| 00:20.11 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:20.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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 |
| 00:34.21 | *** join/#asterisk fchaverri (c87aa505@gateway/web/freenode/ip.200.122.165.5) |
| 00:34.37 | fchaverri | Hi good evening |
| 00:35.56 | fchaverri | I'm wondering if someone could help me troubleshoot an issue I have |
| 00:37.21 | fchaverri | I´m trying to set up a new install from scratch of asterisk to replace an existing one that is using an old version of free pbx |
| 00:37.39 | fchaverri | this new install is on dedicated hardware, the old one was a VM |
| 00:38.13 | fchaverri | however in the new setup there is a really odd behavior, outgoing calls work fine |
| 00:38.45 | fchaverri | incoming calls go to the queue, I can hear the welcome message and then the MOH |
| 00:39.38 | fchaverri | however if one of the agents (hard phones, polycom 331) pick the call the phone looks unresponsive and the MOH stops, as if the call was answered, but there is only silence |
| 00:40.22 | fchaverri | here's the funny bit: If i use a softphone instead (3cx windows client) I can answer the calls |
| 00:41.07 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 00:42.24 | fchaverri | and here's another weird thing going on, if I try to call an extension configured on a desk phone from the soft phone I have on my computer the exact same thing happens |
| 00:42.39 | fchaverri | I have all the devices on the same network |
| 00:43.52 | fchaverri | Asterisk is configured properly, the external address is fine as well as the local network ranges |
| 00:45.57 | Samot | What is the status of the peers? |
| 00:46.05 | fchaverri | the only thing that I think is different is that in the new setup I used pjsip rather than chan_sip for each extension |
| 00:47.17 | Samot | Question is still the same. |
| 00:47.37 | fchaverri | the trunk is online |
| 00:47.44 | Samot | The phones. |
| 00:48.27 | fchaverri | and the extensions looks fine too Contact: 7000/sip:7000@172.16.3.102:5060 2d36c29dac Avail |
| 00:48.52 | Samot | Show a call that does this.. |
| 00:48.55 | Samot | asterisk -rvvvvvvvvv |
| 00:48.59 | Samot | pjsip set logger on |
| 00:49.03 | Samot | ~pb |
| 00:49.03 | infobot | pastebin is, like, 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. |
| 00:49.22 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 00:50.24 | fchaverri | I'm on it samot, I'll get back to you in a moment |
| 01:00.18 | fchaverri | This is the endpoint status http://pastebin.com/A3wC1Gud and this is an extract from set logger on http://pastebin.com/3Kq6NSeW |
| 01:00.28 | fchaverri | let me know if you need a longer trace |
| 01:02.06 | Samot | That's not what I asked you to do. |
| 01:02.15 | Samot | I want the live output from the CLI |
| 01:02.46 | Samot | There's not even a call in this. |
| 01:03.26 | Samot | 8:48:58 PMÂ <Samot>Â asterisk -rvvvvvvvvv |
| 01:03.33 | Samot | 8:49:01 PM S<Samot> pjsip set logger on |
| 01:04.13 | Samot | Make the call, everything from the start of the first INVITE for the call on until the end of the call should be copy and pasted. |
| 01:05.33 | fchaverri | hold on I´m going to do that placing a test call |
| 01:09.02 | *** join/#asterisk mknooihuisen (~mknooihui@adsl-108-212-224-196.dsl.klmzmi.sbcglobal.net) |
| 01:10.39 | fchaverri | I´m going to unplug all the other phones and leave just one, the console output gets too cluttered with the info status of the other extensions |
| 01:11.07 | *** join/#asterisk u0m3 (~u0m3@79.115.167.77) |
| 01:11.37 | mknooihuisen | Hey all, quick sanity check: If I'm receiving this console message every 20 seconds, is that a hack, or FlowRoute trying to connect? [Mar 18 21:09:20] NOTICE[20769]: chan_sip.c:15751 sip_reg_timeout: -- Registration for âtech_prefix_here@sip.flowroute.com' timed out, trying again (Attempt #65) |
| 01:11.58 | *** join/#asterisk u0m3_ (~u0m3@79.115.167.77) |
| 01:12.06 | Samot | That would be Asterisk not getting a reply to registration attempts to Flowroute. |
