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| 01:44.28 | saint_ | hi all |
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| 02:07.50 | ChannelZ | ahoyhoy |
| 02:14.28 | [TK]D-Fender | chips |
| 02:18.39 | *** join/#asterisk deo_ (~deo@203.177.214.75) |
| 02:18.45 | carrar | California Highway Patrol? |
| 02:19.45 | carrar | Ponch and Joh WTF |
| 02:19.54 | carrar | s/joh/john/ |
| 02:22.47 | saint_ | is this correct: GotoIf($[${DIALSTATUS} = DONTCALL]?xxx) ..? |
| 02:23.07 | saint_ | the ${DIALSTATUS} = DONTCALL , is this a correct test to test the value ? |
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| 02:35.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:35.49 | [TK]D-Fender | saint_: logically yes, except that is a reserved variable and that is not a value it should ever hold |
| 02:36.41 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 02:38.56 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:41.09 | linocisco | hi all |
| 02:41.57 | linocisco | how have done configuration with Grandstream HT702 + asterisk? |
| 02:42.04 | linocisco | who have done configuration with Grandstream HT702 + asterisk? |
| 02:48.29 | saint_ | anyone would have an example of torture menue ? |
| 02:48.36 | saint_ | s/menue/menu |
| 02:49.07 | saint_ | never mind |
| 02:50.14 | [TK]D-Fender | 10 PRINT "I AM GOING INSANE!!!" |
| 02:50.18 | [TK]D-Fender | 20 GOTO 10 |
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| 03:12.21 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 03:15.53 | linocisco | I am testing asterisk on QNAP with Grandstream HandyTone HT702. but call from one phone to another is with much delay. it should be quick dial as it is calling one extension to another. how could I configure? |
| 03:16.22 | linocisco | I am testing asterisk on QNAP with Grandstream HandyTone HT702. but call from one phone to another is with much delay. it should be quick dial as it is calling one extension to another internally. how could I configure? |
| 03:16.51 | [TK]D-Fender | is the VOICE delayed, or is it slow to acknowledge what you DIAL? |
| 03:17.33 | linocisco | [TK]D-Fender, actually. dialing tone is delayed to hear in my ear. |
| 03:17.56 | [TK]D-Fender | LinSo you pick up the phone and it takes too long to start hearing the tone? |
| 03:17.58 | linocisco | [TK]D-Fender, actually. I feel like calling overseas. |
| 03:18.06 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 03:18.08 | [TK]D-Fender | before you dial? |
| 03:18.09 | linocisco | [TK]D-Fender, sure |
| 03:18.24 | [TK]D-Fender | linocisco: Then check your HT... it is what generates the tone |
| 03:18.26 | linocisco | [TK]D-Fender, no after dial |
| 03:18.41 | [TK]D-Fender | After dial delay is ALSO the HT's doing |
| 03:18.59 | [TK]D-Fender | Then there is whatever gts added by how you dial out from there |
| 03:19.37 | linocisco | [TK]D-Fender, I dont understand |
| 03:20.03 | linocisco | [TK]D-Fender, I dont know exactly what to look at either on HT or asterisk |
| 03:22.27 | linocisco | [TK]D-Fender, I dont know how to check. There are so many parameters or settings/options on HT's web GUI |
| 03:24.33 | [TK]D-Fender | look at when * actually starts processing a call. All that time is the HT's doing |
| 03:24.35 | *** join/#asterisk bytemaster (~ewrewr@host81-150-217-168.in-addr.btopenworld.com) |
| 03:24.49 | [TK]D-Fender | Then lok at the time from the start of * actually processing and calling out. |
| 03:25.00 | [TK]D-Fender | Which you didn't tell us what it's doing |
| 03:27.42 | linocisco | [TK]D-Fender, what is "*" |
| 03:28.03 | [TK]D-Fender | * = a symbol you should know on your keyboard. |
| 03:28.08 | [TK]D-Fender | Do you know how to say it? |
| 03:28.19 | [TK]D-Fender | Asterisk <---- |
| 03:29.01 | [TK]D-Fender | Which is why it's their logo |
| 03:29.04 | linocisco | [TK]D-Fender, I am using asterisk system on WebGUI. |
| 03:30.15 | linocisco | [TK]D-Fender, asterisk system on one settop box calle QNAP. I have not check on Linux CLI of asterisk commands because I dont know asterisk commands. that is why I used only WebGUI like freepbx or asteriskNow |
| 03:37.11 | [TK]D-Fender | linocisco: You will need to learn to watch * CLI |
| 03:37.15 | [TK]D-Fender | ~book |
| 03:37.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:37.23 | [TK]D-Fender | asterisk -rvvvvvvvvvvvv |
| 03:38.25 | linocisco | [TK]D-Fender, to just know one thing, need to read the whole book start to end |
| 03:38.27 | linocisco | ? |
| 03:39.23 | [TK]D-Fender | Have you read the table of contents? |
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| 03:58.00 | linocisco | [TK]D-Fender, I dont even know what to look at. meaning what causes what problems. |
| 03:59.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 04:00.22 | tompaw | chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. << what does this mean? |
| 04:03.30 | [TK]D-Fender | Or he could just leave without answering a simple yes/no question... |
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| 04:05.08 | tompaw | http://pastebin.com/2FVf7pWV |
| 04:05.19 | tompaw | Gents, that's something that turned my day today into a disaster. |
| 04:05.32 | tompaw | Any idea what could be causing this behavior? Google doesn't help much. |
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| 04:10.55 | tompaw | chan_sip.c: Probably a DNS error for registration to 772032@, << why isn't it showing any domain name / ip address? |
| 04:23.49 | [TK]D-Fender | maybe you should look at your register line.... |
| 04:25.50 | tompaw | [TK]D-Fender: I don't have a register line... out of a sudden asterisk started acting strange today and now it's trying to register all of my realtime peers... for no reason! |
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| 04:28.05 | tompaw | [TK]D-Fender: what do you mean by "register line"? |
