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| 00:05.31 | granta | I have a X100P card, and asterisk is set to just do an echo test when I call in (from a different POTS line). I hear the echo test annoucements fine, but all asterisk seems to receive is noise, not me talking. Any suggestions how to troubleshoot this? |
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| 00:54.35 | cmdln | good evening |
| 00:55.19 | bkrus1 | cmdln: hello |
| 00:56.12 | *** join/#asterisk abes (n=abes@rn-v1w2a13.uwaterloo.ca) |
| 00:56.42 | cmdln | i need some help figuring out asterisk |
| 00:57.15 | bkrus1 | cmdln: http://voip-info.org |
| 00:57.17 | cmdln | i dont quite understand what all i need to get from my carrier |
| 00:57.35 | bkrus1 | just go to voip-info and poke around, its a GREAT information source |
| 00:58.23 | cmdln | thanks |
| 00:58.26 | cmdln | im sure ill be back |
| 00:59.17 | bkrus1 | cmdln: k, just let me know, PM me if you wish, my aim is kruzweb and my jabber is bkruse@digium.com |
| 00:59.24 | bkrus1 | :] |
| 00:59.29 | bkrus1 | happy reading! |
| 00:59.59 | docelmo | What is the easiest way in asterisk to limit concurrent channels? |
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| 01:01.29 | bkrus1 | docelmo: uh....not sure |
| 01:01.40 | bkrus1 | you could make a counter in the dialplan, but im sure theres greater ways of doing it |
| 01:01.45 | bkrus1 | gotoif's |
| 01:01.51 | *** join/#asterisk nvictor (n=nvito@avenou.cafe.tg) |
| 01:01.58 | nvictor | hello |
| 01:02.04 | nvictor | I'm a php developper |
| 01:02.08 | nvictor | -p |
| 01:02.22 | nvictor | my co-workers user asterisk |
| 01:02.30 | nvictor | and I'm building their interface |
| 01:02.50 | bkrus1 | nvictor: ive written TONS of stuff for asterisk in php |
| 01:02.58 | bkrus1 | you want my class file? i can get it tomorrow |
| 01:03.09 | nvictor | bkrus1: cool |
| 01:03.16 | nvictor | bkrus1: I want it |
| 01:03.22 | bkrus1 | nvictor: because php does sockets so cool |
| 01:03.38 | nvictor | you see, I want to see which calls are fowarded |
| 01:03.45 | nvictor | which numbers sorry |
| 01:03.51 | bkrus1 | k |
| 01:03.59 | bkrus1 | youve been messing with the managers interface i hope write? |
| 01:04.01 | bkrus1 | right* |
| 01:04.22 | nvictor | well somehing like that |
| 01:04.31 | nvictor | they've written the scripts |
| 01:04.35 | nvictor | I do the interface |
| 01:04.42 | nvictor | send variables to the scripts :D |
| 01:04.55 | bkrus1 | k |
| 01:05.01 | nvictor | but while testing the forwarding, I've messed up a bit with things |
| 01:05.05 | bkrus1 | it should be in the managers interface, have you checked out ajam? |
| 01:05.14 | bkrus1 | you need my php class file |
| 01:05.15 | nvictor | ajam? |
| 01:05.19 | nvictor | yes |
| 01:05.19 | bkrus1 | i have one for 1.4 and 1.2 |
| 01:05.22 | nvictor | I need it |
| 01:05.33 | nvictor | what is ajam? |
| 01:05.38 | bkrus1 | *getting link* |
| 01:05.45 | nvictor | ok :D |
| 01:05.49 | bkrus1 | http://www.asterisk.org/node/73 |
| 01:06.05 | bkrus1 | youlll also want this.... |
| 01:06.05 | bkrus1 | http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) |
| 01:06.10 | bkrus1 | and this |
| 01:06.11 | bkrus1 | http://www.voip-info.org/wiki-Asterisk+manager+API |
| 01:06.30 | nvictor | bkrus1: cool |
| 01:06.32 | nvictor | :D |
| 01:06.46 | bkrus1 | nvictor: youlll love my php class, let me see if i can vpn in real quick.... |
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| 01:07.10 | nvictor | bkrus1: can you send it to me? |
| 01:07.23 | bkrus1 | of course! |
| 01:07.32 | bkrus1 | open source :] |
| 01:07.43 | bkrus1 | i found php classes awesome for sockets |
| 01:07.55 | bkrus1 | because i can do something like make the socket name the server name |
| 01:08.02 | bkrus1 | then address it in your main php file as |
| 01:08.23 | bkrus1 | $asterisk->call_in_agent($server, $agent_array, $delay etc etc) |
| 01:08.42 | nvictor | coool! |
| 01:08.54 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 01:08.55 | nvictor | that will make my interface easy to write |
| 01:08.58 | bkrus1 | indeed. |
| 01:09.14 | bkrus1 | $asterisk->login($server, $password) |
| 01:09.24 | bkrus1 | for convenience, i set all my server passwords to the same in the managers interface |
| 01:09.35 | bkrus1 | but if its used in a production envirnment, its best not to do that ;] |
| 01:10.20 | nvictor | bkrus1: it is said that ajam is for version 1.4 |
| 01:10.24 | bkrus1 | k |
| 01:10.30 | bkrus1 | these classes will do just fine |
| 01:10.33 | bkrus1 | how much php you know? |
| 01:10.44 | bkrus1 | i have a version in my email, not sure how old it is :X |
| 01:10.49 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 01:11.20 | bkrus1 | and its not the 1.4 version.......the only difference in the 1.4 version of the php file is a little syntax changing for the command line |
| 01:11.20 | nvictor | bkrus1: well, I have been coding in php for 4 or 5 months |
| 01:11.21 | *** part/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 01:11.21 | bkrus1 | are you using 1.2.13 i hope? |
| 01:11.21 | bkrus1 | nvictor: have you done class/object oreiented coding? |
| 01:11.22 | nvictor | 1.2.3 yes |
| 01:11.28 | bkrus1 | k |
| 01:11.36 | bkrus1 | and are you fimiliar with class coding |
| 01:11.43 | bkrus1 | class { |
| 01:11.43 | bkrus1 | functions yay |
| 01:11.43 | bkrus1 | } |
| 01:11.49 | nvictor | nvictor: well a bit, but I can learn.. |
| 01:11.54 | nvictor | yes |
| 01:11.57 | bkrus1 | k |
| 01:12.03 | bkrus1 | (looking in email) |
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| 01:13.06 | nvictor | ok |
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| 01:19.36 | slePP | whee |
| 01:19.40 | bkrus1 | slePP: hello |
| 01:19.43 | slePP | 'lo |
| 01:19.49 | bkrus1 | :] |
| 01:20.56 | nvictor | bkrus1: have you found the code? by the way can you explain me some things?? |
| 01:21.17 | slePP | bkrus1: common belief thinks you're... not tired? |
| 01:21.25 | bkrus1 | slePP: yes. |
| 01:21.31 | bkrus1 | nvictor: what? |
| 01:21.32 | bkrus1 | oh! |
| 01:21.56 | bkrus1 | nvictor: someone else IM'd, and ive been talking to him about php for about 5 minutes |
| 01:22.01 | nvictor | well, I only hear my co-workers discuss asterisk... never search on it myself... |
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| 01:22.07 | bkrus1 | slePP: being overworked ;[ |
| 01:22.12 | bkrus1 | nvictor: if you know php you will be fine |
| 01:22.21 | slePP | bkrus1: aren't we all? |
| 01:22.46 | bkrus1 | slePP: yes ;] |
| 01:22.47 | nvictor | how does asterisk work to make calls? |
| 01:22.50 | bkrus1 | <3 insomnia |
| 01:22.58 | bkrus1 | nvictor: that i do not want ot answer |
| 01:23.06 | nvictor | why? |
| 01:23.08 | bkrus1 | hahaha |
| 01:23.21 | bkrus1 | nvictor: how does asterisk make calls? |
| 01:23.39 | bkrus1 | nvictor: can you IM me tomorrow in the hours of 1-5 pm? |
| 01:23.43 | bkrus1 | i can get you the files then. |
| 01:23.49 | nvictor | ok |
| 01:23.51 | nvictor | I'll |
| 01:23.55 | bkrus1 | but for now i must go, hit me up on jabber or aim |
| 01:24.03 | bkrus1 | bkruse@digium.com && kruzweb |
| 01:24.22 | nvictor | won't you log into irc? |
| 01:24.32 | bkrus1 | ill be to busy probably |
| 01:24.34 | bkrus1 | but i will try |
| 01:24.41 | nvictor | I don't have jabber |
| 01:24.44 | bkrus1 | aim? |
| 01:24.45 | bkrus1 | msn? |
| 01:24.46 | bkrus1 | yahoo? |
| 01:24.47 | nvictor | I don't have aim |
| 01:24.50 | nvictor | yahoo |
| 01:24.53 | nvictor | :D |
| 01:24.57 | bkrus1 | whats ur yahoo |
| 01:25.08 | nvictor | noagbodjivictor@yahoo.com |
| 01:25.17 | nvictor | let me connect |
| 01:25.22 | bkrus1 | and i can get that to ya....i have one in my email, but its super old, and I do not think it works |
| 01:25.46 | nvictor | oh, I'll wait for the new one |
| 01:26.24 | bkrus1 | k, good |
| 01:26.52 | nvictor | see you tomorrow then |
| 01:27.01 | bkrus1 | k, bye |
| 01:27.02 | *** part/#asterisk bkrus1 (n=root@69.73.127.92) |
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| 01:38.56 | DavoFrom818 | anyone here ever use the endpoint manager for asterisk |
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| 01:49.30 | tim27 | anyhere how SIP provider babytel.ca handle DID |
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| 01:59.51 | steve26 | ~book |
| 01:59.55 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
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| 02:09.10 | Newbie___ | hi all, is CAC bank II good for * ? |
| 02:09.25 | DavoFrom818 | anyone here can help me please? |
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| 02:17.56 | tim27 | you there Davo |
| 02:23.58 | converx | what is the cmd for sending voicemail to more than one mailbox? |
| 02:24.31 | [TK]D-Fender | converx : "show application voicemail" |
| 02:30.28 | hmmhesays | harmonic minor modes are interesting |
| 02:30.43 | slePP | i bet they are |
| 02:32.08 | [TK]D-Fender | hmmhesays : is that the one thats a 3rd & 7th half-tone drop off the major? |
| 02:33.44 | hmmhesays | http://www.myguitarsolo.com/sc_harm.htm |
| 02:34.08 | hmmhesays | i'm building myself a scale lesson book |
| 02:38.51 | [TK]D-Fender | hmmhesays : I tend to mix the normal & harmonic minors. the 7th note drop differencemakes the transition span octaves better as a start/finish. |
| 02:39.49 | hmmhesays | if you're playing at warp speed your passing notes don't really matter |
| 02:40.21 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 02:40.27 | [TK]D-Fender | hmmhesays : I guess somewhat.... made a warp-speed sample for me to listen to? |
| 02:41.32 | *** part/#asterisk rtcg-tx (n=chatzill@mail.richardthecomputerguy.com) |
