| 00:21.41 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 00:28.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:28.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:40.06 | *** join/#asterisk moy_ (~moy@74.12.102.18) |
| 00:43.04 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 00:51.31 | *** join/#asterisk chuckf_ (~chuckf@ubuntu/member/chuckf) |
| 00:51.31 | *** join/#asterisk lirakis_ (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 00:52.38 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 00:52.45 | *** join/#asterisk DarkNet (~FreeNoden@courriel-quebec.com) |
| 00:56.05 | *** join/#asterisk ybit2 (~quassel@unaffiliated/ybit) |
| 00:56.55 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:03.08 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 01:03.32 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:04.13 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 01:04.58 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 01:30.12 | *** join/#asterisk nitram (foo@superblob.com) |
| 01:57.41 | *** join/#asterisk arielb27 (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 02:00.21 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 02:20.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:58.37 | *** join/#asterisk Kyosh (whoa@pool-74-101-225-118.nycmny.fios.verizon.net) |
| 03:00.11 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 03:10.37 | chopp | I have attach=yes under [genera] in voicemail.conf, but isn't working for some reason.l |
| 03:11.58 | ChannelZ | you need to look at mailcmd and see if A. it's sensible for your setup and B. it works if you do something similar yourself |
| 03:24.51 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 03:28.28 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 03:32.29 | chopp | ChannelZ: I have 'mailcmd=/usr/bin/sendmail -t', and it's sending an e-mail, there just isn't an attachment. |
| 03:34.13 | chopp | sendmail is attaching the .wav locally, but aplay outputs silence with them. |
| 03:47.45 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 04:01.46 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 04:02.46 | AliRezaTaleghani | hi |
| 04:03.06 | AliRezaTaleghani | can some one give me a hint how to enable H323 on asterisk 1.6? |
| 04:27.46 | *** join/#asterisk devdvd (~twister19@c-71-61-188-154.hsd1.wv.comcast.net) |
| 04:28.32 | ChannelZ | ChannelZ: make sure whatever attachfmt= you are using is also being used as a normal format= |
| 04:30.32 | devdvd | i'm using asterisk 1.6.2.10 and trying to configure srtp. I am looking at this http://www.voip-info.org/wiki/view/Asterisk+SRTP but im guessing it is out of date as the svn link to the srtp code for asterisk is no longer valid. Is SRTP built into asterisk in 1.6.10? if not where can i get the code and preferably some updated documentation on getting it working |
| 04:32.25 | drmessano | devdvd: You'll want 1.8, which is in beta |
| 04:32.42 | devdvd | why? does 1.6 not support it at all? |
| 04:33.47 | drmessano | Nope |
| 04:33.55 | devdvd | hmm |
| 04:36.11 | devdvd | native in 1.8? |
| 04:36.21 | drmessano | yes |
| 04:36.25 | devdvd | ok |
| 04:40.36 | *** join/#asterisk Alagar (~Administr@122.164.166.235) |
| 04:48.40 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-21.lv.lv.cox.net) |
| 04:48.45 | *** join/#asterisk brendansterne (~brendanst@cpe-70-124-61-17.austin.res.rr.com) |
| 04:49.47 | brendansterne | Greetings. I'm trying to debug SIP using the 'sip show history <callid>' command. But cannot figure out what the <callid> parameter should be. |
| 04:50.12 | brendansterne | The actual Call-ID value from the SIP headers does not seem to work |
| 04:50.30 | brendansterne | Anyone used this command and know what this parameter should be? |
| 04:53.25 | Besticles | yeah |
| 04:55.09 | brendansterne | ? |
| 04:55.29 | Besticles | you turned on sip call history right? |
| 04:55.37 | brendansterne | Yes |
