| 00:03.52 | lanning | helllllllllllllllllllllloooooooooooooooooooooooooo asterisk. |
| 00:04.08 | lanning | we should fix that echo |
| 00:04.11 | lanning | :) |
| 00:05.24 | carrar | hrmmm |
| 00:05.30 | carrar | I don't hear any echo |
| 00:06.05 | carrar | Just a loud ringing sound in my head |
| 00:06.40 | lanning | hrmmm |
| 00:06.43 | lanning | I don't hear any echo |
| 00:06.46 | lanning | Just a loud ringing sound in my head |
| 00:06.53 | lanning | :P |
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| 02:29.20 | cgoodwin73 | hello to all! |
| 02:33.33 | cgoodwin73 | Has anyone seen "dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN)" problems when calling a Ring Group under AsteriskNow 3.0.0? We're seeing it now with a server that's supposed to be deployed tomorrow. We've seen this failure with every phone we've tested so far - Digium D40 & D70, Polycom 550. Directly dialing the same extensions works normally. Any ideas, guys? We appreciate any help on this!! |
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| 02:39.21 | cgoodwin73 | We are using DPMA on the Digiums, all Digium-specific features, etc., are working. We simply can't find anything wrong with our config. Any thoughts? |
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| 02:40.25 | igcewieling | cgoodwin73: make sure call counter is enabled in sip.conf |
| 02:43.17 | hebber | Hi, where do I find a how-to to correctly report an possible Asterisk bug? |
| 02:44.01 | andross | are you sure its a bug, whats happening |
| 02:44.59 | hebber | Thats why I wrote its a possible bug - here is the scenario: It's consistent and possible to replicate |
| 02:45.49 | hebber | Its realtime ACL where we use an ATA which then reports an error in in the ACL container everytime there is an incoming call |
| 02:46.10 | hebber | it works, but annoying in the CLI |
| 02:46.35 | hebber | not static realtime btw |
| 02:46.49 | cgoodwin73 | I'll pass that to my programmer, igcewieling, thanks. He posted to the asterisk users mailing list last night, but no reposnse at all there.... |
| 02:47.23 | hebber | hmm, can I subscribe to that list? |
| 02:47.27 | igcewieling | cgoodwin73: since dialparties.agi is really a freepbx thing, you might have better luck there or reproduce the issue outside of freepbx |
| 02:47.51 | hebber | BTW, I have had this "bug" the last two months updating Asterisk three times |
| 02:47.58 | igcewieling | hebber: you can even search it, add site:lists.digium.com to your google search. sign up at lists.digium.com |
| 02:49.43 | cgoodwin73 | igcewieling: good point, I had the same thought, signing up there now |
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| 04:26.51 | rhineheart_m | is there a way to connect skype to asterisk? |
| 04:29.51 | netman | I think there is a propietary module for doing that |
| 04:31.20 | igcewieling | These toll fraud hackers are actually being helpful. They are finding all the routers we don't have on the "list of routers", and so were not updated with new ACLs. 8-| |
| 04:32.28 | igcewieling | rhineheart_m: Stype does not want Asterisk users using their service. There are a couple of hacks, most of them likely no longer work . |
| 04:33.04 | rhineheart_m | igcewieling: I see.. now I know. Thanks. |
| 04:33.30 | rhineheart_m | Any recommendation for a cheap DID in the US? |
| 04:33.37 | rhineheart_m | DID provider I mean.. |
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| 05:21.51 | andross | yeah anyone with a recommendation on a cheap sip trunk provider |
| 05:22.01 | andross | for like 2 incoming and outgoing |
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| 06:33.44 | ChannelZ | What kind of usage? |
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| 06:44.33 | eject_ck | hi, how can I make call, (Dial then playback(my-recording) ) from shell ? |
| 06:46.29 | ectospasm | eject_ck: I'd use a call file. |
| 06:47.01 | eject_ck | ectospasm: that's what I'm using now |
| 06:47.26 | eject_ck | there is a way to make call from asterisk cli ? |
| 06:47.50 | ectospasm | you can execute a specific section of dialplan (e.g., Goto...) |
| 06:48.02 | ectospasm | oh, |
| 06:48.46 | kaldemar | eject_ck: core show help channel originate |
| 06:48.47 | ectospasm | Not easily. I'd use a call file for that, then !cp callfile.txt callfile, !mv callfile /var/spool/asterisk/outgoing/ |
| 06:49.16 | ectospasm | yeah, but will originate allow eject_ck to execute dialplan when it connects? |
| 06:49.22 | kaldemar | sure |
| 06:49.33 | kaldemar | just like any other means of origination. |
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| 07:59.51 | eject_ck | thank you all! |
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| 08:06.30 | slav3_kitten | for/ |
