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| 02:33.01 | kmyst | is anything special needed to pass callerid? i'm having a tough time making it do anything :/ |
| 02:36.27 | WIMPy | no |
| 02:37.04 | kmyst | huh, all i get is unknown or depending on whatever option i fiddle with pieces of what i set it to |
| 02:38.37 | WIMPy | From where to where? |
| 02:39.03 | WIMPy | And "pieces" sounds very strange. |
| 02:39.23 | kmyst | outbound asterisk sip call to my cell phone |
| 02:40.08 | WIMPy | Are you allowed to send caller ID? |
| 02:40.40 | kmyst | WIMPy: i tried Set(CALLERID(all)="Test User <1234567890>") as called myself and got (234) 567-890 |
| 02:40.52 | kmyst | yes |
| 02:41.42 | WIMPy | Assuming your mobile is located in NA, that looks ok to me. |
| 02:41.44 | *** join/#asterisk doop (~doop@colostomy.club) |
| 02:41.47 | WIMPy | What do you expect? |
| 02:43.04 | kmyst | wait maybe that wasn't right, i've tried so many combos i forget but i did get it to pass that much one time only |
| 02:43.54 | WIMPy | Make sure you use the expected format. |
| 02:44.20 | kmyst | ok what do you mean by that? |
| 02:44.56 | WIMPy | Send it the way your provider expects it. |
| 02:46.32 | kmyst | ok in that case i know that it's based on the following header fields in order of preference: P-Asserted-Identity, Remote-Party-ID, or From....i tried the first one and nothing and last one and nothing there as well |
| 02:55.15 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 02:59.19 | kmyst | whoa think that test was successful |
| 02:59.35 | kmyst | apparently it likes num not name |
| 02:59.35 | *** join/#asterisk robmal (r@wporzo.pl) |
| 03:02.28 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:05.17 | [TK]D-Fender | <kmyst> WIMPy: i tried Set(CALLERID(all)="Test User <1234567890>") <- wrong |
| 03:05.23 | [TK]D-Fender | quotes only goes around the name |
| 03:05.31 | [TK]D-Fender | <> goes around the number |
| 03:06.48 | kmyst | [TK]D-Fender: ah |
| 03:07.00 | kmyst | well it'll pass the number now but no name |
| 03:07.14 | [TK]D-Fender | Pass to where? |
| 03:07.18 | kmyst | my cell |
| 03:07.34 | [TK]D-Fender | You almost never get to pass a name to the PST at all. |
| 03:07.47 | [TK]D-Fender | And I'm not sure about your area, but our cells get no name period |
| 03:07.59 | [TK]D-Fender | unless you have them in the phone directory |
| 03:08.52 | kmyst | you know, good point...i just thought since i get robo calls showing up with both name and number on my home phone i could do similar |
| 03:16.39 | kmyst | ok so newb question but what's the cleanest/easiest way to set this once and not have to drop it in every exten prior ti Dial()? |
| 03:18.03 | WIMPy | Set it in the device definition. |
| 03:18.48 | kmyst | in the sip context? |
| 03:19.10 | WIMPy | yes |
| 03:21.03 | [TK]D-Fender | Use macros |
| 03:21.32 | [TK]D-Fender | And stop calling dial directly. Do the dial in the macro alnog with CID setup |
| 03:21.57 | kmyst | hrm |
| 03:22.26 | kmyst | [TK]D-Fender: just learning, been at it a few days :) |
| 03:23.03 | WIMPy | Macros are deprecated. Use Gosub instead. |
| 03:23.41 | [TK]D-Fender | WIMPy, in theory I suppose but they aren't going away it seems and I've never really had any consequence |
| 03:23.49 | [TK]D-Fender | WIMPy, but why not :) |
| 03:24.30 | kmyst | so to put it in sip.conf as a device definition is the format different or is it still Set(CALLERID....? |
| 03:25.01 | WIMPy | It's callerid= as shown in the example. |
| 03:25.31 | kmyst | example? |
| 03:25.54 | WIMPy | the sample sip.conf |
| 03:26.43 | [TK]D-Fender | I'd recommend just chaging your dialplan for this |
| 03:27.03 | [TK]D-Fender | and doing it in a macro and compact the rest of your dialing needs |
| 03:27.08 | [TK]D-Fender | (or gosub) |
| 03:28.11 | kmyst | [TK]D-Fender: agreed but gotta work up towards that, right now learning out of the book and wiki and using barebones files |
| 03:28.37 | *** join/#asterisk robmal (r@wporzo.pl) |
| 03:28.58 | kmyst | WIMPy: ah those, yeah i have those squirreled away, with all of them in /etc/asterisk it spewed a bunch of notices and errors all the time |
| 03:29.39 | [TK]D-Fender | kmyst, PB your dialplan and I'll give yo a quick sample |
| 03:29.51 | WIMPy | They are smples, meant for reference, not defaults ment to be used. |
| 03:30.37 | kmyst | WIMPy: just followed the install on using centos binaries that's what i wound up with installing asterisk-configs but yeah i hear ya that's why i did what i did |
| 03:32.24 | kmyst | [TK]D-Fender: http://pastebin.com/W8kuaKJY |
| 03:32.45 | kmyst | [TK]D-Fender: and yeah it looks newbish :) |
| 03:34.46 | WIMPy | The only thing I find interesting is that you used a variable for your area code, but hardcoded your country code. :-) |
| 03:35.05 | kmyst | laziness :) |
| 03:35.34 | kmyst | 1 is less typing than ${blah} |
| 03:35.34 | WIMPy | But only sometimes? |
| 03:36.03 | kmyst | right only sometimes, occasional method to my madness |
| 03:36.06 | WIMPy | 337 IS SHORTER THATN ${LOCAL} AS WELL. |
| 03:36.08 | WIMPy | oops |
| 03:36.33 | kmyst | good point, just wanted to do it |
| 03:37.36 | kmyst | so callerid= put into [flowroute] did nada guess i gotta put it into each device entry for phones? |
| 03:38.13 | WIMPy | Yes, it's for incomming calls. |
| 03:38.39 | WIMPy | But you can use templates. |
| 03:38.48 | kmyst | woot! |
| 03:39.02 | kmyst | yeah i did use it in template for phones |
| 03:39.08 | [TK]D-Fender | kmyst, http://pastebin.com/8PsTPugU |
| 03:39.20 | [TK]D-Fender | Merry Christmas |