| 01:12.47 | *** join/#asterisk u0m3_ (~u0m3@79.115.167.77) |
| 01:13.10 | mknooihuisen | Samot: I blocked SIP registration on the flowroute end, so I'm guessing that's a hack attempt, then? |
| 01:13.30 | Samot | Coming from your box? |
| 01:13.47 | *** join/#asterisk u0m3_ (~u0m3@79.115.167.77) |
| 01:13.58 | mknooihuisen | Heh, good point |
| 01:13.59 | Samot | So the hackers are trying to register your flowroute trunk, on your box? |
| 01:14.44 | Samot | Is your Flowroute peer set with a register string? |
| 01:17.30 | mknooihuisen | If by that, you mean something like this, then yes: |
| 01:17.41 | mknooihuisen | [outgoing] |
| 01:17.41 | mknooihuisen | exten => _1NXXNXXXXXX,1,Dial(SIP/${TECHPREFIX}*${EXTEN}@flowroute) |
| 01:17.58 | mknooihuisen | @flowroute is also defined, of course |
| 01:20.55 | Samot | No. that's dialplan. |
| 01:21.03 | Samot | In sip.conf |
| 01:21.15 | Samot | Do you have a register string in the Flowroute peer. |
| 01:22.26 | Samot | You disabled registrations on their end. Did you add your WAN IP to your allowed IP lists for outgoing calls? |
| 01:22.48 | Samot | And did you update your DIDs (if any) to route to the WAN IP of the PBX? |
| 01:22.48 | mknooihuisen | Got it. Yes, I do. |
| 01:23.15 | Samot | So in your peer, you don't need defaultuser/username or secret. |
| 01:23.38 | Samot | Or a registration string. |
| 01:25.12 | mknooihuisen | Most of this config was done a month or so ago (and worked then) so I'm just double checking everything |
| 01:29.55 | mknooihuisen | Samot: Okay, the Flowroute system configurator is including those values, but I trust you more. I'll go ahead and remove them. |
| 01:30.34 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-dmkpptclzpncjukd) |
| 01:34.03 | Samot | https://www.irccloud.com/pastebin/oyCmdBO7/ |
| 01:34.10 | Samot | That works fine for me. |
| 01:35.33 | Samot | t38pt_udptl=yes <-- You'll need to configure updtl.conf and other t38 stuff to use that. So you can remove it. |
| 01:40.53 | fchaverri | http://pastebin.com/ahFpibUF at line 525 you´ll see the interesting activity when I dialed ext. 7000 from the softphone x8000 |
| 01:43.15 | Samot | OK, I thought you said you replaced an old FreePBX server with an Asterisk server? |
| 01:43.27 | Samot | Why am I seeing FreePBX generated context and dialplan in this? |
| 01:44.46 | mknooihuisen | Samot: Okay, that seems to have cleared the warning, and Basic hello world works again. Thanks! Back to work for me :) |
| 01:45.12 | fchaverri | I moved from the old frepbx version to the latest one that has Asterisk 13 |
| 01:45.18 | mknooihuisen | No idea why the sip.conf I had worked a month ago and just decided to stop, but oh well |
| 01:48.04 | Samot | fchaverri: That's not what you said. There is a difference between a pure Asterisk setup and a FreePBX setup. |
| 01:48.08 | Samot | Those are important to know. |
| 01:48.25 | Samot | Mainly because FreePBX isn't support here. That's what #freepbx is for. |
| 01:49.30 | Samot | fchaverri: Fix your codecs. You have one device trying to only use g729 and the other device doesn't. And it's looking like g729 isn't enabled as a codec. |
| 01:50.13 | fchaverri | Sorry about the confusion, i meant that I'm moving from the virtual machine to new dedicated hardware |
| 01:50.21 | Samot | The calls doesn't have audio because one device is using a codec that is not enabled. |
| 01:50.43 | Samot | Make everything use ulaw/g711u |
| 01:50.46 | fchaverri | thanks for the input Samot I´m going to check why is that happening |
| 01:50.50 | Samot | On the PBX and on the devices. |
| 01:51.07 | Samot | The devices need to have the right codecs enabled on them. |
| 01:51.25 | Samot | PolycomSoundPointIP-SPIP_331-UA |
| 01:51.38 | Samot | a=rtpmap:18 G729/8000 <-- It's set to only use g728. |
| 01:51.41 | Samot | a=rtpmap:18 G729/8000 <-- It's set to only use g7289 |
| 01:51.44 | Samot | ugh. |
| 01:51.46 | Samot | g729 |
| 01:52.31 | Samot | g729 doesn't appear to be an acceptable/enabled codec on the PBX. |
| 01:56.42 | fchaverri | <PROTECTED> |