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| 08:28.42 | Freeaqingme | hi. I'm looking for a softphone (linux) that displays a sip-log. any ideas? |
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| 08:41.31 | fling | [Apr 5 15:41:16] WARNING[17882]: pbx.c:4458 pbx_extension_helper: No application 'GoSub' for extension (hh, +73852350000, 1) |
| 08:41.38 | fling | what am I doing wrong? |
| 08:42.27 | x1user | http://codepad.org/4OdBKes2 In this context it doesnot play the Background sound, but the same context works on different version of asterisk... anyone who knows what is the problem? |
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| 09:12.45 | kaldemar | fling: what version are you using? |
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| 10:34.55 | bogrd__ | Hello, I am getting null values for $VM_DATE and $VM_DUR variables after the VoiceMail has finished on Asterisk 11.2.1. Any idea why this is happening ? |
| 10:41.10 | kaldemar | where and why are you expecting to see non-null values? |
| 10:42.27 | bogrd__ | kaldemar: My dialplan for an extension looks like this : |
| 10:42.35 | ectospasm | ~pb |
| 10:42.36 | 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. |
| 10:44.01 | bogrd__ | kaldemar: http://paste.kde.org/716720/ |
| 10:44.59 | kaldemar | why are you expecting to see values in the variables? |
| 10:45.58 | bogrd__ | VM_MESSAGEFILE gave the path of the Voicemail, likewise was expecting a value for $VM_DUR too |
| 10:46.30 | kaldemar | the app does not set those variables on the calling channel. |
| 10:47.18 | bogrd__ | kaldemar: any way by which I can get the voicemail duration ? |
| 10:50.37 | kaldemar | afaik, no. |
| 10:51.26 | bogrd__ | kaldemar: ok thanks ' |
| 10:57.54 | x1user | http://codepad.org/4OdBKes2 In this context it doesnot play the Background sound, but the same context works on different version of asterisk... anyone who knows what is the problem? |
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| 11:20.29 | kaldemar | x1user: pastebin a CLI output of a call with verbosity enabled. |
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| 11:40.38 | Greenlight | Is there an established method to implement a "queue callback" - whereby callers can be called back when they reach top of the queue ? |
| 11:45.40 | skrusty | anyone ever used an SPA122 with a fax machine in the UK? |
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| 11:48.57 | skrusty | seems when i call the fax machine behind an spa122, the call gets put on hold |
| 11:49.18 | skrusty | so inbound fax hears moh, while the fax machine tries to negotiate fax tones |
| 11:49.34 | skrusty | really odd :/ |
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| 12:42.59 | chuckf_ | skrusty: out of curiousity, what happens if you point that call to a sip phone? |
| 12:44.18 | srp_ | Hi, I'm using Asterisk ARA for storing SIP users details (like username, secret, etc.) in MySQL database. I want to know how do I set the md5 hash salt (for the sip user secret) so that Asterisk uses the same salt while calculating the md5 sum while the user is registering... |
| 12:45.27 | skrusty | chuckf_: it gets answered when picked up |
| 12:45.53 | skrusty | nothing odd at all when pointed directly to a sip phone |
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| 13:01.03 | seik0 | Hi everyone. I've got another question for you. I have DAHDI, 8-port BRI card and some incoming ISDN calls. I realised today, that each call goes into same exten (into configured for card context). It means, that TelCo doesn't provide DID for me? Or what else I can check? |
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| 13:02.51 | seik0 | One options is to configure input point for each card port, but then I become dependant on if someone very smart will switch cords between card and NT-terminators. But now it's only option I see. |
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| 13:03.42 | kaldemar | seik0: what extension it that? s? |
| 13:03.50 | seik0 | No |
| 13:04.09 | seik0 | But goes to "s" as specific exten is not define |
| 13:04.11 | seik0 | d |
| 13:05.04 | seik0 | In fact, that exten is even not an msn |
| 13:05.26 | seik0 | the only common part is city code |
| 13:06.01 | WIMPy | seik0: Which channel? |
| 13:06.33 | seik0 | don't understand question ) |
| 13:06.48 | WIMPy | channeltype |
| 13:06.58 | WIMPy | what drivers |
| 13:07.21 | seik0 | dahdi |
| 13:08.08 | WIMPy | It's MSNs, not DDI? |
| 13:08.15 | WIMPy | Do you perhaps have alwaysimmediate enabled? |
| 13:09.20 | seik0 | DID, of course |
| 13:09.27 | seik0 | stop |
| 13:09.39 | seik0 | yes, DID is right |
| 13:10.00 | WIMPy | That's a difference. If you have DDI, you want immediate=yes. |
| 13:10.36 | seik0 | immediate yes forces to use "s" extension, no? |
