| 00:08.15 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 00:17.03 | *** join/#asterisk mmlj4 (1000@ip98-163-252-185.no.no.cox.net) |
| 00:18.43 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 00:39.55 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 00:41.49 | *** join/#asterisk nimbius (~nimbius@putrid.dev1ce.com) |
| 00:42.08 | nimbius | anyone have documentation or experience with SIP and SMS? specifically getting SMS across to an asterisk gateway? |
| 00:44.22 | nimbius | nm |
| 00:44.23 | *** part/#asterisk nimbius (~nimbius@putrid.dev1ce.com) |
| 00:49.18 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 00:50.48 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-mig-svr.ad.v10networks.ca) |
| 01:15.50 | Samot | Wow. Two minutes. |
| 01:21.04 | drmessano | SMS doesnt take long |
| 01:22.21 | Samot | I figured it was going to be like the guy the other night. |
| 01:22.55 | Samot | "What do you mean I have to use an SMS gateway and connect to other SMS gateways?" |
| 01:24.10 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 01:25.29 | drmessano | Sounds like something freebs would ask |
| 01:25.41 | freebs | relax fella |
| 01:28.41 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:31.45 | Samot | Yeah. |
| 01:31.48 | Samot | It does. |
| 01:31.56 | Samot | freebs...psssh. |
| 01:32.10 | Samot | He runs his Asterisk on loopback x2 |
| 01:32.27 | Samot | 154.0.0.2 |
| 01:32.37 | Samot | 254.0.0.2 |
| 01:50.03 | Akari | So what's the alternative to freenum.org today? |
| 01:53.55 | Akari | Eh nvm |
| 01:58.26 | drmessano | lol |
| 01:58.35 | drmessano | freenum and all that are just... |
| 01:59.29 | drmessano | sip://ginger@spiceworld.com |
| 01:59.33 | drmessano | Thats how I roll |
| 02:01.47 | Akari | It looks like freenum is completely dead anyway |
| 02:02.53 | drmessano | Anything that substitutes dialing numbers for dialing numbers is just another idiotic system |
| 02:03.07 | drmessano | Another problem |
| 02:03.13 | Akari | What |
| 02:03.33 | drmessano | Like ENUM |
| 02:04.10 | drmessano | e164.org et al |
| 02:04.31 | Akari | The purpose was to resolve numbers to SIP URIs |
| 02:04.55 | drmessano | Yep |
| 02:07.08 | Akari | So that's not assigning numbers to numbers, because URIs often as you noted involve alphabetical characters. |
| 02:10.02 | drmessano | e614.org was supposed to be "owned" PSTN numbers to SIP URIs.. there were others that offered a unique number that looked up a SIP URI.. Which to me was idiotic, because you're adding a new thing to a middleware shim |
| 02:10.09 | drmessano | e164.org* |
| 02:13.10 | Akari | Yeah because everyone has a QWERTY keyboard on their deskphone >_> |
| 02:13.17 | drmessano | No |
| 02:13.33 | drmessano | You can do the same thing locally |
| 02:14.06 | Akari | You mean by typing in the IP address? |
| 02:14.11 | drmessano | I think it would have been neat, 10 years ago, to create a Web UI Address Book so you can manage some PSTN <> SIP URI translation that's added to the dialplan |
| 02:14.42 | drmessano | No, literally a local extensions.conf entry for the remote SIP URI |
| 02:15.53 | *** join/#asterisk matrix1233 (~matrix123@197.0.85.47) |
| 02:30.46 | drmessano | So anyway |
| 02:45.06 | Samot | derp. |
| 02:45.23 | drmessano | You know.. |
| 02:45.41 | drmessano | e164.org and FreeNUM.. two examples of how they getcha |
| 02:45.52 | drmessano | If you think about it.. |
| 02:47.22 | drmessano | I could just "translate" numbers to URI's in my PBX |
| 02:47.34 | drmessano | Which is in my control.. |
| 02:47.42 | drmessano | or go Opt-In to someone elses database |
| 02:47.50 | drmessano | Which is exactly how they get ya |
| 02:48.19 | drmessano | Now they have your DID and a SIP URI for you.. all in one little database entry |
| 02:48.55 | Samot | Yeah... |
| 02:49.27 | drmessano | They get DNS lookups for every number you call |
| 02:49.46 | drmessano | Because if you're going to do ENUM, you're going to lookup EVERY call |
| 02:50.03 | Samot | Oh yeah. |
| 02:50.05 | drmessano | So I am now collecting all of these call logs from everyone |
| 02:50.44 | drmessano | So even better |
| 02:51.08 | drmessano | FreePBX had an "ENUM Trunk" type, right |
| 02:51.15 | drmessano | I assume it's still there |
| 02:51.49 | drmessano | I think it did lookups on e164.com/org/net at one time.. I think that was culled down to .org because it was the only valid one after some time |
| 02:51.58 | drmessano | Now it appears that one is dead |
| 02:52.25 | Samot | I'm not sure it still exists. |
| 02:52.29 | drmessano | But regardless.. Nothing stops the average whoever from adding it because "It looks like a cool thing" |
| 02:53.07 | drmessano | It's in 13 |
| 02:53.16 | drmessano | So I add an ENUM trunk |
| 02:53.29 | drmessano | Just blasting those lookups out there |
| 02:53.34 | drmessano | Who owns e164.org ? |
| 02:53.59 | Samot | Yeah you can add on..but there's not options for any details. You can name it and have pattern matching.. |
| 02:54.09 | Samot | Shrug |
| 02:54.38 | Samot | Hrm. |
| 02:54.42 | drmessano | The domain info is hardcoded |
| 02:54.55 | Samot | According to the whois, he lives like 40 minutes from me. |