| 02:43.03 | hmmhesays | lol no |
| 02:43.41 | [TK]D-Fender | hmmhesays : Gimme a sec, I'mm make on for you :) |
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| 02:49.31 | [TK]D-Fender | hmmhesays : http://aocomputing.net/E-Harmonic-Minor.mp3 |
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| 03:08.00 | darqchild | may i trouble someone for some support? :) |
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| 03:08.27 | [TK]D-Fender | darqchild : Ask a specific question and you may get a specific answer... |
| 03:08.36 | *** join/#asterisk phoem (n=phoem@jn.isprime.com) |
| 03:08.49 | Newbie___ | any one has any idea which model of CAC CB does not require a T1 card |
| 03:09.32 | darqchild | [TK]D-Fender: Dial() won't generate a ring tone when i use the 'r' flag. |
| 03:10.22 | darqchild | [TK]D-Fender: I answer a call coming in on an IAX2 trunk, and attempt to dial an SIP trunk. the 'm' flag will give me music, but the 'r' flag gives me nothing. Have i missed something in the docs? |
| 03:10.23 | [TK]D-Fender | darqchild : You shouldn't be using that option execpt in extreme cases. What are on both ends of your call? |
| 03:10.44 | [TK]D-Fender | darqchild : You should not put "r" in taht case at all. |
| 03:11.12 | darqchild | [TK]D-Fender: i don't get a ring without it either |
| 03:11.13 | [TK]D-Fender | Newbie___ : CB is by std definition a T1/E1 device.... |
| 03:11.33 | [TK]D-Fender | darqchild : What is on the SIP end? |
| 03:12.09 | darqchild | [TK]D-Fender: xlite |
| 03:15.45 | darqchild | [TK]D-Fender: the soft-phone, rings, and i can complete a call. everything else works fine. |
| 03:15.56 | DrCron | hmm, so what sort of streams can asterisk use as MoH |
| 03:17.04 | [TK]D-Fender | darqchild : X-Lite rings, but the IAX side doesn't hear it? |
| 03:17.33 | darqchild | Correct. |
| 03:17.36 | *** part/#asterisk alerios (n=alerios@190.24.97.151) |
| 03:18.33 | darqchild | [TK]D-Fender: Here is my full dial command: Dial(SIP/xlite1|60|dr) |
| 03:20.05 | [TK]D-Fender | darqchild : remove the ,dr |
| 03:20.24 | darqchild | [TK]D-Fender: okay ... |
| 03:21.18 | darqchild | [TK]D-Fender: Still no ring. |
| 03:21.54 | [TK]D-Fender | darqchild : hrm. |
| 03:23.15 | darqchild | I haven't seen anything in the debugging output that gives me any clues either. I can pass music while i dial, that's not a problem. |
| 03:27.25 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.164) |
| 03:28.37 | HaMYaI | how do we know how long a sip client will be recognized by asterisk? |
| 03:29.07 | HaMYaI | is it from the "Expire" value in "sip show peer <peername>"? |
| 03:32.10 | [TK]D-Fender | HaMYaI : Until then unles it re-registers earlier to renew. Failing a qualify will kill it off too, even if it comes backa while later, or only just dropped off. |
| 03:34.04 | HaMYaI | [TK]D-Fender: you mean it will not be renewed if qualify has not been set to "yes"? |
| 03:35.03 | [TK]D-Fender | HaMYaI : no, if it IS set to "yes" and it fails. |
| 03:35.23 | HaMYaI | [TK]D-Fender: right |
| 03:36.20 | *** join/#asterisk wolrah (n=yearight@cblmdm72-240-156-158.buckeyecom.net) |
| 03:36.41 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:37.27 | HaMYaI | [TK]D-Fender: I'm using Eyebeam and I saw the values Max/Min register time there |
| 03:38.43 | HaMYaI | [TK]D-Fender: don't know if it has anything to do with this |
| 03:40.26 | HaMYaI | [TK]D-Fender: I set my client to re-register every 300 secs but even the client isn't registered, asterisk still recognizes the client |
| 03:40.49 | HaMYaI | even the staus is "UNREACHABLE" |
| 03:47.52 | HaMYaI | [TK]D-Fender: what I was trying to say is that even sip client fails to qualify, asterisk still send my call to the right context |
| 03:49.42 | [TK]D-Fender | HaMYaI : You mean a call FROM your SIP phone, not TO it. |
| 03:50.34 | HaMYaI | [TK]D-Fender: yes, not to it |
| 03:51.26 | [TK]D-Fender | HaMYaI : You need to remember that reistering has NOTHING to do with authorizing you to place calls. All it does is tell the registration server what IP to send your phone's calls TO. |
| 03:51.53 | HaMYaI | [TK]D-Fender: so, I'm wondering if "Expire" is applicable for this case too |
| 03:52.06 | darqchild | [TK]D-Fender: I've done some more poking around here, and it seems that if i use Ringing(), Congestion() etc anyone on the other end of the IAX trunk gets nothing. |
| 03:52.15 | HaMYaI | [TK]D-Fender: I see |
| 03:52.31 | *** join/#asterisk bmg505 (n=leon@c1-42-3.rndf.isadsl.co.za) |
| 03:52.34 | darqchild | [TK]D-Fender: is this a local configuration issue? Or should i be calling my voip provider? |
| 03:52.55 | [TK]D-Fender | darqchild : I'd double check what kind of indication you put on that IAX2 channel if I were you... pastebin the peer setu (mask only the user/pass) |
| 03:52.56 | [TK]D-Fender | ~pb |
| 03:52.59 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:53.37 | [TK]D-Fender | HaMYaI : The expire bit should oly affect how often your SIP client is requested to rerister (to keep track of possibly changing IP's) |
| 03:53.53 | file | [TK]D-Fender: stick your head in the microwave and give yourself a tan |
| 03:54.54 | [TK]D-Fender | file : Bake, not fried :) |
| 03:55.03 | file | :D |
| 03:55.04 | [TK]D-Fender | file : Got your call this afternoon :) |
| 03:55.11 | file | my friend dialed the wrong number |
| 03:55.18 | file | so it ended up at your speed dial entry |
| 03:55.19 | [TK]D-Fender | lol |
| 03:55.55 | HaMYaI | [TK]D-Fender: I understand that now but from what I found here. Asterisk sometimes send my out-going calls to from-sip-external context which I set it for un-authorized sip calls |
| 03:57.56 | HaMYaI | with normal settings, so I thought we need to be registered to place a call |
| 03:59.11 | HaMYaI | [TK]D-Fender: haven't you had this problem before? |
| 03:59.56 | HaMYaI | ohh |
| 04:00.03 | [TK]D-Fender | gibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBERgibberGIBBER |
| 04:00.04 | file | I hear him typing |
| 04:00.08 | file | oh, yup - there we go |
| 04:00.50 | HaMYaI | file: is that about what you talked to him? =) |
| 04:01.03 | Qwell | file: the PHONE?! |
| 04:01.05 | Qwell | why? |
| 04:01.45 | [TK]D-Fender | Qwell : We don't want to meet your mom! |
| 04:01.47 | *** join/#asterisk dkf3s (n=hare@58.69.158.114) |
| 04:01.58 | Qwell | eh? |
| 04:02.11 | file | Qwell: We just want... |
| 04:02.16 | [TK]D-Fender | file : ! ! ! |
| 04:02.19 | file | WOOT! |
| 04:02.40 | file | oh god |
| 04:02.57 | file | that was close. |
| 04:03.01 | [TK]D-Fender | Qwell : Since you weren't in on the joke - http://www.starterupsteve.com/swf/Group_X_video.html |
| 04:03.06 | *** join/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 04:03.48 | Narkov- | whats the easiest way to split calls over two separate net connections? |
| 04:04.05 | Narkov- | two IAX trunks over two ADSL connections to the same provider |
| 04:05.32 | Narkov- | i cant figure out how to split/route the packets over the two DSL connections |
| 04:06.02 | file | [TK]D-Fender: So did you hear about the SIP phone that couldn't place calls? Yeah, it conformed strictly to specs. |
| 04:13.37 | blitzrage | Narkov-: you don't do that with Asterisk, you do that at with something like IPtables |
| 04:14.28 | Narkov- | any hints blitzrage? is it even possible? |
| 04:14.58 | blitzrage | yah... I just gave you a hint :) |
| 04:14.59 | *** join/#asterisk sbingner (n=thanotos@adsl-699.flex.com) |
| 04:15.13 | blitzrage | Narkov-: you don't do that with Asterisk, you do that at with something like IPtables |
| 04:15.46 | Narkov- | hehe...thanks blitzrage |
| 04:16.12 | Narkov- | i'm just a bit unsure where to start given that they both register to the same IP address |
| 04:16.37 | Narkov- | so its not as if I can chuck in a static route because they all go to the same IP |
| 04:17.30 | linlin | anyone know what kind of monitor i'd have to use for this? http://cgi.ebay.com/HP-VISUALIZE-C180-C-180-WORKSTATION-A4200A-A4231A_W0QQitemZ270064785848QQihZ017QQcategoryZ11221QQcmdZViewItem |
| 04:17.55 | *** join/#asterisk sbingner (n=thanotos@adsl-699.flex.com) |
| 04:18.08 | darqchild | [TK]D-Fender: I have to go now, thanks for your help. |
| 04:18.21 | blitzrage | linlin: umm... a normal one? |
| 04:18.27 | blitzrage | (LCD) |
| 04:18.40 | blitzrage | or get a DVI connector |
| 04:18.51 | linlin | the description the guy states it was giving out of ranger errors |
| 04:19.08 | linlin | just wondering if anyone had experence with this line of HP systems and if they know from experence |
| 04:19.48 | blitzrage | linlin: this room is not appropriate for that question |
| 04:20.06 | linlin | just wonderin...seemed dead anyways |
| 04:21.18 | *** join/#asterisk diclophis-work (n=jbardin@c-69-181-70-186.hsd1.ca.comcast.net) |
| 04:24.37 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 04:29.24 | HaMYaI | <PROTECTED> |
| 04:29.30 | *** part/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 04:30.05 | HaMYaI | do you have any idea what causes this? |
| 04:31.35 | *** join/#asterisk X-Rob (n=rob-x@dsl-58-6-69-193.vic.westnet.com.au) |
| 04:33.37 | *** join/#asterisk nays85 (i=nays85@got.root3d.net) |
| 04:34.22 | *** join/#asterisk apardo (n=apardo@87.217.147.146) |
| 04:34.54 | DrCron | asterisk can do transcoding, right? |
| 04:35.04 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 04:35.23 | bcnl | DrCron: yup |
| 04:39.14 | Supaplex | are $(CDR(DST)} and "exten => _" after the _ suppose to be the same on in incomming call? |
| 04:39.53 | DrCron | how many terminations can the iaxy drive? |
| 04:39.56 | slePP | bcnl: boo? |
| 04:40.20 | Corydon76-home | Supaplex: an underscore means it's a pattern |