| 13:10.56 | WIMPy | That should only happen in NT mode. |
| 13:11.15 | seik0 | i'l try |
| 13:11.29 | WIMPy | overlapdial=yes and immediate=yes |
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| 13:13.11 | seik0 | tried, looks same |
| 13:13.58 | seik0 | em |
| 13:14.24 | seik0 | kaldemar asked if it's "s" extension. Now I think yes |
| 13:14.35 | seik0 | it's always only s extension |
| 13:14.51 | igcewieling | seik0: then you are not receiving the DDI info. |
| 13:15.00 | seik0 | number i thounght was a did was CallerID |
| 13:15.00 | WIMPy | Unusual. Put a WaitExten there. |
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| 13:18.39 | seik0 | I checked again, it's not a CallerID, excuse me =). So it's really DDI, it differs from callerid |
| 13:18.57 | seik0 | So, DDI is always the same |
| 13:19.16 | seik0 | immediate, overlapdial and WaitExten don't change anything |
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| 13:19.58 | WIMPy | And what is the exten? Is that just your base number or what? |
| 13:20.44 | igcewieling | immediate=yes is almost never needed |
| 13:20.50 | seik0 | base numbers are 83422410026, 83422410027, 83422410028. Exten is 3422700725 |
| 13:22.46 | seik0 | Should run now. Thanks for help. Now I still think it's TelCo issue. |
| 13:23.02 | [TK]D-Fender | igcewieling, It's kind of a fact of life with BRI |
| 13:23.34 | igcewieling | [TK]D-Fender: but not PRI? |
| 13:24.41 | WIMPy | igcewieling: I'd never do without. |
| 13:24.58 | WIMPy | seik0: And that list looks more like MSNs thatn DDI. |
| 13:28.17 | [TK]D-Fender | igcewieling, not so commonly |
| 13:29.01 | jmetro | so if i have exten => 1 that means the user can press 1 as an option...but asterisk is ignoring everything except my "S" extension.. |
| 13:29.06 | [TK]D-Fender | WIMPy, I'm forgetting on a term used for that. "[blank] dialing" when it checks as you dial in-line... |
| 13:29.22 | [TK]D-Fender | jmetro, PASTEBIn <- show us |
| 13:29.29 | [TK]D-Fender | jmetro, Code & CLI |
| 13:29.34 | WIMPy | overlap |
| 13:29.48 | [TK]D-Fender | WIMPy, thanks. |
| 13:29.50 | jmetro | [TK]D-Fender: @,@ give me a bit |
| 13:29.56 | [TK]D-Fender | igcewieling, Yes, "overlap dialing" |
| 13:30.05 | WIMPy | And that can happen in all directions. |
| 13:30.16 | [TK]D-Fender | igcewieling, Starts routing in-line... its just how things are in much of europe. |
| 13:32.47 | jmetro | http://pastebin.com/x0zb9P4j is my dialplan |
| 13:33.15 | igcewieling | [TK]D-Fender: ah, I was not thinking of inband dtmf for overlap dialing |
| 13:33.33 | igcewieling | still rather odd, ISDN normally sends the entire DNIS in enbloc |
| 13:33.41 | WIMPy | igcewieling: How do you get to DTMF now? |
| 13:34.03 | WIMPy | igcewieling: Only for MSNs, not for DDI. |
| 13:34.08 | igcewieling | WIMPy: our ISDN inbound calls do not transmit the dialed number using DTMF |
| 13:34.27 | WIMPy | Off course not. |
| 13:34.28 | igcewieling | they transmit the dialed number in the setup message. |
| 13:34.52 | jmetro | http://pastebin.com/JnM9Lqjc is my dial |
| 13:34.55 | WIMPy | Yes, hence the question what makes you mention DTMF. |
| 13:35.20 | igcewieling | WIMPy: your siggestion for immediate=yes and wait exten implies you are expecting DTMF |
| 13:35.36 | WIMPy | igcewieling: No |
| 13:36.05 | WIMPy | jmetro: You're missing the priority. |
| 13:36.12 | WIMPy | exten => 1,1,Verbose... |
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| 13:36.58 | WIMPy | igcewieling: WaitExten works perfectely before accepting the call. |
| 13:37.36 | jmetro | Wimpy: Oh jebus. |
| 13:37.52 | igcewieling | WIMPy: immediate=yes immediatly answers (accepts) the call |
| 13:38.12 | WIMPy | igcewieling: No. Only if the dialplan does. |
| 13:38.55 | WIMPy | If you just do a s,WaitExten(), the call won't be accepted until it has a match in the dialplan. |
| 13:40.34 | WIMPy | Asterisk and dialling as a very tough topic, but it is possible :-) |
| 13:40.51 | WIMPy | s/as/is/ |
| 13:42.41 | jeffspeff | how can i tell what codec a channel is using? |
| 13:43.42 | WIMPy | jeffspeff: <channeltype> show channel[s| <channel>] |
| 13:43.47 | igcewieling | jeffspeff: "sip show channels" |
| 13:44.05 | jeffspeff | haha! thanks |
| 13:44.11 | jeffspeff | haven't had all my cofee yet. :) |
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| 13:45.43 | igcewieling | coffee is 2 doors down the hall. Both cups and IV drips are provided. |
| 13:46.16 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 13:46.17 | WIMPy | No networked coffee? |
| 13:46.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 13:58.53 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-30.usfamily.net) |
| 14:01.30 | lorsungcu | anyone experience that packet of death people were talking about on supermicro machines a while back? |
