listen on 5060/udp or 5061/tls, but not both. So I can't have a mixture of TLS and non-TLS phones. Still experimenting with that. |
| 17:54.54 | _abc_ | drmessano: oh. Okay. |
| 17:55.11 | _abc_ | drmessano: is there another which works now? |
| 17:55.16 | drmessano | No clue |
| 17:55.18 | Samot | Show me what you have. |
| 17:55.28 | drmessano | But if you want support for it, contact the author of the blog |
| 17:55.32 | drmessano | or login and comment |
| 17:55.40 | _abc_ | Samot: http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html seriously scroll down to [astsms] |
| 17:56.07 | Samot | OK then show me the output from the console. |
| 17:56.10 | _abc_ | drmessano: good idea, except he must have moved on a bit meanwhile. |
| 17:56.36 | _abc_ | Samot: that is too long to paste here, will try some more things first. |
| 17:56.40 | Samot | No |
| 17:56.46 | Samot | Put it on pastebin and give the link |
| 17:58.18 | igcewieling | I'm glad I've never received a request for sip messaging from my users. |
| 17:59.08 | _abc_ | SIP messaging can be very useful to pass urls and so on to peers, who may be using some other type of phone. Works on many desk phones too and android. |
| 17:59.30 | _abc_ | Spelling with NATO alphabet or pidgin versions thereof is so 1900s |
| 17:59.56 | Samot | OK. |
| 18:00.12 | igcewieling | Why not just send an instant message? |
| 18:00.13 | _abc_ | Will take a while need to save the scrollback and edit it. |
| 18:00.15 | Samot | I based my current MESSAGE sending on that link you provided _abc_ |
| 18:00.24 | _abc_ | Samot: ? |
| 18:00.27 | igcewieling | _abc_: don't edit the scrollback. |
| 18:00.31 | Samot | So for the last time, show me the output of this "not working" message. |
| 18:00.38 | _abc_ | igcewieling: do you MIND if I remove private info? |
| 18:00.54 | igcewieling | _abc_: the only "private" info you should remove is PASSWORDS. |
| 18:00.57 | _abc_ | Please wait a few 10s of seconds |
| 18:01.03 | _abc_ | igcewieling: and ips on nat |
| 18:01.34 | igcewieling | well if you don't want any help, I guess you could edit them out, but it makes much of the info usless. |
| 18:01.55 | igcewieling | I'm not going to dig through the scrollback regardless. |
| 18:01.56 | _abc_ | Anonymizing and deleting are 2 different things |
| 18:02.12 | drmessano | No, they are not |
| 18:02.25 | drmessano | Difficult to check for simple errors if you obfuscate the data |
| 18:02.45 | Samot | Well, I tried. |
| 18:02.51 | Samot | I have that example working |
| 18:02.56 | Samot | But hey, whatevs. |
| 18:03.29 | drmessano | #cv |
| 18:03.48 | Samot | It's worse than being rickrolled. |
| 18:03.58 | Samot | At least you can dance to that. |
| 18:05.35 | _abc_ | https://termbin.com/dhpf |
| 18:05.47 | _abc_ | I said wait, I did not say flame & rant :) |
| 18:05.58 | Samot | That's not what I asked for. |
| 18:06.02 | igcewieling | for fuckssake, that isn't the CLI output. |
| 18:06.05 | _abc_ | hm? |
| 18:06.12 | Samot | I want the output from the Asterisk CLI |
| 18:06.20 | _abc_ | That is CLI output for sip set debug peer 4001 and 4002 |
| 18:06.29 | Samot | I want to see Asterisk processing this. |
| 18:06.31 | Samot | No. |
| 18:06.33 | Samot | FFS. |
| 18:06.41 | _abc_ | What debug settings should I set |
| 18:06.43 | Samot | core set verbose 10 |
| 18:06.47 | drmessano | lol |
| 18:06.54 | _abc_ | eh. |
| 18:07.07 | igcewieling | THIS is CLI output https://pastebin.com/Z53s1jX2 |