| 03:40.33 | kmyst | [TK]D-Fender: ty :) |
| 03:41.19 | kmyst | lol |
| 03:41.30 | kmyst | that only took me two evening and 36 test calls |
| 03:43.57 | kmyst | ugh and now all my internal numbers show up the same |
| 03:48.33 | [TK]D-Fender | the same where? |
| 03:48.54 | [TK]D-Fender | were you looking for your phones to have THEIR outgoing calls look different? |
| 03:49.46 | kmyst | well i haven't used the macro thing yet i was referring to when i put callerid= into the template for phones in sip.conf |
| 03:50.18 | kmyst | and yeah i guess i was looking for if it's outgoing tag it with callerid i set else if it's internal don't |
| 03:53.03 | [TK]D-Fender | ok, show me that line you set. |
| 03:53.23 | [TK]D-Fender | (one of them) |
| 03:53.28 | [TK]D-Fender | since you did for multiple |
| 03:53.38 | kmyst | no just set one |
| 03:54.03 | [TK]D-Fender | pate it |
| 03:54.17 | kmyst | callerid=<my number> |
| 03:54.42 | kmyst | in a template for phones so all my phones inherit that |
| 03:55.53 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 03:56.28 | [TK]D-Fender | is that what you want? |
| 03:56.32 | [TK]D-Fender | oh wait... |
| 03:56.39 | [TK]D-Fender | that kills INTER PHONE CID too |
| 03:56.39 | kmyst | no |
| 03:56.44 | kmyst | yeah |
| 03:56.47 | [TK]D-Fender | would be nice to set it for OUTBOUND |
| 03:56.51 | [TK]D-Fender | So, do thisL |
| 03:56.52 | [TK]D-Fender | : |
| 03:57.27 | [TK]D-Fender | setvar=outboundcid="Test User" <1234567890> |
| 03:57.39 | [TK]D-Fender | in each for whatever you want their outbound to be. |
| 03:58.00 | [TK]D-Fender | you don't HAVE to do this but I'll give you a sample that takes advantage if you did |
| 03:58.18 | WIMPy | Yes, setvar is very hand, where available. |
| 03:59.26 | kmyst | can i set that once for all in the template? |
| 03:59.33 | [TK]D-Fender | you COULD |
| 03:59.38 | kmyst | but? |
| 03:59.46 | [TK]D-Fender | you won't have to. |
| 03:59.56 | [TK]D-Fender | watch my new sample |
| 03:59.57 | *** join/#asterisk rewzn (~rewzn@p5794B69B.dip0.t-ipconnect.de) |
| 04:01.24 | [TK]D-Fender | http://pastebin.com/UiVdgTcq |
| 04:04.18 | WIMPy | No need for a GotoIf if you use ExecIf. |
| 04:04.52 | kmyst | oh thats awesome |
| 04:05.20 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 04:06.45 | [TK]D-Fender | WIMPy, too lazy to look up the syntax, but yes, |
| 04:06.45 | [TK]D-Fender | could ahve been smaller |
| 04:06.54 | [TK]D-Fender | kyBut you should be getting the idea about to abstract your setup |
| 04:07.18 | kmyst | yes i am |
| 04:07.44 | kmyst | slowly :) |
| 04:10.32 | *** join/#asterisk protem (~protem@unaffiliated/protem) |
| 04:11.09 | kmyst | i have that definitive asterisk book, while very good and heavy in detail it is kind of lacking in certain examples of stuff like "set callerid" |
| 04:12.12 | kmyst | between that and the wiki thats how i got that dialplan going to the point i did |
| 04:16.43 | [TK]D-Fender | ~asteriskwiki |
| 04:16.43 | infobot | from memory, asteriskwiki is http://wiki.asterisk.org |
| 04:16.53 | [TK]D-Fender | SERIOULSY keep to this one. |
| 04:17.06 | kmyst | i am |
| 04:17.14 | kmyst | got like 5 tabs on it open :) |
| 04:17.19 | [TK]D-Fender | There is a decript old one whose information is dated and is often vague about what versions any given sample is for. |
| 04:17.23 | [TK]D-Fender | ~wikis |
| 04:17.23 | infobot | it has been said that wikis is VoIP Wiki covering FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners. For Asterisk, see ~asteriskwiki |
| 04:17.26 | [TK]D-Fender | This one |
| 04:17.31 | [TK]D-Fender | Tread lightly there |
| 04:17.49 | [TK]D-Fender | use it to supplement what you can't find elsewhere and take witha grain of salt |
| 04:18.01 | kmyst | yeah i've learned that too |
| 04:21.31 | kmyst | actually i've noticed a lot of what is out there is out of date relatively speaking |
| 04:22.02 | kmyst | i figured the book i got covering version 11 and i'm running 13 was a good jumping off point to learn with |
| 04:22.22 | [TK]D-Fender | yup |
| 04:24.02 | kmyst | but since i'm not concerned with voicemail, sql, etc. at this point in time i was like wtf trying to work out this callerid |
| 04:24.20 | kmyst | should probably spend a few nights reading those sample files |
| 04:27.00 | snadge | what determines whether a peer is lagged or unreachable? |
| 04:27.19 | snadge | i understand qualify = yes means 2000.. but apparently the lag time can go well beyond that.. is there a formula for that or something? |
| 04:27.56 | WIMPy | That IS the limit for lagged. |
| 04:28.05 | learath | argh phonepower! Why do you hate me! |
| 04:29.25 | bhans | How do I know if I configured my Asterisk properly in CentOS 6.6 |
| 04:29.37 | learath | Does it work? |
| 04:29.46 | WIMPy | By trying if it does what you want. |
| 04:30.09 | kmyst | hehe |
| 04:30.10 | bhans | I'm figuring out if it works or not :-/ I just followed how-to-install |
| 04:30.34 | bhans | But now, im hanging from testing if it works |
| 04:30.43 | bhans | How could I test this out? |
| 04:31.02 | kmyst | try the hello world example? |
| 04:31.24 | kmyst | which STUN tripped me up on that for an hour |
| 04:34.02 | *** join/#asterisk mbecroft (~user@ak2.becroft.co.nz) |
| 04:36.35 | [TK]D-Fender | <bhans> How could I test this out? <_ you use it. |
| 04:36.45 | [TK]D-Fender | bhans, There is no "test" things happen in REAL TIME |
| 04:36.53 | [TK]D-Fender | testing = doing |
| 04:37.25 | bhans | I don't know how to do it |