| 14:05.51 | bulkorok | read about that... |
| 14:07.37 | igcewieling | yes. |
| 14:08.05 | igcewieling | that was a VERY bad week, all three of our newly deployed core asterisk switches had the problem. |
| 14:08.43 | lorsungcu | what were the symptoms, igcewieling |
| 14:08.48 | lorsungcu | think it just happened to me |
| 14:09.41 | igcewieling | lorsungcu: the most obvious symptom was large numbers of dropped packets or errored packets in the output of "ifconfig" |
| 14:09.50 | lorsungcu | yeah |
| 14:09.53 | lorsungcu | thats what i saw |
| 14:09.55 | lorsungcu | like 4 billion |
| 14:10.08 | lorsungcu | although rebooting fixed it |
| 14:10.09 | igcewieling | they come in bursts. are you sure you have the chip with the issue? |
| 14:10.18 | lorsungcu | yeah just checked |
| 14:10.33 | lorsungcu | Intel 82574L |
| 14:10.37 | igcewieling | then get a new card ASAP (add-on) |
| 14:10.44 | lorsungcu | going to today |
| 14:11.13 | lorsungcu | damn that sucks. should have checked before deploying the thing |
| 14:11.29 | lorsungcu | just got everything stable and working for this customer |
| 14:11.33 | lorsungcu | thing takes a dump |
| 14:14.15 | igcewieling | lorsungcu: same here |
| 14:14.27 | lorsungcu | what phones? |
| 14:14.44 | igcewieling | but we have on average 200+ calls active across the two systems |
| 14:15.05 | igcewieling | No phones, adtran media gateways |
| 14:15.09 | lorsungcu | there are really not that many calls on this system |
| 14:15.33 | lorsungcu | not totally sure how many, but i'd say 20, max |
| 14:16.37 | *** join/#asterisk AlHafoudh (~alhafoudh@85.248.11.120) |
| 14:16.38 | AlHafoudh | hi all |
| 14:16.47 | AlHafoudh | does anyone have experience with avaya IP office here? |
| 14:17.17 | jmetro | ew |
| 14:17.49 | AlHafoudh | i know :( |
| 14:17.49 | Greenlight | I replaced one with Asterisk once, does that count? :) |
| 14:18.05 | jmetro | ^ only one? XD |
| 14:18.26 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 14:18.50 | Greenlight | Only one I've came accross yet, so 100% record :) |
| 14:19.08 | AlHafoudh | i have a strange problem, when recieving call from SIP trunk and putting it into H323 trunk in avaya ip office, the calling number (caller id) is like real caller number concatenated with source IP address withut dots :) |
| 14:20.07 | lorsungcu | AlHafoudh: try in #avaya |
| 14:20.36 | AlHafoudh | yeah :) i tried |
| 14:21.22 | Greenlight | Avaya's a commercial system - can't you call the installer or Avaya themselves ? |
| 14:23.11 | AlHafoudh | installer is me |
| 14:23.31 | AlHafoudh | avaya support with reply in two weeks with suggestions like "did you try to turn it off and on again" ? |
| 14:23.35 | AlHafoudh | ... |
| 14:25.20 | Greenlight | http://www.wickes.co.uk/powastrike-sledge-hammer-7lb/invt/167858/ ? |
| 14:25.38 | AlHafoudh | :) |
| 14:26.21 | jmetro | I know an avaya installer in my area, taking a class on how to install and use asterisk =) |
| 14:27.32 | Greenlight | I guess you want to take a quick SIP trace, and check what's being passed over, and they play around with the settings on Avaya till it looks right |
| 14:27.47 | Greenlight | Latest firmware etc if you've not already |
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| 14:37.55 | *** join/#asterisk rgsteele (~chatzilla@12.150.6.65) |
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| 14:41.08 | rgsteele | I've got one VoIP provider with whom I get audio drops several times a day 20-30 secs after the call connects. Upon further investigation, it appears as though it's due to Asterisk sending out a REINVITE (this is while the users are talking, no media stuff such as on-hold or background music is involved). |
| 14:41.43 | rgsteele | Although I have canreinvite=no, I'm aware that doesn't stop all reinvites. Is there any way to force Asterisk to prevent sending ANY reinvites for ANY reason for a specific SIP peer? |
| 14:42.00 | Greenlight | What version are you using ? |
| 14:42.47 | rgsteele | 1.8.11.1-1digium1~lucid on Ubuntu Lucid. |
| 14:42.55 | Greenlight | Then it should be directmedia=no |
| 14:43.24 | Greenlight | I also think session timers will cause reinvites sometimes |
| 14:43.25 | rgsteele | FYI, here's what we see in the logs when it happens: [Apr 4 23:29:35] WARNING[1576] chan_sip.c: just did sched_add waitid(3477831) for sip_reinvite_retry for dialog 1899061346fb931e4bc6082c2bd5bc69@1.2.3.4 in handle_response_invite |
| 14:43.31 | rgsteele | Greenlight: I've disabled session timers |
| 14:44.01 | Greenlight | What about rtptimeout ? |
| 14:44.47 | igcewieling | rgsteele: that is a harmless message and not the cause of any issues you may be having |
| 14:45.03 | Greenlight | Change canreinvite=no to directmedia=no, and see if that helps. |