| 04:40.21 | slePP | you are who i think you are, i'm sure. |
| 04:40.34 | Corydon76-home | slePP: !!! |
| 04:40.37 | slePP | coyrdon :> |
| 04:40.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:41.06 | Corydon76-home | LTNS |
| 04:41.11 | slePP | yessir |
| 04:41.14 | slePP | been working, etc. |
| 04:41.31 | Corydon76-home | voiceovers for pornos? ;-) |
| 04:41.48 | slePP | heh. i wish |
| 04:41.53 | slePP | that'd be far more fun |
| 04:42.04 | Corydon76-home | You have the voice for it... |
| 04:42.10 | slePP | i know ;> |
| 04:42.15 | slePP | been a while since i did that, though |
| 04:42.22 | Supaplex | I have no idea how to identify sip calls based on where they want to go. ${CDR(dst)} is empty in these situations. |
| 04:42.37 | Corydon76-home | Supaplex: was the call answered? |
| 04:43.03 | Supaplex | yes, it dropped into my bitbucket handler. (eg operator intercept) |
| 04:43.19 | Corydon76-home | and it bridged? |
| 04:43.42 | Corydon76-home | If it bridged, you should have a dst |
| 04:43.50 | Supaplex | nay |
| 04:44.13 | bcnl | slePP: yes boo |
| 04:44.17 | Corydon76-home | If it didn't bridge, well, that's why there's no destination |
| 04:44.24 | slePP | bcnl, that is. |
| 04:44.33 | bcnl | heh |
| 04:44.37 | Supaplex | didn't match any known numbers. I use pattern matches on the dids to direct to a specific project/queue etc. |
| 04:44.49 | bcnl | don't do that |
| 04:44.53 | Corydon76-home | slePP: whip me, beat me, make me vote Republican |
| 04:44.55 | bcnl | I got beats enough last week from the move |
| 04:44.59 | slePP | heh |
| 04:45.01 | slePP | me too, don't worry |
| 04:45.15 | bcnl | yea you poor bastard |
| 04:45.23 | file | wow - it's slePP |
| 04:45.23 | slePP | heh |
| 04:45.24 | slePP | hi josh |
| 04:45.26 | bcnl | I'll come up there again soon and take you to the pub |
| 04:46.33 | bcnl | bye bye file |
| 04:46.44 | orlock | Does anybody know if asterisk buffers rtp streams? |
| 04:47.21 | Corydon76-home | orlock: uh, no... |
| 04:47.29 | orlock | you dont know, or it doesnt? |
| 04:47.33 | bcnl | that would be b...........................ad |
| 04:47.35 | Corydon76-home | It shouldn't |
| 04:47.51 | orlock | yeah, i was thinking it might be |
| 04:47.55 | orlock | having some call quality issues |
| 04:47.58 | file | "Please wait... buffering phone call." |
| 04:48.06 | Corydon76-home | Any buffering would cause the conversation to lag... |
| 04:48.08 | linlin | any guides around to teach me how to link two asterisk boxes together with iax? |
| 04:48.11 | Supaplex | powered by real media |
| 04:48.11 | orlock | does anybody know of any good ways to test/verify rtp? |
| 04:48.22 | orlock | i was thinking just a jitter buffer |
| 04:48.22 | Corydon76-home | "rtp debug" |
| 04:48.48 | DrCron | the asterisk book focuses on connections to the ptsn, is there somewhere i can read about setting up an IP only system? |
| 04:48.51 | *** join/#asterisk sbingner (n=thanotos@adsl-699.flex.com) |
| 04:48.51 | docelmo | I HAVE DIDs! |
| 04:49.10 | Corydon76-home | DrCron: it's the same concept |
| 04:49.19 | orlock | Corydon76-home: that gives me the same data i can get with ethereal+tcpdump |
| 04:49.37 | Corydon76-home | Well, not the same data |
| 04:49.41 | Corydon76-home | but close |
| 04:49.44 | Supaplex | durring sip debug, I do see "To: <sip:012341234@sip.example.com"... it'll answer and since I cannot match ${CDR(dst)} it falls through |
| 04:50.13 | Corydon76-home | Supaplex: did you intend to match the EXTEN ? |
| 04:50.13 | Supaplex | s/match \S* /match / |
| 04:50.26 | Supaplex | yea, 1s |
| 04:50.27 | *** join/#asterisk sbingner (n=thanotos@adsl-699.flex.com) |
| 04:50.28 | orlock | Corydon76-home: yeah, same stuff |
| 04:50.31 | Corydon76-home | Supaplex: or is the fallthrough intentional |
| 04:50.32 | orlock | timestamp, length, etc |
| 04:50.42 | orlock | ethereal has some quite nice RTP analysis stuff now |
| 04:50.49 | converx | in cmd voicemail, if I do voicemail(box1@context&box2@context&box3@context) -- the field in extensions_table is limited to 128char. |
| 04:51.22 | Corydon76-home | converx: please see #openpbx |
| 04:52.02 | file | if you were using a jitterbuffer it would buffer it some... |
| 04:52.09 | converx | this is for * realtime. |
| 04:52.16 | Supaplex | Corydon76-home: it's sort of intentional because I don't know how to match it. all my iax2 registry entries work fine. for the conext= from sip.conf in extensions.conf: exten => _012341234,1,Goto ... |
| 04:52.29 | orlock | file: yeah |
| 04:52.39 | Corydon76-home | Supaplex: _X. will match pretty much anything |
| 04:52.46 | orlock | clients are hearing quality issues when one packet goes missing |
| 04:53.02 | bcnl | slePP: so find anymore metaswitch bugs today? |
| 04:53.09 | slePP | acually, yes |
| 04:53.11 | orlock | upstream rovider has said, ok, lets eliminate asterisk as one missed packet should not cause a noticable call issue |
| 04:53.24 | Supaplex | maybe there's a different variable I can dump for debugging. |
| 04:53.55 | converx | anyone in here has the right sendmail.cf file to work with asterisk? |
| 04:53.55 | Corydon76-home | converx: why are you loading extensions from a table in the first place? |
| 04:54.05 | bcnl | hah |
| 04:54.09 | Corydon76-home | converx: there is no "right" file |
| 04:54.52 | converx | corydon -- because my extensions.conf is 8000 lines .. doh! |
| 04:55.12 | Corydon76-home | converx: so? |
| 04:55.35 | converx | corydon -- so you go and edit an 8k line text file!!! |
| 04:55.41 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 04:55.47 | Corydon76-home | converx: I do. all the time. |
| 04:56.03 | Corydon76-home | converx: In fact, I edit 100,000 line text files all the time. |
| 04:56.54 | Corydon76-home | converx: what it really speaks to is that you're probably not using patterns to your advantage |
| 04:57.51 | Corydon76-home | converx: there is zero reason to use realtime extensions. None. It's a waste of resources. |
| 04:58.30 | bcnl | slePP: next time you're at TB kick derek into updating the stuff on vids/tv |
| 04:58.31 | converx | corydon -- easier update/search/debug. |
| 04:58.59 | Corydon76-home | converx: Doesn't fly. |
| 04:59.21 | *** join/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 04:59.24 | DrCron | realtime or db based, there is a diffrence, iirc |
| 04:59.31 | Corydon76-home | The only reason it was ever created was to make it easier to edit from a web browser |
| 04:59.33 | converx | speed is meaningless. |
| 04:59.47 | DrCron | you can still use a db then run reloads, right? |
| 05:00.31 | slePP | and now, it's time for food |
| 05:00.32 | Corydon76-home | DrCron: actually, the better course of action is to use a static dialplan which does lookups from a database... a truly dynamic dialplan |
| 05:00.59 | converx | is there a problem in redefining the field structure in * realtime. for example, increasing length of field from 128char to 256char. |
| 05:01.28 | Corydon76-home | converx: nope, the only reason it was 128char in the first place is because you're using openpbx. |
| 05:01.34 | DrCron | my real interest is using a mysql backend for voicemail, blobs and all |
| 05:02.05 | converx | corydon-- you keep mentionning openpbx. I am not using openpbx. |
| 05:02.05 | Corydon76-home | DrCron: can be done already, in 1.2 |
| 05:02.21 | Corydon76-home | converx: why is your table named extensions_table ? |
| 05:02.47 | converx | thats how I named it. |
| 05:02.49 | DrCron | i know it can be done, i'm just tring to figure out how to get it working on my system :) |
| 05:03.08 | Corydon76-home | converx: so it's complete coincidence that that is the name used by #openpbx |
| 05:03.31 | Corydon76-home | converx: and since you created the table, you should know whether or not 128 char was a limit, right? |
| 05:03.40 | Corydon76-home | converx: it doesn't fly |
| 05:04.14 | converx | corydon -- voip-info has the realtime table definition -- Get a clue! |
| 05:04.32 | Corydon76-home | converx: Hah, a clue... |
| 05:04.43 | Corydon76-home | converx: you seem to be the one who needs a clue |
| 05:05.42 | file | easy you two |
| 05:05.53 | Corydon76-home | file: sorry |
| 05:06.08 | file | Corydon76-home: ... or else you'll have to be punished! |
| 05:06.25 | Corydon76-home | file: spank me? |
| 05:06.42 | file | never! |
| 05:06.49 | Corydon76-home | Awwww |
| 05:07.08 | orlock | hmm.. |
| 05:07.28 | orlock | anybody have any suggestions on debuggig rtp? |
| 05:07.38 | *** join/#asterisk diclophis-work (n=jbardin@c-69-181-70-186.hsd1.ca.comcast.net) |
| 05:07.49 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 05:07.52 | orlock | i am thinking maybe using a high quality WAV file, using asterisk to play it back, and then dumping the wav file |
| 05:07.59 | orlock | from the packet dump |
| 05:08.13 | Corydon76-home | orlock: what is the issue that you're having? |
| 05:08.29 | orlock | and then viewing the original wav file, and wav file from the payload, and viewing the waveform |
| 05:08.46 | orlock | Corydon76-home: inbound audio quality glitches |
| 05:09.07 | Corydon76-home | orlock: dropped packets/collisions perhaps? |
| 05:09.14 | orlock | we are going to remove asterisk from the equation at one of the sites, for one of the DID's |
| 05:09.34 | Corydon76-home | orlock: perhaps try ilbc as your codec |
| 05:09.52 | orlock | Corydon76-home: one of the calls i tcpdump'd has two missing inbound rtp packets, and they noticed the audio glitch when it was missed |
| 05:09.59 | *** join/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 05:10.06 | orlock | using g729 currently |
