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| 01:04.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:20.55 | UnixDev | is there a way to send a sip notify from the dial plan? similarly to sip notify on the console? |
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| 03:33.24 | bdemers_ | how do i get the headers of my current call to show up in the Asterisk console ? |
| 03:33.50 | bdemers_ | i've tried starting asterisk with: asterisk -cvvvvvvvvvvvvvvv |
| 03:34.16 | bdemers_ | and changing the logging in logging.conf to: console => notice,warning,error,debug |
| 03:36.40 | *** join/#asterisk junmin (~junmin@189.180.177.70) |
| 03:42.17 | junmin | hello all, asterisk cannot transfer calls successfully, can hear the message "transfer", and "log: started music on hold", but can not hear the music. after keyed the extention number, just nothing happed, looks like call "drops", and log syas "bling transfering and stopped musi on hold ..." |
| 03:43.16 | junmin | extension doesn't ring ... no audio in both side. :s |
| 03:47.07 | bdemers_ | ahh, found what i was looking for: |
| 03:47.07 | bdemers_ | sip set debug on |
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| 04:04.49 | vader-- | what do you guys setup for remote extensions that require something like tftp for configuration? |
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| 04:44.48 | *** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com) |
| 04:48.31 | Sean-Der | I am running Debian wheezy and I have func_curl in my /var/lib/modules and I have autoload=yes |
| 04:48.51 | Sean-Der | What else can i debug for the curl option not existing in my dialplan |
| 04:49.40 | Sean-Der | config.c: Registered Config Engine curl |
| 04:49.50 | Sean-Der | thats my only curl related log message |
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| 04:55.02 | Sean-Der | module show like curl is giving it back to me also? |
| 04:55.03 | Sean-Der | Huh |
| 04:57.04 | Sean-Der | core show function CURL also is good |
| 04:57.06 | Sean-Der | what the heck |
| 05:02.28 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-fwcyjbkkhsakxcwa) |
| 05:03.11 | kaldemar | maybe something with you syntax. show the extension and the output of its execution. |
| 05:08.01 | Sean-Der | exten=>start,1,CURL(http://google.com/anstoheuona) |
| 05:08.10 | Sean-Der | [Jul 9 00:56:56] WARNING[16258]: pbx.c:4218 pbx_extension_helper: No application 'CURL' for extension (curlTest, start, 1) |
| 05:08.36 | Sean-Der | I have also tried curl, Curl. The Asterisk wiki says CURL so I think thats right |
| 05:11.04 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 05:15.47 | Sean-Der | == Registered custom function 'CURL' |
| 05:15.57 | Sean-Der | <PROTECTED> |
| 05:16.08 | Sean-Der | so it loads and everything |
| 05:21.28 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:25.03 | Sean-Der | (${CURL(http://www.google.com/oaheu)} |
| 05:25.14 | Sean-Der | This is the syntax now? |
| 05:25.23 | Sean-Der | Somebody should change the existing docs |
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| 05:28.42 | *** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com) |
| 05:29.14 | Sean-Der | is anyone running asterisk 10? I don't know if what I am seeing is a bug, and I can't grab svn at this time |
| 05:33.07 | kaldemar | Sean-Der: your problem is the dialplan syntax. |
| 05:33.27 | Sean-Der | kaldemar: whats the correct syntax for the curl function? |
| 05:34.03 | kaldemar | Sean-Der: CURL is not an application but a function. an extension always must have an application, so you must embed the CURL function inside an application. what are you trying to do? |
| 05:34.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 05:35.09 | Sean-Der | kaldemar: I was just trying to verbose out curl. I actually have it working now |
| 05:35.16 | Sean-Der | Verbose(${CURL(http://www.google.com/oaheu)}) |
| 05:35.47 | Sean-Der | So all functions need to be inside applications with that syntax? |
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| 06:00.25 | kaldemar | seanbright: yes |
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| 06:30.27 | bulkorok | hello |
| 06:39.18 | *** join/#asterisk psteyn (~pieters@dsl-185-166-229.dynamic.wa.co.za) |
| 06:39.31 | psteyn | any way to get the MSISDN number of a vgsm sim ? |
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| 07:19.48 | jkroon | hi guys, after connecting to the AMI (asterisk 1.8.13.0 - will probably upgrade to 1.8.13.1 this evening), I'm seeing a situation where the connection will simply get shut down from the asterisk side for no obvious reason. I connect using two connections, then initiate an Originate action on the one, both connections will terminate. |
| 07:20.04 | jkroon | The last received event is almost always a SetVar one. |
| 07:22.57 | wdoekes | upgrade from 1.8.12? |
| 07:23.35 | wdoekes | or.. was there a previous version that did work? |
| 07:31.50 | jkroon | wdoekes, not actually sure, busy writing a new application that functions via AMI and only just picked it up now |
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| 08:18.02 | fling | hello! how to connect to my isp over g.711? |
| 08:19.06 | fling | something like [ToMyISP] /n type=friend /n port=1720 /n allow=g711 ? |
| 08:19.24 | fling | or type peer? |
| 08:23.31 | jkroon | type=peer most likely |
| 08:25.04 | fling | I have also set username, secret, codecs, etc… how to check if I'm connected? |
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| 08:28.39 | bulkorok | fling: in cli type 'sip show peers' |
| 08:28.44 | fling | 'sip show peers' says Status Unmonitored and looks like wrong port is used |
| 08:29.17 | fling | right, no port in the sip.conf :p |
| 08:29.32 | bulkorok | add qualify=yes to [yourprovider] |
| 08:29.47 | fling | why? |
| 08:31.05 | fling | with port 1720 it says UNREACHABLE |
| 08:31.41 | bulkorok | what port does your provider want to use!? |
| 08:33.10 | fling | bulkorok: they don't say, they say 'no port needed, you should have open port' |
| 08:33.33 | *** join/#asterisk Lantizia (~lantizia@cpc25-stok15-2-0-cust110.1-4.cable.virginmedia.com) |
| 08:33.34 | bulkorok | std SIP port is 5060 |
| 08:33.52 | Lantizia | Lo, can anyone help me debug? *8 isn't working |
| 08:34.00 | fling | bulkorok: 5060 is closed, I will scan all the ports and then call them again |
| 08:34.10 | bulkorok | k |
| 08:34.20 | bulkorok | Lantizia: ~ask |
| 08:34.24 | bulkorok | ~ask |
| 08:34.24 | 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. |
| 08:34.58 | Lantizia | Well 3649 rings... 3648 dials *8 - it gets zip all |
| 08:35.15 | Lantizia | nothing shows on the console for 3648 unless I enable SIP debug |
| 08:35.35 | psteyn | any way to get the MSISDN number of a vgsm sim ? |
| 08:35.41 | bulkorok | and what does it sohw?! is there an INVITE? |
| 08:35.45 | Lantizia | both peers are in the same callgroup/pickupgroup - and yeah pickupexten=*8 in features.conf |
| 08:36.02 | Lantizia | bulkorok, yeah it shows *8@<ip of pbx> is dialed |
| 08:36.21 | elliot98 | can encryption be done on a per device basis or only global? |
| 08:36.23 | elliot98 | waves high |
| 08:36.49 | fling | bulkorok: looks like 'nmap 10.222.2.10 -sU -p 1-65535' takes too much time… |
| 08:36.51 | Lantizia | bulkorok, i asked this very question on friday in here - all in one line, all very specific - and got no answers... that's the only reason I kept it short and sweet this time |
| 08:37.04 | bulkorok | :-) |
| 08:37.25 | bulkorok | Lantizia: I don't think I can help with pickupgroup... |