| 14:45.15 | igcewieling | Though you are welcome to open a bug in jira. harmless messages are not supposed to be warnings |
| 14:45.20 | rgsteele | Greenlight: Cool, will give it a shot! |
| 14:45.35 | mjordan | pretty sure we already killed that 'warning' |
| 14:45.37 | rgsteele | igcewieling: If this doesn't help, I'll probably do that. |
| 14:45.52 | rgsteele | (Assuming I can't find an existing bug and/or it hasn't been fixed in newer revisions) |
| 14:46.13 | igcewieling | rgsteele: setting directmedia=no will stop the message, since the message only happens with directmedia |
| 14:46.47 | mjordan | hm. Nope, we didn't :-) |
| 14:47.36 | mjordan | Although that message does imply that a re-INVITE was sent while another request was pending, so that's a tad odd. |
| 14:47.58 | rgsteele | Yeah, I've tcpdump'ed the SIP traffic, it's definitely due to a reinvite |
| 14:48.07 | rgsteele | As soon as that happens, all audio drops |
| 14:48.28 | rgsteele | End up having to hang up & retry the call. It's intermittent, which is kinda weird |
| 14:48.41 | rgsteele | I just changed it to use directmedia=no, though, so we'll see what happens |
| 14:49.04 | rgsteele | Guess I missed that setting when upgrading this old box (used to be 1.4) to 1.8 |
| 14:53.19 | Greenlight | directmedia can be um .. funky at times |
| 14:53.41 | rgsteele | Does 1.8 still honor the canreinvite option, then? |
| 14:53.50 | rgsteele | I don't see any complaints about it being present in sip.conf |
| 14:54.16 | mjordan | yes. canreinvite simply maps to directmedia. |
| 14:54.47 | rgsteele | So, changing canreinvite=no to directmedia=no probably won't result in any change at all, then. |
| 14:54.52 | mjordan | I would guess that if you've set canreinvite to no, you are getting a re-INVITE for a different reason. |
| 14:55.16 | mjordan | canreinvite never actually squelched re-INVITES completely, hence the name change (also, directmedia actually says what it does, etc.) |
| 14:56.48 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:58.52 | rgsteele | mjordan: So, is there any way I can instruct Asterisk to not send any re-INVITEs for any reason? |
| 14:58.57 | mjordan | you may get a re-INVITE if you have SIP session timers enabled; if a hold frame is processed by one of the channels; if the call was going from audio to T.38 and/or back; in certain transfer scenarios; if party identification is changed; and probably a number of other scenarios that I'm not remembering |
| 14:59.06 | rgsteele | mjordan: I have session timers disabled |
| 14:59.13 | mjordan | rgsteele: nope. It would break pretty much all of the scenarios I've outlined. |
| 14:59.26 | rgsteele | Disabling session timers fixed the drops that were > 30min |
| 14:59.32 | mjordan | well, that's goo |
| 14:59.33 | rgsteele | But, all the <30sec drops are still there :-/ |
| 14:59.33 | mjordan | good |
| 14:59.58 | mjordan | two things to check |
| 15:00.27 | mjordan | (1) what is different in the re-INVITE then the original INVITE? |
| 15:00.43 | mjordan | well, okay one thing. |
| 15:01.06 | mjordan | Without knowing what the re-INVITE is trying to update in the dialog, we're kind of just guessing as to what's going on. |
| 15:01.27 | mjordan | a DEBUG log usually tells you why Asterisk is sending the re-INVITE as well. |
| 15:01.33 | rgsteele | Yeah, I'm trying to find the pcap that had the evidence in it |
| 15:02.19 | *** join/#asterisk BlackIceXS (~BlackIceX@195.189.150.77) |
| 15:02.35 | rgsteele | I just re-enabled debugging on the peer we're having issues with. If I can't find the records I captured a few weeks ago, I'll get some new data to chew on. |
| 15:03.54 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:04.46 | BlackIceXS | hello, is there a way for me to have the Read app to terminate on star instead of the pound key ? |
| 15:04.56 | Greenlight | rgsteele: You might want to enable debug logs while you're at it |
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| 15:10.54 | gttuna | http://pastebin.com/rpsVi6Fe anyone have any ideas what would cause this? |
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| 15:26.33 | danfromuk | Hi, Is it possible to auto-delete old voicemail messages after 30 days? |
| 15:26.41 | mjordan | gttuna: you have an audiohook on a channel (MixMonitor?) and it's not getting any audio. |
| 15:26.43 | mjordan | [2013-04-05 11:06:59] DEBUG[20531] res_rtp_asterisk.c: Received frame with no data for RTP instance '0xb1e78e8' so dropping frame |
| 15:27.30 | *** join/#asterisk RyanTG (~Thunderbi@mailer.americassuppliers.com) |
| 15:27.30 | mjordan | without knowing more about what's occurring, it's hard to say. The 'pretty quick' debug statement is rather low level and is not, on it's own, indicative of any problem |