| 05:10.22 | Corydon76-home | orlock: that should alleviate the trouble for dropped packets, as long as they don't exceed a certain threshold |
| 05:10.24 | Narkov- | is it possible to set the peer port on a peer by peer basis in IAX? |
| 05:10.36 | Corydon76-home | Narkov-: yes, it is |
| 05:10.45 | orlock | Corydon76-home: provider seems to think one acket wont be perceptibloe, so has asked us to remove * |
| 05:10.50 | Corydon76-home | port=1234 in the peer definition |
| 05:11.05 | Corydon76-home | Narkov-: but you need to set the bindport on the remote host to match |
| 05:11.07 | Narkov- | thanks Corydon76-home....voip-info didnt have that in the iax.conf page |
| 05:11.33 | Narkov- | oohhh...so it cant be 4569-> 4570 ? |
| 05:11.53 | Corydon76-home | Narkov-: it can but the remote host must be listening on 4570 |
| 05:12.17 | Narkov- | ...and my local listen port can be 4569? |
| 05:12.35 | Corydon76-home | Yes, that's the bindport definition |
| 05:12.40 | converx | How can i set a voicemail greeting for each mailbox? |
| 05:12.41 | Narkov- | ahhh.ok |
| 05:12.43 | Narkov- | thanks mate |
| 05:12.52 | Corydon76-home | bindport = local listen port; port in peer = remote connection port |
| 05:13.22 | Corydon76-home | ~book |
| 05:13.28 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:13.28 | Narkov- | is it possible to set my local port for sending? |
| 05:13.40 | Corydon76-home | Narkov-: no, that's your bindport |
| 05:14.01 | Corydon76-home | converx: most of your questions can be answered by reading the book |
| 05:14.43 | Narkov- | hrmm...i want to split two peers (both the same provider/IP) using some kind of IP differentiation |
| 05:14.54 | Narkov- | ..so I can route over two DSL connections |
| 05:15.02 | linlin | whats the easiest way to link two asterisk machines together to act as one machine |
| 05:15.03 | converx | corydon - name? |
| 05:15.12 | Corydon76-home | converx: see above |
| 05:15.39 | Corydon76-home | Narkov-: not sure I understand |
| 05:16.18 | Narkov- | i have two accounts with the same IAX end point...two DSL connections...and I want to be able to force peer1 to go over dsl 1 and peer2 over dsl2 |
| 05:16.30 | Corydon76-home | Narkov-: I think you'll want to look at the Linux routing tables, which Asterisk observes |
| 05:17.03 | rob0 | iproute2 if installed, "man ip" with recent versions. |
| 05:17.28 | rob0 | BUT you have to know a lot about IP routing to be able to understand that. :) |
| 05:17.37 | Narkov- | yeah I'm pretty sure i'll need iptables but I dont know how i can differnetiate the two connections given they go to the same IP/port combo |
| 05:17.55 | rob0 | nono I did not say iptables, I said ip(8) |
| 05:18.03 | orlock | Narkov-: ip, not ip tables |
| 05:18.11 | Corydon76-home | Narkov-: sounds like connection bonding... |
| 05:18.27 | orlock | you can set up routes based on ranges |
| 05:18.35 | Corydon76-home | Narkov-: in any case, Asterisk is not involved at that point; that's advanced networking |
| 05:18.41 | Narkov- | sure but it goes to the same IP/port combo orlock |
| 05:18.48 | rob0 | LARTC HOWTO, lartc.org |
| 05:19.23 | orlock | what rob0 said |
| 05:19.23 | Narkov- | thanks guys..i'll have a look at lartc |
| 05:20.01 | Sed[PCT] | anyone here using a 7960 phone with sip image... do you have speed dial lines setup in your tftp config? |
| 05:20.10 | orlock | Sed[PCT]: nope |
| 05:20.19 | Sed[PCT] | hmm ok |
| 05:20.36 | Sed[PCT] | I know it can be done on the phone.. was wondering if it could be done with the SIPXXXXXX.cnf file on the tftp server... |
| 05:20.40 | monsted | Sed[PCT]: i use them, but not with speed dial |
| 05:20.54 | Sed[PCT] | ah |
| 05:24.18 | DrCron | how are incoming calls from unknown sources (ie direct sip calls) handled? |
| 05:24.47 | Supaplex | yea, like DrCron said, http://rafb.net/paste/results/A3IkmB85.html |
| 05:24.54 | Supaplex | Corydon76-home: ^^^ :) |
| 05:25.17 | Supaplex | nono not unknown sources (me), unknown dst :-x |
| 05:25.37 | Supaplex | I'm putting the grey matter to bed real soon now. |
| 05:25.54 | DrCron | oh, sorry, i was asking a new question. I dont quite understand where they go |
| 05:31.44 | *** part/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 05:33.56 | Newbie___ | anyone using a cmg card ? |
| 05:37.44 | docelmo | Say anyone got time for a couple questions.. 1.. Where can I find a list of the USA area codes? And also whats a simple way to control concurrent calls on a peer? |
| 05:39.17 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 05:40.24 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.mn.comcast.net) |
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| 05:46.44 | DrCron | google nanpa |
| 05:47.07 | DrCron | http://www.nanpa.com/area_codes/index.html |
| 05:49.31 | docelmo | thanks |
| 05:49.43 | docelmo | any ideas on the concurrent calls? |
| 05:49.55 | hads | call-limit |
| 05:50.09 | hads | (in sip.conf) |
| 05:50.28 | docelmo | hmm.. Any other way to set it w/o the peer? |
| 05:50.38 | docelmo | Cause I am using SER as my front end which uses SER's IP |
| 05:51.10 | hads | Possibly some configuration there, but I don't know SER. |
| 05:53.14 | *** join/#asterisk nedhelp (n=pcgultia@202.84.109.215) |
| 05:59.10 | Sed[PCT] | orlock and monsted you have the mwi working with asterisk? |
| 06:04.11 | monsted | mwi? |
| 06:05.24 | Sed[PCT] | the light..message waiting indicator |
| 06:07.04 | *** join/#asterisk darqchild (n=e@206-248-138-220.dsl.teksavvy.com) |
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| 06:13.44 | *** part/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 06:17.35 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:18.41 | Sed[PCT] | ah ha.. nm |
| 06:27.47 | *** join/#asterisk Mother_ (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 06:34.50 | DrCron | I'm trying to figure out how to set asterisk up to accept incoming calls from anywhere over sip, and make sure it comes in with the correct context |
| 06:37.34 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:38.19 | *** join/#asterisk Dustyservers (n=eric@S01060060979872cb.ed.shawcable.net) |
| 06:38.58 | Dustyservers | hello every one do I need an fxo or and fxs to connect and ip phone to my asterisk server am still new with this stuff.. |
| 06:39.38 | DrCron | no |
| 06:40.14 | DrCron | ip phones only need some sort of ip connection, 802.3 802.11 something like that |
| 06:40.25 | Dustyservers | oh |
| 06:40.29 | lowlevel | ;) |
| 06:40.32 | Dustyservers | then what are fxo/fxs cards for then |
| 06:40.39 | DrCron | fxo and fxs are for connections to the analog network |
| 06:40.45 | lowlevel | for interfacing to real phone lines and real phone.s |
| 06:40.46 | DrCron | and to analog phones |
| 06:41.01 | DrCron | "real" |
| 06:41.03 | lowlevel | ;) |
| 06:41.14 | JT | well they are real |
| 06:41.14 | Dustyservers | ok so whihc one I need to connect my telus phone line to the box then? |
| 06:41.22 | JT | digital isdn lines are even more real :P |
| 06:41.34 | lowlevel | jt; no, those are sureal |
| 06:41.35 | DrCron | Dustyservers, ick ick ick telus |
| 06:41.37 | lowlevel | heh |
| 06:41.42 | JT | uhuh |
| 06:41.44 | Dustyservers | I know I hate telus too |
| 06:41.45 | Dustyservers | but yea |
| 06:41.46 | Dustyservers | lol |
| 06:42.03 | DrCron | fxs iirc |
| 06:42.10 | Dustyservers | fxs for telus lines then? |
| 06:42.32 | Sed[PCT] | no |
| 06:42.38 | Dustyservers | fxo? |
| 06:42.45 | Sed[PCT] | fxo is what you plug your telephone line into from the wall |
| 06:42.49 | Sed[PCT] | fxs is what you plug a analog phone into |
| 06:42.51 | lowlevel | wait till he finds out the fxo talks fxs and vice versa ;) |
| 06:42.55 | Dustyservers | aww ic now |
| 06:43.06 | Sed[PCT] | so you want a fxo to pull your landline phone into asterisk |
| 06:43.08 | Sed[PCT] | lowlevel: lol |
| 06:43.23 | Dustyservers | aww make sence now |
| 06:43.29 | DrCron | what does fxo stand for? |
| 06:43.38 | Sed[PCT] | FXO: Foreign Exchange Office |
| 06:43.45 | Sed[PCT] | FXS: Foreign Exchange Station |
| 06:43.46 | Sed[PCT] | iirc |
| 06:43.52 | Sed[PCT] | and spelling too probably |
| 06:44.08 | DrCron | see, that seems totaly backward to me |
| 06:44.21 | Sed[PCT] | DrCron: wanna know whats really confusing? |
| 06:44.31 | DrCron | it is totaly backward? |
| 06:44.31 | Sed[PCT] | fxo's talk with fxs signalling.. |
| 06:44.38 | Sed[PCT] | and fxs's talk fxo signalling |
| 06:44.50 | Sed[PCT] | :p |
| 06:44.53 | Dustyservers | make sence |
| 06:45.08 | Sed[PCT] | yup |
| 06:45.34 | *** join/#asterisk groogs___ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
| 06:45.44 | DrCron | Dustyservers, if you have a broadband connection you could drop your landline all together, and use a voip provider for incoming and outgoing calls |
| 06:46.46 | DrCron | is there a list of bastard npa codes that lead to unregulated markets? |
| 06:49.17 | Dustyservers | thanks for all the help |
| 06:51.05 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:56.18 | *** join/#asterisk wd (n=wd@82.99.185.7) |
| 06:57.00 | *** join/#asterisk wd (n=wd@82.99.185.7) |
| 06:59.54 | wd | would anybody mind, if I'll ask question not directly related to Asterisk? |
| 07:00.38 | Sed[PCT] | I guess it depends what it is.. and if anyone can answer it :p |
| 07:01.02 | wd | I would like to have one PBX connected to more VoIP providers |
| 07:01.33 | Sed[PCT] | ok |
| 07:01.41 | wd | if provider no1 will be unavailable, the PBX should use the 2nd etc. |
| 07:01.56 | wd | my question is: |
| 07:02.40 | wd | what number will I use for outgoing calls? |
| 07:03.05 | DrCron | you mean CID? |
| 07:03.22 | wd | what is CID? |
| 07:03.27 | Qwell | ~cid |
