| 00:19.03 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:19.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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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| 10:25.12 | VTheThinker | I am trying to receive an SMS (sent from my phone) on an Asterisk account; nothing is detected in the web app where I'm connected with the credentials (using SIPml5 JS library) but I can see the message in the Asterisk device web interface (SMS Inbox) |
| 10:25.46 | VTheThinker | can it be an Asterisk settings issue? I can give you more details from Asterisk Logs |
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| 12:15.44 | Samot | What SMS inbox? |
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| 12:51.30 | VTheThinker | Samot, it's just a web interface from the device that I use (GSM Gateway from Openvox); not related to Asterisk |
| 12:52.49 | Samot | So what is the issue? |
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| 13:19.35 | mysth | Dont know if someone is here who helped me on incrediblepbx and Asterisk configuration yesterday(https://pastebin.ca/3840666) but by VOIP provider dont see any kind of register attemp being made. I also tried different router, did not work. |
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| 13:22.34 | Samot | mysth: Show your trunk/peer settings for your provider |
| 13:22.47 | Samot | mysth: Also the output of 'sip show settings' |
| 13:22.49 | Samot | ~pb |
| 13:22.49 | infobot | rumour has it, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:22.54 | Samot | ^^^ pastebin them both. |
| 13:25.30 | file | the "__sip_xmit returns error for pkt data" message is output if the operating system returns a failure when sending the UDP packet |
| 13:25.52 | file | bad file descriptor, host unreachable, interface down, network failure, or ICMP port unreachable |
| 13:25.52 | Samot | Yeah, it could be a bad SIP setting. |
| 13:25.57 | Samot | So I want to check it out.. |
| 13:26.18 | Samot | Responds to pings.. |
| 13:26.34 | Samot | So connectivity is there.. |
| 13:26.49 | Samot | mysth: That stuff coming? |
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| 13:27.06 | [TK]D-Fender | Or firewall on the server is preventing the packet from hitting the stack |
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| 13:28.15 | Samot | mysth: Just a heads up, it is Monday. That means we don't have 45 minutes to just kill & chill while we wait for your replies. |
| 13:28.34 | Samot | Had that problem yesterday. |
| 13:29.18 | mysth | Ok, dont need that attitude now. I`m getting them but i also have to look after my children. |
| 13:29.33 | Samot | I'm not giving you attitude. |
| 13:29.36 | Samot | I'm giving you facts. |
| 13:29.46 | mysth | Yes, im too. |
| 13:36.36 | mysth | Allright, https://pastebin.ca/3840941 |
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| 13:37.14 | [TK]D-Fender | mysth, "iptables --list" |
| 13:39.27 | mysth | https://pastebin.ca/3840946 |
| 13:40.28 | [TK]D-Fender | I would trash that first after setting "Chain INPUT (policy DROP)" to Accept first |
| 13:41.02 | [TK]D-Fender | and you were asked to ping, traceroute, etc your provider host from CLI to which you hadn't responded |
| 13:41.24 | [TK]D-Fender | Do that first, and if that happens to fail adjust the firewall and retest again |
| 13:41.26 | [TK]D-Fender | show ALL attempts |
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| 13:42.09 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 13:44.19 | mysth | Do you mean ping when i entered "asterisk -r" command? |
| 13:44.50 | Samot | First. |
| 13:44.56 | Samot | Please show your trunk settings. |
| 13:45.03 | Samot | As requested before. |
| 13:45.30 | mysth | Is there a command so i can copypaste them from CLI? |
| 13:45.51 | Samot | Straight from sip.conf |
| 13:45.57 | Samot | Well this is a GUI |
| 13:46.04 | Samot | So go into the GUI and in the trunk settings... |
| 13:46.40 | Samot | Are you using Westlink? |
| 13:46.45 | Samot | In Finland? |
| 13:46.50 | mysth | Yes. |
| 13:46.52 | Samot | K |
| 13:47.08 | Samot | They don't have a weppi.fi FQDN for sip as far as I can tell.. |