| 04:55.49 | Besticles | Turn on sip debugging, show that it shows in the console |
| 04:55.52 | Besticles | as the call happens |
| 04:55.59 | brendansterne | I've done that |
| 04:56.00 | Besticles | you'll see the Call-ID on the actual call. |
| 04:56.14 | Besticles | the alphanumber string? |
| 04:56.15 | brendansterne | like this: Call-ID: 20bb30ab7fc74e166d2d550c47d096c2@209.189.233.166 |
| 04:56.17 | brendansterne | ? |
| 04:56.20 | Besticles | yeah |
| 04:56.22 | Besticles | not working? |
| 04:56.41 | brendansterne | No such SIP Call ID starting with '20bb30ab7fc74e166d2d550c47d096c2@209.189.233.166' |
| 04:56.46 | brendansterne | is what I get back |
| 04:58.27 | brendansterne | I did a 'sip set history on' |
| 04:58.30 | Besticles | hmm, not sure. i haven't used it in a long time. Let me see if i can get it to work on mine. |
| 04:58.47 | brendansterne | Does it only work for a call that is still 'live' |
| 04:58.51 | brendansterne | ? |
| 04:59.04 | ChannelZ | I imagine it's the asterisk call ID, like "-- SIP/vitelity-out-00000013 answered" |
| 04:59.16 | chopp | ChannelZ: still not working, this is my voicemail.conf http://pastebin.slackadelic.com/p/fPDgSG35.html |
| 05:01.10 | ChannelZ | chopp: where is your attachfmt ? |
| 05:01.27 | ChannelZ | oh you're doing it per mailbox |
| 05:01.57 | brendansterne | ChannelZ... I see in my log -> -- SIP/qaproxy1-0000003f answered Local/dial@dialcall-b2d2;2 |
| 05:02.09 | brendansterne | But |
| 05:02.17 | ChannelZ | the mailcmd should be up in the [general] section though I'm not positive it matters |
| 05:02.28 | brendansterne | this fails -> sip show history SIP/qaproxy1-0000003f |
| 05:03.28 | ChannelZ | chopp: and what version of asterisk are you running? the mailbox syntax doesn't look right |
| 05:04.31 | AliRezaTaleghani | ChannelZ: hi man |
| 05:04.31 | chopp | ChannelZ: it's 1.6.2.11 |
| 05:04.46 | AliRezaTaleghani | are u interested with h323? |
| 05:04.47 | brendansterne | OK... so while the channel is live... if I do a 'sip show channels' I get a list of live channels |
| 05:04.54 | brendansterne | One of the columns is Call ID |
| 05:05.35 | brendansterne | But its an id like '399b34364cf9abe' which doesn't appear to have anything in common with the Call-ID header, or the SIP channel id |
| 05:06.05 | brendansterne | I wonder what that call id (399b34364cf9abe) is? Any is there any way to see it from the logs, or do you have to run 'sip show channels'? |
| 05:07.15 | ChannelZ | brendansterne: ah. That is the call ID. sip show history will only show the history of active calls - it's the SIP history for the call, but not history as in you can do it 15 minutes later and see what went on |
| 05:08.16 | ChannelZ | And hello AliRezaTaleghani |
| 05:08.42 | ChannelZ | chopp: but anyway you were saying earlier it is working, it's sending a mail, and attaching something that is just blank? |
| 05:09.04 | brendansterne | ChannelZ: ok. but then it might be hard for me to get a 'sip show history' for the bug I want to report because it happens during call setup and I'm not sure there will be time to do a 'sip show channels' then a 'sip show history <callid>' |
| 05:09.32 | ChannelZ | if you're making a bug report I think you'd want the actual SIP.. |
| 05:09.36 | ChannelZ | sip set debug on |
| 05:09.54 | brendansterne | Yep.. I have that on... the bug guidelines also say to get a sip show history |
| 05:09.59 | ChannelZ | History just shows a condensed version of each message that went back and forth |
| 05:10.05 | brendansterne | Ahhh... ok |
| 05:10.09 | brendansterne | do |
| 05:10.23 | brendansterne | so sip debug should be good enough |
| 05:10.39 | brendansterne | along with a .pcap perhaps |
| 05:11.03 | chopp | ChannelZ: it's being emailed out to gmail without an attachment, and sending mail to my local user with an attachement. |
| 05:11.06 | ChannelZ | if the call setup is failing as you say yes getting a history is hard to do |