| 07:03.35 | jbot | TCP client/server Caller-ID system, including server and Tk GUI client.. URL: http://www.tummy.com/cid |
| 07:03.35 | Sed[PCT] | depends how you have your dialplan setup... if you manage to have the same number with both providers (hard to do)... usually your outgoing number will be the one from the respected provider your going out |
| 07:03.57 | Qwell | stupid bot |
| 07:04.00 | russellb | wtf |
| 07:04.00 | Sed[PCT] | he's lagging... be kind :p |
| 07:04.12 | russellb | jbot: cid is CallerID, you nub |
| 07:04.13 | jbot | ...but cid is already something else... |
| 07:04.24 | Sed[PCT] | job: no, cid is CallerID |
| 07:04.26 | Qwell | jbot: no, cid is CallerID |
| 07:04.27 | jbot | Qwell: okay |
| 07:04.34 | Qwell | jbot: cid is also a TCP client/server Caller-ID system, including server and Tk GUI client.. URL: http://www.tummy.com/cid |
| 07:04.35 | jbot | Qwell: okay |
| 07:04.37 | Sed[PCT] | ops :p |
| 07:04.37 | DrCron | do you need the caller id set properly? or do you need the actual out bound number set |
| 07:04.41 | *** join/#asterisk X-Rob (n=rob-x@dsl-58-6-69-193.vic.westnet.com.au) |
| 07:04.52 | wd | ok...give me few seconds to absorb these informations |
| 07:04.55 | wd | :) |
| 07:04.57 | Sed[PCT] | lol |
| 07:05.15 | Sed[PCT] | if your provider allows you to set your caller-id on outbound.. you can set it to whatever number you want |
| 07:05.29 | DrCron | if you need the demarcation to ss7 to have the same info, you are most likely SOL |
| 07:05.48 | DrCron | if you just need the caller-id information to be the same, well thats easy |
| 07:05.59 | wd | OK. Small example: |
| 07:07.04 | wd | usually I'll use no1. Anyone who calls me will use no1 network and servers to reach me. But if its network/server crashes... |
| 07:07.29 | wd | ...how will incoming calls reach me via no2 network/servers? |
| 07:07.47 | DrCron | ah, incoming |
| 07:08.08 | DrCron | incoming backup paths are more difficult |
| 07:08.28 | wd | my potential customers shouldn't know anything about network failures... |
| 07:08.54 | JT | needs to be setup on your provider's end |
| 07:09.05 | wd | they should dial/receive calls independently on network failures. |
| 07:09.43 | DrCron | yhea, short version, its a pain in the ass, and requires your provider to work with another provider, and well, it can be ugly |
| 07:09.46 | JT | incoming backup paths have to be setup at whoever is providing the DIDs |
| 07:10.07 | wd | I'm not so familiar with dial plan - I prefer IP. Is the routing process the same? |
| 07:10.23 | JT | IP, what does that have to do with the dial plan? |
| 07:10.40 | wd | my provider says: number xyz is on this IP address? |
| 07:11.02 | wd | I just tried:) |
| 07:11.02 | JT | yeah with SIP usually |
| 07:11.09 | JT | if they terminate to the pstn |
| 07:11.21 | JT | they won't normally be informing callers of your ip |
| 07:11.21 | DrCron | do you want protection against your provider going down. or one of your servers |
| 07:11.30 | JT | just routing a phone connection in their switching matrix |
| 07:11.48 | wd | DrCron: providers server |
| 07:12.04 | JT | to put it basically |
| 07:12.08 | JT | unless you're a telco |
| 07:12.11 | JT | you're dreaming |
| 07:12.14 | JT | to get reliable backup |
| 07:12.21 | JT | for inbound |
| 07:12.23 | wd | JT: :( |
| 07:12.26 | JT | your provider is meant to work |
| 07:12.28 | DrCron | how much are you willing to spend? |
| 07:12.46 | JT | if reliability is a concern, get PRIs for your inbound lines, don't use voip |
| 07:12.54 | JT | voip is less reliable than POTS/TDM |
| 07:13.14 | wd | DrCron: it depends on reliability:) |
| 07:13.31 | wd | JT: I know that... |
| 07:13.41 | wd | I'm ISP |
| 07:13.43 | JT | then why talk about inbound voip |
| 07:13.51 | DrCron | if you want it for less then a PRI, dream on |
| 07:14.00 | DrCron | PRI's would be much cheaper |
| 07:14.09 | wd | I work with no1 provider, but he has problems |
| 07:14.25 | JT | then don't work with him if the problems are too great |
| 07:14.28 | wd | my customers don't understand the meaning of word problems |
| 07:15.07 | wd | so I'm trying to solve it |
| 07:15.13 | JT | anyway, long and the short of it is you should get inbound PRIs |
| 07:15.32 | wd | I have PRI |
| 07:15.40 | hads | Use it? |
| 07:16.07 | wd | but I was thinking about VoIP |
| 07:16.07 | DrCron | your pri provider has problems? ick |
| 07:16.25 | hads | This conversation is going in circles. |
| 07:16.26 | wd | DrCron: no...the VoIP provider |
| 07:16.34 | wd | hads: jop |
| 07:16.34 | JT | well if reliability is super important, you shouldn't be thinking about voip for inbound |
| 07:16.37 | DrCron | stick with the pri |
| 07:16.54 | wd | OK. thanks everybody |
| 07:16.58 | DrCron | use voip for outbound |
| 07:17.53 | DrCron | anyways, is there a list of npa codes that go to, how shal we say, nasty places? |
| 07:18.15 | wd | DrCron: but outgoing calls will not use the same number as incoming calls |
| 07:18.33 | JT | shrug, it's a way to save money, just block CID sending |
| 07:18.40 | JT | otherwise use PRI for outbounds too |
| 07:18.50 | DrCron | or you can have the CID set as the same |
| 07:18.57 | JT | not always |
| 07:19.16 | DrCron | some providers can |
| 07:19.18 | JT | sensible telcos have sanity checking on CIDs set by customer PRIs |
| 07:19.46 | JT | or SIP (which would end up at a PRI anyway) |
| 07:19.53 | orlock | ours does |
| 07:20.11 | DrCron | att/sbc doesnt, at least not for residential |
| 07:20.30 | DrCron | and cingular doesnt |
| 07:20.57 | JT | i don't know of any providers in .au that let you set CID to a number that's not yours |
| 07:21.07 | wd | Thanks. I'll think about it few hours:) I'll be back ;) |
| 07:21.16 | DrCron | they happily report what ever i tell voipjet to use |
| 07:22.08 | DrCron | now, I've only used my own numbers, i wonder if it would complain if I set it to the white house tour information number |
| 07:22.43 | JT | well that's not a real test |
| 07:22.54 | JT | if you've only used your own numbers |
| 07:22.59 | JT | of course you can set that |
| 07:23.32 | DrCron | voipjet doesnt know they are my numbers |
| 07:23.43 | JT | hmm |
| 07:24.11 | DrCron | yup just set the cid as the white house comments line # |
| 07:24.18 | DrCron | never trust your CID |
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| 07:27.40 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 07:28.15 | yxa | hi how can I let Voicemail not prompt for the mailbox and only the password? the mailbox should be the same as the extension which called voicemail() |
| 07:29.02 | hads | VoiceMailMain(${CALLERID(num)}) |
| 07:31.07 | Sed[PCT] | I wish my teliax cid would work |
| 07:32.16 | yxa | hads thanks! |
| 07:34.09 | DrCron | i seem to remeber a voip provider that offered free outgoing 800 number calls |
| 07:35.47 | *** join/#asterisk lorinc (n=ang@213.178.121.201) |
| 07:38.47 | EmleyMoor | DrCron: FWD? |
| 07:39.54 | russellb | iaxtel ... *should* work for that, too ... |
| 07:40.15 | wd | summary: I should turn off CID sending (easiest way). Or I have to ask my providers to cooperate (then I can use them as backup for outgoing/incomming calls). Are my conclusions right? |
| 07:41.08 | DrCron | they dont just have to cooperate, it would cost them $$, and so would cost you $$ |
| 07:41.59 | wd | DrCron: allways a catch:) |
| 07:42.27 | wd | thanks for help |
| 07:43.12 | DrCron | outbound cid setting shouldnt be a problem though, |
| 07:43.13 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:43.45 | wd | but it wont help with inbound... |
| 07:44.01 | DrCron | nope |
| 07:44.29 | DrCron | if you need high 9's reliability inbound use pri |
| 07:44.52 | DrCron | and then save money by routing outbound over voip providers |
| 07:45.15 | wd | DrCron: ok |
| 07:45.57 | shellshark | wd: i can provide you outbound calling very resonably |
| 07:46.21 | wd | DrCron: is there a way how to find out if my providers allows me to set any CID? |
| 07:46.41 | DrCron | easiest is to ask |
| 07:46.44 | shellshark | wd: we allow all of our business accounts to set CID |
| 07:46.47 | DrCron | other wise, just try it |
| 07:46.54 | wd | DrCron: :))) |
| 07:47.02 | *** join/#asterisk ComPuTeR (n=Toyota@88.240.41.123) |
| 07:47.10 | Chris-NB | hi |
| 07:47.11 | Chris-NB | anyone got some experience with AoC records? |
| 07:47.22 | wd | shellshark: can you send me some contacts? |
| 07:47.34 | shellshark | wd: http://www.shellshark.net/ |
| 07:47.41 | shellshark | wd: sales@shellshark.net |
| 07:48.49 | wd | shellshark: ok. thanks |
| 07:48.57 | shellshark | wd: no problem |
| 07:49.23 | shellshark | wd: we just have a couple standard plans listed on the site, but we can custom-tailor to anything you need |
| 07:50.51 | DrCron | hmm, so i'm trying to set asterisk up to do uri calls inbound |
| 07:51.30 | shellshark | DrCron: should not be too difficult as long as the extension is defined in extensions.conf and you expose port 5060 to the world |
| 07:52.02 | shellshark | sip:someextension@your.sip.gateway.dom |
| 07:52.05 | wd | shellshark: I will send you an e-mail. I need to know more and more people (my boss;) ) |
| 07:52.27 | shellshark | sure :) |
| 07:52.29 | DrCron | how does asterisk handle incoming calls, ie what context do they go under? |
| 07:52.31 | wd | shellshark: ...and ask more... |
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| 07:52.51 | shellshark | DrCron: whatever context you desire |
| 07:53.38 | DrCron | um... yhea, thats where i'm having a bit of a problem, how do i set up the sip.conf for anon incoming calls |
| 07:53.49 | DrCron | or iax for that matter |
| 07:54.30 | DrCron | would it be [guest]? |