| 04:37.52 | bhans | That's why I am asking you.. I tried connecting to it using ZOIPER but it wont connect |
| 04:38.00 | bhans | What am I missing here? |
| 04:38.10 | [TK]D-Fender | Don't know how to do what? |
| 04:38.14 | [TK]D-Fender | You configured your server |
| 04:38.16 | [TK]D-Fender | PLACE CALLS |
| 04:38.22 | [TK]D-Fender | that is "using your server" |
| 04:38.34 | kmyst | bhans: i had to disabled STUN on zoiper for it to start working |
| 04:38.47 | [TK]D-Fender | if it won't connect... well something is wrong. What is the reason for "no connect"? |
| 04:38.58 | [TK]D-Fender | Are packets making it to your server? |
| 04:39.02 | bhans | I'll try that kmyst.. thanks! |
| 04:39.10 | [TK]D-Fender | Where are these 2 devices relative to one another? |
| 04:39.11 | bhans | [TK]D-Fender: I really don't know |
| 04:40.21 | kmyst | not saying that's the problem just know i had that issue one i started testing |
| 04:40.57 | [TK]D-Fender | [TK]D-Fender> Where are these 2 devices relative to one another? |
| 04:50.08 | bhans | [TK]D-Fender: can you give me a guide on how to setup this thing? I'll try to setup from scratch again. |
| 04:52.32 | [TK]D-Fender | I just asked you a question. |
| 04:52.32 | [TK]D-Fender | twice |
| 04:53.04 | [TK]D-Fender | Things have to match your circumstances. |
| 04:53.16 | [TK]D-Fender | So start from the beginning |
| 04:53.20 | bhans | What you mean by 2 devices? |
| 04:53.29 | bhans | My PCs? |
| 04:53.46 | bhans | relative to one another: they are in the same network |
| 04:54.25 | [TK]D-Fender | Server and Zoiper |
| 04:54.46 | bhans | im using windows for zoiper and installed asterisk to centos |
| 04:55.03 | bhans | which are in a different computers but in the same network |
| 04:55.27 | bhans | I'll just start from the beginning, please guide me |
| 04:56.33 | bhans | I have CentOS 6.6 |
| 04:56.40 | bhans | Any asterisk version is ok with this? |
| 04:58.39 | kmyst | [TK]D-Fender WIMPy: thanks for the help |
| 04:58.54 | kmyst | bedtime calls |
| 04:59.03 | *** part/#asterisk kmyst (~eric@74.193.224.215) |
| 05:00.16 | [TK]D-Fender | bhans, Any can run. |
| 05:00.23 | [TK]D-Fender | You'll want 11 or 13 only though |
| 05:00.28 | [TK]D-Fender | 13 preferred |
| 05:00.36 | [TK]D-Fender | but 11 will do |
| 05:00.41 | [TK]D-Fender | you already have this |
| 05:00.46 | [TK]D-Fender | So it's LOCAL LAN, right? |
| 05:00.53 | [TK]D-Fender | How did you install *? |
| 05:00.55 | [TK]D-Fender | Is it running? |
| 05:01.55 | bhans | I just uninstalled everything |
| 05:02.12 | bhans | and got a fresh CentOS too |
| 05:02.29 | bhans | So, we'll do * 13 |
| 05:02.49 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 05:04.21 | [TK]D-Fender | You trashed in these past 7 minutes? |
| 05:04.22 | [TK]D-Fender | eek |
| 05:09.58 | bhans | yes |
| 05:10.09 | bhans | i just ran back-up |
| 05:10.37 | bhans | Can we start now? |
| 05:12.55 | bhans | [TK]D-Fender: yuhoooo |
| 05:13.38 | bhans | do I need to install PJSIP too?> |
| 05:13.45 | [TK]D-Fender | sure just get things wher you thought they should be. |
| 05:13.54 | [TK]D-Fender | disable that for now |
| 05:13.56 | bhans | Okay |
| 05:14.05 | bhans | I have asterisk-13.3.2 |
| 05:14.25 | bhans | dahdi-linux-complete-2.10.2+2.10.2 |
| 05:14.30 | bhans | jansson-2.7 |
| 05:14.34 | bhans | libpri-1.4.15 |
| 05:14.55 | bhans | am I lacking something else or there's useless on that list? |
| 05:15.36 | [TK]D-Fender | looks fine |
| 05:15.41 | [TK]D-Fender | so get your configs in place |
| 05:15.54 | [TK]D-Fender | And verify that * is runing and your peers are loaded, etc |
| 05:16.02 | [TK]D-Fender | and check your FIREWALL on it |
| 05:16.09 | [TK]D-Fender | many stock settings may filter it |
| 05:16.14 | bhans | got it |
| 05:19.34 | bhans | configure is successful |
| 05:19.41 | bhans | ill do make now |
| 05:23.02 | bhans | I have error with 'PRI' |
| 05:23.20 | bhans | PRI dependency was previously satisfied but is now unsatisfied |
| 05:25.53 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 05:28.38 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 05:34.21 | bhans | [TK]D-Fender: installation complete |
| 05:34.26 | bhans | What to do next? |
| 05:37.35 | [TK]D-Fender | Setup your stuff |
| 05:37.41 | [TK]D-Fender | installing != configuring |
| 05:37.47 | [TK]D-Fender | Setup your zoiper peer |
| 05:37.51 | [TK]D-Fender | set up your dialplan. |
| 05:37.55 | [TK]D-Fender | start testing |
| 05:38.57 | bhans | wt |
| 05:39.39 | bhans | I was asking you how to set it up correctly |
| 05:39.48 | bhans | I don't even know how to do the account stuff |
| 05:40.42 | [TK]D-Fender | then there was nothing to test |
| 05:40.55 | [TK]D-Fender | Zoiper can't magically connect when you didn't set up an account for it to auth on |
| 05:41.13 | [TK]D-Fender | And it can't dial anything if you didn't configure a dialplan to process calls it will place |
| 05:41.20 | bhans | That's my point, I don't know where do I set the account |
| 05:41.35 | [TK]D-Fender | Sounds like you had expectations of this coming with something FUNCTION to test with |
| 05:41.40 | [TK]D-Fender | Toss that idea mostly out the door |
| 05:41.46 | bhans | You are not really helping. |
| 05:41.50 | [TK]D-Fender | And you need to learn how to configure * |
| 05:41.53 | [TK]D-Fender | Got that book? |
| 05:41.57 | bhans | You're just being an asshole. |
| 05:41.57 | [TK]D-Fender | You need to start reading NOW |
| 05:42.01 | bhans | bye asshole |
| 05:42.03 | *** part/#asterisk bhans (~briguer20@221.253.69.146) |
| 05:42.04 | [TK]D-Fender | No, thiws is important |