| 02:54.57 | drmessano | So some guy owns it |
| 02:55.04 | drmessano | NOW.. |
| 02:55.09 | drmessano | what happens when he drops it |
| 02:55.26 | Samot | Oh...you'll never see it. |
| 02:55.27 | drmessano | and say, whoever buys it up.. puts a DNS server on it |
| 02:55.41 | drmessano | Starts collecting lookups |
| 02:55.45 | drmessano | *WOKE* |
| 02:56.52 | drmessano | Really, FreePBX just makes it EASY.. lots of documentation that still shows up in Google for adding e164.org lookups to vanilla Asterisk dialplans |
| 02:57.11 | drmessano | Im sure on other platforms as well |
| 02:58.02 | drmessano | So this is why I hate things any Middleware DID <> URI translation nonsense |
| 02:58.15 | drmessano | -things (changed thought) |
| 02:58.49 | drmessano | "Call my SuperNUM number at 34938739874837" |
| 02:58.54 | drmessano | Yep, sure thing Mr Soros |
| 03:01.03 | Akari | drmessano: The problem with translating URIs in your PBX is that you have to set it up for every PBX you reference. So that's never going to be a global or even broadly local solution. |
| 03:01.47 | drmessano | Akari: Neither is expecting every other PBX admin on the planet to OPT-IN to your middleware thingo |
| 03:02.07 | drmessano | Actually, thats more ridiculous |
| 03:02.18 | Akari | It's at least optional. |
| 03:02.34 | Akari | And scalable. |
| 03:03.09 | Samot | There is no reason for my users to be able to dial "SIP URIs" |
| 03:03.17 | drmessano | So ME adding 50 URI's to my PBX or a group of PBXs is more work than going to 50 admins and asking them to create accounts and OPT-IN to some middleware |
| 03:03.19 | drmessano | No way |
| 03:03.20 | Samot | Which is dumb anways. |
| 03:03.35 | Samot | 3135551212@sip.domain.com <- URI |
| 03:03.46 | Samot | Which is basically how all SIP is handled. |
| 03:04.13 | Akari | Yes but what if you wanted to call ray@rayromano.com or whatever |
| 03:04.47 | drmessano | Akari: You call Ray's admin and ask him to opt-in to e164.org so your calls will go over SIP? |
| 03:04.49 | Samot | You ask like that's a common thing. |
| 03:04.49 | drmessano | LULZ |
| 03:04.56 | drmessano | exten => 3135551212,1,Dial(SIP/samot@spicegirls.com) |
| 03:04.58 | Akari | And who's going to be routing that call |
| 03:05.10 | Samot | Huh? |
| 03:05.10 | Akari | It's not |
| 03:05.15 | Akari | Because it's too difficult |
| 03:05.19 | Akari | Because it's not a common thing |
| 03:05.25 | Akari | Because it's too difficult |
| 03:05.27 | Akari | Turtles all the way down |
| 03:05.40 | Samot | Because it's nothing that ever caught on as being practical. |
| 03:05.55 | Akari | Because it doesn't have a big enough market share. |
| 03:05.56 | Samot | It allows your users to send calls randomly over the Internet. |
| 03:06.01 | drmessano | The OPT-IN/Create Account/Verify numbers in some middleware is horrible impractical. |
| 03:06.17 | Samot | Akari: It hasn't had a big enough market share in over a decade. |
| 03:06.24 | Samot | Because it's not practical. |
| 03:06.45 | drmessano | But if a user came to me and provided a SIP URI, I could add that to the dialplan.. or as I said above, you could create an address book WEBUI to do it |
| 03:07.43 | drmessano | e164.org and FreeNUM were dead before they even started |
| 03:08.35 | drmessano | Both for requiring some opt-in, and additionally the later for creating some new number scheme which basically introduced more useless numbers |
| 03:09.45 | *** join/#asterisk matrix1233 (~matrix123@197.0.85.47) |
| 03:09.51 | drmessano | Additionally for requiring a high level of trust, considering in order to be practical, ALL of your calls needed to perform a lookup |
| 03:11.20 | Samot | Also weren't they really just some solutions to the fact SIP was still young and not as adopted by the carriers. |
| 03:11.58 | drmessano | No, they purely existed as middleware to facilitate SIP <> SIP calls outside the PSTN |
| 03:12.39 | Samot | Oh yeah.. |
| 03:12.50 | drmessano | I had a good little list of SIP URI's that I made shortcuts for in a couple PBXes |
| 03:13.00 | Samot | There's also the fact you would have to let your PBX accept calls from basically ANYWHERE |
| 03:13.04 | drmessano | Custom Extensions in FreePBX |
| 03:13.23 | Samot | Since you would have no idea where the call is originating from. |
| 03:13.24 | drmessano | and to be honest |
| 03:13.32 | Samot | Secure. |
| 03:13.51 | drmessano | I didnt mind the e164 numbering so much.. because I already knew their PSTN numbers |
| 03:14.17 | Samot | At the ITSP I was at, we used ENUM internally. |
| 03:14.21 | drmessano | I actually started off giving them unique short extensions that fit into my extension scheme |
| 03:14.34 | Samot | To do internal routing from the PSTN or to the PSTN. |
| 03:14.34 | drmessano | But then I couldnt remember if Steve was 301 or 303 |
| 03:14.46 | drmessano | So I created an extension that matched his cell number |
| 03:14.59 | drmessano | Had it call his PBX |
| 03:15.22 | drmessano | That was a lot of fun.. and didnt require a middleware |
| 03:22.05 | *** join/#asterisk KJ4IPS (aaronhaung@gateway/shell/matrix.org/x-tbbnpifotxnowbgm) |