| 04:29.57 | tompaw | [TK]D-Fender: ? |
| 04:34.22 | tompaw | [TK]D-Fender: ? |
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| 04:43.06 | tompaw | Anyone - any ideas why asterisk tries to REGISTER all of my sippeers? |
| 04:45.50 | [TK]D-Fender | tompaw: You're not even loking at the full sip comm... |
| 04:45.53 | [TK]D-Fender | looking* |
| 04:47.34 | tompaw | [TK]D-Fender: there is no SIP message. Asterisk stubbornly tries to initiate the REGISTER all by itself. Even if I kill all the network interfaces. |
| 04:48.01 | tompaw | IT started doing it today on 11.0, after few months of no problem. I updated to 11.2, but it's still doing it. |
| 04:49.07 | tompaw | If I disable realtime sippeers, it shuts up and restores normal operation, but I don't feel like moving thousands of users to a .conf file. |
| 04:49.29 | [TK]D-Fender | tompaw: are you going to show us your configs at some point? |
| 04:49.58 | tompaw | [TK]D-Fender: which one? |
| 04:51.04 | [TK]D-Fender | ... |
| 04:51.19 | tompaw | http://pastebin.com/9GDa5riC |
| 04:52.43 | tompaw | http://pastebin.com/bFbn0mzU << added sip.conf |
| 04:53.37 | tompaw | What can cause this auto-registration? I've never ever experienced this before... |
| 04:53.47 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:54.24 | tompaw | IT seems like asterisk goes through each sippeer, one by one, and tries to register them as "1234@", where 1234 is the actual peer's name. |
| 04:57.49 | tompaw | And those ghost REGISTERs appear in debug as: |
| 04:57.50 | tompaw | [Feb 18 23:56:48] DEBUG[28099] chan_sip.c: Allocating new SIP dialog for 703889cd2a467d1265b84d6941a093b9@(null) - REGISTER (No RTP) |
| 04:58.28 | tompaw | But there's no actual SIP traffic at all, this is all Asterisk talking to itself... |
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| 05:02.40 | tompaw | Actually, it seems like Asterisk is trying to use defaultuser during registration. |
| 05:02.46 | tompaw | But the question remains - WHY? |
| 05:10.58 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 05:11.20 | tompaw | Is it a hardware failure or a bug in asterisk? |
| 05:12.22 | coreyf1513 | tompaw: what version did you upgrade from? |
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| 05:12.48 | tompaw | coreyf1513: 11.0 to 11.2, but this actually started to happen this morning at 11.0 |
| 05:13.10 | tompaw | Since then I was banging my head against the wall, because google mentions NOTHING about this issue. |
| 05:13.25 | coreyf1513 | what version worked correctly? |
| 05:13.30 | tompaw | If I disable realtime sippeers, it goes back to normal. |
| 05:13.37 | tompaw | coreyf1513: 11.0, for a few months. |
| 05:13.51 | tompaw | Then we had a power outage last night, I restarted the server this morning to find this... |
| 05:14.55 | tompaw | Could it be, than a presence / a lack of presence of a certain column (field) in sippeers is causing this? |
| 05:15.06 | coreyf1513 | tompaw: i would suggest doing a backup of your database, check for corruption |
| 05:15.42 | tompaw | coreyf1513: it's a simple 1 table, I actually logged the sql queries and they run perfectly normal. |
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| 05:15.56 | coreyf1513 | tompaw: sorry i don't use realtime so I don't have any great suggestions, but I don't see anything in your sip.conf. |
| 05:16.19 | tompaw | one sec, let me add my sql to the picture |
| 05:17.30 | tompaw | http://pastebin.com/X4JStnZM << there |
| 05:17.51 | tompaw | could it be one of those fields forces asterisk to try and to the stupid register? |
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| 05:19.28 | coreyf1513 | tompaw: from cli if you run: 'sip show channels' does it show the dialog for the invalid register? |
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| 05:20.05 | tompaw | coreyf1513: nope, and sip debug doesn't show them, either |
| 05:20.54 | coreyf1513 | i'm guessing the issue isn't your schema if that has been working, more likely a value is wrong |
| 05:21.18 | tompaw | a value? like in a single row? |
| 05:21.44 | coreyf1513 | have you tried testing with a small sample of peers (backup and clear the table, then restore 10 users) |
| 05:22.01 | tompaw | not really, but that sounds like a good idea. |
| 05:22.04 | tompaw | I will try that, thanks. |
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| 05:22.26 | coreyf1513 | i'm guessing a value got weird when your server shutdown improperly |
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| 05:23.32 | tompaw | coreyf1513: this is something that actually does make sense, vs everything else I was considering today. |
| 05:23.49 | coreyf1513 | when you find the row that caused this you might want to report a bug.. in that case your db would need to be corrected, but asterisk should catch the problem before trying to create an invalid register |
| 05:23.51 | tompaw | OK, need a quick nap and will return my findings first thing tomorrow. |
| 05:24.07 | tompaw | yep, will definitely do it. |
| 05:24.25 | tompaw | (db was not affected by the power outage, tho, as it's 10 000 km away) |
| 05:26.25 | coreyf1513 | could still be an unusable combination of values at time of outage.. seems crazy using a remote realtime db, but that's just my opinion |
| 05:29.05 | tompaw | coreyf1513: this is only for peer registrations, that's hardly any sql traffic, and the db is a part of a bigger crm. |
| 05:34.26 | lorsungcu | tompaw: still there |
| 05:34.27 | lorsungcu | ? |
| 05:35.13 | tompaw | yes |
| 05:35.16 | tompaw | coreyf1513: bingo |
| 05:35.37 | tompaw | it is one of the rows... it's gonna be interesting tomorrow morning to pinpoint which. |
| 05:37.19 | tompaw | but whoa |
| 05:37.22 | tompaw | asterisk is still doing it |