| 02:00.46 | Samot | Oh you can look in the Asterisk SIP Settings |
| 02:01.07 | Samot | How about just fix the phone to use g711u |
| 02:01.19 | freebs | tomas |
| 02:02.45 | Samot | Yes? |
| 02:03.04 | freebs | hi |
| 02:03.18 | freebs | just sayin |
| 02:05.35 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 02:07.28 | igcewieling | fchaverri: yes. You could leave off the recalc option. |
| 02:08.32 | Samot | Help in configuring his PBX should be handled in #freepbx |
| 02:08.42 | Samot | Because this is all stuff to be done correctly in the GUI. |
| 02:21.41 | fchaverri | I just checked the Polycom codecs configuration, G.711Mu and G.711A are enabled on the phone |
| 02:21.59 | fchaverri | and in position 1 and 2 |
| 02:23.45 | Samot | Make sure that ulaw is the top position in the audio codecs list in Asterisk SIP Settings |
| 02:23.57 | Samot | And g729 is unchecked. |
| 02:40.18 | fchaverri | This is how the Polycom phone should look, right? |
| 02:40.20 | fchaverri | a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 |
| 02:55.11 | fchaverri | Well, i think I'll have to boot the old vm and take this and a phone back to home and check what the heck is going on, g711 is the only codec enabled on the phone and in asterisk sip settings but the issue persists |
| 02:55.42 | fchaverri | thanks anyway everyone, specially Samot for the help today |
| 02:55.48 | fchaverri | have a good one guys |
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| 11:55.46 | starbuck | hi guys, it takes my Cisco 7970 with the latest firmware more than 5 minutes to boot till it is funtional. is thins normal or rather long? |
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| 13:36.21 | adnauseum | seems long. Did you ask Cisco? |
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| 14:33.32 | avb | oh, let me take advantage of the cisco topic. maybe anyone have ready cisco 8800 provisioning files ready for asterisk? :) |
| 14:33.40 | avb | im a bit lazy to do them from scratch |
| 14:37.33 | Samot | Not really a topic. |
| 14:37.48 | Samot | But you're talking about the FXO/FXS gateway? |
| 14:39.20 | Samot | Oh, not the SPA. The new UCM or whatever they call it now phones. |
| 14:39.22 | avb | no, a video phones |
| 14:39.27 | avb | right |
| 14:39.39 | avb | you have to provision them via this weird xml files |
| 14:40.39 | Samot | That's not weird. |
| 14:40.50 | Samot | XML is a common format for phone configs. |
| 14:40.56 | Samot | Especially with Cisco. |
| 14:41.00 | avb | anyway |
| 14:41.33 | avb | the weird part about cisco is that its a challenge to find config description |
| 14:41.44 | avb | not an xml format itself |
| 14:41.58 | Samot | What version? |
| 14:42.20 | avb | its something recent, i need to check it out |
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| 14:47.11 | Samot | Well there's three versions available and I'm not sure if the configs change between them. Because I'm too lazy to look. |
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| 14:51.04 | avb | :) haha |
| 14:51.06 | avb | np |
| 14:52.19 | avb | Samot: are you based in US? |
| 14:53.23 | Samot | Business, yes. Residence, no. |
| 14:53.33 | avb | i se |
| 14:53.35 | avb | i see |
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| 15:01.07 | echo083 | hello |
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| 17:47.54 | monsterco | does anyone know if Mitel (old Aastra) phones are lockable using provision so that they never open again without permission of provider? |
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| 18:54.15 | voipmonk | 6800i ? |
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| 19:02.43 | monsterco | voipmonk - so mitel can provisioned and if user even resets them they can go back to provisioning server but if # and 9 (i think) is held then fireware flashes and user has to tftp a new firmware to it |
| 19:03.09 | monsterco | i am wondering if i can stop the user from doing that at boot process so that the phone totally locks to us as we provide the phones free until they use our service |