| 07:54.32 | shellshark | DrCron: you have a guest user in both that allows calls from anywhere, ie. no host and no secret declaritive |
| 07:54.38 | shellshark | yep |
| 07:54.53 | DrCron | thats too simple |
| 07:54.59 | shellshark | IAX URI's include the username |
| 07:55.23 | shellshark | iax2://guest@your.sip.gateway.dom/someextension |
| 07:55.26 | *** join/#asterisk shinux_ (n=shinux@196.220.26.158) |
| 07:56.28 | DrCron | shellshark, do you agree that setting up an inbound backup for calls would be a huge pain? (same inbound number) |
| 07:56.43 | shellshark | absolutely |
| 07:56.48 | shellshark | you'd need SS7 to do it |
| 07:56.58 | DrCron | thats what i thought |
| 07:57.02 | DrCron | thats what wd wanted |
| 07:57.12 | shellshark | right |
| 07:57.21 | shellshark | you're talking major costs at that point |
| 07:57.28 | DrCron | I've had friends set that up, but they had money to burn |
| 07:57.54 | wd | DrCron: how much is it? |
| 07:58.16 | shellshark | DrCron: doubt it... to get SS7 trunks you have to be a CLEC (in the USA at least) |
| 07:58.34 | DrCron | i mean they had it set up thorugh their providers |
| 07:58.52 | DrCron | not personaly, they dont have that much $ |
| 07:59.45 | DrCron | you dont want to do that over ip |
| 07:59.59 | DrCron | you want that done by a local pri provider |
| 08:00.10 | DrCron | over two phys lines if you realy need it |
| 08:00.43 | DrCron | so, cost of pulling new cable, maint fees, |
| 08:01.08 | DrCron | how much reliability do you need, how many 9's will you pay for |
| 08:01.41 | wd | DrCron: 100 :))) No. I think it's not a good idea |
| 08:02.01 | wd | DrCron: I have to ask my providers |
| 08:02.22 | DrCron | 5 9's is usualy standard, |
| 08:02.39 | DrCron | 99.999% uptime is good enough for almost everyone |
| 08:02.46 | shellshark | the people who own the number can do alternative routing on it without having to do SS7 |
| 08:03.39 | wd | DrCron: It is more than enough |
| 08:04.35 | DrCron | they were in palo alto i remeber them doing something complicated with both the local fiber loop and a t-1 for their phone and data service it required 2 providers to work together though, that much i do remeber |
| 08:05.27 | DrCron | wd, then stick to pri for your inbound that would almost certainly be the cheapest way to go |
| 08:06.46 | wd | DrCron: it looks like the only way... |
| 08:07.07 | DrCron | just call your pri provider, see what their uptime guarantee is |
| 08:07.29 | DrCron | you might be suprised, will most likely be four or five 9's |
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| 08:12.21 | DrCron | does asterisk match sip.conf by placment in the file or by specificity? |
| 08:15.59 | *** join/#asterisk Jurian (n=magic@243-253-045-062.dynamic.caiway.nl) |
| 08:17.04 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 08:17.28 | Jurian | Good day, I have a question, how can I specify the outgoing SIP accountname to use? Currently, it appears my phones try to use their internal extension as the outgoing account, which is (obviously) rejected by our provider |
| 08:17.45 | Jurian | oh, and there's an on join spammer here: <aLeX> http://www.videoturk.tr.gs Free Porno Videos |
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| 08:22.20 | hads | Jurian: Account? If you mean callerid then you can Set(CALLERID(num)=123) before you dial the provider. |
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| 08:27.01 | pukkita | good day |
| 08:27.01 | EmleyMoor | Jurian: Surely outgoing calls go via your *? |
| 08:27.39 | pukkita | I'd like to use distinctive ringing depending on the calling source |
| 08:28.07 | EmleyMoor | I believe it's possible but I'd like to know more about that too |
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| 08:28.25 | pukkita | I see this is done using the _ALERT_INFO var but wonder how to integrate that in the dial plan |
| 08:28.37 | zepmantra | hello, our other phoneline comes down (telco problems), this is my test asterisk line, how can i playback using xten the process when the channel rings, is it possible? |
| 08:28.52 | DrCron | Set(_ALERT_INFO)? |
| 08:29.08 | pukkita | but where |
| 08:29.44 | DrCron | pukkita, in your dial plan where you handle incoming calls |
| 08:30.29 | DrCron | pukkita, look at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf << defining extensions |
| 08:30.54 | pukkita | yes, I have them set up |
| 08:31.04 | DrCron | by calling source you mean CID? |
| 08:31.13 | pukkita | I have SIP and a mix of digital lines |
| 08:31.18 | DrCron | ah |
| 08:31.48 | pukkita | mix = digital telco lines, digital "lines" connected to the PBX |
| 08:31.49 | DrCron | set up individual contexts for the ones you want to have distinctive ring on |
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| 08:32.10 | Jurian | hrm, not sure how to explain this, I have 10 numbers with our voip provider, but depending on which phone is used, I want to use a different outgoing number, right now, * appears to be sending my phone's SIP account username as the "from", and the voip provider is rejecting it, cause it needs to be one of the 10 numbers |
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| 08:32.37 | Jurian | SIP/Belcentrale-0819d4c0 is making progress passing it to SIP/802-b6d3b588 |
| 08:32.47 | Jurian | 802 is my internal number |
| 08:32.49 | Jurian | it should be external one |
| 08:32.55 | X-Rob | qwell, Qwell[], file, Corydon-w, Corydon76-home, the user Alex is doing spam /msg's on joins. |
| 08:33.03 | Jurian | setting callerid and cdr(accountcode) doesn't appear to help |
| 08:33.14 | tparcina | ARI - where can I download it? I have check www.littlejohnconsulting.com but it gives me an error. |
| 08:33.15 | DrCron | Jurian, this is dialing thorugh an asterisk server right? |
| 08:33.20 | Jurian | yes |
| 08:33.49 | Jurian | my phone --SIP--> my asterisk --SIP--> provider system |
| 08:34.12 | Jurian | exten => _0[1-9].,n,Dial(SIP/31${EXTEN:${TRUNKMSD}}${SIP2}) |
| 08:34.30 | Jurian | is what I have in extensions.conf |
| 08:34.45 | tparcina | ARI - Asterisk Recording Interface; does anybody know where I can download it (except from - www.littlejohnconsulting.com which doesn't work right now)? |
| 08:34.55 | pukkita | DrCron, I have a HW PBX integrated with my *. When someone from a traditional extension dials e.g. 5XXX the PBX "dials" out thinking it's attached to a phisical line, then my * catches it and dials the SIP exten. I'd like the SIP phone receiving that call use a distinctive ringing. |
| 08:34.59 | Jurian | along with Set(CALLERID(number)=anumber) and Set(CDR(accountcode)=anumber) |
| 08:35.41 | DrCron | you are going to need to modify the 31${EXTEN:${TRUNKMSD}}${SIP2} part |
| 08:36.11 | pukkita | it shouldn't be difficult but right now I'm still digesting all I've read |
| 08:36.27 | Jurian | ${SIP2} is @voipprovider |
| 08:36.40 | DrCron | so, calls going from pbx ->analog line -> *-> sip should do the distinctive ring |
| 08:36.49 | pukkita | yes |
| 08:36.57 | DrCron | Jurian, try entering the details manually |
| 08:37.22 | DrCron | well, directly on that line anyways |
| 08:37.35 | Jurian | how do you mean? |
| 08:38.16 | Jurian | is there a way to, like, Dial(SIP/destination@provider, mynumber) |
| 08:38.29 | pukkita | my problem is the dialplan is always clear to me on where are you going, but not where are you coming from :) |
| 08:38.29 | DrCron | well, i'm using voipjet and instead of using variables i have the info on that line ie: exten => _1NXXNXXXXXX,2,Dial,IAX2/nnnnn@voipjet/${EXTEN} |
| 08:39.01 | DrCron | pukkita, the easiest way to fix that is have the calls coming in from the pbx use a diffrent context |
| 08:39.18 | DrCron | well, easiest way that i see |
| 08:39.24 | DrCron | i'm a noob myself |
| 08:40.14 | pukkita | aaaahh!! |
| 08:40.38 | pukkita | and just do a Set(_ALERT_INFO) there? |
| 08:40.58 | DrCron | thats what i would try, yhea |
| 08:41.04 | pukkita | thx |
| 08:41.36 | DrCron | there is probably a cleaner way to do it |
| 08:42.11 | pukkita | that looks fine |
| 08:42.28 | pukkita | that var is inherited in the following contexts? |
| 08:42.41 | DrCron | i think so |
| 08:42.47 | DrCron | try it |
| 08:42.54 | pukkita | gimme a sec |
| 08:43.21 | DrCron | you wouldnt be here if you weren't willing to fool around with the configs and see if it worked |
| 08:43.34 | hads | show application Set |
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| 08:53.56 | pukkita | hmmm |
| 08:54.25 | pukkita | if I use an s extension, will it be processed, or will * jump to the more specific ones in that context? |
| 08:54.40 | pukkita | for the Set(_ALERT_INFO) |
| 08:55.09 | EmleyMoor | s is for calls entering a context without an extension, so the more specific ones will be used |
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| 08:57.06 | pukkita | then what should be used? I'm looking to execute a Set() everytime we enter that context, then go with the rest of extensions |
| 08:57.28 | HaMYaI | anyone knows what causes a problem where out-going sip calls are redirected to default context? |
| 08:58.00 | HaMYaI | it only happens to me every once in a while |
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| 09:04.18 | DrCron | fwd is traditionally called by the npa 700, correct? |
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| 09:20.53 | Newbie___ | hi, can a single TE110P handle a single adit600 with 48 FXS ? |
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| 09:36.08 | mattfletcher | Hello, does anyone know how to disable the extra virtual lines on a Aastra 480i IP Phone. I want it to act as if it were only a single-line phone |