| 03:24.45 | KJ4IPS | So, I just updated to 14.1.1 and I am seeing a long delay (4+ seconds) before SIP invites are acted upon. My trunk provider (flowroute) gives up first, buy my assorted softphones, and my cisco 7940 (sip firmware) still work if you wait it out. Shaking wreshark at the problem shows the INVITES coming in, and some time later, multiple 100 trying's going out... |
| 03:24.49 | KJ4IPS | Any ideas? |
| 03:26.08 | Samot | INVITES coming in from the endpoints or Flowroute? |
| 03:26.19 | KJ4IPS | Both. |
| 03:27.58 | KJ4IPS | https://gist.github.com/anonymous/f711391227a9ce79865179c1fbe5cc48 |
| 03:28.22 | KJ4IPS | .102 is the ep, asterisk is on .1 |
| 03:29.22 | Samot | Show a real SIP debug from Asterisk. |
| 03:29.47 | KJ4IPS | Incoming... |
| 03:34.54 | *** join/#asterisk somepoortech (~somepoort@72-0-128-202.static.firstlight.net) |
| 03:34.59 | KJ4IPS | https://gist.github.com/anonymous/f66abdbeb7cf81ea74907a06e49509d7 |
| 03:35.06 | KJ4IPS | and |
| 03:35.06 | KJ4IPS | https://gist.github.com/anonymous/d5bcdaddf2242cb1dd219b1f409c7c32 |
| 03:35.37 | KJ4IPS | The first is the cisco unit, which waits it out, the second is from flowroute, who does not. |
| 03:37.20 | Samot | That's it? |
| 03:37.29 | Samot | Where's all the timed out SIP messages? |
| 03:37.44 | Samot | Your debug needs to show support of your statement. |
| 03:38.11 | KJ4IPS | The cisco does not time out...... |
| 03:38.21 | KJ4IPS | Whoops, cropped the flowroute one a bit harsly. |
| 03:38.24 | Samot | Your debug shows the INVITE |
| 03:38.26 | Samot | That's it. |
| 03:38.32 | Samot | No replies, nothing. |
| 03:38.38 | KJ4IPS | (That is my fault). |
| 03:38.38 | Samot | It's one packet of the call. |
| 03:41.35 | KJ4IPS | https://gist.github.com/anonymous/e7d669c8cd1846a42665ac9f3e484021 |
| 03:41.57 | KJ4IPS | That is from the time I place the call from the PSTN, to the time that ast gives up trying retransmission. |
| 03:42.21 | KJ4IPS | (or at least according to the error messages, the finer details still frustate me)... |
| 03:45.50 | Samot | Fix your NAT |
| 03:45.51 | Samot | o=root 882818471 882818471 IN IP4 192.168.0.1 |
| 03:45.59 | KJ4IPS | There is no nat, |
| 03:46.07 | Samot | Your sending replies to Flowroute with private IPs. |
| 03:46.12 | KJ4IPS | That machine is connected directly to the modem, |
| 03:46.21 | KJ4IPS | Oh... |
| 03:46.24 | Samot | So it has a public IP? |
| 03:46.26 | KJ4IPS | Dual homing hell |
| 03:46.54 | Samot | The PBX has a public WAN IP? |
| 03:46.59 | KJ4IPS | It is not sending the correct IP, there are about five interfaces on that box. |
| 03:47.31 | KJ4IPS | It's a solitary home server. |
| 03:47.44 | Samot | That's not the answer to my question. |
| 03:47.48 | Samot | I have no idea what that means. |
| 03:47.54 | KJ4IPS | Yes, the PBX does have a public IP. |
| 03:47.56 | Samot | Does the server have a public WAN IP? |
| 03:48.01 | Samot | And is Asterisk listening on it? |
| 03:48.03 | KJ4IPS | It also has a private IP. |
| 03:48.21 | KJ4IPS | Asterisk is listening on both, the firewall only allows incoming packets from Flowroute on sip from the wan. |
| 03:49.27 | KJ4IPS | I eventually plan to move it onto a VM, once I get an ESXi box running, but for now it's on bare metal. |
| 03:50.32 | Samot | I'm not sure how that has bearing on the network settings not being correct. |
| 03:53.13 | KJ4IPS | So, I've fixed the bind address, apparently it was still binding to the interface, and was somehow working pre-upgrade despite being configured wrong. |
| 03:54.07 | Samot | Has that fixed the issue? |
| 03:54.20 | KJ4IPS | Sadly no.. |
| 03:57.18 | Samot | Show a new call. |
| 03:58.15 | KJ4IPS | https://gist.github.com/anonymous/520be78a3b7b86f216a8a696e9d2a9c7 |
| 03:58.29 | KJ4IPS | First reply is on 179 |
| 04:01.16 | Samot | Show your Flowroute peer, mask any secrets. |
| 04:01.26 | Samot | Ignoring this INVITE request <-- This is happening. |
| 04:02.56 | KJ4IPS | https://gist.github.com/anonymous/c5fa8c3b14ef688649b9cd75bc5edae4 |
| 04:07.38 | Samot | Which IP should the calls be going to? |
| 04:08.16 | KJ4IPS | They should be coming to 68.119.116.168 |
| 04:08.54 | KJ4IPS | From either 70.167.153.130 or 216.115.69.144 |
| 04:12.30 | Samot | It's sending replies to the INVITE and there's never a reply back to it. |
| 04:12.47 | Samot | Are you peering or using registration with Flowroute? |
| 04:12.55 | KJ4IPS | Reg. |
| 04:13.03 | KJ4IPS | I'm stuck with a dynamic IP for my WAN. |
| 04:13.41 | KJ4IPS | I can see the registration both in the asterisk console, as well as in flowroute's site. |
| 04:14.22 | Samot | Well they're sending you an INVITE you're sending back a 100 Trying and that's it. |
| 04:14.33 | Samot | There's nothing in response or follow up to that 100 Trying. |
| 04:15.11 | Samot | Then Flowroute is sending the INVITES again and again but Asterisk is ignoring the subsequent INVITES on the same CSeq. |
| 04:15.59 | Samot | But you're presenting the right IP this time... |
| 04:16.10 | KJ4IPS | Let me check something... |
| 04:16.14 | Samot | You might have to work with Flowroute to see if they are getting replies back from you on the Invites. |
| 04:21.05 | KJ4IPS | The timing of things just does not seem right |
| 04:21.10 | KJ4IPS | Take a look at this: |