| 05:42.23 | *** join/#asterisk bhans (~briguer20@221.253.69.146) |
| 05:42.37 | [TK]D-Fender | Wow.... |
| 05:42.46 | [TK]D-Fender | I am not being an asshole. |
| 05:42.58 | [TK]D-Fender | there is nothing of any real usability to "test" in there |
| 05:43.27 | [TK]D-Fender | You need to understand those basics to set up your ssytem |
| 05:46.08 | bhans | nah.. I know you're better than this. You just aint helping me. |
| 05:46.25 | [TK]D-Fender | It isn't "help" if you're starting from zero |
| 05:46.42 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 05:46.44 | bhans | ignored* |
| 05:46.45 | [TK]D-Fender | it's me typing out full configs and you still not understanding any of it unless I type out al that explanation as well |
| 05:55.34 | [TK]D-Fender | Ah the joy of impatient entitlement..... |
| 05:59.49 | *** join/#asterisk tpdmoore (~tpdmoore@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 06:00.57 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 06:14.58 | *** join/#asterisk CeBe (~CeBe@port-92-200-209-159.dynamic.qsc.de) |
| 06:17.09 | *** join/#asterisk kayfox (~kayfox@orca.zerda.net) |
| 06:24.22 | ChannelZ | ICYMI: the domain nasty.pizza is available |
| 06:24.41 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 06:24.59 | [TK]D-Fender | Well that answers my silent suspicions on that ;) |
| 06:26.13 | ChannelZ | Hmmm.. some other good ones.. nasty.engineer |
| 06:27.22 | ChannelZ | jesus the number of TLDs are insane |
| 06:46.49 | *** join/#asterisk bulkorok (~b.tietz@89.245.151.228) |
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| 07:14.03 | MaliutaLap | I want the dont.care domain - so much potential for $$$'s selling subs |
| 07:14.14 | MaliutaLap | or are.cheap |
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| 07:53.18 | dadrc | "Contact me at mail@i.dont.care" <3 |
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| 08:19.34 | rl1 | Hi guys. Need yer help. |
| 08:19.35 | rl1 | [Jul 22 10:57:15] DEBUG[1333][C-00016522] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: False Text flag: False |
| 08:19.36 | rl1 | [Jul 22 10:57:15] DEBUG[1333][C-00016522] chan_sip.c: ** Our prefcodec: (g729) |
| 08:19.57 | rl1 | why does asterisk select g729 when I explicitly allowed only PCMA/PCMU? |
| 08:21.34 | rl1 | it doesn't even have support for g729, why the hell does it prefer g729 over g711 |
| 08:24.36 | *** join/#asterisk jetlag (~jetlag@pool-71-168-242-50.cmdnnj.east.verizon.net) |
| 08:29.08 | ChannelZ | What did the remote end request? |
| 08:35.46 | ChannelZ | actually.. |
| 08:38.48 | rl1 | ChannelZ, http://pastebin.com/LLVPWguH here's the sdp of the calling party's INVITE |
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| 08:47.10 | rl1 | ChannelZ, see, the remote end also supports PCMA |
| 08:47.31 | rl1 | now, why the hell, why the hell does asterisk prefer g729 over g711?? :( |
| 08:47.52 | rl1 | as it is not installed, asterisk drops the call afterwards |
| 08:48.05 | rl1 | saying that it could not translate the path |
| 08:51.24 | ChannelZ | I'm not actually sure what that debug is meaning. |
| 08:54.31 | rl1 | ChannelZ, as i understand it, asterisk is trying to use g729 with the calling party, but then, as the codec is not installed neither supported by the sip provider, it drops the call with " No path to translate from SIP/pstn-000055b3 to SIP/101-000055b2" |
| 08:55.07 | ChannelZ | in the sip debug, look for a line like "Capabilities: us - (ulaw|g723|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)" |
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| 08:59.56 | ChannelZ | or.. what 'allow'/'disallow' lines do you have in sip.conf, exactly? in [general] and the peer? |
| 09:03.38 | rl1 | ChannelZ, Capabilities: us - (ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) |
| 09:04.09 | rl1 | eer, the peer is not the same, so it doesn't have g729 in the capabilities |
| 09:04.20 | ChannelZ | yeah, but you do. |
| 09:05.01 | rl1 | disallow=all |
| 09:05.01 | rl1 | allow=alaw |
| 09:05.01 | rl1 | allow=ulaw |
| 09:05.44 | Chainsaw | would write that as allow=!all,alaw,ulaw |
| 09:06.01 | rl1 | i like to do it the old way |
| 09:07.25 | ChannelZ | I don't know why g729 is even listed there if that's true |
| 09:08.55 | rl1 | yeah |
| 09:09.20 | rl1 | sip show settings shows: |
| 09:09.21 | rl1 | Global Signalling Settings: |
| 09:09.21 | rl1 | --------------------------- |
| 09:09.21 | rl1 | <PROTECTED> |
| 09:09.21 | rl1 | <PROTECTED> |
| 09:09.23 | ChannelZ | what does 'sip show settings' show for Codec under the Global section? What about for 'sip show peer xxxx' |
| 09:10.28 | rl1 | in the sip show peer: |
| 09:10.28 | rl1 | <PROTECTED> |
| 09:10.28 | rl1 | <PROTECTED> |
| 09:10.44 | ChannelZ | yeah so.. fix that.. |
| 09:11.13 | rl1 | by explicitly setting allow/disallow on the peer? |
| 09:11.18 | ChannelZ | you're either using a template that has it in there that you're not realizing, or it's flat out set that way |
| 09:11.25 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 09:11.37 | ChannelZ | Or removing the allow/disallow for the specific peer so it uses the globals... whatevs |
| 09:14.21 | ChannelZ | bed time |
| 09:14.21 | rl1 | well, we use realtime asterisk with database |
| 09:14.44 | ChannelZ | I was going to ask that. Or if it was FreePBX |
| 09:15.17 | rl1 | nah it's pure asterisk |
| 09:16.00 | dan_j | rl1: As far as I'm aware, if both sides of the call are using g729 and asterisk doesn't have to re-encode the call, no g729 license is required. That may be a possible reason why it's trying to accept g729 and then failing when it discovers that the other side is not g729. |