| 18:07.38 | igcewieling | see how it is executing applications and running dialplan? THAT is what is useful. |
| 18:09.22 | _abc_ | https://termbin.com/zrne |
| 18:09.42 | _abc_ | core verbose level is 10 as you asked |
| 18:09.52 | _abc_ | Again, it fails succesfully. I hate that. |
| 18:11.16 | Samot | The SUCCESS is if the MessageSend() executed successfully |
| 18:11.17 | _abc_ | So the script does what it should, the bug is elsewhere. |
| 18:11.24 | Samot | Not that the message was successfully sent. |
| 18:11.33 | Samot | Well your from is wrong |
| 18:11.38 | _abc_ | exit non zero means error sending? |
| 18:11.42 | Samot | There shouldn't be any <> |
| 18:11.55 | _abc_ | This is what linphone puts there. I have no control over it? |
| 18:12.15 | _abc_ | Aren't uri's allowed to have the <> in them? |
| 18:12.21 | seanbright | no |
| 18:12.26 | igcewieling | _abc_: exit non-zero is an internal message and means nothing. |
| 18:12.32 | _abc_ | Also, note these linphones are set up to talk to each other and they work fine in the same setup |
| 18:12.42 | _abc_ | igcewieling: ah. |
| 18:12.45 | Samot | _abc_: Then you need to parse the from header better. |
| 18:12.49 | seanbright | wait |
| 18:13.01 | seanbright | where is the specific dialplan you are using? |
| 18:13.07 | _abc_ | Samot: delimiting them, no? Foo: <this@uri> "and some noise" |
| 18:13.35 | _abc_ | seanbright: the dialplan contains just the default echo and moh numbers fromt he demo and two extensions. |
| 18:13.41 | *** join/#asterisk miralin (~Thunderbi@178.34.160.50) |
| 18:13.43 | seanbright | i want to see the dialplan |
| 18:13.57 | _abc_ | All of it?! |
| 18:14.01 | igcewieling | seanbright: the result dialplan: https://termbin.com/zrne |
| 18:14.11 | seanbright | i see the output |
| 18:14.22 | seanbright | what is in the astsms context? |
| 18:14.37 | seanbright | the dialplan, not the output of a test |
| 18:14.43 | seanbright | i want to see the dialplan |
| 18:14.44 | _abc_ | The script from the link I pasted. 4th time. I'll pastebin if ffs. |
| 18:14.45 | seanbright | dialplan |
| 18:15.06 | seanbright | _abc_: thank you for your patience with us |
| 18:15.27 | _abc_ | I thank you |
| 18:15.39 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 18:16.26 | seanbright | ok, i found your dialplan |
| 18:16.26 | seanbright | it |
| 18:16.34 | seanbright | it's identical to what is on the page? |
| 18:17.12 | seanbright | hello? |
| 18:17.13 | _abc_ | https://termbin.com/sn87 I cut out all the comments and bracketed out sections |
| 18:17.24 | *** join/#asterisk javi404 (~quassel@unaffiliated/javi404) |
| 18:17.25 | _abc_ | Sorry I can't type while doing something in another term. |
| 18:17.52 | seanbright | ok, so i see the dialplan and i see the output |
| 18:18.02 | seanbright | what problem are you trying to solve exactly? |
| 18:18.08 | _abc_ | It's identical to that page except all _. are now _X. per asterisk startup message hint when loading the original. |
| 18:18.18 | seanbright | great |
| 18:18.20 | _abc_ | seanbright: I can't send text messages between 2 clients. |
| 18:18.32 | seanbright | gotcha |
| 18:18.36 | _abc_ | Resulting in the SIP debug logs I pasted 1st |
| 18:18.55 | seanbright | i didn't see the SIP debugs but i will scroll back and look for them |
| 18:18.56 | _abc_ | The phones call each other etc all is well. |
| 18:19.07 | seanbright | these? https://termbin.com/dhpf |