| 04:21.10 | KJ4IPS | https://gist.github.com/anonymous/a509e3b1e94448c94e030efda9e58af7 |
| 04:21.17 | KJ4IPS | Note the timestamps, they are in seconds. |
| 04:21.31 | KJ4IPS | (2nd column) |
| 04:23.21 | Akari | If you can see sip message logs on the Flowroute site, it might help to see those alongside the Asterisk reg. |
| 04:27.11 | KJ4IPS | I'm going to bounce back the zfs snapshot to before the upgrade, and take a new swing at this tomorrow. |
| 04:39.29 | drmessano | I hate Windows |
| 04:57.37 | *** join/#asterisk matrix1233 (~matrix123@197.0.85.47) |
| 05:05.51 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 05:06.18 | Samot | Uuuughh. If my sinuses fill up anymore I think my eyes and ears are going to pop out like that old school squeeze toy.. |
| 05:10.42 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 05:13.53 | drmessano | So we deployed SCCM |
| 05:14.03 | drmessano | Got a machine I installed new |
| 05:14.07 | drmessano | Windows 7 SP1 |
| 05:14.43 | drmessano | SCCM policy for WSUS defaults to some ultra low amount of metadata any workstation can exchange per session |
| 05:14.57 | drmessano | So if you have like 4 years of updates to catch up on |
| 05:15.06 | drmessano | Because you installed SP1 level |
| 05:15.13 | drmessano | <PROTECTED> |
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| 07:13.17 | Micc | I've added the syslog.local0 line in my logger.conf, and my rsyslog is sending *.* to an rsyslog server, but when it receives it, it doesn't seem to have the facility of local0 anymore and gets logged in a catchall file even though local0.* I'm logging to a specific file, it's not sending it there. |
| 07:13.39 | Micc | I feel like I must be missing something here. |
| 07:14.09 | Micc | using syslog.local0 should log everything with the local0 facility. |
| 07:30.19 | Micc | turns out it's using user, not local0 |
| 07:30.35 | Micc | is that a bug in asterisk or a bug in rsyslog? |
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| 11:21.32 | wdoekes | Micc: https://issues.asterisk.org/jira/browse/ASTERISK-25510 perhaps? |
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| 11:31.07 | danielyk | Hi, I have a interesting problem with pjsip. Following situation: Our voicemail greeting is in gsm and slin codec saved. Our trunk supports alaw and g722. If both are selected as allowed (for the trunk), I do not hear the greeting at the other end (g722). If I set the trunk ONLY allows alaw OR g722, I can hear the greeting at the other phone. |
| 11:36.53 | danielyk | Has this something to do with asymetric RTP codec? |
| 11:39.55 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 12:09.11 | danielyk | @file Could you say something to this? |
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| 12:30.05 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 12:30.07 | wasanzy | hi |
| 12:30.46 | *** join/#asterisk danielyk (~danielyk@p5B21BC6C.dip0.t-ipconnect.de) |
| 12:31.50 | wasanzy | I have a file containing list of phone numbers. I want to write a diaplan to read the file and if a match is found in the file, an action should be taken |
| 12:33.16 | [TK]D-Fender | wasanzy, "core show function SHELL" <- |
| 12:33.30 | wasanzy | ok |
| 12:35.41 | wasanzy | what of this? ReadFile |
| 12:36.50 | [TK]D-Fender | Naturally no |
| 12:37.01 | wasanzy | ok |
| 12:38.34 | wasanzy | [TK]D-Fender: I haven't gotten much info on the SHELL func |
| 12:38.55 | [TK]D-Fender | Then you aren't thinking in shell terms |
| 12:39.54 | [TK]D-Fender | there is virtually nothing to know about SHELL. It's what you DO with it that matters |
| 12:40.14 | wasanzy | I am, but my confusion is, whether to write a shell script or just run the command |
| 12:40.39 | [TK]D-Fender | You clearly don't need a whole script to see if a line of text is there |
| 12:41.36 | *** join/#asterisk bof23 (~Thunderbi@185.13.183.107) |
| 12:42.37 | danielyk | Hi, I try to set preferred_codec_only for a pjsip endpoint but get "Could not find option suitable for category '2000' named 'preferred_codec_only' at line 27 of". What do I wrong? |
| 12:43.11 | [TK]D-Fender | Stop trying to wrong. |
| 12:43.17 | wasanzy | I can use the grep command right |
| 12:43.18 | [TK]D-Fender | Start trying to right :) |
| 12:43.27 | [TK]D-Fender | wasanzy, Yes. |
| 12:43.47 | wasanzy | ok |
| 12:44.47 | danielyk | @[TK]D-Fender I try... According to https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample it should be right... |
| 12:45.17 | [TK]D-Fender | danielyk, You aren't showing your config. |
| 12:45.45 | wasanzy | [TK]D-Fender: if there is a match, the exist status is 0 right? |
| 12:45.55 | wasanzy | exit* |
| 12:46.58 | [TK]D-Fender | Since when does it care about exit status? |
| 12:47.06 | [TK]D-Fender | You don't seem to have read the basic instructions at all |
| 12:47.12 | *** join/#asterisk jkroon (~jkroon@196.14.130.133) |
| 12:47.21 | [TK]D-Fender | [TK]D-Fender> wasanzy, "core show function SHELL" <- |
| 12:47.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 12:49.41 | danielyk | @[TK]D-Fender That should be enough http://pastebin.com/kMwrZkpw |
| 12:50.57 | [TK]D-Fender | danielyk, I see stuff clearly missing VS the one above it |