| 05:11.15 | brendansterne | Can I submit a bug agains 1.6.2.10, or does it need to be against the latest 1.6 trunk version |
| 05:11.16 | brendansterne | ? |
| 05:12.09 | drmessano | It is fixed in SVN? |
| 05:12.12 | ChannelZ | brendansterne: They'd probably prefer you test against the latest release to make sure it's something that hasn't been fixed |
| 05:12.14 | drmessano | Is it* |
| 05:12.21 | drmessano | ^^^ |
| 05:12.21 | Besticles | I've downloaded the new version of X-Lite. When I connect a call to it, and click Answer, it appears to pick up the call. The other end, only hears ringing. I have fiddled and fiddled with it, I can't seem to figure out what is causing this symptom. Any suggestions? |
| 05:12.44 | ChannelZ | chopp: maybe gmail is hosing something up |
| 05:12.45 | drmessano | Does the old version of X-Lite work? |
| 05:13.19 | AliRezaTaleghani | ChannelZ: hi man, i have some problems, and don't know the exact or best way to solve, i need to trunk between a cisco AS5300 and my asterisk.. |
| 05:13.29 | brendansterne | ok.. so I need to get the latest 1.6 from svn before submitting the bug? |
| 05:14.01 | ChannelZ | chopp: based on the syntax I see in your pastebin however, it would be emailing 'chopp', and trying to send a page to me@gmail.com |
| 05:14.06 | AliRezaTaleghani | i tried SIP trunk, but cos the AS5300 was not able to make registered users, that way didn't work |
| 05:15.03 | ChannelZ | well I have no clue about your AS5300 |
| 05:15.25 | Besticles | The old version works on my computer @ work. I am vpn'ed into the office network and decided to work on my project. So I downloaded xlite here, and noticed its a new version, it looks totally different. |
| 05:15.52 | AliRezaTaleghani | ChannelZ: umm |
| 05:15.59 | AliRezaTaleghani | How about H323? |
| 05:16.04 | AliRezaTaleghani | on asterisk i mean |
| 05:16.38 | ChannelZ | chopp: your problem is your syntax |
| 05:17.47 | chopp | isn't that correct though? extension_number => voicemail_password,user_name,user_email_address,,user_option(s) |
| 05:17.49 | ChannelZ | AliRezaTaleghani: never used h323. There's a channel driver for that.. |
| 05:17.58 | AliRezaTaleghani | ChannelZ: tnx |
| 05:18.01 | AliRezaTaleghani | :0 |
| 05:18.03 | AliRezaTaleghani | :) |
| 05:18.13 | ChannelZ | chopp: yes but you have 1 => 69ab4,1,chopp,me@gmail.com,,attach=yes|attachfmt=wav |
| 05:19.02 | ChannelZ | chopp: password=69ab4 username=1 user_email_address=chopp pager_address=me@gmail.com (that's what's between the ,, in your example) |
| 05:19.11 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 05:19.29 | ChannelZ | AliRezaTaleghani: I think there's one in asterisk-extras but I could be lying |
| 05:20.08 | AliRezaTaleghani | :D why lying? |
| 05:20.19 | AliRezaTaleghani | no problem i will search more ;) |
| 05:20.21 | AliRezaTaleghani | && tnx |
| 05:20.24 | Besticles | Dammit, I figured out right after I asked the question. My bad. |
| 05:20.59 | ChannelZ | AliRezaTaleghani: because I don't remember |
| 05:21.32 | ChannelZ | In fact it's "asterisk addons", not 'extras'. See? Lies! |
| 05:23.24 | AliRezaTaleghani | :) ok, i get it |
| 05:26.27 | chopp | ChannelZ: well with out that 1 in there, it worked as it should. Thanks for the patients. :) |
| 05:27.33 | ChannelZ | sure |
| 05:33.52 | Besticles | cRY |
| 05:34.11 | ChannelZ | sobs |
| 05:38.42 | *** part/#asterisk brendansterne (~brendanst@cpe-70-124-61-17.austin.res.rr.com) |
| 05:52.55 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 06:09.11 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 06:41.02 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 07:15.35 | *** join/#asterisk bmg505 (~leon@196-209-120-122.dynamic.isadsl.co.za) |
| 07:21.28 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 07:32.39 | ChannelZ | So are Besticles like testicles only better? |
| 07:32.48 | Besticles | thats right. |
| 07:33.11 | ChannelZ | Alright. |
| 07:33.19 | Besticles | Dunno was a name in a MMORPG that I played for a bit. |