| 08:37.53 | fling | bulkorok: I have spent about a week to get some info from them, now I know right ip address and usernames… |
| 08:38.26 | bulkorok | do you hav to use this provider... I don't think you'll get much support either... |
| 08:38.29 | fling | bulkorok: they say protocol is g.711, is it enough to connect? |
| 08:38.36 | bulkorok | not really |
| 08:38.44 | fling | because it is codec? |
| 08:38.55 | bulkorok | g.711 is just the codec for the RTP... |
| 08:39.04 | bulkorok | do get or send calls to this provider!? |
| 08:39.07 | fling | and alab is a variant |
| 08:39.19 | bulkorok | law... yes |
| 08:39.22 | fling | s/alab/alaw |
| 08:39.33 | bulkorok | hates his keyboard... |
| 08:39.44 | fling | bulkorok: I use qgmlwb |
| 08:40.54 | bulkorok | the hardware self... need a new one soon |
| 08:41.47 | fling | bulkorok: grep -A5 xkb .xinitrc > https://gist.github.com/3075151; .Xmodmap > https://gist.github.com/3075142 |
| 08:42.22 | fling | bulkorok: I use mitsumi classic black, I want DasKeyboard, but can't buy it here :( |
| 08:42.51 | bulkorok | hu!? cyrillic?!.. nice :-) |
| 08:43.27 | fling | bulkorok: I will change it, need to use carpalx on it and move some symbols… |
| 08:44.51 | fling | bulkorok: like to set ь and ъ on a same key, because here is no capital Ь and Ъ in Russian, and some other modifications… |
| 08:45.10 | fling | bulkorok: nmap is still scanning :p |
| 08:45.26 | bulkorok | well then the one without labels would be the best ;-) |
| 08:46.02 | fling | bulkorok: yeah, silent one |
| 08:48.08 | *** join/#asterisk Samuel_CTX (~svanderp@62-213-207-59.colo.housingcenter.be) |
| 08:54.14 | kaldemar | fling: port 1720 sounds like H.323. is that what you have bought? |
| 08:54.18 | fling | bulkorok: omg they can't say it to me again :p |
| 08:54.53 | fling | kaldemar: idk, they say protocol is 711a |
| 08:55.01 | kaldemar | fling: actually, where did you come up with port 1720? |
| 08:55.14 | fling | kaldemar: nmap said it is open |
| 08:55.35 | fling | it is still scanning udp |
| 08:55.41 | kaldemar | maybe you should ask them what protocol to use. |
| 08:55.58 | kaldemar | and don't settle with an answer for 711a. |
| 08:55.58 | fling | trying, still nothing atm |
| 08:59.32 | fling | kaldemar: nmap says 5060/udp open|filtered sip |
| 09:00.48 | fling | kaldemar: and the same for all other ports, so looks like it is closed and or ip address is wrong again :p |
| 09:13.38 | fling | bulkorok: kaldemar: if they use sip then 5060/udp _should_ be opened? |
| 09:15.10 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 09:15.58 | fling | oh! now it shows status OK (10 ms) |
| 09:17.27 | fling | I'm connected! :p |
| 09:25.22 | fling | exten => 600,2,Playback(demo-echotest) |
| 09:25.44 | fling | how to use /var/lib/asterisk/sounds/ru/demo-echotest.alaw instead of /var/lib/asterisk/sounds/en/demo-echotest.gsm ? |
| 09:26.41 | fling | language=ru in [general] ? |
| 09:28.31 | fling | works! |
| 09:30.16 | fling | delay is too long |
| 09:35.16 | *** join/#asterisk black187 (~black187@77.234.149.118) |
| 09:36.03 | elliot98 | is the encryption parameter in sip.conf for outgoing or incoming calls? |
| 09:36.52 | black187 | Hello guys, does anybody know how ti disable Asterisk/Dahdi fax detection ("detected a CED tone towards the network.", "echo canceler disabled its NLP."). I have fax problems on Dahdi PRI, so maybe disabling fax detection will work... |
| 09:39.09 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 09:39.53 | fling | bulkorok: what extensions section to use for exten => _9.,1,Dial(SIP/${EXTEN:1}@mysipprovider-out,30,r) ? |
| 09:42.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 09:46.56 | kaldemar | fling: open|filtered is all nmap can tell about a UDP port. |
| 09:48.25 | kaldemar | elliot98: did you read what it says next to the parameter in the sample config? |
| 09:49.51 | fling | kaldemar: I have connected and I'm trying to call but == Everyone is busy/congested at this time (1:0/1/0) |
| 09:51.41 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:57.42 | kaldemar | fling: look at sip debug when making a call. "sip set debug on" |
| 09:58.33 | elliot98 | kaldemar: it's states only for outgoing, but incoming calls are failing that require srtp...where is that enabled? |
| 10:00.57 | kaldemar | elliot98: just make the originating end use SRTP. how are the calls failing? |
| 10:02.07 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.12) |
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| 10:05.39 | elliot98 | kaldemar: it's responds with a 488 Not Acceptable Her |
| 10:06.53 | *** join/#asterisk din3sh (~din3sh@41.212.200.147) |
| 10:15.17 | elliot98 | kaldemar: does something need to be set up for SRTP to work? |
| 10:18.29 | elliot98 | kaldemar: the res_srtp module looks loaded |
| 10:22.30 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 10:26.18 | elliot98 | perhaps the crypto needs to be changed |
| 10:27.26 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 10:30.19 | fling | kaldemar: [Jul 9 17:29:47] NOTICE[3629] chan_sip.c: Outbound Registration: Expiry for metro is 60 sec (Scheduling reregistration in 45 s) |
| 10:42.13 | fling | but `sip show registry´ says registered |
| 10:44.04 | kaldemar | and? |
| 10:45.39 | din3sh | how do we enable callerid on attended transfer? |
| 10:45.52 | din3sh | asterisk 1.8 |
| 10:46.04 | fling | kaldemar: registration successful :p |
| 10:46.27 | fling | will continue tomorrow, thanks for your help |
| 10:46.34 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 10:48.56 | aberrios | this is probably a stupid Q. If i Transfer() Instead of Dial() another SIP device (gateway), Asterisk lets go of the call? |
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| 11:00.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 11:13.37 | bulkorok | aberrios: what do you want to achieve with them? less load on asterisk machine!? |
| 11:15.10 | *** join/#asterisk jkroon (~jkroon@dsl-244-33-70.telkomadsl.co.za) |
| 11:16.16 | jkroon | hi guys, i've got a variable containing the name of a potential label (for goto purposes), in some cases this label may or may not be valid, so after checking the source for VALID_EXTEN and DIALPLAN_EXISTS it became clear that they convert the priority fields to integer values before doing their work - is there any way to determine if a label is valid? |
| 11:18.28 | aberrios | bulkorok: well at the moment a gateway sends calls to *, * does its stuff in dialplan then afterwards sends the call back to the gateway which sends the call onto a legacy PBX. This results in 2 SIP calls per converstation on the gateway. I'd rather just have the 1.... |
| 11:19.42 | bulkorok | aberrios: I don't think this will work... |
| 11:20.15 | bulkorok | * gets INVITE from gateway and sends an INVITE back to gateway... => 2 channels |
| 11:21.20 | aberrios | bugger. |
| 11:21.21 | bulkorok | maybe transfer would work... I don't know... |
| 11:21.22 | *** join/#asterisk gusto (~gusto@DSL01.83.171.177.118.ip-pool.NEFkom.net) |
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| 11:27.45 | kaldemar | aberrios: asterisk should send a 302 redirect with Transfer. |
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| 11:39.42 | Jo_Bell | I have a requirement to connect to 3rd party SIP enabled video devices from my video conferencing application and pass the video information to those devices from more than one user at a time. Is there any uniform way to do this irrespective of the 3rd party SIP device provider, such as polycom. I want to make minimal change between the device providers. |