| 12:51.05 | [TK]D-Fender | danielyk, An incomplete definition there.... |
| 12:51.14 | [TK]D-Fender | like TRANSPORT for starters |
| 12:51.50 | danielyk | I have to copy &paste all together (because I use FreePBX and all is in different files) |
| 12:52.17 | danielyk | Should I copy all together? |
| 12:53.28 | [TK]D-Fender | danielyk, Look at what you just provided |
| 12:54.50 | danielyk | Is there a mistake? |
| 12:55.30 | [TK]D-Fender | <[TK]D-Fender> danielyk, An incomplete definition there.... |
| 12:55.31 | [TK]D-Fender | <[TK]D-Fender> like TRANSPORT for starters |
| 12:55.45 | wasanzy | 'Set(foo=${SHELL(echo \bar\)})' the output is stored in foo right? |
| 12:57.49 | [TK]D-Fender | wasanzy, are you having trouble understanding what the instructions are saying? |
| 12:58.09 | [TK]D-Fender | [Description] |
| 12:58.09 | [TK]D-Fender | Collects the output generated by a command executed by the system shell |
| 12:58.15 | [TK]D-Fender | what is so hard to follow here? |
| 12:59.40 | wasanzy | collect, fine, and stored where? that is my confusion. that was why I asked if it stored in the variable |
| 13:00.17 | danielyk | [TK]D-Fender http://pastebin.com/uVjD2aSS |
| 13:00.55 | [TK]D-Fender | wasanzy, Do you also have NO clue about channel variables and what Set() is? Go read the book. |
| 13:00.57 | [TK]D-Fender | ~book |
| 13:00.57 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:02.39 | [TK]D-Fender | danielyk, I've told you twice you're missing a TRANSPORT for [2000]. Is there a point where you're going to fix this and try again? |
| 13:05.16 | danielyk | oh, have not seen this. The files are generated by FreePBX, but I add it manually and try it again |
| 13:07.52 | danielyk | [TK]D-Fender, I have added transport=0.0.0.0-udp, but I still get the ERROR for both endpoints. |
| 13:08.37 | [TK]D-Fender | danielyk, You also never said anything about "both" endpoints. Your description and debug arenot complete |
| 13:09.50 | danielyk | Yes, you are right and sorry for this. What can I do for help you? |
| 13:10.41 | file | the preferred_codec_only option only exists in master, it does not exist in 13 or 14 |
| 13:13.26 | danielyk | file, thanks for clarification. I have also tried to set asymmetric_rtp_codec but I get the same error. Is asymmetric_rtp_codec also not available? |
| 13:13.44 | file | that feature is not yet in a release |
| 13:13.59 | file | it will be in 13.13.0 and 14.2.0 |
| 13:14.59 | danielyk | Okay, thank you. Following situation: Our voicemail greeting is in gsm and slin codec saved. Our trunk supports alaw and g722. If both are selected as allowed (for the trunk), I do not hear the greeting at the other end (g722). If I set the trunk ONLY allows alaw OR g722, I can hear the greeting at the other phone. |
| 13:15.39 | danielyk | file, I think my provider does not support asymmetric_rtp_codec. So I have to wait for preferred_codec_only or is there something other for fixing my problem? |
| 13:16.07 | file | if you provide console log and debug someone here may be able to answer that question |
| 13:16.12 | file | otherwise it's guessing |
| 13:19.46 | danielyk | Is there someone who could have a deeper look into my problem? If yes I would collect logs and give it to you |
| 13:20.57 | file | there's noone in particular, it's community support ^_^ |
| 13:23.14 | *** join/#asterisk Rico (~Rico@unaffiliated/rico29) |
| 13:23.16 | Rico | hi |
| 13:23.40 | Rico | I have a problem with asterisk13 which does not wan to start when launched via systemd ... |
| 13:23.42 | Rico | http://fpaste.scsys.co.uk/538967 |
| 13:24.19 | Rico | running asterisk the same way as the way it's launched via systemd works fine |
| 13:24.24 | Rico | asterisk logs are empty |
| 13:26.53 | *** join/#asterisk Oatmeal (~Suzeanne@2602:306:3676:c60:413e:4703:801:3756) |
| 13:27.39 | danielyk | Okay, file and all other of the community. Here is my log (verbose 3 and debug 3). I only hear beep.g722 (and do not hear e.g. vm-theperson.gsm, all digits, vm-intro.gsm and so on). http://pastebin.com/XnKvGKt3 |
| 13:31.31 | Rico | I think the problem is related to asterisk, because systemd debug says : Received SIGCHLD from PID 1071 (asterisk). |
| 13:38.39 | Samot | danialyk: What's the issue again? |
| 13:43.59 | danielyk | Samot, Our voicemail greeting is in gsm and slin codec saved. Our trunk supports alaw and g722. If both are selected as allowed (for the trunk), I do not hear the greeting at the other end (g722). If I set the trunk ONLY allows alaw OR g722, I can hear the greeting at the other phone. |
| 13:44.48 | Samot | And which version of it was the debug? |
| 13:46.02 | Samot | Because transcoding is happening. |
| 13:46.28 | Samot | That debug shows things going from gsm -> alaw, alaw->g722 and back and forth... |
| 13:46.57 | danielyk | 13.12.1 |
| 13:47.05 | Samot | No. |
| 13:47.13 | Samot | Is this the debug with two codecs set or just one? |
| 13:47.28 | danielyk | two codecs |
| 13:47.32 | Samot | You know, the two sceneraios you presented. |
| 13:47.43 | danielyk | and I have not heard the gsm file |
| 13:47.59 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 13:48.01 | Samot | Well if there was a codec issue or a transcoding issue, the call would die. |