| 18:19.08 | _abc_ | https://termbin.com/dhpf seanbright |
| 18:19.12 | _abc_ | yes |
| 18:19.13 | seanbright | ok |
| 18:19.29 | _abc_ | You can ignore the protocol not supported messages/answers that is not present in asterisk apparently |
| 18:19.50 | _abc_ | There should be some more logging in asterisk somewhere? On why it fails on sending perhaps? |
| 18:20.09 | seanbright | so in that trace, the second message is the one coming in to asterisk, correct? |
| 18:20.16 | seanbright | that is what the logs indicate |
| 18:20.19 | _abc_ | I just pared the config down to just this, everything else is disabled/turned off/not connected |
| 18:20.34 | Samot | Show the 4002 peer |
| 18:20.41 | Samot | The config for it. |
| 18:20.45 | _abc_ | seanbright: The 1st message sent is test the 1st request |
| 18:20.50 | seanbright | so lingphone is sending garbage to asterisk and you want asterisk to send that garbage elsewhere? |
| 18:20.50 | _abc_ | sip.conf? |
| 18:20.57 | Samot | Yes |
| 18:21.10 | _abc_ | No, I want it to react properly on the 1st message which is text/plain |
| 18:21.18 | seanbright | the first message is not from linphone |
| 18:21.23 | _abc_ | And the answer which is typed on the other phone which is also text/plain 'what?' |
| 18:21.26 | seanbright | it's from asterisk |
| 18:21.59 | seanbright | the message 'test' is from asterisk |
| 18:22.03 | seanbright | we're all in agreement on that? |
| 18:22.05 | _abc_ | I must have cut something too short. I'll look again. I have saved the scrollback which is about 10,000 lines. Need to edit that a bit. |
| 18:22.12 | _abc_ | Yes agree. Just a minute. |
| 18:22.14 | seanbright | ok |
| 18:22.24 | seanbright | there should be two MESSAGEs |
| 18:22.33 | seanbright | something to asterisk and then something from asterisk |
| 18:22.50 | seanbright | this log is the other way around, which does not match your dialplan on desired functionality |
| 18:22.53 | Samot | There should be an incoming message from 4001 and an outgoing message to 4002 |
| 18:23.26 | _abc_ | Ok. I can't find the 1st msg probably wrongly turned on debugging late. |
| 18:23.32 | _abc_ | Let's focus on ^what? |
| 18:23.39 | _abc_ | which is the same thing but from phone 2 to 1 |
| 18:23.43 | _abc_ | phone 2 is 4002 |
| 18:23.53 | _abc_ | If you find it in the log, it's there, from linphone |
| 18:24.08 | seanbright | i need some logs |
| 18:24.26 | _abc_ | Ok, wait, the cat / pastebin truncated my paste on garbage. Just a second. |
| 18:24.38 | seanbright | so you're saying that asterisk is receiving the message from linphone but not sending anything out in response to that? |
| 18:25.03 | _abc_ | Something like that, or, it is sending it but it is confusing the client somehow. Give me a second. |
| 18:25.15 | seanbright | ok |
| 18:25.15 | igcewieling | seanbright: I think he wants one phone to send a message via asterisk to another phone. |
| 18:25.21 | seanbright | i get that |
| 18:26.26 | seanbright | Content-Type: message/imdn+xml |
| 18:26.50 | seanbright | Content-Encoding: deflate |
| 18:27.00 | seanbright | so the linphone message is gzipped? |
| 18:27.05 | _abc_ | https://termbin.com/3jwgp here, focus on ^what\? |
| 18:27.14 | seanbright | i can say with 100% certainty that asterisk would not know what to do with that |
| 18:27.20 | seanbright | at least not chan_sip |
| 18:27.26 | _abc_ | seanbright: the idiotic "x is typing now" is deflated with gzip since it is xml |