| 07:36.48 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:37.07 | drmessano | Do you have the Best Testicles? Some chix are IN to that |
| 07:38.29 | drmessano | Hmm.. My Gwibber is horribly broken.. Shocking |
| 07:40.30 | ChannelZ | That's what SHE said |
| 07:50.12 | *** join/#asterisk brc007 (~Brian@ip68-104-158-197.ph.ph.cox.net) |
| 08:07.36 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 08:37.44 | *** join/#asterisk karthik (~karthik@122.183.17.90) |
| 08:55.07 | xheliox | middle of the night conversation on #asterisk -- never lacking that disturbing quality. |
| 09:06.24 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 09:09.25 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 09:17.28 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 09:17.50 | *** join/#asterisk amaache_ (c42efc32@gateway/web/freenode/ip.196.46.252.50) |
| 09:18.30 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 09:21.28 | odenkos_ntbk | hi all, when asterisk says it needs gsm(E) to compile codec_gsm, what exactly does it need? |
| 09:36.19 | *** join/#asterisk Rajmohan (~raj@122.164.180.203) |
| 09:40.00 | xheliox | odenkos_ntbk: Probably need to install libgsm-devel |
| 09:42.35 | odenkos_ntbk | xheliox: thanks, I'll try that |
| 09:51.57 | *** join/#asterisk eastr (~eastr@ti0013a380-dhcp2018.bb.online.no) |
| 09:52.23 | eastr | I am looking for video based conferencing solution, would also be interesting to look at commercialized solution. anyone have any recommandation and with a good reason for it? |
| 10:10.47 | *** join/#asterisk gain_ (~gain@host68-76-dynamic.7-79-r.retail.telecomitalia.it) |
| 10:31.09 | *** join/#asterisk louben (~louitzi@212-70-216-131.ath.static.tee.gr) |
| 10:33.24 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 10:51.20 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.150) |
| 10:56.39 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:56.47 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:57.45 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:57.58 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 11:00.50 | *** part/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 11:08.02 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:34.56 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cm-7-7b.dynamic.ziggo.nl) |
| 11:47.20 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 12:08.34 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:10.14 | AliRezaTaleghani | :) who have time to help me? i am looking for a way to play the Agent-ID or Agent name, to the caller of a quque |
| 12:10.19 | AliRezaTaleghani | :-/ |
| 12:32.11 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:56.36 | *** join/#asterisk NiugeS (~NiugeS32@5e0c4557.bb.sky.com) |
| 12:56.40 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 12:59.15 | *** part/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 13:23.39 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 13:49.26 | *** join/#asterisk ariel_ (~chatzilla@184-203-68-68.pools.spcsdns.net) |
| 14:10.58 | *** join/#asterisk Flashtek (~neil@flashtek-uk.com) |
| 14:12.58 | Flashtek | All: I am hitting a bit of an issue with regards linking vTiger with my asterisk server.. |
| 14:13.13 | Flashtek | has anyone here git any pointers ? |
| 14:16.22 | Flashtek | http://pastebin.com/bU5DvmmE |
| 14:23.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:23.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:25.09 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 14:25.11 | Flashtek | waves to leifmadsen |
| 14:33.04 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 14:33.05 | *** join/#asterisk SoulRaven (SoulRaven@89.41.157.171) |
| 14:35.57 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:37.34 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 14:39.57 | *** join/#asterisk BANSAL (~bansal@117.199.114.171) |
| 14:45.48 | *** join/#asterisk SoulRaven (SoulRaven@89.41.157.171) |