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| 11:53.58 | ijpalmer | good day, I have a queue setup that often has calls waiting as thereare no agents available. I need to be able to log an agent into this queue and have the cagent be presented with the longest waiting call instantly, is there a way of doing this? |
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| 12:33.40 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 12:39.12 | carrar | *Y*A*W*N* |
| 12:46.32 | ijpalmer | good day, I have a queue setup that often has calls waiting as thereare no agents available. I need to be able to log an agent into this queue and have the cagent be presented with the longest waiting call instantly, is there a way of doing this? |
| 12:47.10 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 12:50.53 | [TK]D-Fender | ijpalmer, nothing natural. You'd have to corcibly hijack the caller out of the queue |
| 12:51.00 | [TK]D-Fender | forcibly |
| 12:52.11 | ijpalmer | D-Fender, thanks, what would be the best way round this would you say? |
| 12:53.16 | Jo_Bell | Hi all: I have a requirement to connect to 3rd party SIP enabled video devices from my video conferencing application and pass the video information to those devices from more than one user at a time. Is there any uniform way to do this irrespective of the 3rd party SIP device provider, such as polycom. I want to make minimal change between the device providers. |
| 12:53.16 | [TK]D-Fender | AMI Redirect |
| 12:54.11 | [TK]D-Fender | Jo_Bell, app_conference is the only tool I know of for this and it "follows the talker", so you'd have to mute the other participants. |
| 12:54.30 | [TK]D-Fender | Jo_Bell, And they'd all have to be using the same codecs |
| 13:03.37 | Jo_Bell | D-Fender: Asterisk will take video samples one at a time from each client and will forward it to the end point serially using function like app_conference. Thats what you mean to say right? |
| 13:07.13 | Digweed | I am on Asterisk 1.6 and would like to send a busy signal to all incoming calls. This does not work: exten => _X.,1,Hangup(17) |
| 13:07.25 | Digweed | Anyone an idea how I should configure it? 17 should mean 486 busy right? |
| 13:07.36 | *** join/#asterisk blindhog_ (~josh@adsl-75-62-197-176.dsl.rcsntx.sbcglobal.net) |
| 13:08.55 | junmin | Digweed: just for test purpose, did you tried _. ? |
| 13:09.15 | Digweed | Well it accepts the call but I do not receive a busy... |
| 13:09.56 | Digweed | Executing [herethetelephonenumber@inbound:1] Hangup("SIP/heremytrunknumber-b647e9d8", "17") in new stack |
| 13:10.00 | Digweed | This is what the console shows. |
| 13:10.26 | junmin | do you need Hangup or Busy cmd? |
| 13:10.54 | Digweed | I thought I should use Hangup? I need to test the busy signalling so the box should simply say busy on incoming calls. |
| 13:11.22 | junmin | first i think you need answer call and then send busy signal |
| 13:11.32 | junmin | http://www.voip-info.org/wiki/view/Asterisk+cmd+Busy this could help |
| 13:12.04 | leifmadsen | yes, to send a busy, you use Busy() |
| 13:12.14 | leifmadsen | a value to busy is how long to play a busy tone prior to hangup |
| 13:12.18 | leifmadsen | Congestion() is a fast busy |
| 13:12.21 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 13:12.37 | Digweed | so basically this |
| 13:12.38 | Digweed | exten => _X.,1,Answer |
| 13:12.38 | Digweed | exten => _X.,n,Busy |
| 13:12.39 | Digweed | exten => _X.,n,Hangup |
| 13:12.45 | Digweed | ow wait the () |
| 13:12.50 | Jo_Bell | Hi. can any one point to some information on integrating SIP video devices to Asterisk? |
| 13:12.52 | leifmadsen | well () is optional |
| 13:12.56 | leifmadsen | I always put it in for style |
| 13:13.05 | leifmadsen | Jo_Bell: what do you mean? |
| 13:13.09 | Digweed | yeah |
| 13:13.10 | Digweed | that's it |
| 13:13.13 | Digweed | awesome! thanks guys |
| 13:13.24 | leifmadsen | there is sip video support, but it won't transcode |
| 13:13.40 | leifmadsen | in Asterisk 10 or later, in ConfBridge() you can have video conferencing too |
| 13:13.55 | *** join/#asterisk MariusKarthaus (~quassel@541F8611.cm-5-8c.dynamic.ziggo.nl) |
| 13:14.37 | junmin | guys ... and here i havea strange problem, asterisk doesnt transfer calls :s, can hear "transfer" message but just nothing happen and call seems dropped. any idea? |
| 13:14.49 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hrgpuzyzskruhwrp) |
| 13:14.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:14.52 | Jo_Bell | leifmadsen: My case is I have n clients each transmitting h264 video to an application server. Now I want to transmit those samples through Asterisk to SIP enabled video devices such as Polycom. |
| 13:14.55 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 13:15.02 | junmin | from log, i can not get nothing useful to resolved this problem |
| 13:15.27 | leifmadsen | Jo_Bell: so the source is not a sip client then? |
| 13:16.14 | Jo_Bell | No I will have an intermediate component which will do the required transcoding. It will talk in SIP to the Asterisk |
| 13:16.27 | MariusKarthaus | If I have multiple asterisk servers (A &B) registering to the same trunk at the same sip provider with muliple DID numbers in that trunk. (How) is it possible to have asterisk A respond to only s ubset of the numbers and asterisk B to another subset of that numbers. Is there a "give up on this match" apllication in the dialplan ? |
| 13:17.25 | junmin | MariusKarthaus: use pattners in extensions |
| 13:17.53 | Jo_Bell | The intermediate component (gateway) will send H264 video from all the clients to Asterisk server. Now I want Asterisk to get it upto the SIP video devices. |
| 13:19.14 | [TK]D-Fender | Jo_Bell, No, * will send the video of the TALKER to all parties |
| 13:19.19 | junmin | MariusKarthaus: and you can use Hangup cmd. |
| 13:19.21 | MariusKarthaus | junmin: yes I know I can match on the env variable that contains the used dial-in numbers and I can pattern-match on the number. But what is the rule or target that I should use in that line? |
| 13:19.45 | leifmadsen | Jo_Bell: then just allow h264 for a sip peer on asterisk, and enable video |
| 13:19.48 | leifmadsen | that's about it |
| 13:20.04 | leifmadsen | if the clients accept video, it will get it |
| 13:20.18 | MariusKarthaus | junmin: but will hangup not cause the line to be hung up on the calling party by server B if server A will be able to pick it up but just be a microsecond slower to repond ?later |
| 13:22.42 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 13:24.10 | junmin | MariusKarthaus: yes. |
| 13:25.29 | Jo_Bell | leifmadsen: When I add different vendor devices such as lifesize, polycom etc, is there anything custom need to be done? Will they just work? Sorry if my question is primitive |
| 13:25.40 | MariusKarthaus | that is not what I need. I deed server B to ignore the call if the called number matches xyz and server A to handle it (for instance by using DIAL to connect to call to a logged in SIP phone) |
| 13:26.11 | leifmadsen | Jo_Bell: well asterisk has pretty primitive support for video... the end points that speak video need to be pretty generic and it should "just work" |
| 13:26.21 | leifmadsen | if there is any custom stuff those endpoints are doing, then it won't work |
| 13:26.26 | leifmadsen | or won't be supported |
| 13:26.45 | Lantizia | Anyone know how to debug *8 call pickup? |
