| 00:03.29 | voxter | is there really just nobody talking in here? |
| 00:04.16 | WIMPy | Not the best time of day. |
| 00:10.36 | jdoe | if you're west coast you'll do better morning to noon-ish. |
| 00:12.00 | jdoe | voxter: should all outbound calls from the pbx be ulaw or just that one specific one? |
| 00:13.27 | citywok | voxter: that's the same experience i've had. it follows top down the allow list, even if it means doing codec translation instead of going to the next codec in the list sans translation |
| 00:13.36 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 00:13.57 | voxter | citywok: yeah. very stupid imo. it seems to prefer the allow preference order instead of finding the path of least resistance |
| 00:14.01 | voxter | ie if a codec matches |
| 00:14.21 | citywok | well for some people that's how they'd rather have it, so i imagine it could be a flag |
| 00:15.25 | citywok | personally i'd rather it go ulaw because i have lots of bandwidth and no compression may sound a tiny bit better. but i do have some phones that i use g729 b/c they are in low bandwidth areas, so g729 passthru would be easier on the server, so in that case i'd prefer it to use g729 instead of ulaw |
| 00:15.55 | voxter | yeah absolutely. |
| 00:16.32 | citywok | so i guess to answer your question i've observed the same behavior, and could understand why it would work the other way, but imagine for that to happen it would need to be made an option in sip.conf |
| 00:18.00 | citywok | this is a channel variable apparently. ${SIP_CODEC}: Set the SIP codec for the inbound (=first) call leg (see channelvariables.txt or README.variables in 1.2); Asterisk 1.6.2 also comes with SIP_CODEC_OUTBOUND for the remote (=second) call leg. |
| 00:20.08 | b14ck | Hey all, is there a dahdi-latest package somewhere on the downloads.asterisk.org site that I'm missing? |
| 00:21.03 | b14ck | Oh wait, I found it: http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz |
| 00:21.06 | b14ck | Nevermind! |
| 00:22.18 | *** join/#asterisk smeet2002 (~smeet2002@dsl-173-248-230-237.acanac.net) |
| 00:22.36 | smeet2002 | hi there |
| 00:26.50 | ChannelZ | OH HAI |
| 00:28.23 | citywok | thar |
| 00:29.40 | leifmadsen | ChannelZ: omghai?! |
| 00:31.51 | ChannelZ | i'm in ur asterisks, modifyin ur extenshuns |
| 00:31.53 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 00:34.08 | joako | Does anyone know where I can download firmware for an Aastra 53i phone? |
| 00:35.43 | shmaltz | joako, did you try the internet? |
| 00:36.32 | shmaltz | http://www.aastra.com/document-library.htm?curr_nav=2 |
| 00:39.30 | citywok | out of curiousity, has anybody ever tried to run * in an EC2 instance? |
| 00:43.14 | joako | shmaltz: Yes and I don't see "53i" phone listed there. |
| 00:44.01 | shmaltz | citywok, list is full of it |
| 00:44.45 | Corydon76-dig | joako: Aastra 6753i |
| 00:44.54 | Corydon76-dig | 53i is just an abbreviation |
| 00:48.45 | citywok | shmaltz: what? |
| 00:48.56 | shmaltz | citywok, ec2 |
| 00:49.03 | shmaltz | check the list |
| 00:49.21 | citywok | asterisk-users i take it you mean? |
| 00:50.41 | shmaltz | citywok, that is correct |
| 00:50.46 | shmaltz | ~lists |
| 00:51.40 | shmaltz | ~list |
| 00:51.40 | infobot | one warez list being sent |
| 00:52.15 | shmaltz | so infobot has no clue what the list is ;) |
| 00:52.18 | shmaltz | :o |
| 00:55.50 | shmaltz | anyone overhere familiar with asterisknow? |
| 00:56.33 | *** join/#asterisk UserReg_CL (~COB@pc-74-76-83-200.cm.vtr.net) |
| 00:56.41 | UserReg_CL | hi !!! |
| 00:58.11 | x86 | ugh |
| 00:59.53 | ChannelZ | !!!!! |
| 01:00.50 | shmaltz | anyone here using asterisknow? |
| 01:01.11 | leifmadsen | probably just the people in #asterisknow :) |
| 01:02.03 | shmaltz | leifmadsen, only 14 ppl in there counting me and the bots |
| 01:03.08 | leifmadsen | true story |
| 01:03.30 | ChannelZ | I ran it briefly in a VM |
| 01:05.21 | UserReg_CL | have asterisk with xlite works fine within my network ... today connect one user xlite from other net (protocol sip) and have problems with sound.. helpme |
| 01:07.09 | shmaltz | ChannelZ ran what? |
| 01:11.06 | ChannelZ | *now |
| 01:13.19 | ChannelZ | UserReg_CL: Was x-lite behing a NAT firewall? If so, is nat=yes set for the peer in sip.conf? Is Asterisk behind a NAT firewall? If so is externip/localnet set properly in sip.conf? And are RTP ports forwarded? |
| 01:14.18 | ChannelZ | BBL going home |
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| 01:29.15 | UserReg_CL | ChannelZ : please need information about port need for xlite... thank ! |
| 01:32.09 | jsolares | lol 425 load average :S |
| 01:32.44 | jsolares | 52mbits of audio in/out |
| 01:33.02 | jsolares | moh is ok, playback is a bit stuttering |
| 01:34.39 | WIMPy | Out of RAM? |
| 01:35.10 | jsolares | 2.20GB used by asterisk out of 24 |
| 01:35.53 | jsolares | too bad there's nothing like iax trunking for sip |
| 01:36.05 | WIMPy | The rest should be enough for some samples, I guess. |
| 01:38.19 | jsolares | using almost 6 cores out of 12, i guess it's just too many udp packets for it to handle ok |
| 01:39.28 | WIMPy | Do you have a kernel that supports receiving on multiple cores? |
| 01:39.57 | jsolares | nope, and that might be a big problem |
| 01:40.36 | jsolares | ubuntu 10.04 worked just fine, rhel 5.5 and 10.10 don't on that regard, but the nic doesn't work correctly on 10.04 |
| 01:40.39 | WIMPy | Ja, I guess that helps a lot. |
| 01:41.47 | jsolares | well we had two machines handling this traffic, one with ubuntu 10.04 the other with rhel 5.5, both crapped out at around the same amount of calls (700) |
| 01:42.15 | jsolares | maybe it's the nic |
| 01:42.29 | WIMPy | What is it? |
| 01:42.37 | jsolares | it was generating almost twice as much interrupts as the other ones |
| 01:43.03 | jsolares | Emulex OneConnect 10Gbps NIC |
| 01:43.42 | jsolares | the other ones have intel gigabits |
| 01:46.17 | jsolares | 914 connected calls |
| 01:46.29 | WIMPy | calls or channels? |
| 01:46.33 | jsolares | calls |
| 01:46.42 | jsolares | 1800+ channels |
| 01:47.41 | jsolares | and still an hour from peak traffic XD |
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| 01:48.46 | boch | hi |
| 02:00.01 | shmaltz | boch, hi |
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| 02:05.33 | dan__t | hrm is MeetMe deprecated? |
| 02:05.53 | WIMPy | Not yet |
| 02:05.54 | dan__t | Looks like I should be using ConfBridge maybe. |
| 02:06.20 | dan__t | its headed that way eh |
| 02:07.29 | dan__t | I liked MeetMe because of the whole password thing. Am I supposed to take care of authentication before sending the call to ConfBridge()? |
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| 03:50.22 | jsolares | well around a 1000 concurrent calls everything started going downhill :( |
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| 07:55.36 | jkroon | hi guys, i'm looking to log some of the rtpqos values. I do, however, have a problem with this, I can get the values on the incoming leg of a SIP call, but not on the outbound leg (I think) via the CHANNEL function. chan_iax2 doesn't seem to have any such values. Other than writing an h extension everywhere where I've got a Dial() function (which I am willing to do if required), is there any way to get to these values from a CDR modu |
| 07:55.37 | jkroon | le in order to log the values? |
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| 08:01.10 | kaldemar | jkroon: IAX2 doesn't use RTP, that's why you don't have such values for the channel. |
| 08:04.37 | kaldemar | jkroon: what version of asterisk are you using? |
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| 09:15.53 | jkroon | kaldemar, using 1.6.2 |
| 09:16.07 | Starscreamer | I have been reading into asterisk recent but can't find an answer to this question... |
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| 09:16.28 | jkroon | kaldemar, yes, i'm aware that iax2 != sip - however, iax2 show netstats shows values that serves a similar purpose - would be useful to be able to log those values for iax2 at the end of a call and the rtp values for sip at the end of a call. |
| 09:17.16 | Starscreamer | I have a Mitel VOIP PBX server and Mitel handsets. Is it likely that Asterisk can replace this mitel server box? Is it user configurable or is it the kind of thing that would need an expert to set up? |
| 09:17.57 | ChannelZ | A little of both, but the more imporant thing would be what language do the Mitel handsets speak |
| 09:18.54 | ChannelZ | If they can speak SIP then you're in business.. if it's some propriatary thing, well, not so much. |
| 09:19.14 | Starscreamer | I'll find out |
| 09:19.34 | Starscreamer | I was hoping for an Asterisk IRC chat to answer the question ha |
| 09:20.16 | ChannelZ | Well considering we don't even know the model numbers of what you have.. I'm not sure what definitive answer you are expecting |
| 09:20.42 | ChannelZ | So at the moment the answer to your quesiton is "Possibly/maybe" |
| 09:20.46 | Starscreamer | I was going to google it but anyway: Mitel IP Phones 5212 and 5224 |
| 09:23.30 | ChannelZ | According to the user guide: http://edocs.mitel.com/UG/EN/33_5212_5224_UG_R7_EN.pdf it looks like yes |