| 15:27.46 | gttuna | yeah, the commonality between the 3 servers where i'm seeing these errors, is that they're all using MixMonitor to record calls (FreePBX) |
| 15:27.59 | mjordan | what version of Asterisk? |
| 15:28.01 | *** join/#asterisk fkurkowski (~fkurkowsk@177.18.173.170) |
| 15:28.03 | gttuna | 1.8.x |
| 15:28.04 | mjordan | and 32-bit or 64-bit? |
| 15:28.09 | mjordan | not specific enough :-) |
| 15:28.29 | gttuna | 1.8.20.1, 1.8.18.0, and 1.8.13.0 |
| 15:28.43 | fkurkowski | Is there any known issues with Digium cards detecting nonexistent incoming calls? |
| 15:29.00 | mjordan | kk. There was a bug that affected timing in audiohooks ages ago, but the latter two versions are recent enough that it wouldn't be causing the problem. |
| 15:29.33 | fkurkowski | By last two you mean 11 and 10? |
| 15:29.59 | gttuna | mjordan, the 1.8.20.1 and 1.8.13.0 are 32bit |
| 15:30.04 | gttuna | the 1.8.18.0 is 64 bit |
| 15:30.21 | gttuna | errr, 1.8.20.1 is 64 bit |
| 15:30.25 | mjordan | k. The issue I was thinking of only affected 32-bit systems, but it doesn't affect call or audio quality. The recording simply failed. |
| 15:30.45 | mjordan | audiohooks most likely aren't causing you audio quality issues anyway. They intercept the audio after the audio has been received by Asterisk. If you look at an RTP debug (either in a pcap or 'rtp debug on'), do you see a consistent stream of audio coming in for those calls/ |
| 15:30.56 | gttuna | the reason i looked into logs in the first place is that we're seeing some outbound call quality issues |
| 15:31.35 | mjordan | yeah, I don't think this would be the cause of the problem. I'd guess that if you took the MixMonitor off the channels, you'd still have a problem - but it might be worth trying |
| 15:33.19 | mjordan | gttuna: what timing source are you using? |
| 15:33.35 | gttuna | i dont know :( lol |
| 15:34.21 | drmessano | mumbles something about pthread |
| 15:34.25 | gttuna | haha |
| 15:34.30 | drmessano | :) |
| 15:34.42 | WIMPy | 'timing test' |
| 15:35.10 | drmessano | I'm telling you.. If you have another timing source, putting a noload on res_timing_pthread is the best thing you can ever do for an Asterisk box. It's like "timing rehab" |
| 15:35.20 | mjordan | drmessano: +10 |
| 15:35.33 | gttuna | well, how can I check what timing sources im using? |
| 15:35.38 | mjordan | 'timing test' |
| 15:35.40 | gttuna | sorry for being a noob :( |
| 15:35.40 | gttuna | oh |
| 15:35.42 | WIMPy | 'timing test' |
| 15:35.47 | drmessano | 'timing test' |
| 15:35.50 | drmessano | Had to.. |
| 15:36.23 | gttuna | http://pastebin.com/DbEbdPVd |
| 15:36.45 | mjordan | k, well, that's good... although 49 timer ticks is a bit odd. |
| 15:37.21 | gttuna | ive done it a few more times, and gotten 50 each time |
| 15:38.42 | mjordan | k. I'd determine first if the audio coming from the various sources is 'good', that is, does it have jitter, are you seeing dropped/out of order packets, etc. |
| 15:39.32 | mjordan | if all of the sources are good, I'd rule out the MixMonitor next - although I doubt it's causing any issues. |
| 15:40.00 | mjordan | If the sources aren't good, then you can apply a jitterbuffer to the affected channels to see if that gets the audio back in line. |
| 15:40.04 | gttuna | best way to monitor the audio just doing a packet capture ? |
| 15:41.13 | mjordan | pcaps are helpful. Wireshark can analyze an RTP stream. |
| 15:41.28 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
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| 15:42.47 | *** join/#asterisk MalMen (~MalMen@bl22-189-55.dsl.telepac.pt) |
| 15:43.30 | MalMen | hello, i have trixbox installed, now i want to record all calls and calltimes to a database, there is any easy tuturial? i didnt found anything yet |
| 15:43.56 | WIMPy | MalMen: Try #trixbox it's not supported here. |
| 15:44.09 | MalMen | but trixbox not use asterisk for base ? |
| 15:44.33 | WIMPy | Somehow. But you use Trixbox, not Asterisk. |
| 15:44.49 | MalMen | hmmm |
| 15:45.46 | drmessano | Trixbox is not only unsupported, but doesn't use a supported version of Asterisk |
| 15:46.23 | drmessano | It went from being an endangered species to extinct when 1.6.whatever went extinct |
| 15:46.26 | MalMen | i never used trixbox before, i have to try change my boss mind to use asterisk |
| 15:46.42 | WIMPy | Tell him about |
| 15:46.47 | WIMPy | ~trixbox |
| 15:46.47 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 15:47.01 | drmessano | :D |
| 15:47.03 | gttuna | haha |
| 15:48.06 | gttuna | hmmm, should I just capture all UDP packets? |
| 15:48.14 | drmessano | and do what with them? |
| 15:49.22 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:49.26 | gttuna | look at them later with wireshark ? |