| 13:48.06 | Samot | So far I see it transcode |
| 13:48.24 | Samot | I see it playback recordings, it opens a file to record the VM message... |
| 13:48.47 | Samot | If you're going to use g722, you might want the g722 recordings. |
| 13:48.58 | Samot | Because it's going to load the recording that best matches. |
| 13:49.33 | pawiecki | I call SIP/number on another *, dialing IAX2/2nd-asterisk/number. How can i check what is the exact IAX2 cause code, of call? |
| 13:50.06 | Samot | If you have thanks.gsm, thanks.wav and thanks.slin and Asterisk needs to play "thanks" it's going to pick one of those that best matches the codec you're using. |
| 13:50.10 | danielyk | Samot, the recordings do not have to be g722. |
| 13:50.18 | Samot | I know that. |
| 13:50.21 | Samot | Just said that |
| 13:50.24 | danielyk | ok |
| 13:50.30 | Samot | What I said was try it. |
| 13:50.36 | Samot | You know, prove it out. |
| 13:50.38 | Samot | Troubleshoot. |
| 13:50.38 | file | the issue may have been fixed in the 13 branch as a result of the asymmetric work |
| 13:50.51 | Samot | True. |
| 13:51.03 | danielyk | So I have to wait for 13.13.0? |
| 13:51.08 | Samot | Perhaps. |
| 13:51.33 | Samot | Or in the interim have the g722 recordings would solve the issue. |
| 13:51.44 | Samot | Try it. |
| 13:51.45 | Samot | Check it. |
| 13:52.09 | danielyk | okay, i try it |
| 13:52.27 | danielyk | Thanks Samot and file for this first look |
| 13:56.40 | danielyk | Samot, you mean I should try to set format=wav in voicemail.conf? |
| 13:56.47 | danielyk | *format=g722 |
| 13:57.57 | Samot | Do you have the .722 files? |
| 13:58.34 | *** join/#asterisk sekil (~sekil@130.180.229.38) |
| 13:58.40 | danielyk | for the playback? |
| 13:58.45 | Samot | No. |
| 13:58.51 | Samot | For your ipod. |
| 13:58.52 | Samot | Yes. |
| 13:59.01 | Samot | Do you have those wideband files? |
| 13:59.05 | danielyk | ... No I have not |
| 13:59.08 | Samot | OK |
| 13:59.19 | danielyk | Asterisk cant pre-transcode all files or? |
| 13:59.21 | Samot | So when I said if you have three formats and it will pick the best.... |
| 13:59.38 | danielyk | gsm is my best format for these files :D |
| 13:59.43 | Samot | What part of "It is transcoding" is hard to understand? |
| 13:59.54 | Samot | What I am saying is, DONT TRANSCODE |
| 14:00.03 | Samot | Offer the g722 files. |
| 14:00.12 | Samot | So it's not taking a .gsm file and trying to do something with it. |
| 14:00.25 | Samot | have thanks.g722 |
| 14:01.18 | danielyk | There are no good german sound files with g722 available |
| 14:01.37 | Samot | Do you need german for testing?! |
| 14:01.54 | Samot | I didn't say make this your permanent setup. |
| 14:02.01 | Samot | This is troubleshooting. |
| 14:02.06 | danielyk | If I sap to english version, all files are played correctly |
| 14:02.14 | danielyk | *swap |
| 14:02.22 | Samot | And do you have g722 files in the english version? |
| 14:03.06 | danielyk | Yes. they are available as g722 and ulaw |
| 14:04.09 | danielyk | sometimes also wav or sln |
| 14:04.47 | Samot | Hrm.. |
| 14:04.54 | Samot | [2016-11-04 14:22:07] DEBUG[20498][C-0000008c]: app_voicemail.c:6661 leave_voicemail: /var/spool/asterisk/voicemail/default/2000/busy doesn't exist, doing what we can <--- What's up with that? |
| 14:05.22 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 14:05.23 | Samot | [2016-11-04 14:22:07] DEBUG[20498][C-0000008c]: channel.c:5590 set_format: Channel PJSIP/023840504584-0000008a setting write format path: gsm -> alaw |
| 14:05.31 | Samot | [2016-11-04 14:22:07] DEBUG[20498][C-0000008c]: res_rtp_asterisk.c:3549 ast_rtp_write: Ooh, format changed from none to alaw |
| 14:06.11 | Samot | <PROTECTED> |
| 14:06.20 | Samot | So that's what happens when the call goes to VM. |
| 14:06.49 | danielyk | Okay, whats up with "format changed from none to alaw" ? |
| 14:07.56 | Samot | That was the first time.. |
| 14:08.07 | Samot | So it means there was no other codec being used. |
| 14:08.17 | Samot | Because if you look, when it plays back 2, 0, 0, 0 |
| 14:08.18 | danielyk | okay, that is okay |
| 14:08.22 | Samot | It doesn't do that. |
| 14:08.33 | danielyk | okay, fine. |
| 14:09.07 | danielyk | That there is no busy directory, should I create it? |
| 14:09.31 | Samot | [2016-11-04 14:22:17] DEBUG[20498][C-0000008c]: res_rtp_asterisk.c:3549 ast_rtp_write: Ooh, format changed from alaw to g722 |
| 14:09.55 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:10.31 | Samot | This has nothing to do with a codec issue in regards to the actual call. |
| 14:10.53 | Samot | Because it's pretty clear your call is moving from alaw, g722, gsm, etc. |
| 14:11.01 | Samot | It's loading all the files, playing them all back. |
| 14:12.14 | Samot | You may want to look at a language pack. |
| 14:12.20 | danielyk | Yes, but why I do only hear the beep.g722? |
| 14:12.34 | Samot | Because you have a g722 file. |
| 14:13.00 | danielyk | Yes, right. But Iif all other files are transcoded correctly I would hear them also |
| 14:13.13 | Samot | Is it only when you are using g722? |
| 14:13.42 | Samot | Offer up both alaw and g722 but make alaw the priority.. |