| 09:31.39 | kaldemar | jkroon: don't know of a neat way to get the IAX stats in the dialplan, only ugly ones. sounds like using a gosub/macro for dialing would suit your purpose. |
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| 09:35.04 | Starscreamer | apologies, I got disconnected |
| 09:35.20 | Starscreamer | so in theory, as long as the headsets support SIP they can connect to Asterisk? |
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| 09:36.04 | Starscreamer | Although I do support our existing Mitel system, it's very expensive for maintenance and even initial purchase was very expensive. I imagine it's just a linux box with an ISDN controller built in. |
| 09:36.44 | Starscreamer | I only do basic maintenance but I assume it might not be too difficult for a novice to configure Asterisk if they have a Mitel system config to reference to |
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| 09:40.48 | joachim_- | I have a problem. When I dial 100 on my asterisk, it forwards to Dial(SIP/101&SIP/102&SIP/103) But then my CDR records shows 100 as the destination number. How can I set the callerid for CDR after the call has been answered? |
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| 09:42.38 | Flumdahl | http://pastebin.com/JG3qLLxC anyone that can explain why my asterisk wont call my cellphone if i dont answer the home phone when incoming call is made? it works when i call from another extension but not from a friends home phone |
| 09:44.01 | joachim_- | I have a problem. When I dial 100 on my asterisk, it forwards to Dial(SIP/101&SIP/102&SIP/103) But then my CDR records shows 100 as the destination number. How can I set the callerid for CDR after the call has been answered? |
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| 09:46.35 | kaldemar | joachim_-: destination is the number that was dialed from asterisk's point of view. that is, the extension. which is 100 in your case. does it not set the dstchannel correctly? |
| 09:47.12 | kaldemar | Flumdahl: ask in #freepbx |
| 09:47.54 | Flumdahl | i have asked there to |
| 09:49.56 | joachim_- | kaldemar: No, after answering the call, the dst in CDR records is set to 100 automatically. Im using a realtime odbc setup for enterprise virtual extensions, So 100 is just a Virtual extensions that calls other sip clients |
| 09:51.09 | joachim_- | kaldemar: And ive set up my dialplan to run a macro upon answering the call, but i dont know what to set in the macro to point out the real callerid after answering |
| 09:51.55 | kaldemar | joachim_-: it is supposed to be set to 100. and it is not the caller id. whose caller id are you talking about? |
| 09:52.51 | kaldemar | the CDR has a separate field for caller id. |
| 09:53.17 | joachim_- | when i call 100, it calls several sip devices, I want to set the callerid for the device that answers the call. |
| 09:53.33 | joachim_- | so that my cdr records know which one who answered |
| 09:53.56 | kaldemar | CDR has a field called dstchannel. what does it have? |
| 09:56.09 | joachim_- | kaldemar: dst=100 dstchannel=SIP/trunkname-00000010 |
| 09:58.07 | kaldemar | when dialing Dial(SIP/101&SIP/102&SIP/103)? that's got to be the "channel", not dstchannel. |
| 10:00.24 | joachim_- | No, channel is: SIP/102-0000000e |
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| 10:01.39 | joachim_- | dstchannel says: SIP/trunkname-00000010 |
| 10:02.07 | kaldemar | feel free to pastebin a cli output of a call and the CDR entry. |
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| 10:09.03 | kaldemar | at least my cdr has entries that have the answered party as the dstchannel in a case when multiple channels are dialed. |
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| 10:20.49 | FlashDeluxe | hi! I have a question, if i connect to my asterisk manager and execute the command Originate, it says, that the command doesn exist anymore, but in the cli the command does exist, has it been replaced? |
| 10:23.03 | FlashDeluxe | i am using asterisk 1.8.1 |
| 10:23.45 | kaldemar | 1.8.1 lists it with manager show commands. |
| 10:24.20 | brast | hello all, I am having some problems with AEL2...sometimes ael reload produces "Mismatched ')' in expression!" but if I issue another ael reload everything is fine....this happens once every 10-15 reloads |
| 10:24.31 | brast | has anyone dealt with such an issue before ? |
| 10:25.18 | FlashDeluxe | kaldemar: yes and in the cli there is the command, but if i telnet my manager interface the command doesn exist |
| 10:25.18 | Nugget | telnet is eeeeeeevil! |
| 10:28.14 | kaldemar | FlashDeluxe: the cli command is separate from manager originate. it works here. how are you making the originate? |
| 10:28.32 | kaldemar | Action: Originate \ Channel: ... ? |
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| 10:29.30 | FlashDeluxe | with a php script, i first set variables and callerid and then at the end i make Action: Originate |
| 10:30.01 | kaldemar | your script must do something wrong. be more precise. |
| 10:30.59 | FlashDeluxe | an Action: Listcommands doesn show the originate either |
| 10:31.32 | FlashDeluxe | here s the script |
| 10:31.33 | FlashDeluxe | http://pastebin.de/12828 |
| 10:31.38 | kaldemar | yes it does. does your manager user have privileges to use Originate? |
| 10:32.28 | FlashDeluxe | it does have read/write = system,call,log,verbose,command,agent,user |
| 10:33.12 | kaldemar | Originate: Originate a call. (Priv: originate,all) |
| 10:34.00 | kaldemar | add originate to write and try again. |
| 10:35.11 | FlashDeluxe | ok |
| 10:36.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:36.37 | FlashDeluxe | wow, now its there =) |
| 10:36.54 | FlashDeluxe | Thanks kaldemar! |
| 10:37.00 | kaldemar | you're welcome. |
| 10:46.50 | brast | can anyone take a look at this ael2 weirdness: http://pastebin.com/bd0Nq4LE please ? |
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| 10:57.06 | cmnky | gm phone geeks |
| 10:58.47 | cmnky | {term of affection ... not derogatory} |
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| 13:11.19 | schmidts | hello |
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| 13:13.40 | DeHackEd | anyone know of a software fax machine capable of better than 14400 ? (currently using iaxmodem) |
| 13:13.57 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 13:14.30 | cmnky | DeHackEd, i have an SPA2102 .. i think it does fax at 28.8 |
| 13:14.47 | DeHackEd | the cisco/linksys ATA |
| 13:14.48 | DeHackEd | ? |
| 13:14.52 | cmnky | DeHackEd, i know thats not exactly the question you asked tho |
| 13:15.04 | cmnky | DeHackEd, yeah, its a linksys, got it off ebay |
| 13:15.22 | DeHackEd | yeah. I have one, but that's a completely unrelated matter |
| 13:15.43 | cmnky | ah, i get it now (i think) |
| 13:16.16 | DeHackEd | I do have the ATA for my regular voice, but for fax I'm using iaxmodem and asterisk's faxdetect |
| 13:16.22 | DeHackEd | and yes, the network is fast and all that |
| 13:16.50 | cmnky | i assumed it was ;) |
| 13:16.52 | *** join/#asterisk nzw (~rychu@85.11.67.125) |
| 13:17.04 | nzw | hi |
| 13:17.36 | nzw | is there a possibility to change CID, but leaving original CID in CDR? |
| 13:18.43 | nzw | i'm trying to do something similar to CALLERID(pres)=prohib |
| 13:19.11 | nzw | but setting CID to something else than anonymous (original CID stays in CDR) |
| 13:19.43 | nzw | any clues ? |
| 13:23.43 | cmnky | DeHackEd, wouldnt asterisk allow you to use a hardware fax modem that supported the higher speed ? |
| 13:24.36 | DeHackEd | in theory... but I don't have one. actually I need a whole bunch. that's why I'm going with software. |
| 13:24.57 | cmnky | gotcha |
| 13:25.16 | DeHackEd | If I'm stuck with 14.4, that's fine. It would be nice to upgrade though. |
| 13:26.00 | cmnky | you could get 1 hardware fax modem, and make it the primary, and run at full speed, then use the slower as fallbacks/cascades |
| 13:26.19 | cmnky | but if your bank of faxes is saturated i suppose it would have marginal benefits |
| 13:27.16 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:27.16 | DeHackEd | It's in the testing phase right now. I don't know how busy it will be when deployed for real... |
| 13:35.21 | cmnky | DeHackEd, looks like its about $100-$110 for a 33.6 USB/RS232 hardware fax modem |
| 13:35.47 | cmnky | the cheap ones max out at 14.4 for fax |
| 13:36.58 | *** join/#asterisk ddickenson (~ddickenso@166.205.8.59) |
| 13:37.00 | DeHackEd | serial, USB or onboard? serial and PCI ports are at a premium, and I've had USB modems flake out on me a lot |
| 13:37.56 | cmnky | DeHackEd, im looking on newegg.com ... search for 'modems' .. sort by price ... multitech has either RS232 or USB models at round $110 |
| 13:38.43 | cmnky | i assumed PCI slots were filled .. but they have that option as well |
| 13:39.15 | ddickenson | Hello all, I have come across something I have never seen before in the 6 or so installations I have running currently. In the call queue I will randomly have different dahdi channels show up as dynamic agents and I don't know what would be causing that. These channels are parts of my incoming T1. Any ideas? |
| 13:40.16 | russellb | ddickenson: sounds like a dialplan error |
| 13:40.27 | russellb | where incoming calls are somehow making it to the extensions used to add queue members |