| 13:48.28 | Samot | Of course, I'm reading translated pages... |
| 13:48.39 | Samot | But this sounds like a peer configuration issue now. |
| 13:50.11 | mysth | Ok, im copy&pasteing Trunk settings, ready soon |
| 13:54.19 | mysth | https://pastebin.ca/3840949 there, dialed number manipulation is untouched/empty too |
| 13:55.34 | Samot | Uhm. |
| 13:55.40 | Samot | You have zero settings. |
| 13:56.46 | mysth | You mean i need to add settings to incoming settings field? I dont need to call out from VOIP |
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| 13:58.03 | Samot | Well only if you want to accept incoming calls. |
| 13:58.10 | Samot | And be able to route them. |
| 13:58.40 | Samot | Let me just clarify what you said earlier.... |
| 13:59.04 | Samot | You did talk to Westlink but they say they are not seeing any REGISTER attempts from you, correct? |
| 13:59.10 | mysth | Yes |
| 13:59.18 | Samot | Do they have an Asterisk guide? |
| 13:59.33 | Samot | Or any thing to show what the SIP trunk settings should be? |
| 14:00.55 | mysth | I have settings, passwd etc, i just edited them out from pastebin. And i know this is not same thing as in raspberry 1, but there these work. |
| 14:01.58 | mysth | I have authentication ID, passwd, User id, proxy/domain and outbound proxy. They are added in the register string. |
| 14:02.49 | Samot | mysth: RasPBX1 or RasPBX3 doesn't matter. |
| 14:02.59 | [TK]D-Fender | Reliably Transmitting (no NAT) to 109.69.124.130:5060: |
| 14:02.59 | [TK]D-Fender | REGISTER sip:weppi.fi SIP/2.0 |
| 14:03.05 | [TK]D-Fender | We see what it resolves to |
| 14:03.16 | [TK]D-Fender | <mysth> Do you mean ping when i entered "asterisk -r" command? <- OS CLI |
| 14:03.21 | [TK]D-Fender | ping is not an Asterisk thing |
| 14:03.21 | Samot | This is Asterisk and the SIP settings haven't really changed. |
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| 14:03.43 | Samot | So if it works on the other PBX and not this one and the settings don't match... |
| 14:03.47 | [TK]D-Fender | mysth, Where are the things I asked you to do? |
| 14:03.51 | Samot | And they are for the same account.... |
| 14:05.01 | mysth | The raspberry1 is using Freepbx, but yes, it uses Asterisk |
| 14:05.20 | Samot | The trunk settings would be the same. |
| 14:05.28 | mysth | Yes |
| 14:05.35 | Samot | You're saying they are not... |
| 14:06.09 | Samot | You're saying the settings you have now, that don't work, do not match the settings of the other system, that does work... |
| 14:06.47 | Samot | Does Westlink have example Asterisk and/or FreePBX configurations? |
| 14:07.02 | Samot | Do they have something showing what the register string should look like? |
| 14:07.12 | mysth | I dont know about that, i can ask them tomorrow. |
| 14:07.21 | Samot | To compare to? Because I believe your trunk/register string is completely misconfigured. |
| 14:07.34 | Samot | Can you look in your Ras1 box? |
| 14:07.50 | mysth | It is same, i copypasted it from there. |
| 14:08.31 | [TK]D-Fender | <Samot> To compare to? Because I believe your trunk/register string is completely misconfigured. <- no |
| 14:08.37 | [TK]D-Fender | <[TK]D-Fender> Reliably Transmitting (no NAT) to 109.69.124.130:5060: |
| 14:08.41 | [TK]D-Fender | mysth, well? |
| 14:09.13 | mysth | [TK]D-Fender: sorry, what command you want me to do? |
| 14:09.19 | [TK]D-Fender | ... |
| 14:09.26 | [TK]D-Fender | PING & TRACEROUTE IT FROM OS CLI |
| 14:10.19 | [TK]D-Fender | <[TK]D-Fender> I would trash that first after setting "Chain INPUT (policy DROP)" to Accept first |
| 14:10.19 | [TK]D-Fender | <[TK]D-Fender> and you were asked to ping, traceroute, etc your provider host from CLI to which you hadn't responded |
| 14:10.19 | [TK]D-Fender | <[TK]D-Fender> Do that first, and if that happens to fail adjust the firewall and retest again |
| 14:10.19 | [TK]D-Fender | <[TK]D-Fender> show ALL attempts |
| 14:10.29 | [TK]D-Fender | Half an hour...... |
| 14:10.46 | mysth | Yes sorry, i was copying Trunk settings, doing it now |
| 14:12.24 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:12.51 | [TK]D-Fender | trunk settings means nothing |
| 14:13.00 | [TK]D-Fender | Your packets are not being allowed on the stack |