| 18:27.48 | _abc_ | So asterisk ignored the deflated message based on unsupported MIME as you can see |
| 18:28.06 | _abc_ | I censed the garbage a bit to avoid confusing the pastebin again it says garbage removed there |
| 18:28.25 | _abc_ | So is there anything wrong with the headers asterisk puts there |
| 18:28.27 | _abc_ | ?! |
| 18:28.49 | _abc_ | The linphones are the SAME version 4.0.1 on both ends, on purpose. One phone is a kitkat 4.4 and the other an oreo 8.1 |
| 18:28.58 | seanbright | ok, i think i see now |
| 18:29.11 | _abc_ | This is the 3rd day (not in sequence) I spend having fun with this. |
| 18:29.14 | seanbright | i think there is an issue related to this |
| 18:29.16 | _abc_ | seanbright: ok, shoot? |
| 18:30.03 | seanbright | https://issues.asterisk.org/jira/browse/ASTERISK-28513 |
| 18:30.12 | seanbright | lol... "I have run into a problem with MESSAGE s and the Linphone Android client." |
| 18:30.30 | seanbright | ah, damn. this is pjsip. |
| 18:30.32 | _abc_ | Err that loads super slow. |
| 18:30.39 | _abc_ | As in stalled. Ok loaded. |
| 18:30.49 | _abc_ | Also that is 13.28 |
| 18:31.03 | seanbright | oh, sorry, what are you running? |
| 18:31.18 | _abc_ | 13.14 which is the default distributed with Stretch. |
| 18:31.22 | seanbright | gotcha. |
| 18:31.25 | seanbright | shouldn't matter. |
| 18:31.30 | _abc_ | I'am actually on devuan ascii but that is Stretch without systemd |
| 18:31.54 | _abc_ | So do you see any problems with the setup? I do not see any. |
| 18:32.22 | _abc_ | The From issue is a problem I do not know how to solve. Does asterisk save the really-from in the dialplan somehow? |
| 18:32.39 | _abc_ | At least the IP or the full SIP header from the origin? |
| 18:33.34 | seanbright | i mean, the problem is that you are trying to extract information that is not in the MESSAGE |
| 18:33.47 | seanbright | 4001 is not in the original message anywhere |
| 18:34.05 | _abc_ | Indeed. But 4002 is sending to 4001 and vice versa |
| 18:34.15 | _abc_ | 4002 is the phone sending "what?" |
| 18:34.27 | seanbright | when is it ${EXTEN} you're after? |
| 18:34.41 | _abc_ | Should be 4002 in that case |
| 18:34.47 | seanbright | then you're SOL |
| 18:35.34 | seanbright | just so that i am clear |
| 18:35.40 | seanbright | https://termbin.com/3jwgp |
| 18:35.45 | seanbright | who "sent" that first message? |
| 18:35.48 | seanbright | 4001 or 4002? |
| 18:36.07 | _abc_ | 4002 sent 'what?' to 4001 |
| 18:36.22 | seanbright | right, and 4001 does not show up in that SIP message |
| 18:36.26 | seanbright | so how could you possibly route it? |
| 18:36.46 | _abc_ | So what is actually weird? The linphone setup?! |
| 18:36.55 | _abc_ | Give me a second I'll check everything and try again |
| 18:37.02 | seanbright | i mean, i dunno |
| 18:37.08 | seanbright | chan_sip is old as dirt |
| 18:37.37 | seanbright | there's a quasi-related linphone android issue: https://github.com/BelledonneCommunications/linphone-android/issues/605 |
| 18:39.01 | seanbright | asterisk is a B2BUA so it is the 'target' when sending MESSAGEs |
| 18:39.25 | seanbright | i'm just looking at it from a dialplan logic standpoint |
| 18:40.05 | seanbright | asterisk only has the information in the MESSAGE message to go on. if your desired recipient doesn't show up in said MESSAGE, how could asterisk ever figure out where to send the message? |