| 13:41.01 | ddickenson | Hmmm, that could be a possibility, I didnt think of that, thanks |
| 13:41.10 | russellb | sure, no problem |
| 13:41.15 | russellb | I can't think of any other way that could happen |
| 13:41.24 | russellb | unless you had a rogue AMI script running that was doing it :-) |
| 13:41.49 | ddickenson | Haha, hopefully not. The other makes sense though |
| 13:43.14 | russellb | If you can't find it, enable the full log in /etc/asterisk/logger.conf and wait for it to happen again, then search for AddQueueMember or whatever it is |
| 13:43.30 | ddickenson | I see now, I had intended to make it a separate context but missed that and it's right behind my time routing rules so I guess something is falling through from that |
| 13:43.32 | russellb | and then hopefully you'll find a call that got messed up and perhaps you can see how it got there |
| 13:43.40 | russellb | ah :-) |
| 13:43.57 | russellb | yes, contexts are a good thing |
| 13:43.58 | ddickenson | Thanks a million that was getting annoying |
| 13:44.04 | russellb | sure, glad I could help |
| 13:44.58 | russellb | luckily the channel name isn't enough to call the original caller back in this case |
| 13:45.02 | russellb | or that could have been interesting |
| 13:45.41 | ddickenson | Ha |
| 13:45.49 | ddickenson | Yeah |
| 13:47.01 | *** join/#asterisk anebi (~anebi@91.207.191.17) |
| 13:47.36 | ddickenson | You know the odd thing is I'm not sure how it would fall through as the last statement I have prior to those extensions to add queue member is a goto statement that sends anything remaining to a "night" context for after hours calls |
| 13:48.54 | ddickenson | Unless there are some DID's coming across my t1 that I'm not aware of! |
| 13:49.28 | russellb | dun dun dunnnnn |
| 13:50.06 | ddickenson | Weird... I may check the logs anyway |
| 13:53.24 | anebi | Hi. I have a question related with asterisk and outbound proxy. My boss's phone is behind a router and on the router is installed milkfish. The phone has set outbound proxy to the router's ip address, so it uses milkfish as outbound proxy. STUN is not set. My boss is complaining that he does not have audio on inbound and outbound calls. My question is - does he need to set as outbound proxy asterisk server's ip address and what are the benefits of doing this |
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| 14:01.53 | Chainsaw | anebi: Easiest fix is to deploy STUN. |
| 14:03.43 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 14:04.22 | anebi | Chainsaw: yes, i think so too. but anyway i want to understand if there is some benefits of using asterisk server as outbound proxy and what we get actually if we use outbound proxy at all |
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| 14:07.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:10.26 | Chainsaw | anebi: Working connections? |
| 14:10.44 | Chainsaw | anebi: It's a proxy server. It sits in between two end-points and it forwards their traffic, because it can't go direct. |
| 14:11.10 | Chainsaw | anebi: Quick fix? STUN! |
| 14:11.24 | anebi | Chainsaw: thanks. |
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| 14:23.08 | iulhk | trying to install zaptel-1.4.12.1. when trying "make menuconfig" it says at the end "Install ncurses to use the menu interface" although i hv already installed lib64ncurses5 libncurses-ruby1.8 libncurses5 libncurses5-dev libncursesw5 ncurses-base ncurses-bin ncurses-term ? |
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| 14:25.17 | *** join/#asterisk cgillogly (~cgillogly@unaffiliated/cgillogly) |
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| 14:41.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:41.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:51.42 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 14:52.22 | i_heart_asterisk | asterisk by default plays SLIN voicemail files, how do you change that to GSM ? |
| 14:58.57 | pabelanger | save them as gsm? |
| 15:00.15 | i_heart_asterisk | no, play GSM instead of SLIN |
| 15:00.41 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 15:00.50 | i_heart_asterisk | the voicemail menu is playing SLIN files, I need it to play the GSM files instead |
| 15:03.01 | i_heart_asterisk | nevermind i figured it out, i jsut deleted the *.g722 and *.wav files |
| 15:11.40 | tuxx- | hi guys, does the application gotoif support a condition with an AND? @ voip-info.org i see that it supports the | for OR, but i wonder if there is an AND (&) operator too? |
| 15:21.20 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.103) |
| 15:22.59 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 15:23.06 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:24.14 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:25.10 | tuxx- | nvm, fixed it with agi :-) |
| 15:26.11 | FlashDeluxe | Hi! I have a question, i have a script written in pcp, which logs into the manager and executes a few commands (http://pastebin.de/12828) but if i execute the script i get an error 'ERROR[14781]: utils.c:1178 ast_careful_fwrite: fwrite() returned error: Broken pipe' If i do it manually with telnet, everything works fine, does anybody have a suggestion? |
| 15:27.08 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 15:29.50 | Chainsaw | FlashDeluxe: PCP? What language is that? |
| 15:30.18 | FlashDeluxe | oh, php sorry Chainsaw |
| 15:30.24 | WIMPy | FlashDeluxe: Maybe you should read sime time inbetween or flush the receive buffer? |
| 15:30.41 | Chainsaw | FlashDeluxe: Note that PHP scripts tend to end when they've drawn the page. Are you suddenly hanging up on Asterisk? |
| 15:32.02 | FlashDeluxe | Chainsaw: yes |
| 15:32.22 | WIMPy | The script not finishing execution would be a good point indeed. |
| 15:32.25 | FlashDeluxe | but asterisk itself doesnt hangup |
| 15:32.35 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:32.47 | Chainsaw | FlashDeluxe: Because it is amazed at your rudeness. |
| 15:32.52 | Chainsaw | FlashDeluxe: The shock takes some time to clear. |
| 15:33.02 | Chainsaw | FlashDeluxe: Hang up on it properly, like a gentleman. It will treat you better. |
| 15:33.06 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:33.35 | FlashDeluxe | Chainsaw: but it hangs up BEFORE any commands are executed |
| 15:34.07 | WIMPy | Are you sure it is Asterik ending teh connection? |
| 15:34.24 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 15:34.26 | FlashDeluxe | yes i am |
| 15:34.26 | Chainsaw | FlashDeluxe: Your main problem is that you are not having a conversation with Asterisk. |
| 15:34.33 | Chainsaw | FlashDeluxe: You are shouting commands down the line and hoping for the best. |
| 15:34.36 | Chainsaw | FlashDeluxe: Telnet isn't like that. |
| 15:34.43 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:35.04 | Chainsaw | FlashDeluxe: AMI sends replies. Always read the reply in and act on it. |
| 15:35.18 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:35.33 | Chainsaw | FlashDeluxe: You are sending a login, which might be wrong. Or it likes it, and you may proceed. It is telling you. Listen to it. |
| 15:35.41 | cmnky | mozilla update, 13 holes plugged, 3.6.13 has fixes |
| 15:36.13 | FlashDeluxe | Chainsaw: the user logs in and then the error appears and the user logs out, thats all |
| 15:36.34 | FlashDeluxe | Chainsaw: i do not see more, with asterisk 1.4 everything works fine |
| 15:36.38 | Chainsaw | FlashDeluxe: I have tried to explain the problem in a friendly, elaborate manner. |
| 15:36.55 | Chainsaw | FlashDeluxe: You are doing the same thing to me as what you are doing to Asterisk. That is, talking too much and listening too little. Best of luck! |
| 15:38.05 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:38.26 | FlashDeluxe | Chainsaw: No, i would listen if the ami would tell me something, but it doesnt :( |
| 15:38.34 | Chainsaw | FlashDeluxe: Broken pipe. |
| 15:43.10 | FlashDeluxe | Chainsaw: If i uncomment everything in the script, execpt the login i get the same error, broken pipe and after that the manager logs off and i dont know what i can do on that issue :( |
| 15:43.21 | Chainsaw | FlashDeluxe: Broken pipe. |
| 15:44.14 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 15:44.51 | FlashDeluxe | Chainsaw: So can you give me a little hint what i can do to get rid of Broken pipe? Please |
| 15:45.52 | Chainsaw | WIMPy: Well, I've done what I can. I think I may have to /part #asterisk if this continues. |
| 15:46.14 | jsolares | broken pipe means your connection is down when asterisk is trying to send you something |
| 15:46.24 | jsolares | ie your script is terminating the connection before asterisk can even send you anything |
| 15:47.43 | jsolares | now i have a question, if i dial out to multiple destinations, how can i know which one actually connected? |
| 15:48.08 | WIMPy | Where? |
| 15:48.34 | jsolares | dahdi/g0/onenumber&dahdi/g0/anothernumber |
| 15:48.34 | WIMPy | I mean where do you want to find out or do something about it? |
| 15:49.04 | jsolares | after the dial in the dialplan would be nice, so i can write it to the userfield |
| 15:49.37 | WIMPy | I haven't been able to try it, yet, but DAHDI is supposed to support COLP. |