| 14:13.08 | [TK]D-Fender | [2017-07-09 20:08:31] ERROR[19421]: chan_sip.c:4194 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 14:13.10 | mysth | https://pastebin.ca/3840957 |
| 14:13.40 | [TK]D-Fender | Now fix the firewall as I told you |
| 14:14.16 | [TK]D-Fender | retest those, and then restart the * core. "core restart now" |
| 14:14.20 | [TK]D-Fender | from * CLI |
| 14:14.31 | [TK]D-Fender | after proving the FW state first |
| 14:15.12 | mysth | Ok, ill try that |
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| 14:27.37 | mysth | https://pastebin.ca/3840965 here it is, dont know if i configured FW correctly.. |
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| 14:30.21 | [TK]D-Fender | I was very explicit when I said to TRASH those rules. |
| 14:30.47 | [TK]D-Fender | Set input/output & forward to "accept" and TRASH everything else |
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| 14:39.13 | mysth | https://pastebin.ca/3840976 |
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| 14:43.02 | [TK]D-Fender | <PROTECTED> |
| 14:43.23 | [TK]D-Fender | ok, I'm off of this... |
| 14:43.36 | [TK]D-Fender | can't focus the time for it. |
| 14:44.32 | mysth | Any idea? |
| 14:46.32 | Samot | Call your ISP and verify they aren't blocking any SIP ports/traffic. |
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| 14:49.44 | mysth | Ok, Thank you all for helping. |
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| 15:10.48 | drmessano | Ditch the ASUS |
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| 16:08.14 | forgotmynick | asterisk keeps randomly crashing (process disappears) but nothing is showing in the log files even with full verbosity. i have been connected to the asterisk console for a few days but it hasn't crashed yet but i'm confident it will. there's plenty of ram, plenty of disk space. can someone point me in the direction on how to debug this? |
| 16:15.45 | salviadud | forgotmynick, what distro are u using, what version of asterisk. |
| 16:18.02 | forgotmynick | salviadud: asterisk 14.5 on ubuntu 16.04 |
| 16:27.47 | salviadud | That's bleeding edge don't ya think? |
| 16:27.52 | salviadud | Is that a production box? |
| 16:36.51 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
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| 16:39.43 | Samot | I get that Asterisk 14.5 is like 6 weeks old but I wouldn't classify a LTS OS that has been released for 18 months "bleeding edge" |
| 16:40.46 | salviadud | Samot, that's just my opinion, I wouldn't go further than version 13 for production. |
| 16:40.55 | Samot | Now, I don't use PJSIP. That being said, I haven't had one issue with 14.5 |
| 16:41.04 | Samot | 14 is *stable* |
| 16:41.35 | Samot | Perhaps using 14.6.x-rc2 or whatever can count as bleeding edge.. |
| 16:41.46 | Samot | But stable releases are not bleeding edge. |
| 16:41.50 | Samot | Because the edge doesn't bleed. |
| 16:41.53 | Samot | It's stable. |
| 16:42.47 | salviadud | Hehe, very well then. |
| 16:42.55 | Samot | Like when cPanel asks me "How do you want to handle updates" |
| 16:43.02 | Samot | There is "Stable Updates" |
| 16:43.08 | salviadud | forgotmynick, what else can you tell us? |
| 16:43.09 | Samot | And there is a "Bleeding Edge" option. |
| 16:43.24 | forgotmynick | salviadud: i like long walks along beaches |
| 16:44.08 | Samot | forgotmynick: Is this a new install or an upgrade? |
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| 16:51.38 | *** join/#asterisk matt_ (~matt@ccpc-buzzer.bath.ac.uk) |
| 16:53.02 | matt_ | hi, i'm really have a hard time tring to figure out how to get asterisk to register to a server using tls on port 5061 with pjsip, does anybody know what I might be missing? |
| 16:53.02 | matt_ | tcpdump it showing tcp packets going to 5060 |
| 16:54.10 | VTheThinker | Samot would Asterisk Logs and SIP Logs be relevant in identifying my problem? the one with the SMS that I was mentioning earlier |
| 16:54.22 | *** join/#asterisk samwierema (~samwierem@82.169.225.211) |
| 16:54.34 | VTheThinker | it's also when trying to send them... |
| 16:56.26 | forgotmynick | Samot: it was a new install but i've found crash dumps which i'm going through now |