| 14:57.27 | *** join/#asterisk whtsup (~tun@WimaxUser3755-147.wateen.net) |
| 14:57.49 | whtsup | i m getting error on intalling asterisk |
| 14:58.13 | whtsup | configure: *** XML documentation will not be available because the 'libxml2' development package is missing. |
| 14:58.13 | whtsup | configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option |
| 14:58.13 | whtsup | configure: *** or install the 'libxml2' development package. |
| 14:58.25 | whtsup | ? |
| 14:59.45 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.150) |
| 15:05.53 | Flashtek | whtsup: install libxml2 then, or disable the xmldocs |
| 15:06.34 | heffer | or just go with the packages of your distro |
| 15:06.38 | heffer | if there are any |
| 15:07.40 | Flashtek | true |
| 15:09.48 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 15:12.56 | *** join/#asterisk nitram (foo@superblob.com) |
| 15:16.11 | p3nguin | They should make those error messages more descriptive. They're so cryptic, it takes a "computer genius" to understand what they mean! |
| 15:17.40 | heffer | p3nguin: regardin the installation error mentioned above? |
| 15:18.11 | p3nguin | right |
| 15:18.48 | p3nguin | I was just thinking about it. |
| 15:19.01 | heffer | p3nguin: you'd only hit that error if you try compiling asterisk from source |
| 15:19.12 | p3nguin | run the 'configure' script with the '--disable-xmldoc' parameter option <--- WHAT COULD THIS MEAN?! |
| 15:19.28 | heffer | and if you're trying to do that you're either already a "computer genius" or you're just not knowing what you're doing |
| 15:19.31 | p3nguin | or install the 'libxml2' development package. <--- WHAT ARE THEY TALKING ABOUT?! |
| 15:20.29 | WIMPy | You need to be a genius to do configure&&make&&make install? |
| 15:20.44 | WIMPy | is a genius |
| 15:20.55 | heffer | p3nguin: the thing is: almost all distros i know have packages for asterisk and if you're a novice you should try using them first |
| 15:20.56 | p3nguin | No, only to read the words on the screen. |
| 15:20.57 | carrar | Never heard of XML? |
| 15:21.07 | heffer | WIMPy: note the "" :P |
| 15:22.08 | WIMPy | OTOH I'm nut sure if you need to be able to read in order to type. |
| 15:25.31 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 15:39.53 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 15:44.48 | Flashtek | lol |
| 16:01.46 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.141.156.dsl.dyn.forthnet.gr) |
| 16:21.07 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 16:25.40 | *** join/#asterisk [SySteM] (~antoine@aqu33-6-88-168-80-163.fbx.proxad.net) |
| 16:26.02 | [SySteM] | Hello.. I search some help.. i try to connecting two incomming call please, what is the best way ? |
| 16:30.21 | Flashtek | [SySteM]: explain more |
| 16:31.09 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 16:32.47 | [SySteM] | i want than two incomming extern call can be talk together |
| 16:32.52 | [SySteM] | is Meetme the only solution ? |
| 16:34.17 | *** join/#asterisk BANSAL (~bansal@117.199.124.72) |
| 16:35.13 | [SySteM] | I got multiple incomming come wich arrive on a public channel (= press 1 to talk to "name save before", press 2 to talk ...), on press on the dynamic incomming call, two personn can talk together |
| 16:36.40 | [SySteM] | all person are caller from extern |
| 16:36.54 | [SySteM] | so.. is Meetme the only way ? |
| 16:43.57 | p3nguin | That's probably what I would use until I gave it a lot more thought. |
| 16:45.37 | [SySteM] | not possible to use a id channel |
| 16:45.41 | [SySteM] | to connect ? |
| 16:46.10 | p3nguin | You want caller B to be able to call in and connect up with caller A, right? |
| 16:49.09 | [SySteM] | yes |
| 16:49.24 | [SySteM] | exatly |
| 16:49.55 | p3nguin | What will caller A be doing? Why is caller A already on a call? And who is caller A talking with? |
| 16:50.25 | [SySteM] | caller A musiconhold |
| 16:50.42 | odenkos_ntbk | why not use queues then? |