| 15:50.00 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 15:50.07 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 15:50.36 | WIMPy | Not sure if you can find something like the channel name after the dial, however. |
| 15:50.38 | nny | so.. just so I don't feel like an idiot. A T1 (and t1 issues) would be pretty straight forward as to what kind of issues arise from where. |
| 15:50.43 | nny | example, volume or static |
| 15:50.47 | nny | analog issue, no? |
| 15:50.50 | nny | since t1 is a data link |
| 15:51.29 | nny | sorry to bombard, have a business partner pulling his hair out over provider issues (calls not hitting the t1 *at all* and call quality issues) and my gut says "upstream" |
| 15:51.48 | WIMPy | Data as in digital. Otherwisr I don't understand your question. |
| 15:52.08 | nny | i guess what I mean is things like static aren't created on the digital link |
| 15:52.20 | WIMPy | Are you talking about a PRI? |
| 15:52.25 | nny | yes |
| 15:52.25 | jsolares | well the channel name after the fact gets stored in dstchannel, but not the actual dialed number :( |
| 15:52.47 | nny | WIMPy: and I mean teloc equipment -> (here) PBX |
| 15:52.51 | nny | telco* |
| 15:53.11 | jsolares | is the static on *ALL* calls out of the pri? |
| 15:53.26 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:53.38 | nny | jsolares: no |
| 15:53.42 | nny | jsolares: certain LD |
| 15:53.43 | nny | :D |
| 15:53.46 | jsolares | volume you can adjust with tx/rx gain |
| 15:53.55 | nny | jsolares: not volume on every call |
| 15:54.03 | nny | jsolares: some just lose quality over time |
| 15:54.19 | jsolares | well then unfortunately or fortunately whichever way you want to look at it XD, it doesn't seem to be your end |
| 15:55.15 | nny | jsolares: lol I think it's schadenfreude at this point. I have seen it happen so many times now, and the carriers always say "it's not our problem" over and over, until you either get rid of them or find some way to convince them it is |
| 15:55.34 | nny | and then another issue comes up, and it's "not their problem" all over again |
| 15:56.16 | jsolares | hehe, we only have 3 carriers down here, so it's alot simpler, but if you have an issue you're cut off with 1/3 of the country XD |
| 15:56.18 | WIMPy | It's probably indeed not them but some of they'r suppliers. But from your point of view that's surely their problem. |
| 15:57.48 | nny | WIMPy: yeah I understand that, just wish i didn't have to charge the customer to prove it, they'd never admit it |
| 15:58.22 | FlashDeluxe | jsolares: Ok thanks for your help! I told the script to sleep for a second and now it works |
| 15:58.59 | nny | thanks for the conversation, just wanted to make sure i wasn't crazy |
| 15:59.08 | WIMPy | Sleep? Ok, so Chainsaw was right. Looks like you must not send commands before being logged in. |
| 15:59.10 | iulhk | trying to install zaptel-1.4 getting this error "Leaving directory `/usr/src/linux-headers-2.6.32-24-generic-pae'"? |
| 15:59.12 | *** join/#asterisk Tim_Toady (~moi@188.4.146.42.dsl.dyn.forthnet.gr) |
| 15:59.25 | jsolares | see chainsaw was right, you weren't waiting for confirmation on what you sent |
| 15:59.33 | nny | iulhk: does that directory exist? |
| 15:59.45 | nny | iulhk: or rather, do you have the source for your kernel installed? |
| 16:00.01 | jsolares | nny: audio problems does tend to make you want to pull out your hair XD |
| 16:00.04 | nny | (check with uname -a, then look in /usr/src/kernels or something to that effect |
| 16:00.29 | FlashDeluxe | but chainsaw could have been a little bit more verbose, just for a better undestanding :/ |
| 16:00.50 | nny | jsolares: yeah, sometimes it's such a simple concept though. The types of issues that can exist out of POTS or analog links are so vastly different from the digital side |
| 16:00.55 | jsolares | time to see what variables dial passes to the macro executed with M(), or go poking around the code and set a variable with the connected number in app_dial |
| 16:00.58 | WIMPy | I think he told quite exactely what was wrong. |
| 16:01.28 | jsolares | FlashDeluxe, he said the exact same thing i did, but with different terms :P |
| 16:01.39 | WIMPy | jsolares: I can't find any information on how to access COLP information (if available at all) from the dialplan. |
| 16:01.51 | nny | jsolares: we replaced a PRI card due to some calls not making it to the PBX *at all* which I know would be pretty tough to call the card out on. It passed all zttool, etc testing, and as I put it, "That's like your ethernet card not being able to go to google" as far as OSI layers go |
| 16:02.11 | FlashDeluxe | jsolares: but for beginners like me, your terms are much better to understand |
| 16:02.16 | jsolares | nny, and that fixed the problem? |
| 16:02.32 | jsolares | FlashDeluxe, i actually thought his were more beginner like XD |
| 16:02.41 | Chainsaw | FlashDeluxe: The patience on the Chainsaw is not infinite. I could work like Fender and start shouting... |
| 16:02.45 | Chainsaw | FlashDeluxe: But I doubt you'd enjoy that very much. |
| 16:03.04 | nny | jsolares: still testing with the new card. |
| 16:03.09 | jsolares | a Chainsaw can cut you, painfully XD |
| 16:03.13 | nny | jsolares: honestly, i doubt it |
| 16:03.14 | Chainsaw | jsolares: Thanks. I was about to say, what simpler words could I use :) |
| 16:03.41 | nny | jsolares: Chainsaw can also make ice sculptures and be juggled! |
| 16:04.15 | Chainsaw | nny: Did you see the Time Warp special on 3 chainsaw juggling? It was epic :) |
| 16:04.55 | nny | watching now heh |
| 16:05.42 | nny | showing that guy his act in slow motion is gonna be the cause of a lost arm. Oh I could do that better! SLICE |
| 16:05.59 | nny | some things are better left to gut instinct lol |
| 16:07.42 | *** join/#asterisk rbd (~rbd@cpe-024-163-112-027.nc.res.rr.com) |
| 16:08.45 | rbd | I'm using Linode and Flowroute toll free DIDs and am getting busy tones sometimes when dialing in. anyone had weird call quality (busies, jitter, etc) when using asterisk on VPSes (especially with meetme?) |
| 16:10.31 | *** join/#asterisk raha613 (~Andy@205.197.201.98) |
| 16:10.44 | raha613 | hey all |
| 16:11.17 | *** part/#asterisk raha613 (~Andy@205.197.201.98) |
| 16:11.50 | kaldemar | iulhk: that is not an error. it's just an indication that the compilation/installation procedure was doing something in that directory and is moving on. |
| 16:12.20 | kaldemar | iulhk: why are you trying to use zaptel instead of DAHDI? |
| 16:14.16 | jsolares | i wonder if DIALEDPEERNUMBER has what i need when using dahdi, time to go find out :D |
| 16:14.28 | *** join/#asterisk raha613 (~raha613@205.197.201.98) |
| 16:15.14 | raha613 | hello all |
| 16:15.52 | WIMPy | raha613: Hello you |
| 16:16.09 | raha613 | WIMPy: Ever used asterisk realtime? |
| 16:16.19 | WIMPy | no |
| 16:17.01 | raha613 | Has anyone every used asterisk realtime? |
| 16:17.16 | kaldemar | yes, many have. |
| 16:17.19 | WIMPy | ~ask |
| 16:17.19 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:17.24 | kaldemar | if you have an issue with it, just ask. |
| 16:18.08 | raha613 | Everything looks like it is working and Asterisk is quering the database but for some reason the query syntax is incorrect... i keep getting |
| 16:18.13 | raha613 | "Failed to query database" |
| 16:18.25 | raha613 | You have an error in your sql syntax |
| 16:20.34 | kaldemar | check that your database table has all required fields. |
| 16:20.56 | raha613 | I am using A2billing on a 3 servers |
| 16:21.08 | raha613 | Table should already have fields |
| 16:22.40 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 16:22.50 | raha613 | Its sending a non specific query to the database |
| 16:23.25 | raha613 | Asterisk Debug: [Dec 14 10:22:45] WARNING[28750]: res_config_mysql.c:375 realtime_mysql: MySQL RealTime: Failed to query database: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '-sip-buddies WHERE name = ''' at line 1 |
| 16:24.01 | raha613 | MySql Query Debug: 101214 10:15:44 140 Init DBmya2billing |
| 16:24.02 | raha613 | <PROTECTED> |
| 16:24.02 | raha613 | <PROTECTED> |
| 16:24.02 | raha613 | <PROTECTED> |
| 16:24.35 | raha613 | its not going to retun anything where name = '' because all sip peers have a name |
| 16:29.21 | *** join/#asterisk Dovid (~Dovid@ool-4355e297.dyn.optonline.net) |
| 16:29.42 | *** join/#asterisk hmmhesays (~hmmhesay@24-116-107-203.cpe.cableone.net) |
| 16:29.58 | Dovid | if I send a=direction:active. that means that I will send RTP first ? |
| 16:30.36 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 16:30.52 | *** join/#asterisk arnotixe (~arnotixe@cl-205.udi-01.br.sixxs.net) |
| 16:31.39 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 16:32.20 | arnotixe | I've got only one PSTN line for external access. Sometimes the line is busy and people of course get a busy tone from asterisk. So people think the remote end is busy, while it's really our own line that's busy. Is there a way to give them a "the external line is busy, try in a few moments" message? |
| 16:32.43 | *** join/#asterisk Tim_Toady (~moi@188.4.146.42.dsl.dyn.forthnet.gr) |
| 16:34.34 | xSmurf | hey, I have a maybe odd question. Is there a way to initiate a call to some extension when someone joins a conference? Basically we have a conference phone and I want it to ring right away if someone joins this conference |