| 05:37.33 | tompaw | it tries to send a REGISTER to my softphone ip... |
| 05:37.42 | tompaw | as per fullcontact field |
| 05:37.43 | tompaw | wtf???? |
| 05:37.53 | lorsungcu | what do you mean by " Actually, it seems like Asterisk is trying to use defaultuser during registration." |
| 05:38.28 | tompaw | lorsungcu: Asterisk was using the username from that field when it was spawning those REGISTERs. |
| 05:38.37 | lorsungcu | ah ok |
| 05:38.39 | tompaw | Now it seems like it's actually the fullcontact column. |
| 05:38.48 | lorsungcu | wanted to be sure it wasn't actually saying 'defaultuser' |
| 05:39.11 | tompaw | Yes, it is, I am 100% sure at this stage. |
| 05:39.26 | lorsungcu | did you add a column since i last saw a debug |
| 05:39.37 | lorsungcu | because full contact happens to be directly after defaultuser |
| 05:39.43 | tompaw | Yep, but that's irrelevant. |
| 05:39.50 | tompaw | Here's EXACTLY what happens. |
| 05:41.12 | tompaw | For a reason unknown to me, Asterisk goes through EVERY SINGLE row in the sipusers/sippeers table and tries to REGISTER using the name from callbackexten and ip from fullcontact columns. |
| 05:41.39 | lorsungcu | yeah, got that. |
| 05:41.41 | tompaw | If there is nothing in fullcontact, it will still spawn the REGISTER internally to a @(null) uri. |
| 05:41.52 | tompaw | The question remains - what the fuck? |
| 05:42.11 | lorsungcu | look into this at all? |
| 05:42.11 | lorsungcu | [Feb 18 16:39:43] [1;33mNOTICE[0m[27244]: [1;37mchan_sip.c[0m:[1;37m30423[0m [1;37mbuild_peer[0m: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' |
| 05:42.39 | lorsungcu | sorry if i missed something, been ask for a bit. |
| 05:42.56 | tompaw | lorsungcu: I don't even have a 'username' column.... |
| 05:43.04 | lorsungcu | so that is weird. |
| 05:43.34 | tompaw | This whole day is weird. It seems like asterisk grew a mind of its own and went rogue on me. |
| 05:43.38 | lorsungcu | unless it just prints it arbitrarily, i suppose. |
| 05:43.53 | tompaw | Yeah it probably does. |
| 05:44.02 | tompaw | So now I got only ONE entry in the table. |
| 05:44.15 | lorsungcu | that is a good start. |
| 05:44.19 | tompaw | This prevents SIP stack from crashing and allows me to actually register. |
| 05:44.31 | tompaw | But here's the thing: it doesn't stop Asterisk from trying to register. |
| 05:45.07 | tompaw | Since now it has an ip in the fullcontact field, it actually SENDS A REGISTER to my sipphone!! |
| 05:45.24 | tompaw | Is there like a global register-everything setting? |
| 05:45.33 | tompaw | registerpeers, or sth? |
| 05:47.23 | lorsungcu | what does your single entry look like |
| 05:47.35 | lorsungcu | pb csv or something |
| 05:49.26 | tompaw | lorsungcu: http://pastebin.com/X4JStnZM is the table, the entry means filling in everything from name to type. |
| 05:49.54 | tompaw | I'm actually gonna drop the callbackextension column, cause I have a strange feeling about it. |
| 05:49.56 | lorsungcu | right, so how are you filling it in? |
| 05:50.29 | tompaw | Excuse me? |
| 05:50.38 | lorsungcu | what values do you have for each column. |
| 05:51.25 | tompaw | test,dynamic,test,no,g729,all,test,friend |
| 05:51.51 | tompaw | ok, I removed that column and it seems to have stopped registering! |
| 05:52.18 | lorsungcu | it wasn't even populated, though? |
| 05:52.34 | tompaw | at this stage, let me remind that I ADDED it today, because some arbitrary message in debug convinced me to do so :F |
| 05:52.43 | tompaw | lorsungcu: it was =name |
| 05:53.05 | tompaw | ok, let me try restoring backup and killing that stupid column, see what happens. |
| 05:53.10 | lorsungcu | hmm |
| 05:53.34 | lorsungcu | did you ever look at the function |
| 05:53.35 | lorsungcu | register_realtime_peers_with_callbackextens |
| 05:55.19 | tompaw | nope, first time I'm hearing the name |
| 05:55.52 | lorsungcu | its in the debug you posted a while ago |
| 05:56.02 | lorsungcu | [Feb 18 16:39:44] [1;33mNOTICE[0m[27244]: [1;37mchan_sip.c[0m:[1;37m5519[0m [1;37mregister_realtime_peers_with_callbackextens[0m: Created realtime peer '772002' for registration |
| 05:56.34 | lorsungcu | might be worth looking at if you care to understand what was happening. |
| 05:56.45 | tompaw | see now I was looking at different debug and it didn't mention that part. |
| 05:57.23 | tompaw | I think I pretty much do. Earlier today, I made a mistake of following the error that complained about this missing column, and everything else was a consequence. |
| 05:57.32 | tompaw | The only two questions that remain unanswered are: |
| 05:57.53 | tompaw | 1) Why was my sip stack frozen, since obviously it wasn't the fucking missing column. |
| 05:58.23 | drmessano | Asterisk 11? |
| 05:58.34 | lorsungcu | yes |
| 05:58.35 | tompaw | 2) Why is the whole realtime thing so very inconsequent? The columns that are not needed should not really throw ERRORs, as this can cause a LOOOT of trouble once you mess around with them. |
| 05:58.39 | tompaw | drmessano: yes. |
| 05:58.41 | drmessano | I keep running into that too |
| 05:58.51 | drmessano | Different scenario |
| 05:59.07 | drmessano | But SIP seems to take a nap on me.. two different boxes running 11 |
| 06:00.13 | tompaw | :/ |
| 06:01.22 | drmessano | Kinda pissing me off.. dies with nothing logged |
| 06:01.39 | drmessano | no debug, nothing at all before the "event" |
| 06:01.59 | drmessano | Its like chan_sip slips out the back door at 11am and nobody knows its gone until 3pm |
| 06:02.07 | tompaw | I guess it's high time to move my confbridge to freeswitch and kiss those problems goodbye once and for all. |
| 06:02.18 | lorsungcu | tompaw: yeah setting that callbackfield to extension sure does make it send a register there |
| 06:02.26 | tompaw | drmessano: precisely what derailed my day today... |