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| 09:39.15 | DrCron | i dont know if you can |
| 09:39.21 | DrCron | why would you want that? |
| 09:40.10 | pif | do you guys let your users hide their caller id ? |
| 09:40.12 | mattfletcher | because when i'm using agents, the queue is seeing the extra lines on the phone, and passing calls to those agents, rather than skipping to the next one |
| 09:40.47 | mattfletcher | * when they are already off hook i ought to say :) |
| 09:41.18 | qwertz | Hi, does anybody know if GotoIfTime is available in * 1.0.10? |
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| 09:42.37 | DrCron | mattfletcher, i'm sure there is a way, i just have no idea how to do it |
| 09:42.48 | DrCron | is it a pretty good phone? |
| 09:43.01 | DrCron | i'm thinking of getting the one with the wireless handsets |
| 09:45.48 | mattfletcher | to be perfectly honest, i don't know how good they are. we bought two, and shipped them down to our other office the day after they arrived. from looking at them briefly they seem to be fine. they replaced some rather nice DECT handsets and I have heard no complaints so far (after a week) |
| 09:46.46 | DrCron | yhea, well i would, admittedly love to see a good dect gateway to iax |
| 09:47.01 | DrCron | because then i could have a wireless phone with bluetooth |
| 09:49.47 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 09:55.13 | zumbush | Have an old Asterisk 1.0.9 box with an Wildcard TDM400P with analog ports |
| 09:55.13 | zumbush | Just installed a new box with Asterisk SVN-branch-1.2-r46258 connected to an 32-channel PRI through an Wildcard TE110P. |
| 09:55.19 | zumbush | Things seem to work well an i can call in and out to the box. But when i try to dial in with my old box to the incoming digital receptionist on the new box it just keeps on ringing and never reacts to my new box Answering and playing the recording. |
| 09:55.23 | zumbush | If i do an direct in dial to a sip-extension everthing works ok but not when i call my main number that is direkted to the digital receptionist. |
| 09:55.23 | zumbush | Any idea what might be wrong. |
| 09:55.32 | *** join/#asterisk spax (n=spax@nat-pool-227.bmcag.de) |
| 09:55.49 | pif | how to detect a prohib flag on a call? |
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| 10:06.15 | yxa | anyone has experienced with Cologne based HFC bri cards with mISDN? |
| 10:06.32 | yxa | ie, Digium B410P |
| 10:09.13 | pif | yep |
| 10:09.28 | pif | HFC-4S |
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| 10:10.27 | yxa | pif i've successful set up incoming calls and outgoing |
| 10:11.20 | yxa | pif but when someone picks up an incoming call and transfer, asterisk and the misdn driver hangs |
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| 10:11.33 | pif | versions? |
| 10:11.49 | *** join/#asterisk Guest77166 (i=Singh@124.62.150.38) |
| 10:12.03 | yxa | Asterisk SVN-branch-1.2-r48192 2.6.15.7 |
| 10:12.18 | pif | misdn? |
| 10:12.31 | Guest77166 | hello guys |
| 10:12.38 | yxa | i just issue make b410p from zaptel/ |
| 10:12.48 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 10:12.57 | Guest77166 | I tried to install asterisk 1.4 |
| 10:13.15 | pif | yxa: that's not misdn |
| 10:13.20 | Guest77166 | but i was unable to install |
| 10:13.48 | Guest77166 | do i need to install any additional package? |
| 10:15.07 | yxa | pif how do i check? |
| 10:15.26 | Guest77166 | anyone can help me in here? |
| 10:15.45 | EmleyMoor | What went wrong? |
| 10:20.47 | yxa | pif its misdn. but i just don't know the version that digium gave |
| 10:26.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:27.21 | puzzled | morning |
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| 10:29.07 | DrCron | hmm, interesting, free US DID |
| 10:29.26 | DrCron | actaully, they pay you |
| 10:30.19 | X-Rob | I'll have a million DID's then, thanks! |
| 10:30.32 | DrCron | only if you can get incoming calls on them |
| 10:30.43 | DrCron | http://www.trxtel.com/ |
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| 10:35.57 | Dovid | morning all |
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| 10:47.46 | pukkita | is there any variable that holds the channel a call comes from? |
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| 10:49.26 | Dovid | yes |
| 10:49.36 | Dovid | i believe channel or channelid |
| 10:49.41 | Dovid | have a look on the wiki |
| 10:49.42 | Dovid | ~wiki |
| 10:50.02 | pukkita | DrCron: the _ALERT_INFO didn't work. It seems the s extension there only gets called if a call coming from there doesn't find any extension |
| 10:50.12 | Dovid | (my connection is too slow to get u the URL) |
| 10:50.54 | Dovid | pukkita: thats what the s extension is for |
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| 10:51.08 | pif | SIPAddHeader(Alert-Info: blah) |
| 10:51.26 | pukkita | Dovid is there any other that always is executed? |
| 10:51.47 | pukkita | I want to set an specific ALERT_INFO depending on the incoming channel |
| 10:52.20 | pif | have your channels use different contexts |
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| 10:53.17 | Dovid | pukkita: for diffrent DID's ? |
| 10:53.27 | Dovid | then do as pif said. create diffrent contexts. |
| 10:53.40 | DrCron | i thought that was what i sugested |
| 10:53.42 | Dovid | so u can add the alert before you send it to the exten or IVR |
| 10:53.55 | pif | or learn GotoIf() |
| 10:54.16 | DrCron | wow, DID's can be cheap |
| 10:54.23 | DrCron | $4 a month? |
| 10:54.49 | DrCron | this could end up saving me a metric butload of cash |
| 10:56.03 | oink | How's Cisco 7912G phones compared to Snom 300 ? |
| 10:56.12 | oink | (Hello! ;-) |
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| 10:56.46 | Dovid | DrCron: Be carefull with whom you go with |
| 10:56.55 | Dovid | cheap doesnt allways = good |
| 10:57.46 | DrCron | do you have any recomendations on a DID for the united states? |
| 10:57.47 | Dovid | i found a provider for $5.00 per DID and i ccan get up to 10 channels at once incoming (although if you keep up heavy traffic they will can you) |
| 10:57.48 | pukkita | DrCron: that's what I've tried, but cannot make it execute it always then go on with the dialplan. |
| 10:58.12 | Dovid | DrCron: myphonecomapny.com |
| 10:58.18 | Dovid | i use em only for inbound |
| 10:58.22 | Dovid | outbound they arent cheap |
| 10:58.39 | Dovid | its not on thier site. you have to email them for a mydeviceplan |
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| 10:59.03 | Dovid | been with them for 2 1/2 years. They were down once for 30 min cause of "network issues" |
| 10:59.35 | Dovid | Pukkita: y cant you split the incoming calls by context ? |
| 10:59.37 | Modcuts | Good morning, anybody know of any alert software that can be used in conjunction with sip phones to alert calls coming in? |
| 10:59.39 | pukkita | DrCron, I already have a context defined for incoming calls through a digital line. but what kind of extension should I declare there so that it's always executed? |
| 10:59.56 | DrCron | i was trying out voipjet for my outbound calls, seemed decent |
| 11:00.16 | Dovid | Modcuts: define alert software |
| 11:00.28 | Dovid | DrCron: Voip jet is good but thier cc sux |
| 11:00.44 | Dovid | i use them as primary (cause of thier rates) and fail over to teliax if they arent up |
| 11:00.52 | DrCron | cc = payment system? |
| 11:01.05 | Dovid | oops |
| 11:01.10 | Dovid | cs = customer service |
| 11:01.13 | pukkita | dovid yes, I can. say that context is [isdn_in_te] then the call goes to a SIP extension. |
| 11:02.00 | DrCron | yhea, i prefer to use a pre paid system, seems like a better idea then monthly |
| 11:02.18 | Modcuts | <Dovid>: you have a call coming in and the number........, use sip hardware phones but i have been asked to get software alerting also any ideas? |
| 11:02.22 | Dovid | DrCron: montly usualy costs more |
| 11:02.26 | pukkita | how do I set the __ALERT_INFO var in [isdn_in_te] so that no matter where that call ends up, that var is set (and inherited in the final context) |
| 11:03.23 | Dovid | Modcuts: dont know off hand |
| 11:03.28 | Dovid | google it |
| 11:04.08 | Dovid | Pukkita: SIPAddHeader(Alert-Info: Variable) |
| 11:04.13 | Dovid | pukkita: I asume now u have something like |
| 11:04.18 | Dovid | exten => s,1,Answer ? |
| 11:04.22 | Dovid | so the next line put in |
| 11:04.33 | Dovid | exten => s,2,SIPAddHeader(Alert-Info: Variable) |
| 11:05.09 | Modcuts | will take a look |
| 11:05.18 | Modcuts | cheers |
| 11:05.26 | DrCron | ick, i guess if i want to do call fowarding it has to go through my asterisk server |
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| 11:05.58 | dlynes_laptop | I'm just curious what the best way would be to solve a problem I'm having |
| 11:06.07 | dlynes_laptop | The problem is that all incoming calls must ring on all phones |
| 11:06.09 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.153.71) |
| 11:06.26 | dlynes_laptop | However, I want it to ring on all phones that have available non-inuse lines |
| 11:06.34 | dlynes_laptop | Each phone has four incoming lines |
| 11:06.59 | oink | Anyone's using 7912G phones along with Asterisk ? |
| 11:07.10 | Dovid | dlynes_laptop: so u want to call only the phones that are not in use ? |
| 11:07.12 | dlynes_laptop | ChanIsAvail() reports the phones as being free, regardless of whether they're busy or not, and completely ignores the call-limit field in sip.conf |
| 11:07.45 | dlynes_laptop | Dovid: No, I want to call the phones that are in use, too (if they have available free lines, and they don't have do not disturb enabled) |