| 19:03.53 | voipmonk | can u control the tftp server in a dhcp method |
| 19:04.49 | voipmonk | are you married to this make/model of phone |
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| 19:07.49 | monsterco | voipmonk - yes married |
| 19:08.07 | monsterco | a user can terminate service, take the phone and do whatever they want via tftp i guess |
| 19:08.08 | Samot | Providers who have "locked" firmware have that firmware as the factory default. It's made for them. |
| 19:10.06 | Samot | Otherwise any user can follow the factory default rules for pretty much any device that offers them through a process that doesn't require admin access. |
| 19:10.34 | voipmonk | Polycom and Cisco have both made my custom firmware -- Im not sure how to do it without Mitels help here - looking for an administrators provisioning guide still relevant for those phone models. |
| 19:11.09 | Samot | Yup, just like the old Linksys would have firmware for certain providers. |
| 19:11.10 | voipmonk | but without a custom firmware the user could have some nice playtime |
| 19:11.24 | voipmonk | a custom firmware build |
| 19:11.38 | Samot | Yup. That's the only way to guarantee the device is yours. |
| 19:11.58 | voipmonk | but we had 15k users, dunno how to approach them without the $ numbers. |
| 19:12.01 | Samot | Otherwise, anyone can pretty much reset the phone and have playtime. |
| 19:12.17 | voipmonk | call and ask |
| 19:12.46 | Samot | You got the $$$ I bet they'll do it. |
| 19:13.08 | Samot | I bet there is some volume buying of hardware involved. |
| 19:14.07 | voipmonk | oh lord |
| 19:14.17 | Samot | I think a development/engineering fee and a volume order will be needed. |
| 19:15.23 | Samot | It's the only thing that makes sense. They'll build the firmware and flash X minimum devices per request. |
| 19:15.25 | voipmonk | 613 592 2122 i think |
| 19:15.40 | voipmonk | i'll call em. |
| 19:15.49 | voipmonk | 7am to 5pm m-f |
| 19:17.08 | Samot | Just be prepared for the sitting on hardware possibility. |
| 19:18.49 | Samot | Or a high engineering/development cost if the volume is low. They're going to make their money. |
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| 19:19.15 | Samot | Locking devices was so much easier 10+ years ago. |
| 19:19.23 | voipmonk | call tomorrow |
| 19:19.43 | Samot | Because the average user didn't know how to find the information and it wasn't as wide spread to get anyways. |
| 19:19.57 | voipmonk | 613 592 2122 guy said call tomorrow - #2 , #3 |
| 19:20.05 | voipmonk | option 2 then option 3 and ask |
| 19:20.24 | voipmonk | but I don think its gonna be free :) |
| 19:20.30 | Samot | Or cheap. |
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| 19:21.51 | Samot | This is why I have a percentage of the user costs cover devices. |
| 19:22.40 | Samot | I either provide them free or with a small setup and then get the cost back over the first 12 months. |
| 19:23.38 | monsterco | so there is a custom firemware for mitel? |
| 19:23.45 | voipmonk | you can have it built |
| 19:24.16 | voipmonk | i have had polycom custom firmware and linksys and cisco custom firmware built - and the guy at mitel said call them tomorrow and ask them |
| 19:24.56 | monsterco | maybe it's linux afterall and open? |
| 19:25.01 | Samot | Just wont be cheap |
| 19:30.24 | drmessano | Linux != open |
| 19:31.04 | Samot | Based on Mitels history, it wont be open |
| 19:31.18 | drmessano | Exactly |
| 19:32.02 | Samot | This is a company that has made its money on owning thier PBXes and the phone for them. |
| 19:33.42 | Samot | All the way back to their PRI/T1/FXS system days. |
| 19:34.06 | Samot | And thats still a big chunck of their users. |
| 19:34.18 | Samot | Because those things are goddamn workhorses. |
| 19:34.51 | Samot | Pretty much all the hotels i have sip gateways at, Mitel PBX |
| 19:35.13 | Samot | That they have had for 20+ years |
| 19:36.16 | Samot | And still pay $500+ a pop for replacement cards |
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