| 06:02.44 | tompaw | lorsungcu: yep, seems so |
| 06:03.03 | tompaw | anyway, thanks for your help guys |
| 06:03.09 | lorsungcu | good luck |
| 06:03.12 | tompaw | I'm gonna enjoy my 4 hours of sleep now |
| 06:03.18 | tompaw | speak tomorrow, good night! |
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| 07:52.16 | madhatt3r | helloAll |
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| 08:00.13 | nunne | does anyone know if it's possible to send callerid with cmd pickup? |
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| 08:04.12 | studybo__ | hi |
| 08:04.29 | studybo__ | I'm setting up an asterisk server |
| 08:04.37 | lorsungcu | go on |
| 08:04.53 | studybo__ | my client connects to asterisk by iax2 |
| 08:05.16 | studybo__ | but timeout error keeps happening |
| 08:05.49 | studybo__ | am i missing any settings? |
| 08:05.57 | studybo__ | i'm using asterisk 10.12 |
| 08:07.00 | studybo__ | the client's working fine with asterisk 1.8 |
| 08:09.27 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:11.19 | studybo__ | I'm also confused by settings in extension.conf |
| 08:11.21 | studybo__ | and iax.conf |
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| 08:11.55 | studybo__ | what is the difference of these 2 settings |
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| 08:12.59 | din3sh | gd mrning ppl |
| 08:13.16 | kaldemar | studybo__: extensions.conf is your dialplan. iax.conf defines how IAX2 is used. when the call comes in, you need to look at asterisk's CLI and enable iax2 debug with "iax2 set debug on". |
| 08:16.10 | studybo__ | how do iax and dialplan relate? |
| 08:16.45 | studybo__ | as I understand I define the system's behavior when there's an incoming call by a dialplan |
| 08:18.18 | studybo__ | ah |
| 08:18.37 | studybo__ | I notice that the default port for iax (4569) isn't opened |
| 08:18.53 | studybo__ | though asterisk is running |
| 08:18.58 | studybo__ | is it alright? |
| 08:18.59 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.120.157) |
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| 08:22.25 | kaldemar | studybo__: dialplan defines what asterisk does with calls. IAX2 is a protocol for calls and iax.conf defines what may connect to asterisk via IAX2 and the relation to dialplan via a context. |
| 08:22.48 | kaldemar | studybo__: how did you notice the port is not opened? |
| 08:23.16 | studybo__ | nmap -p 4569 hostname |
| 08:23.48 | kaldemar | IAX2 does not run on TCP. your assumption is wrong. |
| 08:24.03 | studybo__ | got a clearer view between a dialplan and iax |
| 08:24.08 | studybo__ | oh |
| 08:24.19 | studybo__ | got it! |
| 08:26.12 | studybo__ | kaldemar |
| 08:26.15 | studybo__ | indeed |
| 08:26.29 | studybo__ | the port is opening (open|filtered) |
| 08:26.39 | studybo__ | so |
| 08:27.04 | studybo__ | what do you think is the reason of connection timeout? |
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| 08:30.13 | ChannelZ | Wrong IP? Firewall on the client side? You'd have to debug some networky things on your own end if you're not seeing the traffic even hit Asterisk |
| 08:32.37 | kaldemar | studybo__: look at asterisk as i told you. you'll get more hints. |
| 08:34.31 | studybo__ | I will check the networking stuffs and enable iax debug mode |
| 08:34.52 | studybo__ | >> kaldemar, ChannelZ: thanks for your help!! |
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| 08:46.33 | studybo__ | I can see a udp request |
| 08:46.37 | studybo__ | hit server |
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| 08:47.08 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:47.41 | studybo__ | I also enable iax debug (at cli: iax2 set debug on) |
| 08:48.08 | studybo__ | but nothing happens at the cli |
| 08:49.21 | kaldemar | see with netstat -nlup that asterisk is listening on the correct interface. if so, check your firewall on the asterisk box. |
| 08:50.19 | studybo__ | udp 3776 0 0.0.0.0:4569 0.0.0.0:* 17567/asterisk |
| 08:50.22 | studybo__ | here is the output |
| 08:50.42 | studybo__ | so I understand that it's ok |
| 08:50.47 | Wiretap | studybo__: redhat? |
| 08:50.51 | Wiretap | make sure system firewall is off |
| 08:51.19 | studybo__ | centos |
| 08:51.23 | Wiretap | same thing |
| 08:51.28 | Wiretap | (I knew you were going to say that) |
| 08:51.33 | Wiretap | cents is essentially |
| 08:51.34 | Wiretap | RHEL |
| 08:51.49 | studybo__ | oh! |
| 08:52.34 | studybo__ | I don't know that |
| 08:53.23 | ChannelZ | iptables -L INPUT if your policy is DROP or DENY (or is it REJECT? I don't remember) you'll need a rule for port 4569 |
| 08:54.15 | ChannelZ | although you said a min ago "I can see a udp request hit server" - how were you seeing that? |
| 08:54.19 | studybo__ | unfortunately |
| 08:54.26 | studybo__ | the policy is ACCEPT for INPUT |
| 08:54.38 | studybo__ | ah ok |
| 08:55.06 | studybo__ | by this command /usr/sbin/tcpdump -nnvvXS udp and dst port 4569 |
| 08:55.51 | ChannelZ | hmm |
| 08:57.48 | ChannelZ | and I assume iptables -L OUTPUT is also default to ACCEPT? |
| 08:58.32 | studybo__ | as you're assuming |
| 08:58.53 | ChannelZ | I would maybe stop and restart asterisk, maybe it barfed binding to the interface/port in some strange way |
| 08:59.39 | ChannelZ | if those packets really are incoming to your server you should see *something* with iax debug on |
| 09:01.44 | studybo__ | I'm afraid that the problem lies in config files |
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| 09:08.07 | yoyop | Hi all |
| 09:08.25 | yoyop | I'm running into a problem where I'm getting this message from my VOIP provider |
| 09:09.18 | yoyop | <PROTECTED> |
| 09:09.54 | ChannelZ | Perhaps you are dialing a number they can't route? |
| 09:10.24 | yoyop | The call works when I use zoiper to call though |
| 09:10.32 | ChannelZ | 'Termination not possible' |