| 16:35.49 | Dovid | does Asterisk not send RTP until it gets some first ? |
| 16:38.00 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 16:38.09 | SuPrSluG | arnotixe: look into DIALSTATUS |
| 16:39.09 | Dovid | it seems that if there is no 183 Asterisk will not send the rtp stream first |
| 16:39.33 | jsolares | sucess DIALEDPEERNUMBER has what i need after dialing multiple destinations |
| 16:45.53 | SuPrSluG | i don't think either side will send RTP until a 200 ok is received |
| 16:46.40 | WIMPy | I do get audio after 183. |
| 16:47.04 | WIMPy | Otherwise I wouldn't get external dialtone or announcements. |
| 16:50.47 | jsolares | Dovid if you need asterisk to send audio before answering a channeling, use Progress |
| 16:51.09 | jsolares | channeling... lol i need coffee |
| 16:51.46 | SuPrSluG | early media? |
| 16:52.15 | *** join/#asterisk akoma1s (quasselcor@metadosis.gr) |
| 16:52.17 | brast | hello all, I am having some problems with AEL2...sometimes ael reload produces "Mismatched ')' in expression!" but if I issue another ael reload everything is fine....this happens once every 10-15 reloads |
| 16:52.32 | brast | can anyone take a look at this ael2 weirdness: http://pastebin.com/De4he2VC please ? |
| 16:53.06 | SuPrSluG | yes. there is no rtp until a 200 is received |
| 16:54.39 | brainiac | hello, does anyone know how I can fix a one-way audio problem with an A200D card? |
| 16:55.45 | arnotixe | SuPrSluG, ok thanks I will. Just came to think - maybe I can just use the "j" flag to the dial command? I call out with 'exten => _5.,1,Dial(SIP/${EXTEN:1}@940)' where 940 is the SIP-PSTN adapter. |
| 16:55.55 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 16:56.59 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 17:01.46 | *** join/#asterisk CraigW76 (~techcaw@cl-427.dal-01.us.sixxs.net) |
| 17:03.27 | SuPrSluG | brainiac: sangoma cards are well supported. Since NAT is not an issue, I would start with the config files |
| 17:04.08 | SuPrSluG | unless your phone are going through NAT to get to the A200D |
| 17:13.05 | *** join/#asterisk p3nguin (~pwne@cl-587.chi-02.us.sixxs.net) |
| 17:17.45 | Dovid | if i have 3 audio cards i will have /dev/snd /dev/snd1 etc ? |
| 17:22.13 | p3nguin | You'll probably have /dev/snd/ just like if you have one card, but you'll have more device files under /dev/snd/ with more cards. |
| 17:22.43 | p3nguin | hwC0D0 hwC1D0 hwC2D0, for example |
| 17:27.22 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 17:34.29 | *** join/#asterisk dajhorn (~dajhorn@adsl-71-158-166-44.dsl.rcsntx.sbcglobal.net) |
| 17:34.44 | *** join/#asterisk nachoguy (~boster@hofner.dreamhost.com) |
| 17:36.04 | nachoguy | 1.4.38 wasn't supposed to be released until https://issues.asterisk.org/view.php?id=18130 was addressed. Does anyone know if it is still going to be looked at? |
| 17:37.24 | russellb | nachoguy: it's pretty high on our priority list, but not at the top. If I were a betting man, I'd say we'll get to it by the end of January. |
| 17:39.40 | rbd | if there are multiple headers (e.g. Via: ) does SIP_HEADER() return the first one it sees? |
| 17:49.25 | dajhorn | How can I get the compile-time options out of the closed chan_skype.so module? |
| 17:49.31 | nachoguy | russellb, Thanks! |
| 17:58.43 | *** part/#asterisk nachoguy (~boster@hofner.dreamhost.com) |
| 18:00.22 | *** join/#asterisk fofware (~Fabian@host210.190-225-14.telecom.net.ar) |
| 18:06.43 | *** join/#asterisk ccesario (~ccesario@linux.unialco.com.br) |
| 18:11.46 | dajhorn | Ah, you can't. The "compile-time options" in a module are actually an MD5 that are tested by the Asterisk loader. |
| 18:12.32 | *** join/#asterisk martinblr (3d0c11aa@gateway/web/freenode/ip.61.12.17.170) |
| 18:24.15 | bn-7bc | hmm is my ISP routing incorectly or is aserisk.org down today? |
| 18:24.30 | tvc123 | up for me |
| 18:24.40 | WIMPy | Works for me |
| 18:24.50 | bn-7bc | hmm not goos |
| 18:24.54 | bn-7bc | thanks |
| 18:25.09 | tvc123 | ip address 216.207.245.33 if that helps |
| 18:26.27 | pabelanger | bn-7bc: http://downforeveryoneorjustme.com/ |
| 18:29.35 | bn-7bc | well that proves the routing problem, thay is theexact ip i got from dns |
| 18:29.42 | bn-7bc | tankseveryone |
| 18:30.11 | *** join/#asterisk darkskiez_ (~dz@62.50.248.69) |
| 18:30.31 | WIMPy | wonders when the first phone will come out where you have to connect a mouse to use it. |
| 18:32.51 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 18:33.37 | *** join/#asterisk bkruse (~Brandon@75.76.105.124) |
| 18:36.28 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 18:44.10 | pabelanger | does iax2.conf support templates? |
| 18:47.06 | *** join/#asterisk gnarf (~gnarf@c-98-213-203-220.hsd1.il.comcast.net) |
| 18:47.18 | gnarf | anyone here use chan_sccp-b and have the conference buttons working? |
| 18:47.23 | gnarf | I'm having some issues with it |
| 18:48.16 | gnarf | also, does anyone know if the phone supports blind transfers? |
| 18:49.11 | pabelanger | I guess, no, templates are not supported |
| 18:53.41 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
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| 19:57.00 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 19:59.45 | candrews | My asterisk server is behind NAT. My soft phone inside the firewall works great with it (incoming/outgoing calls work). Outside the NAT, the phone rings / can place calls, but doesn't get any audio, and disconnects seemingly randomly. I have ports 5061, 5061, and 10000-20000 forwarded to the asterisk server |
| 20:01.26 | candrews | I've also configured STUN in res_stun_monitor.conf |
| 20:03.22 | *** join/#asterisk n1gak (~lawrence@189.162.186.156) |
| 20:04.21 | n1gak | hello all |
| 20:05.12 | n1gak | I'm having trouble getting inbound calls working from a DID vendor - I want to do some testing w/ someone to verify that I have my configuration more-or-less correct before squawking to the vendor |
| 20:05.41 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:09.28 | n1gak | ping? |
| 20:10.05 | candrews | n1gak, I don't have access to SIP right now (see my question above) so I can't help :-( |
| 20:10.41 | n1gak | Greetings, candrews ... your msg of 19:10Z is the only I've seen in > 5 mins |
| 20:10.58 | candrews | My asterisk server is behind NAT. My soft phone inside the firewall works great with it (incoming/outgoing calls work). Outside the NAT, the phone rings / can place calls, but doesn't get any audio, and disconnects seemingly randomly. I have ports 5061, 5061, and 10000-20000 forwarded to the asterisk server |
| 20:11.02 | candrews | I've also configured STUN in res_stun_monitor.con |
| 20:11.09 | candrews | sorry, I just noticed I sent those before you joined |
| 20:11.40 | n1gak | well -- I'm doing better than that ... I'm running my asterisk on my colo box in ... somewhere in the US ... Washington DC, I think |
| 20:12.07 | n1gak | at home I'm testing with a softphone (Twinkle) and can successfully dial internal ports in my Asterisk instance, including myself |
| 20:12.21 | candrews | I'm considering doing that at some point. or running asterisk on my openwrt router. |
| 20:12.21 | n1gak | *AND* I've got out-bound traffic flowing -- I can make calls to the PSTN |
| 20:12.32 | candrews | so then what's the problem? |
| 20:12.47 | n1gak | if someone dials my DID, it rings a couple of times and then plays busy |
| 20:12.50 | *** join/#asterisk haakonnes (~haakonnes@137.174.34.95.customer.cdi.no) |
| 20:13.11 | n1gak | tailing the logs on the asterisk switch I see no indication of inbound attemp |
| 20:13.33 | n1gak | BUT, I'm not sure what precisely to look for in the sip debug stream, because there is a non-trivial amount of 'chattiness' between my Ast. and theirs |
| 20:14.00 | n1gak | so -- it very well COULD be that they ARE trying to send traffic to me, and it's getting lost in a malconfiguration at my end |
| 20:14.10 | n1gak | OR it could be malconfiguration at their end (They've been doing this for about 3 weeks now) |
| 20:15.22 | n1gak | A third possibiliity is: The inbound traffic is being lost somewhere upstream of them |
| 20:15.50 | haakonnes | I'm having some really weird issue here. I have a AGI script that plays an audio file, records your voice, and then plays an audiofile again to get your input about the recording. Everything works great if I call in to asterisk. But if I use a .call file, it works well until after the recording. The next audiofile starts playing, but after 100ms or so, the audio stops and the call halts completely :( |
| 20:16.35 | haakonnes | The audio just stops working |
| 20:17.37 | haakonnes | I've even tried calling the user first, and then calling another asterisk instance (latest 1.6.2), and it still manages to stop functioning 100ms into the last playes message |
| 20:17.52 | haakonnes | anyone seen similar behaviour before? |
| 20:17.59 | n1gak | always exactly 100mS? |
| 20:18.02 | haakonnes | yes |
| 20:18.15 | haakonnes | not sure if it is 100ms, but it is always the same amount of time |
| 20:18.42 | haakonnes | and i've tried with different audio files, and i've tried sending the call via another external asterisk via sip |
| 20:19.48 | haakonnes | i've even tried exec playfile instead of stream_file, but same thing |