| 16:51.30 | p3nguin | You could use queues and make caller A an agent who logs in and sits listening to MoH. MeetMe would do the same thing, though. |
| 16:51.45 | [SySteM] | its the same problem : can a caller be caller on a queue |
| 16:51.52 | [SySteM] | * be called |
| 16:52.08 | p3nguin | not unless you make that person an agent, also. |
| 16:52.17 | [SySteM] | ok understand for agent |
| 16:52.27 | [SySteM] | but cant create queue dynamicly |
| 16:52.39 | [SySteM] | destination is to create a chat with lots of caller |
| 16:52.55 | p3nguin | When an agent logs into his queue, he listens to MoH until he receives a call. When the caller hangs up, the agent listens to music again. |
| 16:53.04 | [SySteM] | and to work with queue, obligation is to create one queue by agent |
| 16:54.14 | [SySteM] | but in reallity .. A or B or C choose to talk to A or B or C |
| 16:54.22 | [SySteM] | so cant put them musiconhold |
| 16:55.34 | p3nguin | I'm thinking MeetMe is a good idea for this. |
| 16:55.37 | [SySteM] | want they have in round : "A name", "B name", "C name".. if anyone press * since "X name" is says, script ask X if i want to take the call.. |
| 16:56.02 | [SySteM] | its a chatroom with only private possibility |
| 16:56.33 | [SySteM] | i thinking Meetme is the good solution too but |
| 16:56.42 | [SySteM] | in my script got for each channel a unique id |
| 16:56.50 | [SySteM] | like SIP/mytrunk-XXXXX |
| 16:56.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:57.13 | [SySteM] | i was thinking than it was possbile to dial SIP/mytrunk-XXXX from another caller |
| 17:04.59 | [TK]D-Fender | ? |
| 17:08.43 | [TK]D-Fender | [SySteM]: Curious as to what you're trying to do based on that line I got coming in. |
| 17:25.51 | *** join/#asterisk moy_ (~moy@bas1-toronto47-1177731709.dsl.bell.ca) |
| 17:31.59 | Flashtek | [TK]D-Fender: can I pick your brains ? |
| 17:32.28 | *** join/#asterisk ariel_ (~chatzilla@m385536d0.tmodns.net) |
| 17:36.10 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn250.78-98-184.t-com.sk) |
| 17:36.18 | [TK]D-Fender | Flashtek: You can coose it, but it's already accounted for and I'm not selling ;) |
| 17:37.21 | Flashtek | lol |
| 17:37.40 | Flashtek | have you used vTiger ? |
| 17:40.28 | [TK]D-Fender | Flashtek: Nope. No CRM |
| 17:41.53 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 17:49.12 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 18:07.09 | *** join/#asterisk jetlag (jetlag@pool-173-61-209-95.cmdnnj.east.verizon.net) |
| 18:29.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:34.22 | *** join/#asterisk guilhermebr (~Guilherme@189.63.50.214) |
| 18:36.04 | *** part/#asterisk louben (~louitzi@212-70-216-131.ath.static.tee.gr) |
| 18:36.58 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:01.09 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 19:22.00 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 19:32.19 | *** join/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 19:39.13 | *** join/#asterisk ahowlader (~Adnan@180.234.49.161) |
| 20:07.11 | *** join/#asterisk deeperror (~deeperror@d149-67-49-94.try.wideopenwest.com) |
| 21:00.11 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 21:17.04 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 21:22.07 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:24.04 | *** join/#asterisk anthony- (~root@li113-241.members.linode.com) |
| 21:24.41 | anthony- | hi |
| 21:25.04 | Flashtek | ho |
| 21:25.21 | anthony- | care to help me out? |
| 21:25.33 | Flashtek | explain the issue.. |
| 21:26.01 | anthony- | My sip carrier isnt passing a 'confort noise'/ringing sound after they get the call |
| 21:26.18 | anthony- | is there a way i can insert my own ringing sound? |
| 21:26.47 | anthony- | doing Ringing() only works till before the sip provider gets the call |
| 21:26.58 | anthony- | but after that, im still stuck. |
| 21:28.43 | anthony- | any ideas or should i RTFM it some more |