| 09:10.48 | ChannelZ | Or the number format you are sending they don't like |
| 09:11.07 | yoyop | I'm sending it like that as well in the zoiper client |
| 09:11.12 | yoyop | tried with 011 and + |
| 09:12.13 | Wiretap | authentication problem |
| 09:12.25 | ChannelZ | well I guess you'd have to ask your ITSP specifically what the error means then. 500 is kind of a generic server error, and that specific message must mean something to them. |
| 09:14.14 | kaldemar | sip debug might reveal some additional info. |
| 09:19.24 | yoyop | sip debug on the CLI? |
| 09:20.46 | ChannelZ | ayup |
| 09:30.14 | yoyop | so this is what it says immediately |
| 09:30.17 | yoyop | <--- SIP read from UDP:69.90.209.57:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 184.173.162.122:5060;branch=z9hG4bK7801504f;rport=5060;received=184.173.162.122 From: "64664514" <sip:5555914476@sip3.voipvoip.com>;tag=as740413f2 To: <sip:01162811789312@sip3.voipvoip.com> Call-ID: 34b93de744b333aa5271003f399852c5@sip3.voipvoip.com CSeq: 103 INVITE Content-Length: 0 |
| 09:30.24 | yoyop | and then I get the not possible following it |
| 09:30.33 | yoyop | <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:69.90.209.57:5060 ---> SIP/2.0 500 Termination not possible Via: SIP/2.0/UDP 184.173.162.122:5060;branch=z9hG4bK7801504f;rport=5060 From: "64664514" <sip:5555914476@sip3.voipvoip.com>;tag=as740413f2 To: <sip:01162811789312@sip3.voipvoip.com>;tag=7f30cfab27f183c281995d8bc71a2ffb-97c2 Call-ID: 34b93de744b333aa5271003f399852c5@sip3.voipvoip.com CSeq: 103 INVITE Content-L |
| 09:32.10 | kaldemar | ~pb |
| 09:32.10 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 09:34.07 | yoyop | oops |
| 09:34.09 | yoyop | sorry |
| 09:34.14 | yoyop | here's the pastebin |
| 09:34.15 | yoyop | http://pastebin.com/AAx3qkLN |
| 09:36.56 | kaldemar | ask your ITSP. something happens between 100 and 500 but they are not sending a cause. |
| 09:43.35 | yoyop | my voip provider is pretty bad :( |
| 09:43.49 | yoyop | will there be a better one that you guys recommend? |
| 09:43.54 | yoyop | I'm using voipvoip now |
| 09:44.04 | yoyop | their support guys just says restart... |
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| 10:12.14 | din3sh | whats a cyclcitest on an * box? |
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| 10:59.44 | _zoom_ | is there anyway to rotate or randomize callerid without using scripts? |
| 11:00.27 | WIMPy | use RAND? |
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| 11:03.18 | _zoom_ | thnx WIMPy |
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| 14:13.03 | Katty | hi lads. |
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| 14:30.53 | chuckf | hi lady |
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| 14:51.45 | Katty | http://tinyurl.com/b5k3lt4 <- feeder is hoppin this morning. |
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| 14:55.33 | lorsungcu | nice, Katty |
| 14:56.29 | Katty | ty (= |
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| 15:10.25 | rogers | is IAX2 an IP Protocol? trying to figure out ACLs for my cisco to allow the traffic |
| 15:12.29 | lorsungcu | rogers: http://tools.ietf.org/html/rfc5456 |
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| 15:13.30 | rogers | ok looks like it works over IP |
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| 15:53.12 | danfromuk | Hi, is there any reason why when I do this.....Dial(LOCAL/07977XXXXXX@feature_outgoing_call,30,treM(announce^/var/lib/asterisk/clientsounds/features/187/343)... the macro is called but ${ARG1} is empty ? |
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| 15:59.32 | Marco-123 | hi all. im making a sip call between two computers on a local network THROUGH an asterisk server which is also on my local network. I'm trying to control two-way voice/video sip calls. problem: one-sidedness. If I set my client's Min Frame Size option to qcif, I get two way video out of asterisk (at qcif). if i set it to cif, i get one way video. question: what other parameters might be at play here (in terms of format negotiat |
| 15:59.39 | Marco-123 | because peer to peer it works fine |
| 15:59.51 | Marco-123 | any ideas? plz and thanks |
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| 16:06.02 | bitwize | Hi everybody! I have an problem which i hope you could help me out with.. |
| 16:06.07 | bitwize | I have a problem implementing a SIP-trunk. When Asterisk sends the initial REGISTER request the registrar reply with "403-Forbidden" (due to the missing "Authorization" header in the REGISTER request). |
| 16:06.13 | bitwize | During my earlier trunk implementations the registrar has always responded with "401 - Unauthorized", in theese cases Asterisk re-register with the "Authorization" header, but not when "403 - Forbidden" is replyed from the registrar. |
| 16:06.18 | bitwize | Somehow I need to inject the Authorization-header in my initial REGISTER request, anybody know how to configure this? |
| 16:09.43 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 16:09.53 | file | authentication is challenge based, you can't respond to a challenge you don't have |
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| 16:12.29 | bitwize | file: ok, thanks. That sound totally logical, I wonder why the ITSP tells me to send the credentials in my initial request.. |
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| 16:22.17 | rogers | having some issues with an IAX2 trunk between 2 sites. We had a working setup so I know the trunk is configured correctly. We since then set up some new routing equipment. From the 2 asterisk boxes I am able to ping, and each return open with an nmap - 4569/udp open|filtered unknown. tcpdump shows communication both ways, but the trunk still says unreachable. What might be the issue? |