| 20:20.09 | *** join/#asterisk wbniv_ (~wbniv@187.154.7.46) |
| 20:30.51 | *** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net) |
| 20:32.07 | devmod | I want to play some video to a channel, are there any options besides Playback with .h26* files and mp4play app? |
| 20:41.27 | tvc123 | I'm trying to create a sip connection between and asterisk box and a switchbox my config is here http://pastebin.com/399FqDvM |
| 20:41.53 | tvc123 | I can call from the switchvox to the asterisk but not from the asterisk to the switchvox |
| 20:42.07 | tvc123 | the errors I'm getting are in the pastebin as well |
| 20:42.12 | tvc123 | any ideas? |
| 20:44.57 | _Raptor_ | 19220x1280 |
| 20:46.26 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 20:52.35 | *** join/#asterisk mac-mini_ (~mac-mini@unaffiliated/macmini/x-648924) |
| 20:54.45 | z4nD4R | somebody here with snom 360 + srtp and asterisk ? |
| 21:04.28 | *** join/#asterisk recurs|ve (~developer@173-165-81-235-Illinois.hfc.comcastbusiness.net) |
| 21:04.35 | recurs|ve | I was using freepbxinaflash for a while due to someone setting up our phone server. We had some issues so I decided to try trixbox. Welcome to phone server 2.0 :). Anyhow I updated all polycom phones to the new firmware, setup a new config file structure and was up and running. Here are the issues I am trying to solve: There are periods of no sound during phone calls and lagged outgoing/incoming/internal calls. Any |
| 21:09.48 | z4nD4R | somebody here to help with snom 360 + srtp and asterisk ? |
| 21:12.06 | recurs|ve | . Any help is appreciated. |
| 21:25.45 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 21:31.04 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 21:31.26 | *** join/#asterisk Dovid (~Dovid@ool-4355e0a4.dyn.optonline.net) |
| 21:36.47 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 21:38.23 | tvc123 | sweet I seem to have resolved my issue ... hrm changed exten => _2XXX,n,Dial(SIP/switchvox/${EXTEN}) to exten => _2XXX,n,Dial(SIP/192.168.x.x/${EXTEN}) and everything seems to be happy now |
| 21:38.49 | tvc123 | is that common that in a extension call you would use the ip or DNS name and not what is in your sip.conf? |
| 21:41.04 | z4nD4R | <PROTECTED> |
| 21:43.53 | JerJer | tvc123: prototype SIP/exten@peer,options |
| 21:44.15 | pigpen | I have an oddity. When a call is dumped into voicemail, I should be able to press "*" or "0" during the greeting/prompts. It does not recognize it. Ideas? Asterisk 1.6.1.20 |
| 21:44.47 | JerJer | pigpen: does those exten's exist in the same context ? |
| 21:45.10 | pigpen | Oh, I have the operator=yes, autofallthrough=yes, I have the exten= 0 and exten =a, and all in the same context from the dialplan through the voicemail.conf |
| 21:45.27 | pigpen | JerJer, sorry, I was still typing. but yea. |
| 21:46.30 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 21:46.38 | SuPrSluG | tvc123:if you had defined the host with a static in the config you could have used your original |
| 21:47.27 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 21:47.41 | P424D0X | I can't registrate my sip phones on "sipgate.de".. What's wrong with my config?! http://pastebin.com/wsf3S0s4 |
| 21:47.59 | tvc123 | SuPrSluG: and JerJer that makes sense it didn't know where to send the call |
| 21:52.52 | pigpen | Well, I have tested on 3 1.6.1.(13,20) boxes with no luck. Identical setup on 1.8, works like a charm. |
| 21:53.01 | pigpen | So I am glad that I am not loosing what is left of my mind. |
| 21:53.05 | pigpen | Upgrade time!! |
| 21:54.51 | haakonnes | hmm.. I am having problems with call files where I do a AGI record_file. The audio is not recorded, and stream_file afterwards stops playing audio after about 100ms or so. |
| 21:55.37 | haakonnes | could it be that asterisk doesn't handle the "answer" state from the outgoing call correctly? I'm calling out to DAHDI directly or via Local, either way, the same thing happends |
| 21:55.54 | z4nD4R | <PROTECTED> |
| 21:56.19 | JerJer | z4nD4R: it is best if you ask a specific question |
| 21:56.41 | z4nD4R | JerJer: ok |
| 21:57.10 | JerJer | haakonnes: have to tried doing s,1,Answer s,n,Wait(1) then go into your AGI ? |
| 21:57.22 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:58.01 | z4nD4R | JerJer: i have asterisk to use srtp ... i have 1 soft phone and snom 360.. if i call from softphone to snom it is ok.. but if i call from snom to softphone ... tell me asterisk " WARNING[7072]: chan_sip.c:8365 process_sdp: We are requesting SRTP, but they responded without it! |
| 21:58.01 | z4nD4R | " |
| 21:58.49 | P424D0X | I can't registrate my sip phones on "sipgate.de".. What's wrong with my config?! http://pastebin.com/wsf3S0s4 |
| 21:58.59 | haakonnes | Jerjer, yes.. |
| 21:59.13 | haakonnes | Seems like it's the other way that is the problem |
| 22:01.25 | z4nD4R | Q: i have asterisk to use srtp ... i have 1 soft phone and snom 360.. if i call from softphone to snom it is ok.. but if i call from snom to softphone ... tell me asterisk " WARNING[7072]: chan_sip.c:8365 process_sdp: We are requesting SRTP, but they responded without it! |
| 22:01.25 | z4nD4R | " |
| 22:01.26 | JerJer | P424D0X: what does asterisk CLI say when it attempts to register? |
| 22:02.20 | JerJer | z4nD4R: is the snom configured to use SRTP ? |
| 22:02.35 | P424D0X | JerJer: My Version is: Asterisk 1.6.2.14 |
| 22:03.01 | JerJer | P424D0X: that's nice, but doesn't help at all |
| 22:03.13 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 22:03.13 | JerJer | is your registration rejected or what ? |
| 22:03.20 | P424D0X | JerJer: I know.. Pse moment.. |
| 22:03.56 | bn-7bc | can anyone point me toa goide on how to install asterisk 1-8-1 with srtp; i have googked and tried a lot of thing but ./configure complains"!libsrtp could not be linked as a shared objec" etc,and yes i have tried yhe tip giveb in the rest of the message |
| 22:04.19 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 22:04.33 | JerJer | bn-7bc: if my memory services me correctly, you need to tell libsrtp to comple a shared object |
| 22:04.49 | JerJer | ie edit libsrtp's Makefile |
| 22:05.07 | *** join/#asterisk errr_ (~errr@fedora/errr) |
| 22:05.12 | *** join/#asterisk CraigW76 (~techcaw@cl-427.dal-01.us.sixxs.net) |
| 22:05.22 | P424D0X | JerJer: Hmm.. I can't see error messages on debug output.. But register messages go out.. |
| 22:05.31 | bn-7bc | JerJer: thanks |
| 22:06.35 | P424D0X | JerJer: Is my config correct?! |
| 22:07.32 | JerJer | no clue - i don't run sipgate :) |
| 22:08.54 | JerJer | bn-7bc: +CFLAGS= -Wall -O4 -fexpensive-optimizations -funroll-loops -fPIC |
| 22:09.03 | JerJer | (remove the leading +) |
| 22:09.12 | JerJer | (i ran a diff -u) |
| 22:09.24 | JerJer | (add -fPIC) |
| 22:10.00 | P424D0X | JerJer: Name/username Host Dyn Nat ACL Port Status 2473867/2473867 217.10.79.9 N 5060 UNREACHABLE 30/2473867 (Unspecified) D 0 UNKNOWN 31/6514036 (Unspecified) D N 5060 UNKNOWN 3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0 offline] |
| 22:10.14 | *** join/#asterisk brack101 (~tbrackett@66.35.2.50) |
| 22:10.19 | brack101 | Hey |
| 22:10.31 | JerJer | w00t |
| 22:10.50 | JerJer | P424D0X: unreachable is bad |
| 22:10.59 | JerJer | firewall blocking it ? can you ping it? |
| 22:11.37 | P424D0X | JerJer: I know.. Can you tell me, what's wrong with my config?! |
| 22:11.49 | P424D0X | JerJer: http://pastebin.com/wsf3S0s4 |
| 22:11.53 | JerJer | it may not be your asterisk confi g |
| 22:11.59 | *** join/#asterisk ccesario (~ccesario@200-207-171-46.dial-up.telesp.net.br) |
| 22:12.11 | JerJer | if your linux box cannot reach 217.10.79.9, then game over |
| 22:14.08 | P424D0X | JerJer: 64 bytes from 217.10.79.9: icmp_req=1 ttl=57 time=63.5 ms |
| 22:15.39 | JerJer | ok then either your local linux has a firewall restricting udp port 5060 or your ISP is blocking udp port 5060 |
| 22:16.17 | P424D0X | JerJer: No on both!! |
| 22:16.20 | P424D0X | :-/ |
| 22:16.46 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 22:17.00 | nny | can someone explain what the PRI option immediate start is and how it relates to asterisk? |
| 22:19.02 | malcolmd | chan_dahdi.conf says "If immediate=yes the dialplan execution will always start at extension 's' priority 1 regardless of the dialed number" |
| 22:19.06 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 22:19.25 | malcolmd | means that it ignores DID and drops it to the s extension |
| 22:21.22 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 22:21.26 | bn-7bc | JerJer: thanks that worked |
| 22:23.03 | nny | malcolmd: not so much in asterisk configs, but on the providers end |
| 22:23.29 | brack101 | So what's a good, cheap device that can run asterisk? |
| 22:24.04 | brack101 | I've heard of people doing it on that seagate hard drive dock |
| 22:24.12 | brack101 | any other options? |
| 22:24.47 | pabelanger | nny: immediate start is for T1 CAS, not PRI |
| 22:27.19 | nny | pabelanger: i guess the two options are immediate or wink |
| 22:27.40 | nny | pabelanger: the provider used the term in regards to provisioning the PRI, may refer to the T1 the pri uses |
| 22:27.52 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 22:29.14 | pabelanger | nny: then your provider is confusing 2 different technologies. Easy question to ask them, how many channels are they providing? 24 or 23+1D |