| 16:57.45 | *** join/#asterisk jkroon (~jkroon@165.16.204.167) |
| 16:58.40 | file | because of the amount of testing we've put into things 14 has been a rather quiet release series |
| 16:58.52 | file | also due to the policy of being able to put new things into 13 too |
| 17:03.53 | igcewieling | file: Is Asterisk 15 still expected to be released in Oct? |
| 17:03.58 | file | yes. |
| 17:04.05 | igcewieling | thanks |
| 17:04.36 | file | is busy getting stuff done for feature freeze |
| 17:04.41 | file | could use a vacation |
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| 17:11.32 | VTheThinker | trying to send SIP message from web app, using SIPml5 with Asterisk credentials: 1) SIP Logs: SIP/2.0 405 Method Not Allowed, CSeq: <xxxxx> MESSAGE 2) Asterisk Logs: Received message to <sip:<recipient-number>@<domain>> from "<Asterisk-number>"<sip:<Asterisk-account-name>@<domain>>;tag=W3Kam6DtzkgdPbHcDbWP, dropped it... Content-Type:text/plain;charset=utf-8 Message: sms test |
| 17:12.59 | VTheThinker | any clue? |
| 17:14.50 | file | that message only appears in 1.8 and earlier, which do not support out of call messaging |
| 17:15.05 | Samot | VTheThinker: What version are you running? |
| 17:16.01 | igcewieling | I stopped reading at "web app" 8-| |
| 17:17.06 | salviadud | Speaking of versions, should I just buy the 4th edition of the book, I don't think any major changes to the dialplan code or config files are coming in any time soon. |
| 17:17.31 | VTheThinker | Samot it's 2.3.2 |
| 17:18.11 | VTheThinker | and we cannot upgrade Asterisk on that device, it;'s readonly |
| 17:18.52 | Samot | 2.3.2? |
| 17:18.52 | Samot | What version is that? |
| 17:18.58 | Samot | There is no Asterisk 2.3.2 |
| 17:19.10 | Samot | Or are you referring the to"GUI" for it? |
| 17:21.29 | salviadud | Is there a 5th edition in the works? I must know |
| 17:21.53 | file | I'm not aware of anyone working on such a thing. |
| 17:22.23 | VTheThinker | Samot sorry, that ws the version of another software, the OS... The Asterisk version is 1.8.20.0 - I know it's old |
| 17:22.25 | [TK]D-Fender | It's been a LONG time since I've seen Leif in here... |
| 17:22.48 | [TK]D-Fender | VTheThinker, Anything older than 13 = DEAD, and that version doesn't support SIP messaging |
| 17:22.50 | file | VTheThinker: that version does not support out of call messaging, the code is literally not in it so no matter what you do it can't be used for that |
| 17:22.51 | [TK]D-Fender | VTheThinker, You're DOA |
| 17:24.22 | VTheThinker | oh, I see |
| 17:25.36 | Samot | Yeah, if you want this to receive SMS/text you need to upgrade. |
| 17:25.56 | VTheThinker | thanks |
| 17:45.15 | igcewieling | Assuming sms_raw="\"SM\",6" char store[32] and int index. I'm trying to parse out the data using sscanf. Here is what I'm using now, but the index is always 0. sscanf(sms_raw, "\"%[^\"]\",%d", store, index); Am I doing anything obviously wrong? I'm not very good with C. |
| 17:45.47 | igcewieling | *sigh* wrong channel |
| 17:56.11 | *** join/#asterisk infinity_ (~brendon@web2.artsopolis.com) |
| 17:56.18 | infinity_ | is there a recommended ATA for astiersk? i need 1 port for a fax machine. (and yea, i know fax isn't great for this) |
| 17:57.36 | infinity_ | oh. not a pci card. an ethernet one :) |
| 17:57.54 | igcewieling | I can't recommend that. |
| 17:58.08 | infinity_ | what? |
| 17:58.49 | igcewieling | infinity_: running a fax on an ATA. |
| 17:58.49 | infinity_ | Oh, yea. I know. I didn't ask that question though. I just offered what I was going to use it for |
| 17:59.06 | igcewieling | Where I work, we install analog lines for fax. If we already have an Adtran on-site we might use T.38, but still no ATA. |
| 18:00.14 | igcewieling | *sigh* I really hate C. 8-( |
| 18:12.12 | drmessano | infinity_: SPA112 |
| 18:12.19 | drmessano | Just run T.38 end to end |
| 18:12.36 | drmessano | Should work +/- 80% |
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| 18:18.40 | rmudgett | igcewieling: You need to pass the address of index: sscanf(sms_raw, "\"%[^\"]\",%d", store, &index) |
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| 18:23.40 | igcewieling | rmudgett: I figured that out a few min ago, but I appreciate the answer anyway. |