| 18:40.06 | drmessano | It's always Linphone |
| 18:40.45 | _abc_ | https://termbin.com/jwpd here's another set. Focus on messages l337-1 l337-2 |
| 18:40.53 | seanbright | focusing |
| 18:41.48 | seanbright | ok, so is the call to MessageSend correct |
| 18:41.54 | seanbright | let me look at the docs |
| 18:42.06 | _abc_ | Looks like it is correct but the From part needs discussion imo |
| 18:42.31 | _abc_ | ${EXTEN} is always set to the target exten, right? What's the origin set to? CALLERID? |
| 18:42.40 | _abc_ | Assuming it is not spoofed |
| 18:43.48 | *** join/#asterisk ih8wndz (jwpierce3@mail.000.srv.trnkmstr.com) |
| 18:45.57 | _abc_ | I am going to force an origin message in the dialplan, "Hello World"? Should that work? Static text and all? |
| 18:45.58 | seanbright | looks like from and to cak both have <>s around them |
| 18:46.00 | *** join/#asterisk zapata (~zapata@2a02:1748:fad4:7260:85a8:7ddc:ec:3f36) |
| 18:46.12 | _abc_ | cak? |
| 18:46.20 | seanbright | can |
| 18:46.32 | _abc_ | I thought the <>s are legal in headers as long as they are not IN the uri |
| 18:47.00 | seanbright | in your log |
| 18:47.06 | seanbright | -- Executing [4001@astsms:6] MessageSend("Message/ast_msg_queue", "sip:4001,<sip:4002@192.168.0.110>") in new stack |
| 18:47.17 | seanbright | and looking at chan_sip.c |
| 18:47.28 | seanbright | those arguments appear to be correct |
| 18:47.41 | _abc_ | Wow. I edited that way long ago on another server I built for someone. Do not remember fondly. |
| 18:47.55 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 18:48.11 | _abc_ | Looking at the SIP dialog as logged it does go out to the peer, right? Into the asterisk and out to the peer |
| 18:48.25 | _abc_ | Both times code 200 |
| 18:48.35 | _abc_ | Sorry 202 |
| 18:49.48 | *** join/#asterisk miralin (~Thunderbi@178.34.160.50) |
| 18:51.01 | seanbright | the 1337-2 message is not being "relayed" |
| 18:51.25 | seanbright | and i am not seeing the incoming message for 1337-3 in your trace |
| 18:51.38 | _abc_ | I set debug peer 4002 then debug peer 4001; apparently only one is debugged at one time? |
| 18:51.52 | _abc_ | Can one set debug a list of peers? |
| 18:51.56 | seanbright | sip set debug on |
| 18:52.11 | _abc_ | ok a minute for another test |
| 18:52.17 | seanbright | can't wait |
| 18:52.28 | _abc_ | ? |
| 18:52.47 | seanbright | OH BOY I CAN'T WAIT |
| 18:52.50 | seanbright | (better?) |
| 18:54.03 | sibiria | :D |
| 18:54.39 | _abc_ | https://termbin.com/q12y focus on nbleh |
| 18:55.23 | seanbright | focusing |
| 18:55.43 | _abc_ | I think the message format is not understood somehow by linphone, after a transformation (?!) by asterisk. I have no other good explanation. Perhaps missing CR LF at end? I have no idea. |
| 18:56.28 | sibiria | linphone understands SIP messaging. source: me, having played around with SIP messaging using linphone |
| 18:56.57 | seanbright | ok, all of that looks fine to me |
| 18:57.19 | _abc_ | I do not see an error anywhere either. |
| 18:57.19 | seanbright | i mean, the request URI is a fucking disaster but i assume that is how linphone registers |
| 18:58.15 | _abc_ | I really have no idea what is going on. |
| 18:58.23 | sibiria | that's a really nice contact field payload |
| 18:58.25 | _abc_ | Maybe use tcpdump to compare frames byte by byte |
| 18:58.35 | seanbright | does linphone have debugging facilities? |
| 18:58.51 | _abc_ | Nothing that I can see. I think there's a debug build edition to work with. |