| 09:16.32 | ChannelZ | Yeah but the capabilities of the peer don't say g729 which is the weird part |
| 09:16.52 | ChannelZ | >> Capabilities: us - (ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw|speex|ilbc) |
| 09:17.09 | ChannelZ | Out of curiosity what version of asterisk? |
| 09:17.17 | dan_j | From experience, asterisk isn't clever enough to work it out and then ends up dropping the call. |
| 09:17.24 | rl1 | 11.6 |
| 09:18.09 | dan_j | rl1: Find the column for that peer thats called disallow and put 'all'. Then find the column that says allow and put 'alaw,ulaw' (remove any reference to g729 from the allow column) |
| 09:18.48 | dan_j | Remember that in 'real time' the SIP Peers are cached so when you make that change you have to prune and reload that peer to get the new settings. |
| 09:19.32 | dan_j | sip prune realtime peer {SIP PEER NAME} |
| 09:19.55 | dan_j | sip show peer {SIP PEER NAME} load |
| 09:20.20 | dan_j | You will also lose the sip registration when you do that so you'll either need to wait for the sip phone to re-register, or restart the sip phone. |
| 09:21.47 | rl1 | Yaaaay |
| 09:21.52 | rl1 | thanks guys |
| 09:21.55 | rl1 | found the problem |
| 09:22.10 | rl1 | in the sippers table it says "allow g729, alaw, ulaw" |
| 09:22.20 | rl1 | how could i be so stupid |
| 09:30.33 | dan_j | I'm sure we've all been there once in our asterisk lives. Good luck. |
| 09:53.39 | esupra | HI, I have an issue where not even google seems to be my friend |
| 09:53.53 | esupra | [Jul 22 11:18:38] WARNING[6292] res_config_mysql.c: MySQL RealTime: Failed to update database: Lost connection to MySQL server during query |
| 09:53.53 | esupra | [Jul 22 11:18:38] ERROR[6292] res_config_mysql.c: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. |
| 09:54.07 | esupra | happens average 180x per day |
| 09:54.10 | esupra | asterisk 11.8 |
| 09:55.11 | dan_j | Check your mysql server is ok and check the connectivity to it is stable. |
| 09:55.19 | dan_j | Doesn't look like an asterisk issue |
| 09:56.01 | dan_j | Try switching to odbc and see if thats more stable. |
| 09:56.04 | dan_j | res_config_odbc |
| 09:56.26 | esupra | running cdr updates to main db server with odbc no issues |
| 09:56.43 | esupra | it is only when sip_regs do update that this happens every once in a while |
| 09:57.00 | dan_j | If odbc is working fine for cdr, then switch realtime to odbc too. |
| 09:57.48 | dan_j | I stopped using res_config_mysql a while ago, and moved to res_config_odbc. |
| 09:57.48 | esupra | will test and see if it helps |
| 09:57.51 | dan_j | I cant remember why |
| 09:58.24 | dan_j | I think res_config_mysql is deprecated. |
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| 09:59.08 | esupra | think so too, this was installed by somebody else though, ubuntu apt-get, so I think it might be issue with the ubuntu version |
| 09:59.36 | esupra | compiled 13 from source, but it has a bug in it that crashes on blank channel transfer on libc-2.19.so |
| 10:00.08 | esupra | so that dont help much, since it runs on clustered with around 6000 extensions on it |
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| 10:00.18 | esupra | but will try odbc and see what it does tonight |
| 10:00.20 | esupra | thanx |
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| 12:09.12 | DivideBy0 | esupra: I know I'm late to the party, but that reconnect message is because your mysql timeout is very low. Look for "wait_timeout" in my.cnf |
| 12:09.36 | DivideBy0 | a db admin probably lowered it to expire stale connections, but went too low |
| 12:09.42 | esupra | sweet, thanx, seen it, was set to 10seconds |
| 12:09.57 | DivideBy0 | so that means every connection will timeout in 10 seconds of inactivity |
| 12:10.09 | esupra | but with 6000 registrations happening every 2 min, it was bound to cause hickups, |
| 12:10.13 | DivideBy0 | which sounds sorta-right with 6000 extensions, but |
| 12:10.14 | DivideBy0 | ... |
| 12:10.15 | DivideBy0 | yeah |
| 12:10.16 | esupra | thanx, will up it tonight and restart the sql |
| 12:10.29 | esupra | thanx for help |
| 12:10.31 | DivideBy0 | I *think* you can change it on the fly with a set global |
| 12:11.28 | DivideBy0 | yeah, you should be able to change it without a restart |
| 12:11.43 | DivideBy0 | but remember, if it works, to change it in my.cnf or you'll lose it on restart (I've done this too many times) |
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| 12:36.22 | sekil | 6k regs on asterisk.. |
| 12:36.23 | sekil | nice |
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| 12:57.21 | Rico | hi |
| 12:57.30 | Rico | I have problem with incoming calls on asterisk 13 (pjsip) |
| 12:57.39 | Rico | my pjsip.conf looks like this : http://pastebin.com/zDk5KS3w |
| 12:57.44 | Rico | sip trunk is registered |
| 12:57.55 | Rico | when doing an incoming call by this trunk I have this error message : |
| 12:58.40 | Rico | [Jul 22 13:40:35] NOTICE[1676]: res_pjsip/pjsip_distributor.c:256 log_unidentified_request: Request from '"067697abcd" <sip:067697abcd@x.x.x.x>' failed for 'y.y.y.y:5060' (callid: 7e5c-466-6222015114035-JANGO-1-x.x.x.x) - No matching endpoint found |
| 13:02.54 | [TK]D-Fender | Rico: I see no AOR in your config |
| 13:03.02 | Rico | [TK]D-Fender: I'm fixing it |
| 13:03.08 | Rico | (trying to) |
| 13:04.05 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:05.33 | Rico | (just broke my register) :D |