| 18:28.47 | infinity_ | drmessano: thanks. |
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| 20:32.11 | bluez_ | hi guys |
| 20:35.12 | bluez_ | i'm using x-lite directly to a sip provider and everything works fine. I try with x-lite -> asterisk -> same provider, and voice... doesnt work x-lite -> far end (dialed number on provider) but works far end -> xlite |
| 20:35.12 | bluez_ | asterisk is on a public ip |
| 20:35.12 | bluez_ | rtp debug shows packets coming from x-lite and it says they're being sent to the provider |
| 20:35.13 | bluez_ | but then they suddenly stop after the call connects it seems |
| 20:38.52 | bluez_ | https://pastebin.com/raw/BUUYvcMk |
| 20:39.05 | bluez_ | so the client is behind nat and <client external ip> is my external public ip |
| 20:39.25 | bluez_ | it seems after this : > 0x7f717800cc50 -- Probation passed - setting RTP source address to <client external ip>:53032 |
| 20:39.36 | bluez_ | it stops receiving voice from the client |
| 20:41.19 | igcewieling | disable direct media in asterisk |
| 20:41.20 | igcewieling | check your localnet settings and your nat= settings |
| 20:41.20 | bluez_ | is it on by default |
| 20:42.01 | bluez_ | https://www.irccloud.com/pastebin/4iNxUNfs/ |
| 20:42.16 | bluez_ | hmm i think this is likely the problem |
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| 21:16.22 | bluez_ | directmedia=outgoing did the trick |
| 21:16.32 | bluez_ | turning it off worked but outgoing is better bandwidth wise |
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| 22:10.43 | Dovid | `/j ##php |
| 22:12.12 | lorsungcu | yes |
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| 22:18.18 | drmessano | MORE #### |
| 22:18.57 | Dovid | lol |
| 22:18.57 | drmessano | '/j ##############php |
| 22:18.57 | drmessano | NEXT LEVEL PHP CHANNEL |
| 22:18.57 | Dovid | has anyone found an issue where phpagi does not die when the user hangs up? |
| 22:19.51 | drmessano | Abacus > PHP > Artificial Intelligence > NODE.JS <-- The evolution of Man |
| 22:20.42 | [TK]D-Fender | <Dovid> has anyone found an issue where phpagi does not die when the user hangs up? <- don't talk like that is everything |
| 22:20.49 | [TK]D-Fender | It has INTERRUPT handlers |
| 22:20.55 | [TK]D-Fender | so does PHP in general |
| 22:20.58 | [TK]D-Fender | Get reading |
| 22:21.11 | [TK]D-Fender | just because the caller quits doesn't mean it's safe to quit the AGI |
| 22:21.16 | [TK]D-Fender | it may still have CLEANUPS to do |
| 22:21.46 | Dovid | TK: I understand. I did try with http://php.net/manual/en/function.pcntl-signal.php but it seems that it never gets the intterupt handler |
| 22:34.53 | bluez_ | does anyone know of a really simple credit management addon for asterisk? |
| 22:35.14 | Samot | What type of credit management? |
| 22:35.34 | bluez_ | i just want to 'add credit' and all calls will cost the same per minute. while calling, decrement credit and cut it off when it runs out |
| 22:36.02 | bluez_ | i could write one but if something exists it'll save me time |
| 22:36.12 | Samot | There are numerous third party solutions for this. |
| 22:36.19 | drmessano | A2Billing turns Asterisk into a money printing machine |
| 22:36.24 | Samot | ^^ |
| 22:36.26 | Dovid | lol |
| 22:36.26 | Samot | Pretty much |
| 22:36.27 | bluez_ | lol |
| 22:36.33 | Samot | Makes you an instant provider. |
| 22:36.33 | bluez_ | yeah but alot of them are overkill |
| 22:36.51 | Samot | Then I guess you are writing your own. |
| 22:36.52 | drmessano | It gives you alllll of the LECs |
| 22:36.58 | lorsungcu | nods |
| 22:37.00 | drmessano | ILEC, CLEC, NEXTLEVELLEC |
| 22:37.12 | lorsungcu | KLEC too |
| 22:37.17 | lorsungcu | which is a russian variant |
| 22:37.23 | drmessano | You can even be a LLLLLLLLLLLL-COMBO-BREAKER-LEC |
| 22:37.42 | drmessano | Its easy |
| 22:37.46 | drmessano | 1. Install Asterisk |
| 22:37.53 | drmessano | 2. Install A2Billing |
| 22:38.16 | drmessano | 3. ?????? |
| 22:38.16 | drmessano | 4. GET MONEY |
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| 22:41.07 | Dovid | if it was only that easy |
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