| 16:22.47 | Greenlight | Credentials ? |
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| 16:23.40 | rogers | credentials in the trunk settings? I haven't touched those and they were working prior |
| 16:24.04 | Greenlight | DOes it specify IP addresses, which might have chagned ? |
| 16:24.29 | Kobaz | mmm |
| 16:24.30 | rogers | all the routing has changed |
| 16:24.41 | rogers | I made sure localnets are updated in sip_nat.conf |
| 16:24.49 | Greenlight | BUt, you're using IAX ? |
| 16:24.51 | rogers | yeah |
| 16:25.01 | Greenlight | Did you check the peer settings ? |
| 16:25.01 | rogers | for the trunk between the two boxes |
| 16:25.13 | rogers | peer settings all look fine |
| 16:26.10 | rogers | I thought it might be an acl rule in the firewall causing problems |
| 16:26.17 | rogers | but the nmap shows open/filtered |
| 16:26.23 | Greenlight | Hmm |
| 16:26.28 | rogers | its like the asterisk box is denying connections |
| 16:26.57 | Greenlight | If it's a stateful firewall, I've seen them get confused before, you tried rebooting it just incase |
| 16:27.22 | rogers | I rebooted one side, the netgear |
| 16:27.39 | rogers | theres 2 other core routers between |
| 16:27.42 | rogers | both ciscos |
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| 16:28.47 | rogers | I'll try the netgear again for kicks |
| 16:29.08 | rogers | The peer info shows |
| 16:29.10 | rogers | Addr->IP : 10.10.20.10 Port 4569 |
| 16:29.10 | rogers | <PROTECTED> |
| 16:29.21 | rogers | should the default addr be set? |
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| 16:37.25 | Marco-123 | can i confirm a few things. is asterisk involved in the negotiation of video codec formats? |
| 16:37.31 | Marco-123 | for sip calls |
| 16:37.43 | file | yes. |
| 16:38.16 | Marco-123 | thanks. 2) can bit rate be a factor in what frame size is negotiated? |
| 16:38.23 | Marco-123 | max bit rate |
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| 16:43.32 | nickfennell | I don't think SIP is adaptive. |
| 16:43.53 | nickfennell | i.e., once the setup has been completed, that's what you'll use for the duration of the call. |
| 16:44.10 | lorsungcu | what do you mean by setup? |
| 16:44.34 | nickfennell | Negotiation of the call. Agreeing of codecs, media ports etc |
| 16:44.36 | jmetro | call setup time. bringing the two peers. |
| 16:44.52 | Marco-123 | thanks. i wonder why asterisk is not negotiating the parameters i have clearly specified in my client. looking at the sip debug log now |
| 16:44.57 | nickfennell | SIP doesn't particularly care if it's audio or video. the process is the same |
| 16:45.04 | Marco-123 | if anyone would have a look at it with me, i'd appreciate it |
| 16:45.16 | nickfennell | Marco-123: What's your actual issue. Describe. |
| 16:46.36 | Marco-123 | my issue is with frame size. when i specify qcif, no probs. when i specify cif (from the client i mean), i get cif on one side, and qcif on the other. QUIRK=if i switch who is the caller/receiver, i get different behaviour. cif one side, no video on other side. |
| 16:46.39 | lorsungcu | http://sofia-sip.sourceforge.net/refdocs/soa/soa_sdp_oa_use_cases.html |
| 16:46.48 | Marco-123 | http://pastebin.com/JXfE15S5 <- that is my log is anybody is interested |
| 16:47.11 | Marco-123 | lorsungcu: perfect! was looking for this |
| 16:47.18 | Marco-123 | thanks nickfennell and lorsungcu |
| 16:48.56 | nickfennell | No problem. Hope it leads to a fix to your issue. |
| 16:51.12 | lorsungcu | nick that link was mostly for you |
| 16:51.19 | lorsungcu | showing SIP update offers and so forth. |
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| 16:53.25 | Marco-123 | hmm. this log is pretty short, would one of you fine gentlemen have a quick look for me? im trying to find where the frame-size is first lost |
| 16:54.25 | lorsungcu | sure, paste bin it |
| 16:54.37 | Marco-123 | http://pastebin.com/JXfE15S5 |
| 16:54.39 | Marco-123 | thank you very much |
| 16:54.53 | Marco-123 | i need all the clues i can muster up |
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| 17:08.37 | fubada | hi folks |
| 17:09.23 | fubada | can someone clue me in on a slightly offtopic question? whats preferred among openh323, h323plus, or OPAL |
| 17:09.37 | fubada | Im trying to build t38modem for use with asterisk for fax |
| 17:09.58 | fubada | it can be linked against either of those 3 |
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| 17:14.07 | Katty | dances with chuckf |
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| 18:32.01 | danfromuk | Hi, is there any reason why when I do this.....Dial(LOCAL/07977XXXXXX@feature_outgoing_call,30,treM(announce^/var/lib/asterisk/clientsounds/features/187/343)... the macro is called but ${ARG1} is empty ? |
| 18:42.09 | kaldemar | danfromuk: no. show a CLI output of the call. |
| 18:47.21 | danfromuk | Just getting it. One moment. |
| 18:48.48 | _Corey_ | Anyone know of a media player for windows that can play g729 files natively? (Doesn't need to be free...) |
| 18:49.02 | leifmadsen | _Corey_: try VLC |
| 18:49.11 | leifmadsen | it seems to play everything |
| 18:49.18 | leifmadsen | _Corey_: (and it's free) |
| 18:49.27 | _Corey_ | tried it, seems they removed the g729 at some point |
| 18:49.34 | leifmadsen | bastardos |
| 18:50.12 | _Corey_ | I tried to find the codec library (free or otherwise) and everybody's just broken links and non-working websites |
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| 18:55.12 | danfromuk | kaldemar: heres the cli output http://pastebin.com/ujxdMkRL |
| 18:56.03 | danfromuk | Line 70 is the dial and line 101 is the ${ARG1} |
| 18:56.34 | leifmadsen | danfromuk: you're calling a Local channel without /n, so it probably can't execute the M() option without that |