| 22:29.24 | nny | pabelanger: 23+1D |
| 22:30.10 | nny | pabelanger: i vetted all this stuff back when we setup the T1, just checking it again |
| 22:30.22 | *** join/#asterisk Squeeb (~squirt@eggwee.co.uk) |
| 22:30.25 | pabelanger | Then immediate or wink start are not options. You will be provisioned D4 / AMI or B8ZF / ESF running DMS 100, 4ESS, 5ESS or NI2 |
| 22:31.04 | Squeeb | Hi, can anyone help me with my UK ISDN2e setup, we have a quadBRI card with 3 of the 4 ports in use, they're connected to individual ISDN2e boxes.. the system was restarted a few times today after a power outage and now all I'm getting is this |
| 22:31.08 | Squeeb | Primary D-Channel on span 2 down |
| 22:31.12 | Squeeb | on span 2 3 and 4 |
| 22:31.15 | Squeeb | it's been doing this for hours |
| 22:31.25 | Squeeb | but the lights on the quadBRI card are green |
| 22:32.01 | Squeeb | Are there any clue's as to what could be sorting this? |
| 22:32.13 | Squeeb | causing* sorry, it's been a long and tiresome day :( |
| 22:32.28 | nny | pabelanger: ok thanks |
| 22:32.31 | pabelanger | Squeeb: Call your provider and see if the D-channel is up. Most telcos will turn it down when in alarm |
| 22:32.55 | Squeeb | Oh :/ |
| 22:33.36 | Squeeb | Do they reset after a bit if I turn asterisk off? |
| 22:34.39 | pabelanger | ask them. My experience with telcos is they will turn down the Dchannel if in an alarm stat. I then have to call them to turn it back up |
| 22:35.29 | Squeeb | ok, This is BT we're talking about, they're reluctant to tell me any information these days |
| 22:35.32 | Squeeb | lol |
| 22:35.49 | Squeeb | What is a D-channel btw? I can't seem to find much information on the subject |
| 22:35.59 | Squeeb | oh wait, wikipedia |
| 22:36.30 | pabelanger | Squeeb: Dchannel, out-of-band signalling |
| 22:37.07 | Squeeb | gotcha, so when I have channels 1-2 4-5 7-8, the 3 and 6 and 9 are the D-channels? |
| 22:38.08 | Squeeb | also, would they turn the Dchannels down even if zttool is reporting no alarms at my end? |
| 22:39.34 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 22:39.52 | atan | Can I force all calls from/to one sip client to be recorded? |
| 22:42.49 | pabelanger | Squeeb: Usually yes, not sure about BRI but with PRI you can setup NFAS so all your span don't need a D-channel. |
| 22:42.55 | Squeeb | atan: I use the Monitor() application in my dialplan for that |
| 22:43.08 | atan | Maybe ;) |
| 22:43.14 | atan | err, wrong channel |
| 22:54.06 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:54.38 | atan | Monitor(wav, /var/spool/asterisk/d/dev-{$TIMESTAMP}${CALLERIDNUM}-${EXTEN}.gsm) isn't recording in there |
| 22:54.42 | atan | Did I miss something? =\ |
| 22:56.27 | Squeeb | do something like core set verbose 100 |
| 22:56.39 | Squeeb | and make the call, you'll see if it's hitting that app in the dialplan |
| 22:56.48 | Squeeb | or at least see which path the call took through the dialplan |
| 22:57.13 | Squeeb | you'll also see if you're permissions are set wrong etc, IE it can't write to the file |
| 22:57.49 | atan | res_monitor.c:620 __ast_monitor_change_fname: Cannot change monitor filename of channel SIP/555-00000021 to |
| 22:57.49 | Squeeb | pabelanger: Is there anything I can do my end to diagnose this problem? I can see configuration requests on each of the active span's but I'm a bit clueless as to what I'm looking for |
| 22:57.55 | atan | It won't change the name of it? hmm |
| 22:58.06 | Squeeb | atan: usually when it can't write to that dir |
| 22:58.19 | Squeeb | also is there anything after "to" |
| 22:58.31 | Squeeb | IE, you might want ${TIMESTAMP} |
| 22:58.36 | Squeeb | not {$TIMESTAMP} |
| 22:59.36 | atan | Interesting. Now I have to channels. |
| 22:59.42 | atan | I wonder if I can combine them somehow =) |
| 22:59.50 | Squeeb | yea there's an option in Monitor for that |
| 23:00.06 | Squeeb | actually I think it's just MixMonitor |
| 23:00.25 | Squeeb | <PROTECTED> |
| 23:00.28 | Squeeb | yea that looks like it |
| 23:01.32 | Squeeb | Holy crap my Lines came up! |
| 23:01.34 | atan | Now the only hitch is the called party gets called back if they hang up |
| 23:01.36 | atan | hmm :D |
| 23:01.37 | Squeeb | pabelanger: thanks for the info anyway |
| 23:01.44 | Squeeb | atan: they called back!? |
| 23:01.52 | Squeeb | they get called back* ? |
| 23:02.02 | atan | Yeah =\ the called person hands up and the phone rings again |
| 23:02.02 | Squeeb | Wierd, chuck a Hangup() at the end of your macro |
| 23:02.09 | Squeeb | for safes |
| 23:02.10 | atan | It indeed is :| strange |
| 23:02.17 | Squeeb | personally I've not had that |
| 23:02.21 | atan | exten =>_NXXXXXX,n,Hangup is anyway |
| 23:02.24 | atan | Maybe it should be () |
| 23:02.29 | Squeeb | I don't think it matters |
| 23:02.47 | Squeeb | is it a phantom call? |
| 23:02.50 | Squeeb | IE, dead when answered? |
| 23:03.26 | Squeeb | Oh Happy day though :) I don't have to face no phones tomorrow at work |
| 23:03.27 | Squeeb | \o/ |
| 23:03.42 | Squeeb | this happens alot with BT I've noticed.. you restart your pbx and you're out of phones for a few hours :/ |
| 23:05.22 | atan | No both parties is there |
| 23:05.33 | atan | On another note though, exten =>_NXXXXXX,1,MixMonitor(dev-${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}.wav) isn't getting saved as that file name |
| 23:05.34 | Squeeb | uhm |
| 23:05.42 | atan | it's just dev---${EXTEN}.wav right now |
| 23:05.53 | atan | changes timestamp |
| 23:05.54 | Squeeb | chuck a few NoOp(${VARIABLE})'s in |
| 23:05.58 | Squeeb | to see what's in those variabels |
| 23:06.09 | Squeeb | they'll be printed to console |
| 23:06.31 | recurs|ve | trying to solve: There are periods of no sound during phone calls during connection and lagged outgoing/incoming/internal calls. Using trixbox 2.8 |
| 23:07.07 | pabelanger | Squeeb: Call the telco first and make sure it is up, without it you cannot do much |
| 23:07.09 | Squeeb | recurs|ve: what type of connection? to a SIP provider externally, within a LAN.. any more information would be helpful |
| 23:07.26 | Squeeb | pabelanger: will do in the future. I appear to be able to make calls and receive them from the office |
| 23:07.37 | Squeeb | so .. best not to touch I suppose .. until at least friday :P |
| 23:08.31 | atan | NoOp("SIP/555-0000002f", "") in new stack |
| 23:08.32 | atan | <PROTECTED> |
| 23:08.43 | Squeeb | so just ${EXTEN} |
| 23:08.52 | Squeeb | oh |
| 23:09.01 | Squeeb | yea, the "" refers to that empty variable |
| 23:09.13 | Squeeb | the first string is the extension with reference |
| 23:09.16 | atan | I don't understand why ${TIMESTAMP} does not exist |
| 23:09.52 | Squeeb | atan: which version of asterisk |
| 23:09.55 | Squeeb | I ask this because of this: |
| 23:09.56 | Squeeb | ${TIMESTAMP}: Current date time in the format: YYYYMMDD-HHMMSS This is deprecated as of Asterisk 1.4, instead use :${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S) |
| 23:10.07 | atan | Will do!! |
| 23:10.14 | Squeeb | :) |
| 23:10.24 | Squeeb | ${CALLERID(num)}: The current Caller ID number - ${CALLERIDNUM} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4. |
| 23:10.28 | Squeeb | may also help |
| 23:10.39 | Squeeb | http://www.voip-info.org/wiki/view/Asterisk+variables |
| 23:10.40 | Squeeb | for more fun |
| 23:12.00 | atan | Yup. That was it! Thanks buddy!! |
| 23:12.05 | Squeeb | np :) |
| 23:12.11 | recurs|ve | Squeeb: i can share any config files necessary on a pastebin, External provider through a Digium card, the delayed connect/ring happens for all calls (LAN and Incoming), same with the connection between lines (takes up to 3 seconds), using polycom phones 560 and 670 with the new firmware |
| 23:12.12 | Squeeb | that site btw is awesome |
| 23:12.29 | Squeeb | recurs|ve: unfortunatly I'm no good with polycom phones |
| 23:12.52 | recurs|ve | Squeeb: well, i can easily setup a softphone and see if it is happening there as well |
| 23:13.00 | Squeeb | I'd suggest setting up a test macro and call into it externally |
| 23:13.04 | Squeeb | to see what effect that has |
| 23:13.32 | recurs|ve | Squeeb: alright... |
| 23:13.46 | Squeeb | and also looking at the output of 'core set verbose 100' for any clues |
| 23:14.20 | Squeeb | perhaps switch codec's on the phones |
| 23:14.24 | atan | I often just log in with asterisk -rvvvvvvvvvv =\ |
| 23:14.32 | Squeeb | atan: yea you can do that too |
| 23:14.40 | atan | Is that the same as verbose 100? or would that be == 10? |
| 23:14.42 | Squeeb | i'm usually already in rasterisk |
| 23:14.57 | Squeeb | atan: I'm not entirely sure of the levels of verboseness, I just use 100 by habit |
| 23:16.08 | Squeeb | recurs|ve: Perhaps make sure all the client phones are using the standard ulaw or alaw codec (g711u, g711a) to bypass any transcoding which maybe taking place on the PBX |
| 23:16.30 | recurs|ve | Squeeb: pasting the output of the verbose output now |
| 23:16.34 | Squeeb | k |
| 23:16.38 | recurs|ve | Squeeb: if you wouldnt mind taking a look at it |
| 23:16.41 | Squeeb | i'm not sure how helpful I can be but I'll take a look :) |
| 23:19.34 | recurs|ve | Squeeb: https://gist.github.com/741316 |