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| 13:06.52 | Rico | seems that asterisk don't like if auth section has name identifier than registration section... strange |
| 13:07.14 | file | Asterisk doesn't care, many endpoints do |
| 13:07.47 | file | your underlying problem is that there is nothing in your PJSIP configuration to match the above request |
| 13:08.00 | file | ah you're using line support |
| 13:08.04 | Rico | just changing [1004] to [1004_auth] (and outbound_auth=1004 to outbound_auth=1004_auth) fixed the register problem |
| 13:08.10 | file | what is the request? |
| 13:08.21 | [TK]D-Fender | Rico: You also seem to have registration settings under the endpoint and not in a proper registration section |
| 13:08.33 | file | yes, that too |
| 13:10.17 | Rico | [TK]D-Fender: I've clean it, let me pastebin the new one |
| 13:12.22 | Rico | [TK]D-Fender, file : http://pastebin.com/KPBNj9jM |
| 13:12.45 | file | with that configuration there is nothing to match that incoming request to an endpoint |
| 13:13.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:15.08 | file | you will either need to create a match section which matches based on source IP address, or use line support which may work (depends on how the remote server behaves) |
| 13:15.45 | Rico | file: ok, I'll take a look at that, thanks |
| 13:16.29 | *** join/#asterisk davlefouAMD (~davlefouA@41.225.44.104) |
| 13:18.24 | Rico | looking for doc about "line" |
| 13:19.06 | Rico | "If you would like to enable line support and have incoming calls related to this registration go to an endpoint automatically the "line" and "endpoint" options must be set. |
| 13:19.11 | *** join/#asterisk quintana (~sylvain@avencall-gw-vbo.alionis.net) |
| 13:27.50 | [TK]D-Fender | contact=sip:my.public_ip_addr:5060 |
| 13:27.57 | [TK]D-Fender | that should be THEM, not YOU iirc... |
| 13:33.21 | rl1 | ehm... How do I turn on verbosity in logs on 11 asterisk? |
| 13:33.48 | rl1 | there now seems to be two verbosity settings which are Root console verbosity, Current console verbosity |
| 13:34.05 | ganbold | core set debug on, core set verbose on |
| 13:34.06 | rl1 | in the old version (like 1.6) there was only one verbose setting |
| 13:35.00 | rl1 | ganbold, core set verbose on is an invalid command |
| 13:35.21 | ganbold | well I meant similar command |
| 13:35.53 | ganbold | or you can maybe run asterisk like -gcvvvvvv |
| 13:36.32 | rl1 | it's already running, so i can't run it with -c |
| 13:37.21 | rl1 | if i core set verbosity to any level, it only applies to the current console and not for the logs |
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| 13:37.54 | rl1 | DEBUG NOTICE WARNING ERROR VERBOSE DTMF FAX i have this in logger.conf |
| 13:38.14 | rl1 | debug is working but verbose not :( |
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| 13:45.35 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:46.48 | rl1 | should i set verbosity on the root console to make it apply for the logger? |
| 13:47.46 | ganbold | maybe, I didn't test much |
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| 14:12.02 | Rico | file: do you have doc about 'line" and 'match' options in pjsip ? |
| 14:12.16 | file | um it would be on the wiki |
| 14:12.54 | file | I know match is part of the example configuration on the wiki for doing ITSP connection |
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| 14:24.58 | Rico | file: can you explain me that : http://pastebin.com/2k89aAm9 |
| 14:25.09 | Rico | -> why is auth 'trunk_sbc' not found ? |
| 14:25.14 | Rico | can't understand |
| 14:25.45 | file | you generally don't want to authenticate calls from an ITSP because they won't authenticate |
| 14:26.00 | file | and I don't know - look at your console logs at startup and it will say why |
| 14:26.10 | Rico | no but I want to authenticate myself to my trunk provider |
| 14:26.24 | file | ah |
| 14:26.34 | file | well, look at your console output at startup |
| 14:27.39 | file | and have you tried following the example for this? |
| 14:27.57 | Rico | [Jul 22 16:27:07] WARNING[1353] config_options.c: Cannot update type 'identify' in module 'res_pjsip' because it has no existing documentation! |
| 14:28.35 | Rico | file: yes I have, https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples and https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip |
| 14:28.45 | file | your installed documentation doesn't match your installed set of modules |
| 14:29.10 | Rico | what what what ? |
| 14:30.23 | Rico | fresh install, only install one version of asterisk, which is the one running |
| 14:30.55 | file | ponders |
| 14:31.01 | file | let me put your config in mine... |
| 14:31.09 | file | what is the complete console output at startup? |
| 14:31.17 | Rico | let me pastebin it to you |
| 14:31.35 | Rico | I'm redoing aé "make all && make installé |
| 14:31.42 | Rico | s/é/"/ |
| 14:31.52 | Rico | slaps infobot |
| 14:34.01 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:34.19 | file | I just put your entire posted config in my Asterisk and it loaded fine |
| 14:34.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:34.59 | file | except registration, let's see... |
| 14:35.10 | Rico | huh... |
| 14:35.13 | Rico | yes |
| 14:38.03 | file | oh, lack of transport |
| 14:38.06 | file | otherwise it loads fine |
| 14:38.14 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-wrbdznidhbxkcvvi) |
| 14:39.08 | Rico | file: I did not put transport in pastebin but it is present in my conf file |
| 14:39.31 | file | do you have the full console output yet? |