| 18:56.56 | danfromuk | leifmadsen: its executing. Just missing the variables. |
| 18:57.00 | leifmadsen | what version? |
| 18:57.01 | danfromuk | I'll try it with a /n |
| 18:57.03 | danfromuk | 1.8.20.1 |
| 18:57.04 | leifmadsen | nah |
| 18:57.13 | leifmadsen | not sure... method looks reasonably sane |
| 18:57.21 | danfromuk | It has worked before so not sure why its stopped working. |
| 18:58.14 | danfromuk | I think i'm going to try downgrading asterisk and see if it resolves the issue. |
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| 19:03.23 | Greenlight | How can I diagnose one way audio after executing a ManagerBridge on two channels? They audio as perfect prior to this bridge, and from what I can see Asterisk ins't trying to remotely bridge them (certianly disabled in the peer settings), but yet there's only 1 way audio. Looking for some pointers where to start looking ? |
| 19:03.25 | kaldemar | CLI output is much easier to read... |
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| 19:15.14 | danfromuk | leifmadsen: when using /n in a dial command with timeout and options, where does it go? at the end of the options or after the number@context? |
| 19:16.39 | kaldemar | danfromuk: you already use it elsewhere in your dialplan |
| 19:17.38 | danfromuk | kaldemar: yes but thats on a line without options |
| 19:17.54 | leifmadsen | that makes no difference |
| 19:18.08 | leifmadsen | suggests asteriskdocs.org |
| 19:19.25 | danfromuk | slash n seems to have fixed it. |
| 19:19.39 | danfromuk | The macro variable is there now. |
| 19:19.55 | danfromuk | Wonder why it stopped working.... or even if it was ever working :-S |
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| 19:20.00 | danfromuk | Thanks for your help with this. |
| 19:20.38 | kaldemar | are you sure you need all those local channels? |
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| 19:24.23 | danfromuk | Its a dial plan that's built automatically by my backend software. Users can change the dialplan themselves. I think the local channels were introduced so that called parties could reject incoming calls while still allowing other called parties to see ringing. I can't remember now and stupidly didnt comment my code. |
| 19:25.10 | danfromuk | For some reason, Local worked better than Goto if i remember correctly. |
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| 19:37.45 | coreyf1513 | _Corey_: ffmpeg.org says v0.9 supports g729 decoding.. I don't know any details just that it's listed as a new feature |
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| 19:40.56 | _Corey_ | coreyf1513: Yeah, I played with that earlier... seem to work OK as a command-line solution |
| 19:44.46 | _Corey_ | coreyf1513: Thanks! |
| 19:44.54 | coreyf1513 | _Corey_: my understanding is VLC uses ffmpeg, so VLC built against ffmpeg 0.9+ in theory should do what you want |
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| 20:20.10 | danfromuk | Is it possible to get asterisk to listen for sip packets on two ports? I have a client whose ISP blocks 5060 |
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| 20:23.02 | jzaw | danfromuk, from a quick google ans=no ... but http://wiki.kolmisoft.com/index.php/Two_SIP_listening_ports_for_single_Asterisk |
| 20:23.28 | jzaw | maybe someone more knowledgeable will correct that |
| 20:24.11 | danfromuk | Yep. I got it aswell. Thanks. |
| 20:24.24 | jzaw | danfromuk, is your client running * |
| 20:24.35 | jzaw | or a sip phone / other device? |
| 20:24.42 | danfromuk | No. Its a softphone over MiFi from Verizon |
| 20:25.05 | jzaw | i run csipsimple on 'roid ... on 5062 |
| 20:25.14 | jzaw | with no forwarding ... just stun |
| 20:25.18 | jzaw | behind nat |
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| 20:34.50 | danfromuk | jzaw: have you used zoiper before? |
| 20:40.41 | danfromuk | are there any softphones for windows that allow you to point to a different server port? |
| 20:42.39 | leifmadsen | yes |
| 20:42.42 | leifmadsen | try jitsi |
| 20:43.35 | danfromuk | Thanks |
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| 20:51.03 | danfromuk | Jisti seems extremely jittery. Have you experienced that? |
| 20:53.39 | artyx | I like zoiper personally |
| 20:53.53 | artyx | fast free, easy, and mostly idiot proof for testing purposes |
| 20:54.06 | artyx | (also nothing in the config really to screw up) |
| 20:54.23 | danfromuk | Yes but i cant find any options to change the sip server port number. |
| 20:54.30 | danfromuk | just the client's port number |
| 20:54.42 | danfromuk | So trying out jitsi which does let you change the port number. |
| 20:54.44 | artyx | dude, in zoiper? |
| 20:54.47 | artyx | You can't find it? |
| 20:57.14 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:59.42 | artyx | in domain, specify server:port |
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| 21:00.09 | artyx | So if your sip server is on port 9999 then under domain your.sip.server.i.p:9999 |
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| 21:00.57 | artyx | What is the boggle? |
| 21:01.36 | danfromuk | I did a google and pages seemed to claim that you couldnt do that in zoiper. I'll give it a test. |
| 21:01.42 | artyx | Well |
| 21:01.51 | artyx | I stuck in :5060 and it works just fine.... but that doesnt mean you annot |
| 21:02.03 | artyx | It just means that adding :5060 didnt cause it to break :P |
| 21:02.14 | danfromuk | 5060 is default so you havent tested anything. |
| 21:02.19 | artyx | Ah, but id id |
| 21:02.24 | artyx | Since it didnt error with the :5060 |
| 21:02.39 | artyx | It proves that the UI is capable of accepting (or at the least not parsing) ip addresses |
| 21:02.46 | danfromuk | However, you are correct. Its working. |