| 23:22.08 | Squeeb | hmm, so that went to voicemail.. was the audio rubbish for the voicemail service? |
| 23:22.33 | recurs|ve | Squeeb: yea, it was correct, sending one with a picked up call |
| 23:22.46 | Squeeb | k |
| 23:23.06 | Squeeb | are the phones on the same subnet as the asterisk system? |
| 23:23.15 | recurs|ve | Squeeb: yes |
| 23:23.19 | Squeeb | k |
| 23:23.48 | recurs|ve | phones are dynamic ip, asterisk is static through dhcp assigned by mac, all computers through phones are static through dhcp |
| 23:23.52 | Squeeb | might be worth checking the asterisk system's firewall's mangle table for anything which might be messing with the TOS fields, or responding to the TOS fields |
| 23:24.09 | Squeeb | iptables -L -t mangle -v -n |
| 23:24.21 | Squeeb | will show you if there's anything processing via TOS field |
| 23:24.58 | Squeeb | APF is a killer for that if you're using that.. it's a set of popular "out of the box" management scripts to handle iptables |
| 23:25.27 | recurs|ve | Squeeb: https://gist.github.com/741330 |
| 23:25.43 | Squeeb | meh, nice and empty |
| 23:25.54 | recurs|ve | Squeeb: the box is behind a firewall and is only accessable internally, i did not setup any firewall rules due to this. |
| 23:26.02 | Squeeb | cool |
| 23:26.20 | Squeeb | did you get the output for the picked up call? |
| 23:26.20 | recurs|ve | Squeeb: working on picked up phone call |
| 23:26.25 | recurs|ve | on it now. |
| 23:27.11 | *** part/#asterisk l2trace99 (~jr@74.118.40.1) |
| 23:27.15 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 23:27.46 | recurs|ve | Squeeb: alright, here is the output and the conditions. 1. call in completed took 11 seconds for the phone to ring, 2. call picked up and took 4 seconds before we could hear eachother |
| 23:29.38 | recurs|ve | Squeeb: https://gist.github.com/741337 |
| 23:29.41 | Squeeb | k |
| 23:30.27 | recurs|ve | so long for find and replace heh, no masking, trying to not have cell publically avail :( |
| 23:30.47 | recurs|ve | whatever, repasting with exact info |
| 23:30.54 | Squeeb | heh, no worries.. I'm not looking at the CLI's :) |
| 23:31.24 | *** join/#asterisk brunner (~Chris@35-137.175-24.bham.res.rr.com) |
| 23:31.41 | Squeeb | I'm really not sure what's causing the problems.. I mean, that dialparties.agi script must be taking up most of the time to do whatever external processing it might be doing |
| 23:32.04 | recurs|ve | Squeeb: i guess its things like line 43 that make me wonder what is going on |
| 23:32.25 | Squeeb | could be a hangup from a preious call |
| 23:32.32 | Squeeb | previous |
| 23:32.40 | recurs|ve | Squeeb: there is no one in the office and no connections |
| 23:32.43 | Squeeb | seeing how the newer call is on span 2-1 |
| 23:32.47 | Squeeb | oh |
| 23:32.55 | Squeeb | so was that a dead call? |
| 23:33.03 | recurs|ve | may have been... |
| 23:33.18 | recurs|ve | how do i list all current calls |
| 23:33.34 | Squeeb | dahdi show channels |
| 23:33.38 | Squeeb | and sip show channels |
| 23:34.06 | Squeeb | I mean, they're all hangup cause 16 |
| 23:34.08 | Squeeb | which is normal clearing |
| 23:34.26 | Squeeb | IE, hung up by the user somewhere |
| 23:34.43 | brack101 | Is transcoding from G.711 to iLBC a pretty CPU intensive operation? Do you need a fast computer to do it? |
| 23:34.45 | recurs|ve | when executing dahdi show channels, that should have as many channels as my trunk right? |
| 23:35.10 | Squeeb | recurs|ve: yea minus how ever many signalling channels are present |
| 23:35.40 | Squeeb | IE if you have 4 x 2 channel spans (1-2,4-5,7-8,10-11) you'll see 8 channels in dahdi show channels |
| 23:35.53 | Squeeb | brack101: it's fairly CPU intensive yes |
| 23:36.11 | recurs|ve | Squeeb: also the sip/dahdi show channels is showing https://gist.github.com/741340 |
| 23:36.26 | Squeeb | *waits for github to stop being slow* |
| 23:36.36 | Squeeb | yea, that looks fine |
| 23:36.38 | recurs|ve | Squeeb: hrm, i only have 23 channels |
| 23:36.39 | Squeeb | no calls present |
| 23:36.48 | Squeeb | <PROTECTED> |
| 23:36.52 | brack101 | Squeeb: So is it realistic that I could repurpose a linksys router to do that? |
| 23:36.55 | recurs|ve | Squeeb: ah |
| 23:37.00 | brack101 | only one stream |
| 23:37.03 | brack101 | I'd be the only client |
| 23:37.07 | Squeeb | brack101: err.. a MIPS CPU may not handle it |
| 23:37.11 | Squeeb | very well |
| 23:37.15 | brack101 | I see |
| 23:37.28 | Squeeb | brack101: there's an artical on sipgate.co.uk about it somwehere |
| 23:37.34 | recurs|ve | Squeeb: alright, so no calls, ill call again and answer and paste again, there should be no hangup case correct? |
| 23:37.38 | Squeeb | as they use iLBC |
| 23:37.54 | Squeeb | recurs|ve: not while you're on the call no |
| 23:37.57 | Squeeb | not until the end :P |
| 23:38.09 | recurs|ve | Squeeb: alrgight giving it another shot |
| 23:38.11 | Squeeb | cool |
| 23:38.54 | Squeeb | brack101: we ran iLBC on a G4 Mac Mini @ 1.6GHz |
| 23:38.59 | Squeeb | there was noticable lag |
| 23:39.06 | Squeeb | from ulaw to ilbc |
| 23:39.08 | brack101 | I'm using Google Voice + Gizmo5 to have a full IP cell phone and it works OK, but I can't use g.711 on 3G because it requires too much bandwidth. |
| 23:39.28 | brack101 | And I have to use their stupid website for outbound calls |
| 23:39.29 | Squeeb | brack101: do you not have the option to use GSM ? |
| 23:39.32 | brack101 | Nope |
| 23:39.37 | Squeeb | :( |
| 23:39.42 | brack101 | The Google voice server ONLY allows g.711 |
| 23:40.08 | brack101 | It's such a great solution otherwise because I have voicemail storage, transcription, free US calling, SMS |
| 23:40.10 | Squeeb | you get what you pay for I guess :) |
| 23:40.20 | brack101 | I'd gladly pay for another service that does the same thing |
| 23:40.21 | brack101 | nobody does |
| 23:40.22 | recurs|ve | Squeeb: https://gist.github.com/741346 |
| 23:40.28 | Squeeb | recurs|ve: looking now |
| 23:41.24 | recurs|ve | Squeeb: interisting, there may be something that is trying to forward |
| 23:41.30 | Squeeb | yea, it looks that way |
| 23:42.11 | brack101 | Unless somebody knows of a service that does all those things |
| 23:42.21 | Squeeb | wierd, without knowing what dialparties.agi does |
| 23:42.24 | Squeeb | not sure I can help :X |
| 23:42.27 | brack101 | sipgate does, but doesn't have SMS and only has one server across the country so latency is unacceptable |
| 23:42.43 | recurs|ve | Squeeb: well ill reset system and take another look, thanks for your help |
| 23:42.51 | Squeeb | np, sorry I couldn't fix :} |
| 23:43.12 | cmnky | anyone running asterisk on a laptop for mobile voip ? |
| 23:43.48 | brack101 | You don't really need it for most "mobile voip" applications |
| 23:44.12 | cmnky | brack101, i was about to suggest just forwarding to my laptop softphone from my office |
| 23:44.29 | recurs|ve | nah thanks for help |
| 23:44.46 | Squeeb | cmnky: You mean using your laptop as a smartphone hooked up to a pbx in the office? |
| 23:44.48 | brack101 | The idea is that you run it as a full time server that clients connect to |
| 23:44.54 | cmnky | what about using a bluetooth headset with the softphone ? is there a special linux app for that other than sp and bt tools ? |
| 23:44.56 | brack101 | ie laptop |
| 23:45.52 | cmnky | Squeeb, no ... not exactly ... more like a mobile extension from my office asterisk |
| 23:46.07 | Squeeb | cmnky: Just use something like x-lite on your linux laptop? |
| 23:46.17 | Squeeb | connected to your asterisk pbx in the office |
| 23:47.11 | cmnky | sure ... i get it ... softphone on laptop ... but do i need some special app to get the laptop softphone to work w/ a bluetooth headset ? |
| 23:47.33 | Squeeb | cmnky: No idea with linux and bluetooth I'm afraid |
| 23:47.49 | cmnky | ok ... thx anyway |
| 23:48.02 | Squeeb | If you can get your bt headset to be recognized as a standard DSP device, then you can use most softphone clients to direct audio over that DSP device |
| 23:48.14 | cmnky | im trying to avoid using the cell phone when away from office |
| 23:49.09 | cmnky | Squeeb, okay ... im real new to it ( my headset isnt even here yet ) ... just getting linux on the laptop now .. thx |
| 23:49.36 | Squeeb | np |
| 23:49.45 | Squeeb | there's tons of reviews on the net for the best linux softphone |
| 23:49.48 | Squeeb | might be worth looking |
| 23:50.54 | cmnky | certainly ... linphone is the only one ive seen so far ... id label it "beta" |
| 23:51.35 | cmnky | was gonna try sflphone next |
| 23:51.48 | Squeeb | it Looks like X-lite is available |
| 23:51.48 | Squeeb | http://counterpath.s3.amazonaws.com/downloads/X-Lite_Install.tar.gz |
| 23:51.49 | Squeeb | :) |
| 23:52.02 | cmnky | x-lite is closed source tho |
| 23:52.10 | Squeeb | works though |
| 23:52.37 | Squeeb | and so's sendmail :P |
| 23:52.39 | cmnky | it might ... but it goes to the bottom of the options list for being closed source ;) |
| 23:52.54 | *** part/#asterisk n1gak (~lawrence@189.162.186.156) |
| 23:53.03 | Squeeb | there's tons of nice free apps out there that are closed source |
| 23:53.29 | cmnky | im sure there are ... the code i write is closed source ;) |
| 23:53.29 | Squeeb | to name a few, the Darwin kernel |
| 23:53.33 | Squeeb | sendmail |
| 23:53.34 | Squeeb | VMware |