| 14:39.52 | Rico | see MP |
| 14:40.09 | file | I don't accept private messages |
| 14:40.21 | Rico | do you accept notices ? |
| 14:40.36 | file | no - any help I give is public |
| 14:40.58 | Rico | I would prefer to not pastebin all the logs in public chan because of public IP addresses etc... |
| 14:41.16 | file | they can be removed |
| 14:43.00 | Rico | file: here it is: http://pastebin.com/ZaeauYq0 |
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| 14:44.47 | learath | anyone rolled centos7 rpms? |
| 14:44.54 | file | what does "pjsip show auths" show? |
| 14:46.14 | learath | anyone familiar with PhonePower and Asterisk 11.7.0? |
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| 14:47.33 | Rico | file: Auth: 1004_auth/1004 |
| 14:47.44 | file | erm |
| 14:48.22 | file | what else did you leave out of the pastebin? |
| 14:48.49 | Rico | mmh the things I have in my config does not match the things CLI is showing... restarting |
| 14:49.46 | Rico | file: nothing |
| 14:49.56 | Rico | just changed ip addresses and hostnames |
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| 14:52.51 | Rico | file: got my trunk registered back, but still can't have incoming calls |
| 14:52.57 | Rico | let me make another pastebin |
| 14:53.09 | Rico | with everything inside |
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| 14:59.22 | Rico | file: http://pastebin.com/ddaMXXPX |
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| 15:00.46 | Rico | file: and logfile : http://pastebin.com/Xf2q1khj |
| 15:01.32 | ak77_ | how to debug locking issues ? (asterisk-chan-dongle locks up) |
| 15:01.48 | file | if your documentation (which is located at /var/lib/asterisk/documentation/core-en_US.xml) does not include documentation for the "identify" type then the configuration won't be read in, that seems to be what is happening |
| 15:02.00 | file | that is done as part of the build process - it's extracted from everything |
| 15:02.14 | file | why it didn't do it for you, I'm not sure, but it's likely environment specific |
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| 15:03.03 | yogg | Hi |
| 15:03.19 | *** join/#asterisk boch (~boch@186.182.125.111) |
| 15:03.22 | boch | hi all |
| 15:05.24 | yogg | I have a strange problem with asterisk "1.8.10.0". I have created some debug outputs. One for the callerid: "### Callerid all: "012345678901" <0012345678901>". But in the sip debug I see this: "From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as686af854" |
| 15:06.22 | yogg | Has someone an idea why asterisk sets anonymous when the caller id is set to an number? |
| 15:06.56 | WIMPy | Because the presentation was set to inhibit? |
| 15:07.32 | WIMPy | Can't remember if 1.8 already used CALLERID(pres) or the old way. |
| 15:07.39 | Rico | file: think I'm getting closer : http://pastebin.com/qNFr2qRC |
| 15:07.49 | Rico | Command 'module load res_pjsip_endpoint_identifier_ip' failed. |
| 15:08.14 | *** part/#asterisk aandrew (foobar@gromit.mixdown.ca) |
| 15:08.25 | yogg | WIMPy: pres is set to "### Callerid num-pres: prohib_not_screened" |
| 15:08.41 | file | Rico, you have to restart |
| 15:08.47 | WIMPy | So that's the explanation. |
| 15:09.51 | Rico | file: Ive done it about 150 times |
| 15:10.09 | Rico | maybe I have to force load in asterisk.conf |
| 15:11.49 | Rico | file : res_pjsip_endpoint_identifier_ip.so PJSIP IP endpoint identifier 0 Not Running core |
| 15:12.26 | yogg | WIMPy: I have this on all my asterisk boxes. Normaly this works without an problem. What else shoud I set here? |
| 15:13.13 | WIMPy | What do you mean? It works as expeced. |
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| 15:17.33 | yogg | WIMPy: ok I think I don't understand this option right. I have to set the "Callerid num-pres" to allowed? |
| 15:18.32 | WIMPy | Yes. "prohib" means it must not be displayed. |
| 15:19.37 | yogg | WIMPy: ok thats strange. I have a second box with exact the same options set and here the number is displayed on the phone. But thanks I will try that |
| 15:20.40 | WIMPy | Where does the caller ID come from? Does your phone set the flag? |
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| 15:25.00 | yogg | No I get an call from external. On the external side I see that "from" is set to the right value. But If the call goes to the internal phone the "from" header is set to anonymous. But I curently try to set "Set(CALLERID(num-prob)=allowed;" |
| 15:26.06 | WIMPy | num-pres, I hope. |
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| 15:35.36 | yogg | WIMPy: typo on my side: Debug: "### Callerid num-prob: allowed" -> From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as24773951 Something is strange |
| 15:36.53 | yogg | and I hve the typo also in my debug messages -.- |
| 15:40.31 | yogg | WIMPy: thx found it. There is "num-prob" and "prob". I have now both on allowed and it seems to work now. |
| 15:40.59 | WIMPy | pres |
| 15:41.32 | yogg | what the hack is wrong with my fingers -.- |
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| 15:57.22 | learath | argh. anyone know how to fix "SIP/2.0 403 From: URI not recognized" from Phonepower? |
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| 16:14.46 | Rico | file: here ? |
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| 16:34.57 | WIMPy | is in a hacking mood today. What do I break next? |
| 16:40.18 | learath | well. that sucks |
| 16:40.30 | learath | WIMPy: want to figure out why the Android default sip works fine, while Asterisk fails? :P |