| 15:50.17 | gttuna | i mean, i have ports 10000-20000 defined in rtp.conf |
| 15:50.29 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 15:50.55 | gnudna | hi guys / girls |
| 15:51.10 | gnudna | is there an updated document on how to secure asterisk properly? |
| 15:51.39 | gnudna | i have inhereted a system and i would like to make sure we have some form of hardening present |
| 15:52.40 | WIMPy | Try the README-SERIOUSELY.bestpractices.txt |
| 15:53.12 | gnudna | is this online somewhere |
| 15:53.45 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 15:53.58 | *** join/#asterisk work (work@82.137.193.222) |
| 15:54.56 | gnudna | thanks i found it online |
| 15:55.21 | MalMen | so, what is the best web gui for asterisk ? :) |
| 15:55.33 | MalMen | i am searching for alternatives to trixbox |
| 15:55.44 | work | please Mixmonitor I could not understand it I excute Mixmonitor start and it ask me chan and range |
| 15:55.54 | WIMPy | The one you do yourself for YOUR needs. |
| 15:56.08 | MalMen | what is the best choises ? |
| 15:56.37 | WIMPy | I think the only one with a bit of support is #freepbx |
| 15:56.51 | work | please how MixMonitor syntax |
| 15:57.05 | drmessano | MalMen: There's AsteriskNOW and the "FreePBX Distro" which both use FreePBX |
| 15:57.10 | work | on CLI |
| 16:01.16 | gttuna | drmessano, so should I unload res_timing_pthread? |
| 16:04.27 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 16:05.08 | drmessano | I believe I saw that you have timerfd loaded. I would unload pthread.. Yes |
| 16:11.32 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 16:13.25 | rrittgarn | Anybody ever have issues where Polycom phones freeze when registering to an 11.3 box? Moved a few phones from a 10 box to an 11 and now they are un-useable. They register, and will even make a call, but about 6-10 seconds in they just freeze. |
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| 16:34.53 | *** join/#asterisk AkkerKid (~AkkerKid@23.31.20.201) |
| 16:35.16 | AkkerKid | HI ALL! |
| 16:35.47 | WIMPy | AkkerKid: lo you |
| 16:36.15 | igcewieling | rrittgarn: only when the phone can't write to the FTP directories |
| 16:38.46 | *** join/#asterisk Free99 (~Free99@ool-4350dd5c.dyn.optonline.net) |
| 16:57.05 | rrittgarn | igcewieling: unfortunately i have other polycom's pushing logs just fine :-/ |
| 16:58.11 | gttuna | mjordan, hmmm, i guess one of my techs put a jitterbuffer in place already. i have a feeling that may be causing audio issues |
| 16:58.40 | gttuna | if it's incorrectly setup, or the audio didn't have problems to begin with |
| 17:04.55 | igcewieling | rrittgarn: make sure you have the latest of what ever branch of phone firmware you are using |
| 17:12.13 | *** join/#asterisk navaismo (~navaismo@189.191.229.207) |
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| 17:20.01 | Free99 | which variable indicates the current context? Something like ${context}? |
| 17:20.55 | jmetro | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 17:20.59 | WIMPy | Almost: CONTEXT |
| 17:21.01 | jmetro | predefined channel vars |
| 17:27.05 | rgsteele | igcewieling: I set logger.conf to catch everything (verbose, debug, dtmf, etc.), set debugging on the SIP peer associated with the VoIP provider we're seeing the reINVITE issue with, and there wasn't any additional info to chew on. I just turned SIP debugging on globally, and I'm setting up a tshark capture on the interface - hopefully will catch more info about the next one (without... |
| 17:27.06 | rgsteele | ...choking off the throughput...) |
| 17:32.38 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
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| 17:44.50 | Free99 | thanks jmetro & WIMPy |
| 17:48.03 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:49.17 | tompaw | Is it possible to have a per-extension call limit? |
| 17:49.48 | WIMPy | See the GROUP functions. |
| 17:50.59 | tompaw | WIMPy: thanks |
| 17:51.14 | WIMPy | BTW: Anyone have a question worth a domation? I'm broke. |
| 17:52.16 | navaismo | feels better now, not the only one broken here just $5USD in the pockets |
| 17:52.34 | tompaw | WIMPy: priv |
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| 18:25.12 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 18:25.44 | clopez | where I can find documentation about the format/fields that "meetme list $conf number concise" outputs? |
| 18:26.03 | clopez | meetme list $confnumber concise |
| 18:28.37 | clopez | ok.. found it on the source code :D straight to the source |
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| 19:48.09 | SuperNull | what does (d) indicate on 'sip show channels' under hold field ? |
| 19:53.19 | tompaw | Guys what do I need to do for REFER to work? As you can see here: http://tompaw.pl/log.txt it's pretty much ignored by Asterisk 11.2 with no errors given. |
