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| 06:03.56 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.1 (2019/10/16) 16.6.1 (2019/10/16) Standard: 17.0.0 (2019/10/28); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 06:12.58 | *** join/#asterisk rue_mohr (~rue_mohr@d50-92-152-244.bchsia.telus.net) |
| 06:13.10 | rue_mohr | console, how do I list sip users? |
| 06:14.29 | rue_mohr | should level 5 verbose shop a sip user trying to log in and failing cause the account its trying to use does't exist? |
| 06:15.25 | rue_mohr | is sip called pjsip? |
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| 07:08.01 | [TK]D-Fender | sip is sip |
| 07:08.08 | [TK]D-Fender | pjsip is also sip |
| 07:08.11 | [TK]D-Fender | but also pjsip |
| 07:08.29 | [TK]D-Fender | If you don't know which you're using, you've got bigger problems... |
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| 08:58.56 | wdoekes | surely everyone uses pjsip by now |
| 08:59.00 | wdoekes | runs away |
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| 09:29.12 | HenrikJott | Hi! I´m running asterisk on a telemarketing company (one of my customers) and recently our ISP wanted us to use a new trunk which requires to register. We reconfigured and it worked fine, but suddenly a lot of harmful request was sent to our asterisk from different IP´s (not from the new trunk). When i delete the peer and the register string, the attack stops immediatly. Is it possible |
| 09:29.12 | HenrikJott | for attackers to "relay" via the new trunk? In my case we did not register to the previous trunk and we have never had a problem there. Our firewall is pretty strict and shouldn´t allow the requests i was getting. Can anybody point me in the right direction? |
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| 12:07.35 | Someone_Else | is it possible to wait for a specific event (a user registers) before dialing that user? |
| 12:10.07 | Samot | Not easily, if at all. |
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| 17:14.41 | monsterco | I have a Mitel 6867i connected to astersik. From time to time user tells me they can't dial out and we have to factory reset it to work. What feature of this phone you think might be responsible? The dialing doesn't get out of the set so it's not an Asterisk issue at all but a phone set one |
| 17:16.53 | [TK]D-Fender | We have no idea. You haven't shown us a failure to look at |
| 17:17.04 | [TK]D-Fender | "My car doesn't work. Why?" |
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| 17:27.10 | TandyUK | "Because its wednesday" of course lol |
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| 17:33.27 | monsterco | when she dials screen shows Call Failed |
| 17:33.41 | monsterco | and nothing is reached by Asterisk so the stupid set has something wrong |
| 17:33.48 | monsterco | but it can dial using a Speed Line |
| 17:33.55 | igcewieling | is NAT involved? |
| 17:34.08 | monsterco | yes |
| 17:34.17 | monsterco | hmmm |
| 17:34.27 | igcewieling | sounds like the router is closing port 5060 when it is idle for too long. |
| 17:34.30 | monsterco | you think phone is not registered when it tries to dial? |
| 17:34.42 | monsterco | I see |
| 17:35.00 | igcewieling | monsterco: phones don't have to be registered to make calls. They only have to be registered when receiving a call. |
| 17:35.21 | igcewieling | To make a call, the phone just needs the correct auth info. |
| 17:35.43 | monsterco | It can dial using Speed Dial when I input a number. I will have to test this one more time though. |
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| 17:36.00 | igcewieling | try setting qualify=10000 and qualifyfreq=20 assuming you are using chan_sip. |
| 17:36.27 | igcewieling | that will ensure activity on port 5060 and prevent the NAT router from closing the connection. |
| 17:43.09 | Samot | This has nothing to do with NAT. |