| 21:29.58 | ManxPower | anthony-, do not answer the call before sending it to the provider |
| 21:31.55 | anthony- | how? |
| 21:33.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:33.38 | ManxPower | anthony-, pastebin the dialplan you use for making a call |
| 21:33.55 | ManxPower | Are you using any Answer, Read, or Playback or Background in your dialplan? |
| 21:36.01 | anthony- | no |
| 21:36.08 | anthony- | one moment while i pastebin |
| 21:36.11 | ManxPower | then I await your pastebin |
| 21:37.33 | anthony- | http://pastebin.ca/1943768 |
| 21:37.50 | anthony- | i have got the important part sectioned off |
| 21:38.01 | anthony- | with a bunch of hashes |
| 21:39.30 | ManxPower | anthony-, BTW, "confort noise" and VAD have nothing to do with ringing. |
| 21:39.58 | ManxPower | anthony-, the easiest thing for you to do is switch to a provider that does not suck. |
| 21:40.27 | anthony- | actually, my problem is the recieving end |
| 21:40.43 | ManxPower | If that is not an option then you have a long, confusing road ahead of you because you will have to learn how to diagnose SIP issues. |
| 21:40.45 | anthony- | vodafone is now selling their own sip service in greece, and they suck. |
| 21:41.04 | ManxPower | anthony-, the person you are calling is not hearing the phone ring? That makes no sense. |
| 21:41.11 | anthony- | no, they are |
| 21:41.22 | ManxPower | They are the "receiving end" |
| 21:41.24 | anthony- | but i am not hearing any ringback while ir's ringing |
| 21:41.52 | ManxPower | what happens if you set up an extension that just runs Ringing and then Wait? |
| 21:41.56 | ManxPower | do you hear ringing then? |
| 21:42.16 | anthony- | if you do a wait command, does the system go to the next step? |
| 21:42.48 | ManxPower | anthony-, It does nothing until the wait timeout happens. This will allow you to determine of it is a local issue |
| 21:42.49 | anthony- | i need it to pass the call and keep sending a ringback till 'answered' comes in |
| 21:43.05 | *** part/#asterisk Flashtek (~neil@flashtek-uk.com) |
| 21:43.09 | anthony- | other numbers work fine. |
| 21:43.13 | ManxPower | anthony-, "pass the call and keep sending a ringback till 'answered' comes in" <-- ASTERISK DOES THIS BY DEFAULT. |
| 21:43.13 | anthony- | its just a certain few. |
| 21:43.27 | ManxPower | Since it is not working for you, you have to determine the problem. |
| 21:43.56 | odenkos_ntbk | anthony-: the problem is, the provider picks it up, therefore asterisk stops ringback, and then calls the other party.. |
| 21:44.03 | odenkos_ntbk | * I think |
| 21:44.05 | anthony- | odenkos_ntbk: thank you |
| 21:44.09 | anthony- | odenkos_ntbk: yes! |
| 21:44.17 | ManxPower | odenkos_ntbk, he would see that as an Answered in the CLI. |
| 21:44.24 | ManxPower | if that is the case there is NOTHING that can be done |
| 21:44.48 | odenkos_ntbk | ManxPower: yes, but it would be answered sooner than the other party picks it up.. |
| 21:44.52 | ManxPower | as far as asterisk is concerned the call is answered. It cannot know if the ITSP answered the call or if the destination answered the call. |
| 21:44.56 | anthony- | odenkos_ntbk: but by the time they pass the call and someone answers, its empty space. |
| 21:45.02 | odenkos_ntbk | yeah.. switch the provider |
| 21:45.04 | ManxPower | anthony-, pastebin the cli output if a failed call |
| 21:45.36 | anthony- | okay, can i get one more shot to re-explain my issue or just pastebin? |
| 21:45.58 | ManxPower | anthony-, the pastbin will tell us more than you could ever tell us. |
| 21:46.08 | odenkos_ntbk | anthony-: we know your problem, and pastebin will tell us even more |
| 21:46.42 | tzafrir_laptop | A friend of mine wants to have some better understanding of a device with Asterisk included |
| 21:47.06 | ManxPower | tzafrir_laptop, that was unusually unparsable for you. |
| 21:47.09 | tzafrir_laptop | He managed to get access to the filesystem there. But not to the shell |