| 13:26.56 | Katty | hello my asterisk does not work at all how to fix plz |
| 13:26.58 | leifmadsen | Jo_Bell: asterisk can only pass through video, but not really do anything fancy |
| 13:27.09 | leifmadsen | Katty: I suggest getting a cat |
| 13:27.20 | [TK]D-Fender | Jo_Bell, no codec conversion, no multiplexing, nothing. * is not a video platform |
| 13:27.25 | Katty | why |
| 13:27.42 | leifmadsen | because it appears cats run the internet |
| 13:27.46 | Lantizia | I have peer 3649 ringing, so 3648 dials *8 and nothing happens (both are in callgroup & pickupgroup 71), nothing shows on the console unless sip debug is on for peer 3648 which shows it had dialed *8@<ip of pbx> in the To: line - but ultimately hangs up... and yes pickupexten is set to *8 in features.conf. Any ideas? |
| 13:27.51 | Jo_Bell | Ok. thanks leifmadsen: polycom will work right? |
| 13:27.55 | leifmadsen | yes |
| 13:27.56 | Lantizia | meow |
| 13:27.57 | leifmadsen | afaik |
| 13:28.03 | leifmadsen | jitsi works for sue |
| 13:28.05 | leifmadsen | sure* |
| 13:28.08 | leifmadsen | it's a softphone client |
| 13:28.28 | leifmadsen | there isn't really a lot of video stuff to configure in asterisk, you enable it, and that's about it |
| 13:28.45 | Jo_Bell | leifmadsen: what about lifesize? any information? |
| 13:28.47 | leifmadsen | other than allowing the peers to use the h264 codec (allow=h264) |
| 13:28.54 | leifmadsen | Jo_Bell: no idea, never used it |
| 13:28.57 | leifmadsen | you'll just have to try |
| 13:29.03 | junmin | leifmadsen: can you help me with my transfer call problem, please? |
| 13:29.28 | leifmadsen | junmin: sorry, was just in a conference call, and now it is done, so back to work for me |
| 13:30.13 | Jo_Bell | ok.. fine. leifmadesen: what about multiple videos then? I mean if the device supports multiple video display, will it be supported by asterisk? |
| 13:30.31 | junmin | leifmadsen: cool. i got a really strange problem, transfering call not working, can not find the cause |
| 13:31.02 | leifmadsen | Jo_Bell: if it can accept multiple streams via multiple connections, then perhaps |
| 13:31.02 | [TK]D-Fender | Jo_Bell, no |
| 13:31.10 | leifmadsen | Jo_Bell: at this point, you'll probably just need to try it |
| 13:31.39 | junmin | leifmadsen: just can hear the "transfer" message, but the other end cannot hear music on hold, and nothing happed, and call seems dropped :s |
| 13:31.53 | leifmadsen | junmin: sorry, not sure |
| 13:31.57 | leifmadsen | junmin: check the sip trace |
| 13:32.07 | leifmadsen | can only assume you're doing sip transfers... |
| 13:32.36 | junmin | leifmadsen: yes it's sip transfer |
| 13:32.55 | *** join/#asterisk bmg505 (~leon@196-209-101-109.dynamic.isadsl.co.za) |
| 13:38.17 | [TK]D-Fender | Jo_Bell, If you send multiple calls to it and the device supports it's own multiplexing then it may work if everything else matches. |
| 13:38.25 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:38.26 | *** join/#asterisk bchia (~Adium@nat/digium/x-jxgwnvbftxceurhy) |
| 13:40.22 | Jo_Bell | D-Fender: My doubt itself is very cloudy. Sorry for that. Just want some information before starting with the code and scripts. The issue that I saw is "Is Asterisk capable to maintain multiple media streams with the same device? " |
| 13:41.18 | junmin | leifmadsen: thank~ |
| 13:41.46 | leifmadsen | Jo_Bell: multiplexing of video is not possible with Asterisk |
| 13:42.01 | leifmadsen | if you multiplex it prior to Asterisk and send a single stream, then that is OK |
| 13:42.06 | leifmadsen | but asterisk will not "mix" video signals for you |
| 13:42.35 | leifmadsen | if you use ConfBridge() in Asterisk 10 or later, it can take multiple streams and switch between them (single stream, latest talker mode) |
| 13:44.10 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
| 13:44.35 | Jo_Bell | leifmadsen: I dont want to mix. |
| 13:44.42 | leifmadsen | ok, then there you go |
| 13:45.29 | Jo_Bell | The separate video data packets from separate clients through a single channel. Is it possible? |
| 13:45.44 | leifmadsen | that sounds like mutliplexing to me.... |
| 13:45.49 | leifmadsen | you can have a single client send a single stream |
| 13:45.56 | *** join/#asterisk serafie (~erin@nat/digium/x-mpokjaarhoeaflbn) |
| 13:46.07 | Jo_Bell | sure.. |
| 13:46.16 | leifmadsen | or you can have multiple clients each sending a single stream |
| 13:46.23 | leifmadsen | which are all separate channels |
| 13:46.31 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 13:47.05 | [TK]D-Fender | Jo_Bell, Yes, * will send whatever the device will accept |
| 13:47.37 | [TK]D-Fender | <Jo_Bell> The separate video data packets from separate clients through a single channel. Is it possible? <- no |
| 13:48.08 | [TK]D-Fender | Jo_Bell, separate clients = separate calls. There is no multiplexing |
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| 14:27.51 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 14:27.55 | mcolombo | hi all |
| 14:30.15 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 14:31.00 | mcolombo | Does asterisk support PRACK/100Rel ?How can i enable it? Thanks |
| 14:32.53 | mjordan | mcolombo: no, but oej has a branch that is currently in testing to support that feature |
| 14:34.49 | mcolombo | Thanks for reply |
| 14:34.51 | mcolombo | Do you think we can manipulate INVITE/OK requests from SBC to add PRACK/100Rel? |
| 14:35.41 | mjordan | mcolombo: I can't speak for the behavior - oej might be able to answer any questions about what the branch supports |
| 14:38.20 | mcolombo | ok, thanks |
| 14:41.55 | leifmadsen | before I go filing an issue, anyone know why asterisk would send a BYE without a nonce? Is the nonce even a requirement? We're seeing a BYE with the nonce on the 1.4 servers, but not 1.8 |
| 14:42.05 | leifmadsen | although this is an older version of 1.8 (1.8.8.2 I believe) |
| 14:42.10 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
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| 15:01.31 | wdoekes | leifmadsen: BYE with a *nonce*? |
| 15:01.49 | leifmadsen | ya that's what one of my engineers is saying is happening... I found it.... odd |
| 15:01.54 | leifmadsen | but I haven't looked at the trace yet |
| 15:01.57 | leifmadsen | let me ask for that :) |
| 15:05.17 | *** join/#asterisk RubyRails (~jaiken@stg.maculon.com) |
| 15:08.05 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:08.37 | oej | Hello! |
| 15:08.55 | oej | mcolombo: I have PRACK support for Asterisk 1.8 |
| 15:09.13 | oej | What do you want to know? |
| 15:12.25 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:13.15 | beardy | What PRACK is. |
| 15:13.16 | kaldemar | leifmadsen: was the call authenticated in the first place? |
| 15:13.32 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 15:13.59 | mcolombo | oej: I need to know how to enable it. Thanks |
| 15:15.10 | oej | If you run my code, it's simple. Just add prack=yes to the general section of SIP.conf. |
| 15:15.18 | oej | But you need to download my code from subversion |
| 15:15.21 | oej | Or apply this patch |
| 15:15.28 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:15.53 | oej | http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/patches/darjeeling-prack-1.8.diff |
| 15:16.37 | autofsckk | hi, i want to have like a little call center, just 2 telephones by now, receiving 2 DID, i want to receive the calls in one telephone, and if it is occupied that the call goes to the other phone, how can i do that? |