| 19:56.47 | Free99 | If I wanted to print a message to the console from my dialplan, how would I do that? I tried using NoOp("Message"), but that doesn't seem to work |
| 19:58.14 | rrittgarn | Verbose(Message) |
| 19:59.01 | igcewieling | if you core set verbose 3 you'll catch all dialplan lines being run. |
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| 20:00.05 | mjordan | tompaw: it isn't being ignored. Asterisk is rejecting it with a 'declined' |
| 20:00.31 | Free99 | thanks rrittgarn |
| 20:00.38 | mjordan | tompaw: are you trying to do an attended transfer where the target of the attended transfer is not in a bridge? |
| 20:01.44 | tompaw | mjordan: I'd actually like to make a blind transfer |
| 20:03.12 | tompaw | mjordan: we are using Zoiper in the office, and that's what happens when I press "transfer" button. Is ther a way to control what type of transfer is performed upon receiving this message? |
| 20:04.04 | mjordan | tompaw: that's up to Zoiper |
| 20:04.33 | mjordan | the only way we send back a 603 on a REFER request is if the channel associated with the dialog is not in a bridge. |
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| 20:07.44 | twitchnln | afternoon everyone, I was wondering if anyone had a recommendation for a good softphone to run on chromebook |
| 20:10.25 | tompaw | mjordan: understood, thanks. |
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| 20:33.40 | gttuna | mjordan, been looking at packet capture stuff all day |
| 20:34.57 | tompaw | Guys, does anyone else have issues with flowroute? They seem to be sending 3 INVITEs / second to me :/ |
| 20:35.33 | gttuna | i found a call where they had audio issues, and found the corresponding RTP stream in my packet capture. http://i.imgur.com/1JayPUL.png |
| 20:37.33 | jmetro | look at dat ping spike |
| 20:38.16 | gttuna | its weird that i dont see a corresponding spike in the reverse |
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| 20:39.33 | gttuna | i guess with the packets after 140193, they're all coming in at the same (relative) time, so the deltas are all super small? |
| 20:41.06 | mjordan | gttuna: looks that way |
| 20:42.22 | mjordan | having a 4 second latency is ... bad. Jitter buffers aren't going to help you - the standard jitter buffer is (IIRC) is 200 ms. You'll blow right through it. Setting it to account for a 4 second delay would make your standard audio crap. |
| 20:55.12 | jmetro | anyone have a listing of polycom error codes? |
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| 21:02.22 | twitchnln | Anyone know of a softphone that will work on Chrome book? |
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| 21:03.37 | jmetro | linux based. |
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| 21:06.42 | Kazuhiro | Is it possible to make calls via Switchvox (location S) -(IAX)-> asterisk IAX broker server (location Z) -(IAX)-> Asterisk (location A)? |
| 21:07.17 | Kazuhiro | Basically using a asterisk system in a central site to have all remote office IAX links terminate into it? |
| 21:07.57 | Kazuhiro | Currently we have a mesh type topology with each office connecting to each other office's switchvox/asterisk system. |
| 21:08.21 | WIMPy | And you want to add a SPOF? |
| 21:09.51 | Kazuhiro | SPOF that lives on a VMware cluster with DRS/live migration etc. |
| 21:09.55 | igcewieling | jmetro: define "error code" |
| 21:18.12 | jmetro | something like 0x501d |
| 21:23.13 | citywok | Kazuhiro: until the cluster fails, or you want to do something with storage, or anything else that NEVER goes wrong goes wrong. the central office loses connectivity & or power for an extended period of time? |
| 21:23.40 | citywok | the mesh seems nice for those occasions, although probably a bit more work to manage |
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| 21:29.48 | Kazuhiro | The mesh setup currently has issues with bandwidth between office sites, broken IAX links even though they say they're up etc. Currently some offices can call into an office, but the receiving office can't dial back etc. So the plan with the central setup is to enable me to actually debug wtf is going. |
| 21:30.14 | Kazuhiro | Switchvox units are in most remote sites, so very limited in terms of what I can actually look at unless I ask an admin in each site to pop root on it so we can start debugging this crap. |
| 21:31.02 | Kazuhiro | a working setup with a SPOF is better than a partially functional setup that no one can get working correctly. |
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| 21:41.34 | igcewieling | SwitchVox us supported by Digium Support |
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