| 18:59.12 | _abc_ | There's also linphone for desktop. I'll try that in a few minutes then I go do chores. |
| 18:59.47 | _abc_ | There's a debug in the apk app. I turned it on. No idea what it does. |
| 18:59.58 | seanbright | god speed |
| 19:05.43 | _abc_ | What is AVPF? |
| 19:07.04 | seanbright | https://tools.ietf.org/html/rfc4585 |
| 19:13.43 | _abc_ | I see no trace of a file log anywhere. |
| 19:14.38 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:11f6:aca2:9efe:7efc) |
| 19:15.13 | _abc_ | Ok, it appears the end point clients are causing this, I do not know how yet. I assume asterisk ALWAYS puts in a terminating CR LF after the content in a packet? Or not? |
| 19:16.43 | seanbright | dunno |
| 19:25.14 | _abc_ | Ok so this is consistent between all android and linux desktop linphones, the receiving end shows an "empty" message as-if the other end is still typing, but not the message proper. |
| 19:25.19 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:11f6:aca2:9efe:7efc) |
| 19:25.57 | _abc_ | Apparently passing that xml is important. I will try directmedia/reinvite, perhaps the silly things can talk to each other directly, then I could tcpdump sniff the traffic and see what the dickens they actually do. But not now. |
| 19:26.01 | _abc_ | Thanks for the patience. |
| 19:30.14 | seanbright | directmedia will have no affect |
| 19:30.24 | seanbright | because a MESSAGE is not "media" |
| 20:17.09 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:108d:9a6e:171:b959:c192:579d) |
| 20:27.34 | _abc_ | re. Is there a simple way to hack asterisk source to pass various "unknown" mime media types as-is? Just copy out? |
| 20:29.06 | _abc_ | Or a catch-all wildcard media copy mode/politcy setting? |
| 20:29.11 | _abc_ | -it |
| 20:29.14 | _abc_ | -t |
| 20:31.54 | _abc_ | Apparently AGI scripting can send text on the cli of asterisk https://www.voip-info.org/asterisk-cmd-sendtext/ |
| 20:47.09 | sibiria | on what channel? |
| 20:47.22 | sibiria | AGI acts on the channel it's called in |
| 21:05.58 | *** join/#asterisk ShellyRoll (~ShellyRol@c-67-170-30-252.hsd1.wa.comcast.net) |
| 21:13.10 | *** join/#asterisk hfb (~hfb@47.139.18.84) |
| 21:13.39 | _abc_ | Ah |
| 21:24.55 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 21:47.44 | *** join/#asterisk miralin (~Thunderbi@178.34.162.113) |
| 21:49.29 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:07.56 | igcewieling | idly ponders how a book which won't be released until Sept 2020 can be on the audible.com best sellers list. |
| 22:45.40 | *** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy) |
| 23:06.47 | *** join/#asterisk ih8wndz (jwpierce3@mail.000.srv.trnkmstr.com) |
| 23:16.57 | seanbright | so let's so that for each outbound call i wanted to rotate through a list of X preconfigured caller IDs |
| 23:17.12 | seanbright | what would be the easiest way to do that with asterisk? |
| 23:18.19 | igcewieling | I'd set up a "fake" array containing the data, then iterate over the data. Use a global index variable to keep track. |
| 23:25.17 | seanbright | like in dialplan? |
| 23:26.33 | *** join/#asterisk tsal (~tsal@i59F52F59.versanet.de) |
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| 23:52.58 | seanbright | and keeping that global index atomically incremented... i guess you would need to lock in dialplan |
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