| 17:43.33 | Samot | If the phone doesn't send digits when they are pressed on the key pad but can send digits when programmed to be sent at once... |
| 17:43.37 | Samot | It's a phone issue. |
| 17:44.02 | Samot | They can't dial 1234 but they can hit a button that present 1234 at once. |
| 17:52.50 | monsterco | they can dial internal extensions but 10 digits doesn't dial apparently |
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| 18:05.22 | [TK]D-Fender | local phone dialplan <- |
| 18:08.38 | monsterco | never knew mitel (old aastra) have a dialplan |
| 18:08.51 | monsterco | or at least have a prob with 10 digit dialing after factory reset |
| 18:11.46 | [TK]D-Fender | If you are ever assuming a "default" then that's a failure as a system admin. |
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| 18:44.20 | monsterco | generally, how do you setup Mitel phones? simply putting the registery info in Global SIP or Line 1, Line 2...? |
| 18:44.41 | monsterco | I have setup in global setting with old Aastra phones and they worked fine - never bothered with each line |
| 18:52.38 | monsterco | what is a local dialplan for mitel that allows calling north america? |
| 18:56.27 | Samot | You're going to have to look at Mitel's documents. |
| 18:57.09 | Samot | Because each brand has their own pattern matching and timeout schemes. |
| 19:10.05 | monsterco | this is the default and supposed to cover all north am: X+#|XX+* |
| 19:10.15 | monsterco | https://www.manualslib.com/manual/1040000/Mitel-6867i.html?page=332#manual |
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| 21:01.21 | bkervaski | Hey guys .. finally migrating all of our stuff to PJSIP .. in learning mode right now .. I'm having an issue with PJSIP_DIAL_CONTACTS on asterisk 13.latest .. If an endpoint is unavailable (i.e., no contacts registered), PJSIP_DIAL_CONTACTS creates a dial string with a prepending '&' and the subsequent Dial() fails. Any advice? |
| 21:02.58 | seanbright | sounds like a big |
| 21:03.00 | seanbright | bug* |
| 21:04.53 | seanbright | but looking at the code i am not sure how that is possible |
| 21:05.11 | seanbright | can you pastebin the console output showing a failed call? |
| 21:05.12 | seanbright | ~pb |
| 21:05.12 | infobot | pastebin is probably 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. |
| 21:05.48 | bkervaski | Yep, one sec ... |
| 21:05.54 | bkervaski | (thanks for looking) |
| 21:08.30 | bkervaski | Dial("PJSIP/heavy.801-000000d7", "&PJSIP/heavy.803/sip:heavy.803@192.168.1.1:5060,15,irxk") |
| 21:08.52 | seanbright | i don't see how that is possible |
| 21:08.58 | seanbright | can you show your dialplan? |
| 21:09.10 | [TK]D-Fender | Shouldn't be a & in front |
| 21:09.14 | bkervaski | So the endpoing PJSIP/heavy.802 was included but offline. I just noticed it will also put an '&' at the end of the dial string depending on whether the last has registered contacts or not. |
| 21:09.17 | bkervaski | Right. |
| 21:09.20 | bkervaski | TK! Wow, years. |
| 21:09.29 | seanbright | [TK]D-Fender: that is the point of the question |
| 21:09.51 | seanbright | bkervaski: pastebin your dialplan |
| 21:10.21 | seanbright | at least the part calling Dial() |
| 21:11.01 | bkervaski | exten => 101,n,Dial(${PJSIP_DIAL_CONTACTS(heavy.802)}&${PJSIP_DIAL_CONTACTS(heavy.803)},15,irxk) |
| 21:11.10 | seanbright | yes |
| 21:11.14 | seanbright | you have a & in there |
| 21:11.56 | bkervaski | Ahh.. so I mistakenly divided my separate conact lists up okay got it, feeling kind of dumb but thankful :) |
| 21:12.16 | seanbright | ${PJSIP_DIAL_CONTACTS(heavy.802)} evaluates (correctly) to empty |
| 21:12.23 | seanbright | which is why you get the leading & |
| 21:12.30 | seanbright | so you'll have to do some juggling to get what you want |
| 21:12.38 | seanbright | but PJSIP_DIAL_CONTACTS appears to be working as intended |
| 21:12.50 | bkervaski | Roger. Thanks for the (in)sanity check! |
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