| 14:13.54 | Samot | Make it so both are offered in the SDP just like now.. |
| 14:14.03 | Samot | But make alaw the first and preferred choice. |
| 14:14.09 | Samot | Do you have the same issues? |
| 14:14.10 | *** join/#asterisk kharwell (kharwell@nat/digium/x-fiphuifwavlsvvzg) |
| 14:14.10 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:14.20 | danielyk | One moment, I will try it |
| 14:15.37 | danielyk | If alaw is first priority, I can hear all sound files |
| 14:16.09 | danielyk | It is only if g722 has first priority |
| 14:16.42 | Samot | So file, would that still be related to the fix you were mentioning? |
| 14:18.08 | danielyk | I hear also all if only g722 is as allowed codec |
| 14:19.19 | igcewieling | sounds almost like you have G723 or G729 enabled. |
| 14:21.48 | danielyk | I do not use G723 or G729 and "core show translation" does not list them |
| 14:24.32 | danielyk | Samot, does this sounds like a bug? |
| 14:27.18 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:27.18 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:27.48 | Samot | Well it works when you use the standard english files? |
| 14:29.12 | danielyk | Yes, but they are saved as g722 and ulaw and sometimes sln and wav |
| 14:30.23 | igcewieling | those don't sound like the standard english files. |
| 14:31.23 | danielyk | These files in g722 and ulaw are in my directory sounds/en/ |
| 14:31.34 | danielyk | The standard files in sounds are in g722 and wav |
| 14:31.43 | danielyk | I do not use them... |
| 14:33.08 | Samot | Where did you get the german pack? |
| 14:34.13 | danielyk | I do not know any more where are they from, I have 3 different german packs for over 2 years now |
| 14:34.32 | igcewieling | What *exactly* is the problem? |
| 14:34.52 | Samot | If he has g722 and alaw offered in the SDP... |
| 14:35.11 | Samot | He doesn't hear the g722 files playback, even though they do. |
| 14:35.18 | Samot | Or more importantly... |
| 14:35.27 | Samot | He doesn't here any recording that isn't in .g722 |
| 14:35.41 | Samot | So it loads thanks.gsm but he can't hear it. |
| 14:35.41 | igcewieling | weird. never heard of that before. |
| 14:36.03 | Samot | But if he offers alaw, g722 then it works. |
| 14:36.12 | Samot | Or if he only offers g722 |
| 14:36.28 | igcewieling | older versions of Asterisk had some significant issues with transcoding g722, but not recently. I assume this is a recent asterisk. |
| 14:36.43 | Samot | Yes but who knows about his language files. |
| 14:36.45 | danielyk | @newtonr, have you got an idea? |
| 14:36.50 | Samot | They could be in the wrong format. |
| 14:36.58 | Samot | Did you ever think of that? |
| 14:37.03 | igcewieling | Samot: no, they could not or they would never work |
| 14:37.16 | Samot | Well the g722's... |
| 14:37.20 | Samot | And how would he know? |
| 14:37.38 | Samot | danielyk: Have the g722 files worked from those language packs? |
| 14:37.44 | igcewieling | Didn't you just say it worked if the DSP only offered g722? |
| 14:37.51 | igcewieling | SDP, of course. |
| 14:38.00 | Samot | According to him. |
| 14:38.19 | danielyk | yes if only g722 offered it works |
| 14:38.42 | igcewieling | what version of Asterisk |
| 14:38.52 | danielyk | Samot, from which language pack? The english one works and I do not have any german g722 |
| 14:39.10 | danielyk | I have also a alaw german language pack, should I try it? |
| 14:39.14 | danielyk | 13.12.1 |
| 14:39.55 | igcewieling | someone else will need to help you then. |
| 14:40.10 | Samot | danielyk: Do another call. |
| 14:40.17 | Samot | core set debug 0 |
| 14:40.28 | Samot | core set verbose 10 |
| 14:40.50 | Samot | I think what is also being failed to be mentioned here... |
| 14:40.52 | Samot | FreePBX. |
| 14:40.59 | danielyk | With the gsm files or with the alaw language pack? |
| 14:41.41 | Samot | Just how it is when it doesn't work. |
| 14:43.46 | danielyk | http://pastebin.com/XTVGhRVR |
| 14:44.26 | *** join/#asterisk Echo6_ (~Echo6@64.136.247.50) |
| 14:45.30 | danielyk | I have tested it now with another alaw language pack and another gsm lanuage pack. Same error. |
| 14:47.21 | Samot | <PROTECTED> |
| 14:47.34 | Samot | ^ That does not set the language for the rest of the call. |
| 14:49.23 | danielyk | What do you mean? |
| 14:50.31 | Samot | Nothing right now.. |
| 14:50.36 | Samot | I saw the rest of the call . |
| 14:50.46 | Samot | OK do the same thing with only g722 |
| 14:51.47 | igcewieling | hugs his Asterisk 11 servers |
| 14:53.09 | danielyk | http://pastebin.com/VXC6pesd |
| 14:54.43 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-hpippwkyuuyircag) |
| 14:54.43 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:55.02 | [TK]D-Fender | danielyk, what does that prove? |
| 14:55.07 | [TK]D-Fender | -- <PJSIP/0123456790543-00000095> Playing 'digits/2.gsm' (language 'de') |
| 14:55.07 | [TK]D-Fender | <PROTECTED> |
| 14:55.15 | [TK]D-Fender | I see it reading back in 'de" |
| 14:55.45 | danielyk | Yes, but I hear not these de sounds |
| 14:57.29 | [TK]D-Fender | Then you didn't install the DE sound files. |
| 14:57.56 | Samot | You said it worked when you just had g722, now you're saying it doesn't? |
| 14:58.38 | danielyk | MY MISTAKE: I heard them |