| 16:42.59 | WIMPy | No. I consider failure the normal behaviour for SIP. |
| 16:43.09 | learath | lol :) it's not that bad. |
| 16:43.35 | WIMPy | Yes, but that's pure chance. |
| 16:44.06 | learath | the android client worked perfectly first try :( |
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| 18:13.40 | learath | hmm. how can I suppress tags in sip registration? |
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| 18:15.24 | janicez | =/win 89 |
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| 19:23.35 | mbowie | Hey folks... appreciate that this seemingly isn't an * thing, just hoping someone here might have come across something similar... we have a volume of Cisco 79x1 running SIP firmware, which have started generating static / crackle during calls. There is no disruption or visible issue on the wire (the stream sounds perfect in playback) so it seems to be with the handsets themselves. |
| 19:25.24 | mbowie | I don't believe we ever had reports of it under 1.8, but don't see why it would just occur on the device if it's not in the RTP stream. Until this began, we were quite happily using the 8.5(4) release, and have tested all the way to 9.4(2SR1), and it persists. |
| 19:25.33 | learath | Can I disable sip tags? (<sip:NUMBERS@sip.phonepower.com>;tag=as7fd2768d) |
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| 20:17.07 | malcolmd | mbowie: we once had a similar audio problem with polycom conference phones that had older power supplies. the supplies had degraded and began affecting the phones audio. |
| 20:28.20 | DivideBy0 | software guy blames the hardware :) |
| 20:28.27 | learath | Is there a recommend sip trunk provider? |
| 20:30.45 | DivideBy0 | learath: I use flowroute over fios like you. I've seen other people here recommend sipstation |
| 20:32.02 | learath | 26$/mo seems high |
| 20:35.05 | bkruse | Currently use flowroute - support has gone downhill, but it's basically level 3 - no real problems with them :) |
| 20:36.11 | DivideBy0 | I pay by the minute. I've never actually tried to figure out what would be cheaper - channels vs minutes |
| 20:36.43 | DivideBy0 | we're really bursty for a few hours a day, so I was nervous about limiting that by # of channels |
| 20:41.24 | mbowie | malcolmd: Good thinking... will investigate that angle! These are all on campus PoE switches, with plenty of load available... but that doesn't mean the phone side rectifier isn't aging poorly. Good tip! :-) |
| 20:43.21 | DivideBy0 | hmm, looks like I'm a complete idiot for paying by the minute |
| 20:45.46 | mbowie | DivideBy0: When it comes to idiocy, I'd say that to be termed as complete would be ultimate achievement. ;-) |
| 20:46.09 | DivideBy0 | funny! |
| 20:46.24 | eschmidbauer | it is level3 |
| 20:46.38 | eschmidbauer | (flowroute) the IPs in the SDP are Level3's |
| 20:46.40 | DivideBy0 | actually, those virtual-pris are only for inbound, nevermind. I'm back down to only partial idiocy |
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| 21:28.07 | garym_000 | Hello #asterisk, I could really use some help here. I have an asterisk/freepbx system in a small medical office that's been going for about 5 months, they are losing calls after they are placed on hold and I can not figure out why. There is just too much stuff going on in the asterisk console with verbosity set to 1 to even keep track. How can I debug this issue? |
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| 21:36.56 | ChannelZ | #freepbx |
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| 21:53.54 | bkruse | eschmidbauer: not exactly level 3 - as the problems I have had have been when flowroute's primary switch goes down, but level 3 is still up - "like" level 3 |
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| 22:06.31 | RobertLaptop | Evening I am attempting to install Asterisk 13 on Centos 6.6 but I am getting a package conflict pjproject-2.3-0.digium3.1_centos6.x86_64 & pjproject-2.3-5.el6.x86_64 |
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| 22:26.19 | learath | flowroute active and tested, nice. |
| 22:26.27 | learath | hopefully not horribly more expensive. |
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| 23:34.37 | axisys | I have a OpenVox A400P 4-port analog card and I compiled dahdi-linux-complete-2.10.0.1+2.10.0.1 and seeing tons of this error message real fast! |
| 23:34.40 | axisys | [1108507.468857] TDM PCI Master abort |
| 23:34.48 | axisys | any suggestion what I am doing wrong? |
| 23:36.06 | WIMPy | Analog is evil! |
| 23:36.28 | WIMPy | Looks lik a PCI issue to me. |
| 23:37.08 | axisys | I am just hobbist .. playing with this cheap openvox card on cheap old ibm hardware.. lspci -vvvv shows it sees the card |
| 23:38.06 | WIMPy | cheap??? |
| 23:38.51 | axisys | I am on ubuntu 14.04.2 LTS running linux kernel 3.13.0-57-generic |
| 23:38.57 | axisys | WIMPy: in comparison |
| 23:40.43 | WIMPy | Could be a BIOS thing. Do you have something else to test with? |
| 23:41.29 | axisys | WIMPy: no |
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| 23:44.49 | axisys | when I stop dahdi .. no more of TDM PCI Master abort |
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| 23:51.46 | axisys | according to lspci -vb this card is sharing IRQ with the audio controller and usb controller |
| 23:51.50 | axisys | yikes |
| 23:52.39 | WIMPy | Try another slot. |
| 23:56.57 | axisys | I am trying to see if I can disable audio and usb controllers as I dont use them with this server |
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