| 15:19.40 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
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| 15:23.11 | mcolombo | Ok! i will try it |
| 15:23.24 | mcolombo | i'll send you a feedback |
| 15:23.30 | mcolombo | Thanks |
| 15:24.29 | junmin | which pattner should i use for "extension not found in context"? is it I? for invalid number |
| 15:25.54 | Qwell | i, but it's only used in certain cases |
| 15:25.55 | junmin | pattern* |
| 15:27.35 | [TK]D-Fender | junmin, for a direct "call in" that doesn't match it will simply 404 on SIP, or error out as the signalling allows |
| 15:27.46 | [TK]D-Fender | junmin, :i: is used for IVR's only really |
| 15:28.41 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 15:28.46 | junmin | [TK]D-Fender: oh... got it,thank you |
| 15:30.28 | pigpen | hi all, i am having an issue w/ 1.8.13.0, parking calls. the call is parked, then hangs up. the only cli sign of this is a 'Parked/SIP/pstn-00000098<ZOMBIE>' |
| 15:30.38 | pigpen | worked fine for a month |
| 15:30.47 | pigpen | had a power outage 34 hours ago. |
| 15:31.06 | pigpen | and naturally, i broke my r-hand, and can hardly type. |
| 15:32.32 | pigpen | well, i think i did, was on way to urgent care when i got the call |
| 15:33.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:33.36 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 15:34.14 | Lantizia | I have peer 3649 ringing, so 3648 dials *8 and nothing happens (both are in callgroup & pickupgroup 71), nothing shows on the console unless sip debug is on for peer 3648 which shows it had dialed *8@<ip of pbx> in the To: line - but ultimately hangs up... and yes pickupexten is set to *8 in features.conf. Any ideas? |
| 15:39.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-127.mobile.uci.edu) |
| 15:50.24 | *** join/#asterisk pagec (~chatzilla@s1.corebizserv.com) |
| 15:52.07 | pagec | Asterisk 1.8.13.1 voicemail records (files exist and play back properly in a wav player) people's unavailable messages but doesn't play them back. It will play back temporary greetings. Anyone have any ideas? |
| 15:53.37 | WIMPy | Temporary overrides all IIRC. |
| 15:55.48 | pagec | is there some setting in voicemail then that turns on/off the playing of custom messages? |
| 15:57.20 | [TK]D-Fender | delete the temporary |
| 15:57.26 | [TK]D-Fender | there is no "switch" |
| 15:57.33 | [TK]D-Fender | if it exists it overides. |
| 15:57.46 | [TK]D-Fender | picking that option in VMM will remove it. |
| 15:57.56 | pagec | yes i did that, sorry, i was saying that feature worked, the unavail and busy aren't working |
| 15:58.15 | [TK]D-Fender | they di if the temp is gone |
| 15:58.19 | [TK]D-Fender | do* |
| 15:58.29 | pagec | the temp greeting is gone and they don't work |
| 15:58.33 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 15:58.36 | [TK]D-Fender | pagec, SHOW us the failure |
| 15:58.37 | [TK]D-Fender | ~pb |
| 15:58.37 | 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. |
| 15:58.39 | [TK]D-Fender | ^^^ |
| 15:59.01 | pagec | i just tried the temp greeting in the course of trouble shooting and saw that it worked |
| 15:59.12 | [TK]D-Fender | along with an "ls -la" dump of the whole folder |
| 16:00.55 | *** join/#asterisk rjvvliet (~rjvvliet@2001:610:600:86a5:226:b9ff:fe96:d6c2) |
| 16:01.42 | junmin | hello, I am serveral out bound SIP trunk, when i define dialplan, i would use whichever availble to make the outbound call, using priorities is the only way to do that? or there is another way eaiser (to not write so much)? |
| 16:02.41 | pagec | there isn't an failure, the expected behavior is that it plays a file /var/spool/asterisk/voicemail/default/805/unavail.wav. the current behavior is it just plays vm-intro.gsm. ls -la: http://pastebin.com/v55JJG44 |
| 16:03.15 | pigpen | hi all, i am having an issue w/ 1.8.13.0, parking calls. the call is parked, then hangs up about 15 sec later. the only cli sign of this is a 'Parked/SIP/pstn-00000098<ZOMBIE>' |
| 16:03.27 | rjvvliet | Hi guys, i am using cdr_adaptive ODBC and the calldate column is not updated, the value stays the default, any one has an idea what could cause this? |
| 16:03.53 | [TK]D-Fender | pagec, I said to show the failure. I want to see the complete CLI of your call including hitting Voicemail |
| 16:03.57 | pigpen | worked fine for a month , had a pwer outage when before ran fine for over a month. |
| 16:05.05 | pagec | ok here is the CLI output at verbose 10 http://pastebin.com/hQEkFpus |
| 16:05.41 | [TK]D-Fender | pagec, "core show application voicemail" <- you aren't following the instructions. You have to tell which to use |
| 16:05.46 | *** join/#asterisk acrg (~aragon@decoder.geek.sh) |
| 16:06.11 | acrg | hiya |
| 16:06.22 | acrg | fun startup error: |
| 16:06.24 | acrg | [Jul 9 12:08:08] WARNING[2054]: file.c:1442 ast_format_str_reduce: ignoring unknown format 'wav' |
| 16:06.27 | acrg | [Jul 9 12:08:08] WARNING[2054]: file.c:1472 ast_format_str_reduce: no known formats found in format list (wav) |
| 16:06.30 | acrg | [Jul 9 12:08:08] ERROR[2054]: app_voicemail.c:12655 actual_load_config: Error processing format string, defaulting to format 'wav' |
| 16:06.56 | acrg | certified asterisk 1.8.11cert4 |
| 16:07.14 | acrg | format_wav is being loaded |
| 16:07.29 | Qwell | acrg: known issue - it's been fixed, but not in any release yet |
| 16:07.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:07.48 | Qwell | If you're only trying to use wav, then you can ignore it. |
| 16:07.55 | acrg | okie |
| 16:07.58 | acrg | thanks Qwell |
| 16:08.39 | *** join/#asterisk mishino (~mishino@bin.bash.sk) |
| 16:08.51 | *** part/#asterisk acrg (~aragon@decoder.geek.sh) |
| 16:09.07 | mishino | hi, 2n voiceblue guru? :) |
| 16:09.43 | pigpen | any handy way of debugging the parking feature? |
| 16:09.55 | pigpen | other then core set debug 9? |
| 16:11.20 | *** part/#asterisk mishino (~mishino@bin.bash.sk) |
| 16:11.22 | pagec | [TK]D-Fender: ty, i was using invoking the old style, that stopped working with 1.8 |
| 16:11.46 | [TK]D-Fender | pagec, no. -- Executing [7805@internal:1] VoiceMail("SIP/cpage-quMc0Dujyea98pwurZlQ-00000047", "805") in new stack |
| 16:12.04 | [TK]D-Fender | pagec, The old style you would put "u" or "b" BEFORE the box #. You didn't do that either |
| 16:12.30 | [TK]D-Fender | pagec, It was simply omitted altogether |
| 16:12.54 | *** join/#asterisk ctaloi (~ctaloi@nat-66-218-1-130.usadatanet.com) |
| 16:13.25 | [TK]D-Fender | And trying to use the old style will gt you a box # mismatch for the polluting char |
| 16:13.38 | rjvvliet | Anyone has a suggestion, i am using cdr_adaptive ODBC and the calldate column is not updated, the value stays the default all other columns are inserted. |
| 16:14.07 | pagec | [TK]D-Fender not sure why that didn't show up there, might have been a few revisions in trying to figure it out, when I went to fix the base extensions file there was leading u and b |
| 16:15.05 | pagec | i forgot to put in I upgraded to 1.8 a few months ago, no one noticed the voicemail was wrong until now though |
| 16:15.15 | [TK]D-Fender | pagec, Either way you should not have the correct syntax to fix your call and should be gettnig the result we are looking for. |
| 16:15.51 | *** join/#asterisk mcf37821 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 16:22.59 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.12) |
| 16:27.31 | rjvvliet | Finally figured it out, cdr_adaptive_odbc is not using the calldate column. it working now. |