| 21:47.30 | odenkos_ntbk | chuckles at ManxPower :) |
| 21:47.34 | *** join/#asterisk shayy (~shayy@109.160.132.106) |
| 21:48.34 | tzafrir_laptop | that device has in /usb/sbin/asterisk something that appears to be asterisk, probably 1.2 |
| 21:49.12 | tzafrir_laptop | That is: the copyright string has: "Asterisk , Copyright (c) 1999 - 2005," |
| 21:49.42 | tzafrir_laptop | Which means that the version string is deleted, and that the version of Asterisk is no later than r8122 / r8123 |
| 21:50.04 | ManxPower | tzafrir_laptop, time for a GPL lawsuit? |
| 21:50.23 | tzafrir_laptop | any idea how to get shell access on the system or whatever? |
| 21:50.50 | anthony- | okay, i cant right now cuz its kinda late in greece and i dont wanna wake anyone. |
| 21:51.01 | tzafrir_laptop | Or even get asterisk to reload its config? |
| 21:51.10 | ManxPower | anthony-, come back when you can do the needed troubleshooting |
| 21:51.18 | ManxPower | tzafrir_laptop, no access to the asterisk CLI? |
| 21:51.22 | anthony- | but another way to explain it would be |
| 21:51.38 | ManxPower | anthony-, I already told you the answer, I just wanted the pastebin to confirm it. |
| 21:51.51 | tzafrir_laptop | No. Only (r/w) access to the file system |
| 21:51.51 | anthony- | ManxPower: okay, ill try another provider. |
| 21:51.54 | ManxPower | your ITSP sucks and your only option is to switch to a different ITSP |
| 21:52.00 | ManxPower | tzafrir_laptop, any JTAG support? |
| 21:52.25 | ManxPower | tzafrir_laptop, if you can write to the file system you should be able to write to /var/run/asterisk.ctl ? |
| 21:53.33 | odenkos_ntbk | anthony-: or, just ignore it |
| 21:54.19 | anthony- | <PROTECTED> |
| 21:54.19 | anthony- | <PROTECTED> |
| 21:54.29 | anthony- | from that time and onward, there is no ringback till i see answerd. |
| 21:54.50 | ManxPower | anthony-, your carrier is not sending back the right messages |
| 21:54.55 | anthony- | and yeah, it looks like ignore it is the only option. |
| 21:55.22 | ManxPower | as you can see there is NO " SIP/voipdiscount8669-09c112c8 answered SIP/06-b54c8900" |
| 21:55.32 | ManxPower | so your ITSP is NOT answering the call |
| 21:55.51 | anthony- | it says answerd when the other party picks up |
| 21:56.08 | ManxPower | anthony-, one option is for your do run Answer in your dialplan, then Ringing, then Dial. Make sure you have a /etc/asterisk/indications.conf or that won't work |
| 21:56.35 | anthony- | okay, i will try that, thanks. |
| 21:56.37 | ManxPower | when you answer then ASTERISK ignores what the far end sends and just plays ringback tones until the call is answered. |
| 21:57.02 | anthony- | cool, that is what i want to do. thanks. |
| 21:57.04 | ManxPower | anthony-, the fact your ITSP does not actually Answer the call is good for you. |
| 21:57.34 | anthony- | actually, the problem is with vodafone in greece, their stuff isnt setup right. |
| 21:57.53 | anthony- | if i call a regular landline owned by "at&t" it works fine. |
| 21:58.04 | anthony- | with the same TISP |
| 21:58.24 | anthony- | its just when i call those numbers, but yes, thanks. ill keep quiet now and let you know how it went. |
| 22:06.40 | anthony- | ManxPower: did i do what you suggested correctly? http://pastebin.ca/1943784 |
| 22:07.02 | ManxPower | anthony-, yes. |
| 22:07.40 | *** join/#asterisk rocknfra (~rocknfra@host183-216-dynamic.16-87-r.retail.telecomitalia.it) |
| 22:07.57 | anthony- | okay yeah that doesnt work for me, looks like i gotta live with it. or think of some other super hack solution. |
| 22:08.17 | anthony- | thanks anyway. |
| 22:34.34 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 22:37.06 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 23:04.58 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 23:34.53 | *** join/#asterisk guilhermebr (~Guilherme@189.63.50.214) |
| 23:53.08 | *** join/#asterisk zzlane (~zzlane@c-98-206-44-7.hsd1.il.comcast.net) |