| 21:03.30 | artyx | If you see a sign saying caution do not stick elbows in ears... do you listen, or do you at least entertain the possibility of what that might do for ya |
| 21:03.39 | artyx | Me? I at least think about it.... |
| 21:05.10 | artyx | Especially when testing software out.. I try to set it up like osmeone who didnt follow the dirs, and see how tolerant it is of user idiocy |
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| 21:05.30 | artyx | Beleive it or not, there are some very lousy instructions out there for applications |
| 21:07.15 | jzaw | artyx, dont put your finger in the fire ;p |
| 21:07.16 | jmetro | anyone here configured an Aastra to "hotline" or "autodial" or "offhook dial" ? |
| 21:07.26 | artyx | too late Jz |
| 21:07.34 | jmetro | aka "automatically dial a number when the handset is picked up" |
| 21:07.35 | jzaw | :D |
| 21:07.39 | artyx | Been there, done that. ive made fires so hot they melted cambpells soup cans... and underwater sprinkler lines ;) |
| 21:07.52 | jzaw | fireman? |
| 21:07.57 | artyx | Chef |
| 21:08.00 | jzaw | ah |
| 21:08.07 | jzaw | my next choice was going to be blacksmith |
| 21:08.17 | artyx | Only recreationally |
| 21:08.29 | artyx | they gave me about 3 hrs to make enough coals to clay bake 18 turkeys |
| 21:08.35 | WIMPy | jmetro: A phone connected to Asterisk, how? |
| 21:09.30 | jmetro | Wimpy: An aastra 6757i, just trying to find the phones local "hotline" or "autodial" settings. I know that past the point of telling the phone to dial a number once the handset is picked up is up to my dialplan to catch and route it |
| 21:09.44 | jmetro | just want to know if anyone knows where the settings are at |
| 21:09.53 | jzaw | danfromuk, i love jitsi on the mac ... but using an svn trunk compile of sflphone on wheezy/kde |
| 21:10.28 | jzaw | danfromuk, if your client can ... why not an iax softphone ? |
| 21:10.35 | WIMPy | Yes, it's entriely up to the phone. Asterisk won;t be involved. |
| 21:11.00 | jmetro | Wimpy: well, it would to the point that your phone can autodial a local extension and you pick it up with regular dialplan from there. |
| 21:15.50 | danfromuk | artyx: thanks . working perfectly with zoiper. |
| 21:16.51 | artyx | But of course it is ;) |
| 21:16.59 | artyx | Hey if you find a better sip client lemme know |
| 21:18.56 | danfromuk | artyx: Been switching to zoiper ever since the latest version of xlite. glad its about to change the port number. was getting worried! |
| 21:19.14 | artyx | I had this other nice softphone, but they installed a ton of other bloatware on your pc and i wanted something nicer |
| 21:22.28 | danfromuk | Personally, i want an open source one that i can brand |
| 21:22.33 | artyx | Lol |
| 21:23.41 | danfromuk | At least then i'll know that my business isn't going to turn upside down every time a company releases a new softphone version. |
| 21:24.19 | artyx | Do they automatically disalbe old versions? |
| 21:24.37 | danfromuk | Xlite did |
| 21:27.21 | artyx | weak |
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| 21:41.28 | danfromuk | After all that hard work, it appears that even though i'm using a random port, Verizon are blocking the call setup SIP packets. It can't be a nat issue because its registering and dialling fine. But the OK INVITE sip packet that contains the agreed codecs isn't being received by the softphone. |
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| 21:42.40 | [TK]D-Fender | Checkout time, BBL |
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| 21:44.11 | danfromuk | Looks like i'm going down the VPN route. Thanks everyone for trying. |
| 21:45.00 | AkkerKid | anyone have code that figures out what the second business day after today is assuming a mon-fri work week? |
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| 22:07.12 | danfromuk | Has anyone seen a guide to setting up a vpn server to connect to asterisk? |
| 22:08.24 | pigpen | hi all, when using dpma where do I load the xml or setting to setup ring-answer? |
| 22:08.35 | pigpen | in xml provisioning, it was, well, the xml file. |
| 22:08.42 | pigpen | now in dpma, unsure. |
| 22:08.55 | malcolmd | in dpma land you need to setup alerts and load those alerts onto the phone |
| 22:09.11 | malcolmd | https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration#DPMAConfiguration-AlertConfigurationOptions |
| 22:09.33 | jzaw | danfromuk, wouldnt any old vpn setup work? |
| 22:09.35 | malcolmd | pigpen: ^ |
| 22:10.46 | pigpen | malcolmd, thanks. I googled around a bit, but no luck. |
| 22:11.13 | malcolmd | np. i'm still waiting on google to figure out that the wiki content there is the definitive resource for configuring digium phones…google seems to be a bit slow on the uptake |
| 22:11.57 | pigpen | malcolmd, ah, so those are setup in the res_digium_phone.conf file? |
| 22:12.06 | malcolmd | yup |
| 22:12.08 | pigpen | very nice. |
| 22:12.19 | pigpen | what? Not each xml file for every phone? haha. |
| 22:12.36 | malcolmd | in dpma-land, the only thing that lives in an xml file (currently) are the contacts - those are still constructed in xml and then loaded onto the phone |
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| 22:19.22 | AkkerKid | Does anyone know how I can calculate the date of the next monday following a given date in dialplan? |
| 22:19.44 | jmetro | add 7 days |
| 22:20.08 | jmetro | oh a given date |
| 22:20.22 | AkkerKid | yeah I'm not quite that daft. |
| 22:21.07 | AkkerKid | ideally i'd build a function that calculates the date of the next business day given inputs of today's date and how many business days to add. |
| 22:21.25 | jmetro | 7 iftime statements |
| 22:21.45 | jmetro | or 6 [or 5 if youre more clever] |
| 22:22.41 | AkkerKid | i wantto be more flexible than that. |
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