| 16:28.23 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 16:28.39 | TheCompWiz | can anyone help me figure out how to determine what queue a call is in from within the dialplan? |
| 16:29.57 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:30.04 | [TK]D-Fender | theYou can't You need to look at deeper levels (AMI, etc) |
| 16:30.13 | [TK]D-Fender | TheCompWiz, ^ |
| 16:30.36 | TheCompWiz | so... no channel variable or anything :( |
| 16:31.08 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 16:31.48 | rjvvliet | TheCompWiz, maybe the queue_variables might help. |
| 16:32.51 | rjvvliet | TheCompWiz, Sorry, does not contain the queue name. (https://wiki.asterisk.org/wiki/display/AST/Function_QUEUE_VARIABLES) |
| 16:33.12 | rjvvliet | TheCompWiz, i should listed better to [TK]D-Fender ;-) |
| 16:33.15 | TheCompWiz | what version of asterisk was that introduced in? |
| 16:34.04 | TheCompWiz | scratch that... not what I needed... /sigh |
| 16:36.09 | rjvvliet | TheCompWiz, what would you,like to do? |
| 16:36.27 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 16:36.38 | TheCompWiz | trigger some stuff after someone has hungup after going through a queue. |
| 16:36.48 | TheCompWiz | logging stuff mostly. |
| 16:37.25 | rjvvliet | TheCompWiz, A, had somthijng similar, Just Set a variablle Before entering the queue. |
| 16:38.11 | TheCompWiz | rjvvliet: I guess that's what I'll have to do. Was trying to avoid re-inventing the wheel.... but apparently nobody created one to start with. |
| 16:38.54 | rjvvliet | TheCompWiz, i needs this for the queue calls waiting in de CallerID, so i set the QUEUENO before sending the call to the queue (and yes working around that new wheel keeps you think in circles) |
| 16:39.15 | rjvvliet | TheCompWiz, Sorry no pun intended.... |
| 16:39.36 | TheCompWiz | lol... accurate as the pun is... |
| 16:42.10 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:43.34 | junmin | how can i know if a installed asterisk has RealTime support? |
| 16:43.48 | [TK]D-Fender | Use it |
| 16:44.15 | pigpen | cli type realtime show <tab> |
| 16:44.24 | pigpen | [TK]D-Fender, man, I need some help |
| 16:44.32 | pigpen | weird one. |
| 16:45.02 | [TK]D-Fender | pigpen, If its the same question you've been repeating every half hour I'm about to need another hand for the count.... |
| 16:45.22 | pigpen | k. sorry, just got new info. |
| 16:45.35 | [TK]D-Fender | pigpen, And I don't have that answer, and you didn't include comprehensive debug that I recall for anyone to help you with |
| 16:45.37 | pigpen | ie: it works fine hitting the same place in the dialplan via iax. |
| 16:46.33 | pigpen | yeah, a I also asked if there is any special debug needed for troubleshooting the feature |
| 16:47.05 | pigpen | but atleast now It seems sip related, so I can at least debut that. |
| 16:47.08 | pigpen | sorry for posting. |
| 16:47.48 | junmin | thx |
| 16:50.06 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 16:50.48 | nny | is it still considered normal practice to end all extension definitions with a Hangup to avoid any accidental loops? |
| 16:51.22 | Qwell | If you've got loops, you've got other problems. |
| 16:51.26 | Qwell | But, sure, it won't hurt anything. |
| 16:52.03 | nny | Qwell: indeed, but I was taught to throw a Hangup() at the end as a simple safety measure. I don't think they're needed, but just not sure if most people practice it or I am being overly paranoid |
| 16:52.37 | leifmadsen | I only put them there as an EOL that shows i mean for things to really hang up |
| 16:52.43 | leifmadsen | more as a visual than anything |
| 16:52.44 | [TK]D-Fender | nny, Yes, good idea |
| 16:52.48 | bchia | Explicit Hangup() is going to save your from strange behavior with pattern matching |
| 16:52.50 | leifmadsen | it's not for loop control |
| 16:53.03 | [TK]D-Fender | nny, Only really a problem if you have autofallthrough disabled. |
| 16:53.44 | nny | cool thanks for the input, sanity check and all that |
| 16:53.54 | nny | is still insane |
| 17:06.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:08.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:10.52 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:31.22 | *** join/#asterisk Coren (~Coren@wikipedia/Coren) |
| 17:32.49 | Coren | Quick Q: the doc says that the lua dialplan is executed every time a channel is brought up. Is it possible that this is no longer true in 10? |
| 17:33.13 | Coren | (Because it seems to be executed once at startup) |
| 17:35.00 | kaldemar | define "is executed" |
| 17:36.15 | Coren | kaldemar: Code at the topmost level is executed, not just a call through the extensions array. |
| 17:36.28 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 17:36.37 | Coren | Quoth the doc: "The extensions.lua file is loaded into memory once when the pbx_lua module is loaded or reloaded. The file is then read from memory and executed once for each channel that looks up or executes a lua based extension." |
| 17:37.09 | Coren | Except that, as far as I can tell, that occurs only exactly once on load. |
| 17:37.20 | Qwell | Coren: It's dialplan. |
| 17:38.53 | kaldemar | reading the dialplan upon startup and using it to process a call are two separate things. |
| 17:40.09 | Coren | kaldemar: I know that, but the doc implies that topmost code gets executed every time a channel is brought up (presumably before the function is called through the table) |
| 17:40.33 | Coren | Oh, wait, it *is*. It's just that the app and channels objects aren't pointing anywhere at that time. |
| 17:41.01 | Coren | just tried an ugly-ass print(). Oldest debug method yet. |
| 17:41.24 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:41.26 | Ice_Strike | Hi |
| 17:42.27 | Ice_Strike | Why SendDTMF is not executing? see http://pastebin.com/s8jgT3FM |
| 17:43.52 | nny | have a strange issue, not sure what to look at explicitly yet. We have an office with some phones that are remote (over VPN). They can't use the phone's conference feature. - When they hit join it dies. The local phones perform this function normally. I am setting up a test now, not sure if i'll see anything since the "join" is on the phone's side. |
| 17:44.15 | Qwell | Ice_Strike: Dial does not return once the call is established. |
| 17:44.45 | Qwell | You'll need to have it call a macro. See core show application dial. |
| 17:44.49 | kaldemar | Ice_Strike: see option D() for app Dial |
| 17:44.58 | Ice_Strike | Qwell once call is established then SendDTMF is not executed? |
| 17:45.24 | Ice_Strike | I see |
| 17:45.27 | Ice_Strike | I think I understand now |
| 17:46.19 | Ice_Strike | thanks |
| 17:47.58 | *** part/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:56.27 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
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| 18:10.41 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
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| 18:35.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:39.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.40.134) |
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| 19:00.07 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 19:02.04 | McBoingBo | I am still trying to get distinctive rings working with my Polycom, I add SIPAddHeader "Alert-Info: Custom 4" <--- which is presetup in the polycom config file, I figured why make a new one. Asterisk console showing me that it is adding the sipaddheader but ring is same everytime, anyone have time to help out a noob? |