| 15:02.17 | danielyk | Should I open a bug? |
| 15:03.03 | [TK]D-Fender | You heard them? You DIDN'T hear them? make up your mind. |
| 15:04.37 | danielyk | Sorry for that mistake. I have not heard them if g722 (First) and alaw (Second) Codec was selected. If I select only g722, I hear them (last pastebin). |
| 15:19.36 | *** join/#asterisk matsk (~matsk@vpnbl.borderlight.net) |
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| 15:25.07 | danielyk | Samot, related to my first log, the soud files are transcoed from gsm to alaw. But the call is g722. Maybe the sound files are send as alaw to the trunk? |
| 15:34.20 | drmessano | Awful lot of language packs |
| 15:46.43 | Samot | Does the trunk support g722? |
| 16:00.20 | igcewieling | The asterisk CLI has a command to convert files between asterisk file formats. |
| 16:00.47 | igcewieling | oddly named "convert" |
| 16:03.19 | [TK]D-Fender | partially |
| 16:04.17 | igcewieling | Obviously it can't convert formats Asterisk doesn't support, such as stereo or higher bitrates files. |
| 16:18.21 | Akari | That's a great book. |
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| 16:32.09 | Akari | The ~book one |
| 16:32.16 | Akari | (I'm reading the backlog) |
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| 17:27.38 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 17:31.35 | hexanol | I've recently had a similar "issue" to the one described in https://issues.asterisk.org/jira/browse/ASTERISK-21752, i.e. accepting SIP INVITE from a peer which use multiple IP addresses, trying to match with a hostname for which there is multiple A records |
| 17:31.50 | hexanol | the solution using chan_sip was to define multiple peers, 1 for each IP address |
| 17:32.07 | hexanol | I was wondering if there was also this "limitation" using pjsip ? |
| 17:37.24 | Samot | No, I believe you can use multiple hosts. |
| 17:38.17 | *** join/#asterisk seiggy (~seiggy@74.203.105.194) |
| 17:45.36 | igcewieling | hexanol: multiple peers is how I used to handle peers with more than one IP. These days I allow guest calls and handle authentication in my AGI. |
| 17:46.17 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 17:46.18 | igcewieling | I've heard that one of the good things about pjsip is that is supports such things like multiple AORs per peer. |
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| 21:46.57 | ctcx | By chance, could someone help a bit in picking a good appliance with asterisknow installed? If it even exists.... |
| 21:47.10 | ctcx | Like for 50 extensions |
| 21:57.09 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fiphuifwavlsvvzg) |
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| 22:06.52 | ctcx | ??? |
| 22:34.58 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 22:34.58 | *** mode/#asterisk [+o file] by ChanServ |
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| 22:59.08 | *** mode/#asterisk [+o file] by ChanServ |
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| 23:36.08 | *** join/#asterisk Viktator (~kvirc@p5DDB23E7.dip0.t-ipconnect.de) |
| 23:36.21 | Viktator | Hi Y'all |
| 23:38.24 | file | hello |
| 23:40.03 | Viktator | my actual mental state is critical due to asterisk. |
| 23:40.31 | Akari | Join the club. |
| 23:41.09 | Viktator | my good ol' ISDN was canceled and now I have the worste VoIP provider imaginable - deutsche telekom |
| 23:42.39 | Viktator | Asterisk did a good job in saving my money as a callback solution. nothing more |
| 23:42.52 | Viktator | that worked fine with ISDN |
| 23:43.27 | Viktator | actually I was able to get asterisk to work/register/dial via a call-back file. |
| 23:43.49 | Viktator | But regardless my wasts of time I am stuck |
| 23:45.23 | Viktator | Situation is: My mobile calls asterisk, * drops the call and initiates a new call to my mobile. so far so good |
| 23:46.28 | Viktator | when I accept the incoming call from * I am getting a dial tone |
| 23:47.05 | Viktator | I tried Dial and disa, but I have no progress at all dialing with dtmf from my mobile |
| 23:47.58 | Viktator | any hints? |
| 23:48.29 | Viktator | I like to mention the concept of Karma, helping me will get you a much better afterlife....or not |
| 23:49.32 | mmlj4 | do you have "dtmfmode=inband" in your sip client definition in sip.conf? |
| 23:49.39 | Viktator | yep |
| 23:49.48 | mmlj4 | and I already have eternal life, but thanks |
| 23:50.18 | Viktator | keep in mind that our univers will collaps sooner or later..... |
| 23:50.58 | Viktator | what would be the command of choice, dial or disa? |
| 23:51.39 | mmlj4 | what are you trying to do? |
| 23:52.42 | mmlj4 | but my Lord's kingdom will never pass away |
| 23:53.30 | Viktator | final target is to be called from * to my mobile, * should than offer me the 2nd fixed line for free local calls. call-back |
| 23:56.05 | mmlj4 | you want disa |
| 23:56.57 | mmlj4 | or agi |
| 23:58.17 | Viktator | better again: my mobile calls my computer running *, who recognise my mobile number and drops the incoming call. |
| 23:58.56 | Viktator | step two: a call file is moved and * calls my mobile, I accept the call. |
| 23:59.54 | Viktator | up to this everything is fine |