| 19:03.37 | McBoingBo | the article I was reading was a little confusing, as it was referencing files like sip.cfg and ipmid.cfg, but I would guess that mine are just named differently |
| 19:13.42 | *** join/#asterisk Hans67521 (~jcputter@41-135-41-111.dsl.mweb.co.za) |
| 19:13.54 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:14.09 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 19:14.30 | *** join/#asterisk mcf37821 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:15.38 | Hans67521 | noob here i need some help, i configure a test lab setup with asterisk, when connecting from outside the network and i phone an extension the call drops after 30seconds with chan_sip.c retransmission timeout reached on transmission |
| 19:15.58 | Hans67521 | now i show i has to do with NAT |
| 19:16.19 | Hans67521 | however i disabled ALG on the router, but still have the problem |
| 19:16.40 | WIMPy | I don't understand the last sentence, but yes, that's a networking issue. |
| 19:16.47 | WIMPy | ~sipnat |
| 19:16.47 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 19:18.45 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 19:19.17 | Hans67521 | here is my sip.conf |
| 19:19.21 | Hans67521 | http://pastebin.com/n2VXLbTH |
| 19:19.39 | Hans67521 | dont laugh to hard.... |
| 19:19.47 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:19.50 | Katty | hello my asterisk does not work at all how to fix plz. |
| 19:20.05 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 19:20.16 | WIMPy | Hans67521: You should change your passwords. |
| 19:20.20 | WIMPy | NOW. |
| 19:20.29 | Hans67521 | OK |
| 19:20.31 | Hans67521 | lol |
| 19:20.45 | sruffell | you lol until you get hit with $1000 in charges.... |
| 19:21.11 | sruffell | ..then it's col. |
| 19:21.22 | Hans67521 | ok done |
| 19:22.09 | Hans67521 | how does that conf file look? |
| 19:22.37 | leifmadsen | line 22 needs a comment |
| 19:22.39 | WIMPy | Looks ok to me. |
| 19:22.56 | leifmadsen | line 22 should start with a semi-colon |
| 19:24.39 | Hans67521 | ya saw that |
| 19:25.04 | Hans67521 | still dont see why i get retransmission timeouts |
| 19:25.26 | *** join/#asterisk Osage_ (~Osage@71-81-57-248.dhcp.slid.la.charter.com) |
| 19:26.19 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 19:27.11 | nny | have a strange issue, not sure what to look at explicitly yet. We have an office with some phones that are remote (over VPN). They can't use the phone's conference feature. - When they hit join it dies. The local phones perform this function normally. I am setting up a test now, not sure if i'll see anything since the "join" is on the phone's side. |
| 19:27.51 | Hans67521 | leifmadsen, i dont know what to try next |
| 19:28.15 | Hans67521 | locally the calls work 100% |
| 19:28.29 | Hans67521 | so i think it does have to do with NAT |
| 19:28.35 | WIMPy | nny: You shouldn't see anything on your side. |
| 19:28.42 | nny | Hans67521: the timeout is basically asterisk saying "well, that ended poorly" |
| 19:28.57 | *** join/#asterisk mimage (~mimage@fedora/mimage) |
| 19:29.00 | leifmadsen | sounds like your firewall/router/nat isn't either receiving an ACK from the other side, or the other sides NAT isn't allowing your 200 OK through (or something similar) |
| 19:29.15 | leifmadsen | if you look at the trace on the sip console, I suspect you'll see a lot of retransmissions that aren't getting acknowledged |
| 19:29.17 | [TK]D-Fender | incomplete NAT setup |
| 19:29.24 | leifmadsen | nothing you can change on asterisk will alleviate that issue -- fix your firewall/nat |
| 19:29.28 | nny | or a sonicwall acting par for the course |
| 19:29.40 | [TK]D-Fender | incomplete NAT setup <- |
| 19:29.40 | Hans67521 | Thanks guys |
| 19:29.51 | leifmadsen | omg </3 sonicwalls |
| 19:29.55 | nny | agreed |
| 19:30.05 | leifmadsen | had nothing but problems with them |
| 19:30.18 | leifmadsen | given it's been 5-6 years since I've used one |
| 19:30.22 | leifmadsen | and that's in purpose |
| 19:30.50 | nny | yeah one and done with me too ha. Every once in a while I get a client with odd issues and that's my first question. |
| 19:35.54 | *** join/#asterisk gandhijee_ (~akp@50.12.169.99) |
| 19:38.03 | gandhijee_ | hey has anyone here gotten a SIP trunk from AT&T for Verizon? |
| 19:42.38 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 19:47.20 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 19:49.27 | jeffspeff | i'm having the weirdest issue... so new asterisk box setup, i setup the SIP provider, however, it won't register with the provider.... I'm sending them options requests, they reply back with a 200 OK then it just repeats! It's about to drive me mad. here's the sip debug log http://pastebin.com/Bzt1rfhr 68.68.118.47 is them and 12.133.5.160 is my public ip. any ideas on what's going on? |
| 19:54.23 | [TK]D-Fender | I see nothing unusal ther other than you set "nat=yes" for a provider peer which you so almost never do |
| 19:55.13 | wdoekes | [TK]D-Fender: nat=yes is default nowadays |
| 19:55.33 | [TK]D-Fender | ... |
| 19:55.37 | [TK]D-Fender | no |
| 19:55.58 | [TK]D-Fender | If you use a vendor who has separate medai VS signalling servers, you are SCREWED |
| 19:55.59 | jeffspeff | [TK]D-Fender, i've tried it with both yes and no, this provider doesn't require registration, they do a straight IP configured way... i have another box at a different location using the same provider and it has nat set to yes and its been working great... i'm out of ideas here, but i have tried that set to both ways |
| 19:56.26 | Katty | hello my asterisk does not work at all how to fix plz. |
| 19:56.28 | *** join/#asterisk sweettea (d1eaa282@gateway/web/freenode/ip.209.234.162.130) |
| 19:56.31 | [TK]D-Fender | "however, it won't register with the provider" .............. "this provider doesn't require registration" |
| 19:56.34 | [TK]D-Fender | W-T_F |
| 19:56.46 | [TK]D-Fender | MAke up your mind |
| 19:56.51 | jeffspeff | ? |
| 19:57.00 | sweettea | Anyone use AsteriskNOW? I forgot which ISO/version I downloaded, any way to tell that via looking at the install? I cant find it |
| 19:57.03 | wdoekes | jeffspeff: it's not adding a to-tag, so asterisk isn't acknowledging the options response.. but that's irrelevant. the options has 0 to do with registration |
| 19:57.16 | [TK]D-Fender | "it won't register" <- you're implying it SHOULD BE TRYING |
| 19:57.29 | [TK]D-Fender | jeffspeff, and then a statment of "it isn't needed" |
| 19:58.00 | jeffspeff | [TK]D-Fender, sorry, when i said that it won't register, i was referring to qualify |
| 19:58.01 | [TK]D-Fender | jeffspeff, You are massively unclear as to what you are doing VS expecting |
| 19:58.21 | jeffspeff | it doesn't require sip registration via a register string, username or password |
| 19:58.26 | wdoekes | as for the qualify: the provider is sending crap |
| 19:58.41 | jeffspeff | when you run sip show peers, the peer shows as unreachable |
| 19:58.54 | [TK]D-Fender | <--- SIP read from UDP:68.68.118.47:5060 ---> SIP/2.0 200 OK |
| 19:59.01 | [TK]D-Fender | They answer what is the problem? |
| 19:59.05 | wdoekes | no to-tag |
| 19:59.14 | jeffspeff | wdoekes, i saw a few places where it was sending to anonymous@68.68.118.47 is that the way it's supposed to be? |
| 19:59.30 | jeffspeff | reference line 49 |
| 19:59.42 | wdoekes | irrelevant |
| 20:00.02 | *** join/#asterisk adeel|work (~adeel@216.183.80.220) |
| 20:00.03 | jeffspeff | ok, line 46 shows the "to" you were referring to doesn't it? |
| 20:00.26 | wdoekes | To: <sip:68.68.118.47> <-- + ";tag=abcdef" |
| 20:01.38 | jeffspeff | wdoekes, so it's something messed up on the providers end? |
| 20:02.05 | wdoekes | does 10.64.0.15 mean anything to you? |
| 20:02.12 | wdoekes | because the call-id is screwed up too |
| 20:02.23 | wdoekes | as is the via.. |
| 20:02.31 | jeffspeff | that's the private IP of my box |
| 20:02.35 | jeffspeff | my * box |
| 20:03.16 | wdoekes | in that case there is some alg going on |
| 20:03.41 | jeffspeff | our provder mentioned something about that |
| 20:03.47 | jeffspeff | does it need to be on or off |
| 20:03.48 | wdoekes | start by removing the externhost=/externip= |
| 20:03.56 | wdoekes | or disable the alg |
| 20:04.18 | wdoekes | (I'd prefer the alg, *if* it works 100%, most of them don't) |
| 20:04.40 | jeffspeff | remove that from where? |
| 20:06.40 | jeffspeff | wdoekes, remove the " externhost=/externip=" from where? |
| 20:06.56 | wdoekes | how does asterisk know your external ip? |
| 20:07.04 | jeffspeff | sip.conf |
| 20:07.37 | jeffspeff | sorry, it's been a long monday... |
| 20:07.52 | jeffspeff | isn't the externip required for natting? |
| 20:08.14 | wdoekes | it generally only confuses the providers that know about nat |
| 20:08.18 | McBoingBo | gah so close to making ringtones lol, I used "sox Untitled.wav -r 8000 -U -c1 Untitled.wav resample -ql" and got the choice on my phone but it plays nothing |
| 20:08.24 | wdoekes | those that don't are better server with an alg |
| 20:08.30 | wdoekes | s/server/served |
| 20:11.27 | jeffspeff | wdoekes, uhm so yeah, i disabled externip and voila it works! |
| 20:12.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:12.54 | [TK]D-Fender | remove ALG, leave the externhost, etc. |
| 20:18.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 20:18.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:23.57 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 20:26.26 | *** join/#asterisk jero (~boo@mtl.savoirfairelinux.net) |
| 20:27.26 | jero | is it possible to tell asterisk to 'lock' itself to the first registered SIP contact in case several are registering using the same account ? |
| 20:30.54 | [TK]D-Fender | jero, No. |
| 20:31.12 | jero | thanks :) |
| 20:31.29 | anonymouz666 | jero: the second client registering is coming from the same network? |
| 20:31.33 | jero | anonymouz666: yes |
| 20:31.38 | anonymouz666 | so, no. |
| 20:31.54 | anonymouz666 | you can do that easily with a sip proxy. |
| 20:32.08 | jero | yeah |
| 20:32.18 | anonymouz666 | just by limiting the max contacts at 1. |
| 20:32.48 | [TK]D-Fender | checkout time, later all |
| 20:33.19 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:39.19 | sweettea | Anyone use AsteriskNOW? I forgot which ISO/version I downloaded, any way to tell that via looking at the install? I cant find it |
| 20:45.33 | *** join/#asterisk gusto (~gusto@DSL01.83.171.177.118.ip-pool.NEFkom.net) |
| 20:45.56 | gusto | hey, i can not log in to asterisk -r with my user account on freebsd, any idea what i am doing wrong? |
| 20:46.12 | gusto | i have asterisk started with -U asterisk -G asterisk and the user is also in the asterisk group |
| 20:47.56 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 20:50.36 | TheCompWiz | look for the pid file... and make sure it exists/can-be-created/etc... |
| 20:54.03 | gusto | yes |
| 20:54.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.52 | gusto | /var/run/asterisk/asterisk.pid |
| 20:55.33 | gusto | and i can access it from the user account as well |
| 20:56.17 | gusto | maybe it's because i have no writing permission to asterisk.ctl? |
| 20:56.37 | jpsharp | That would be it. |
| 20:57.02 | gusto | well it works now |
| 20:57.22 | jpsharp | asterisk.ctl is the pipe that lets you connect to the console. |
| 21:03.58 | *** join/#asterisk fling (~fling@unaffiliated/fling) |
| 21:25.02 | gusto | in the meantime i figured out that there is a parameter to set it, so that it stays so |
| 21:25.34 | gusto | because just changing the permissions is gone after first restart, but asterisk.conf has a parameter in [files] astctlpermissions = 0660 |
| 21:27.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hrgpuzyzskruhwrp) |
| 21:38.10 | PaybackTony | So, I was having an issue with agi scripts, for the life of me I couldn't figure out why I was getting a file does not exist on my php and custom built perl agi scripts. I just found out what the issue was. I used Aptana in windows and uploaded the scripts and the "DOS Format" (line endings etc) was giving the asterisk interpreter problems. No idea how this correlates or if it would qualify as a "bug", but simple copying m |
| 21:38.26 | Qwell | Not a bug. |
| 21:38.30 | Qwell | Don't do that. |
| 21:39.01 | PaybackTony | I don't necessarily see it as a bug either, but I definitely spent too much time on google trying to figure it out before figuring it out on my own |
| 21:39.47 | Qwell | So don't use Windows. :) |
| 21:40.16 | PaybackTony | Then tell EA to release more of their games for Linux and I won't :-p |
| 21:40.26 | leifmadsen | VMs ftw |
| 21:40.27 | Qwell | wine works wonders |
| 21:40.30 | Qwell | or ^^ |
| 21:40.41 | leifmadsen | virtualbox your woes away |
| 21:41.30 | PaybackTony | I use Mac here at the office, which I actually like working on |
| 21:41.50 | jpsharp | Get a Wintendo box for playing games, a Mac or Linux box for real work. |
| 21:42.07 | drmessano | Wintendo.. LOL |
| 21:42.35 | PaybackTony | I have osx86 installed on a drive in my home computer but it's a pain to boot out, change bios settings and boot into often |
| 21:43.03 | PaybackTony | probably a good idea to virtualbox it though (linux) |
| 22:05.35 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:05.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:12.47 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.12) |
| 22:24.27 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 22:24.46 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:37.53 | *** join/#asterisk JonCup (~Jon@static-108-0-128-28.lsanca.fios.verizon.net) |
| 22:41.56 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-7-139.bois.qwest.net) |
| 22:50.00 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 22:50.26 | kuku | Anyone have a working sip.cfg working for dual firmware? For the 331 and 330 firmware? |
| 23:02.20 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 23:07.34 | pavlz_ | is there a way to use SIP, without to pay ? |
| 23:08.52 | pavlz_ | if i am phoning, and i am using my ADSL, i don't understand, why to phone, i have to use SIP and to pay per phone |
| 23:09.32 | pavlz_ | i am just paying the ADSL, why to pay another ISP to use SIP and to phone ? |
| 23:09.46 | junmin | what ... |
| 23:09.54 | junmin | cheaper? |
| 23:09.57 | pavlz_ | isn't there a way don't pay per SIP ? |
| 23:10.38 | junmin | no free lunch |
| 23:10.53 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.12) |
| 23:11.15 | pavlz_ | ISP's have SIP, how they obtain the possibility to use SIP's ? |
| 23:11.50 | pavlz_ | they phone for free, and they sell us the service based on SIP |
| 23:12.11 | pavlz_ | i want to use SIP, but i don't want to pay them |
| 23:14.17 | JonCup | hey guys, cany suggest some troubleshooting steps, i've got 5 phones in a Queue but one of the phones isnt ringing witht he que, cant figure out why |
| 23:17.28 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:29.10 | pavlz_ | how to become SIP provider ? watch: http://forum.voxilla.com/threads/how-to-become-a-sip-provider.13064 |
| 23:31.42 | WIMPy | ? |