| 11:07.54 | dlynes_laptop | Dovid: each phone registers five accounts |
| 11:07.55 | benjk | ChanIsAvail() is evil, try to avoid it |
| 11:08.19 | dlynes_laptop | Dovid: So I want it to try account #1, if that isn't free, try account #2, and so on and so forth |
| 11:08.36 | Dovid | hmm |
| 11:08.56 | Dovid | i simple work around would be to set a variable per pone each time it gets a call |
| 11:09.00 | dlynes_laptop | Dovid: but say if user 1 called out, they might have called out on line 3 (on the phone), and wanted to conference in another user that they have on line 2 (on the phone) |
| 11:09.29 | dlynes_laptop | Dovid: Yeah...I was just curious if it would be simpler to solve this with AEL, AEL2, AGI, or writing my own module |
| 11:09.29 | Dovid | if the variable = 4 then u know that all lines are in use |
| 11:09.38 | pukkita | Dovid, no, I don't |
| 11:10.05 | Dovid | pukkita: what do u have ? |
| 11:10.19 | dlynes_laptop | benjk: yeah...it returns available, every time....it seems to be unwavering |
| 11:10.30 | shellshark | f |
| 11:10.44 | pukkita | that line is connected to a hw PBX. Users from regular hw extensions can call sip extensions from theirs |
| 11:10.45 | benjk | even if it "works" its still better to avoid it |
| 11:11.02 | dlynes_laptop | benjk: I guess you wouldn't know which of the four methods would be the easiest to solve that problem, eh? |
| 11:11.09 | pukkita | what I want is SIP users get a distinctive ring when they're called from the hw PBX |
| 11:11.14 | dlynes_laptop | benjk: I just want the solution that's the most maintainable and easiest to debug |
| 11:11.31 | dlynes_laptop | benjk: and the least buggy |
| 11:11.34 | benjk | I usually dial out and make the dial command to continue in the dialplan, then check DIALSTATUS |
| 11:11.36 | pukkita | I know how is done, but don't know how to set the _ALERT_INFO per each incoming channel |
| 11:11.44 | dlynes_laptop | benjk: that's not an option though |
| 11:12.02 | dlynes_laptop | benjk: using that method it'll ring on four of the five phones, if that channel is busy on the fifth phone |
| 11:12.02 | Dovid | pukkita: create diffrent channels for the diffrent lines |
| 11:12.06 | benjk | anything where you check in advance is subject to glare and race conditions |
| 11:12.14 | Dovid | so line A goes to context A etc |
| 11:12.48 | pukkita | I've done that already |
| 11:13.09 | pukkita | what statement should I put in those context to set the var always |
| 11:13.12 | dlynes_laptop | benjk: so i'm guessing agi doesn't give you any better call checking? or modules for that matter? |
| 11:13.32 | Dovid | like i said above |
| 11:13.36 | Dovid | sipaddheader |
| 11:13.51 | Dovid | pukkita: can you pb your configs please ? |
| 11:13.53 | Dovid | ~pb |
| 11:13.57 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 11:13.58 | benjk | not in this particular case, it gives you better performance and more syntactic freedom, but not any relief from race conditions |
| 11:13.58 | Dovid | !pb |
| 11:14.01 | dlynes_laptop | I'm guessing I'm going to have to go with a module app |
| 11:14.03 | *** join/#asterisk merbanan (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 11:14.12 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 11:14.49 | dlynes_laptop | race conditions? I can check the status of the phones reliably from a module though, right? |
| 11:15.19 | benjk | chanisavail and its underlying mechanism is subject to race conditions |
| 11:15.52 | benjk | and if you don't lock, then its subject to glare |
| 11:16.01 | dlynes_laptop | benjk: Yeah, but if I write my own application module 'Dial2', where it takes a regex for each channel parameter |
| 11:16.09 | *** part/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 11:16.15 | benjk | damned if you do, damned if you don't kind of scenario |
| 11:16.24 | dlynes_laptop | benjk: Then, theoretically I have access to the underliying data structure for the channel information, no? |
| 11:16.48 | benjk | not really, because ultimately asterisk maintains the channel structures |
| 11:17.15 | dlynes_laptop | And the channel structure isn't reliable? |
| 11:17.43 | benjk | there are plenty of locking issues with the internal storage in asterisk |
| 11:18.00 | dlynes_laptop | If I type 'show channels', I can reliably see what channels are in use, though |
| 11:18.11 | benjk | part of the problem is that everything is in a linked list |
| 11:18.29 | dlynes_laptop | benjk: And asterisk doesn't synchronize access to the data stores? |
| 11:18.44 | benjk | most of the show foobar (list all list elements) commands in the CLI will make everything else stop |
| 11:19.14 | dlynes_laptop | Ah, and that introduces call quality issues too, I would imagine |
| 11:19.21 | dlynes_laptop | but actually |
| 11:19.27 | dlynes_laptop | that shouldn't make everything else stop |
| 11:19.31 | benjk | if you ever use the CLI over a slow modem dialup link, you'll find out that all calls stop (no audio) while your CLI command is printing the list |
| 11:19.31 | dlynes_laptop | You're just reading, not writing |
| 11:19.36 | pukkita | http://pastebin.ca/269205 |
| 11:19.47 | dlynes_laptop | You shouldn't need to synchronize reads |
| 11:19.49 | dlynes_laptop | Only writes |
| 11:20.13 | benjk | well, asterisk uses what I call Rocky Mountain locking |
| 11:20.14 | pukkita | DrCron, can you have a look at that pb? |
| 11:20.21 | dlynes_laptop | ? |
| 11:21.00 | benjk | there are certainly many places where you could lock more fine grained but the reality is that they err on the other side of the spectrum and lock bigger scopes and often they lock when they don't even need to |
| 11:21.47 | benjk | anyway, the potential for glare (or race conditions) is there with chanisavail |
| 11:22.30 | dlynes_laptop | yeah...chanisavail is already out of the picture |
| 11:22.40 | dlynes_laptop | As soon as I tried it, I realized it was useless |
| 11:23.04 | dlynes_laptop | it doesn't even return an appropriate status |
| 11:23.20 | dlynes_laptop | and sippeer:status is just as useless for my application |
| 11:23.28 | dlynes_laptop | but at least sippeer:status is reliable |
| 11:23.34 | benjk | see, many of us have stopped trying to do smart things in the dialplan (agi or otherwise) because we found that we should first fix such obstacles in the core |
| 11:23.51 | benjk | that's why there is FS and OPO |
| 11:23.58 | dlynes_laptop | Yeah, but regardless, I need to do this today |
| 11:24.07 | benjk | good luck with that :) |
| 11:24.23 | dlynes_laptop | Not three months from now, or whenever the core gets fixed elsewhere :0 |
| 11:25.00 | benjk | FS might already be there, in this particular aspect, go and ask them |
| 11:28.53 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 11:30.18 | Guest77166 | hello guys |
| 11:31.05 | Guest77166 | I am installing asterisk 1.4 |
| 11:31.09 | Guest77166 | i got an eror |
| 11:31.21 | Guest77166 | checking for ptlib-config... no |
| 11:31.21 | Guest77166 | Cannot find ptlib-config - please install and try again |
| 11:31.21 | Guest77166 | [root@localhost asterisk-1.4.0-beta3]# ] |
| 11:31.22 | Guest77166 | checking for ptlib-config... no |
| 11:31.22 | Guest77166 | Cannot find ptlib-config - please install and try again |
| 11:31.22 | Guest77166 | [root@localhost asterisk-1.4.0-beta3]# ] |
| 11:31.44 | dlynes_laptop | Guest77166: do exactly what it says |
| 11:31.47 | dlynes_laptop | Guest77166: install ptlib |
| 11:32.14 | Guest77166 | how can I installed ptlib |
| 11:32.23 | *** join/#asterisk xnon (n=xnon@200.8.85.221) |
| 11:32.39 | DrCron | is there a short list of npan numbers that are dangerous to dial? (crazy fees and the like) |
| 11:33.04 | Guest77166 | from red hat cd? |
| 11:34.18 | *** join/#asterisk UnderMine (n=paddy@host81-149-176-190.in-addr.btopenworld.com) |
| 11:34.29 | dlynes_laptop | Guest77166: I have no idea...I don't use redhat |
| 11:34.35 | dlynes_laptop | Guest77166: Try asking on #redhat, or #fedora |
| 11:34.46 | DrCron | pukkita, sorry for the delay |
| 11:35.03 | dlynes_laptop | DrCron: npan? As in North American Numbering Plan Authority (NANPA)? |
| 11:35.26 | DrCron | dlynes_laptop, yhea, which area codes arent safe |
| 11:35.40 | dlynes_laptop | DrCron: well, it woudl all depend on what you call 'not safe' |
| 11:35.47 | dlynes_laptop | DrCron: they're all 'safe' afaik |
| 11:35.48 | DrCron | i guess that would be npa numbers |
| 11:36.03 | benjk | 900 numbers are not safe |
| 11:36.10 | DrCron | exactly |
| 11:36.11 | benjk | not safe for your finances |
| 11:36.15 | *** join/#asterisk beyond (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:36.19 | dlynes_laptop | Ah...you mean that kinda crap |
| 11:36.23 | DrCron | yup |
| 11:36.31 | dlynes_laptop | 1-976, 1-876, 1-900 |
| 11:36.38 | dlynes_laptop | Those are all pay per use services |
| 11:36.47 | dlynes_laptop | Usually for party and sex lines |
| 11:36.57 | benjk | DrCron, for that I have PREMIUM_RATE_BARRED=YES in my dialplan |
| 11:37.14 | DrCron | pukkita, change the s to _. |
| 11:37.21 | dlynes_laptop | And some NXX codes within certain Carribbean NPAs, too |
| 11:37.33 | benjk | those aren't premium rate though |
| 11:37.42 | benjk | they are to be rated as international |
| 11:38.14 | dlynes_laptop | benjk: so the sex lines in carribbean countries just charge you regular carribbean rates? |
| 11:38.19 | *** part/#asterisk peterme2005 (n=petere@browse.net-serv.co.uk) |
| 11:39.01 | benjk | I maintain a db entry for each extension which has flags for tollfree, emergency, local, mobile, national, premium-rate, international and sattelite |
| 11:39.30 | benjk | so for every one I can control what they can call |
| 11:39.34 | *** join/#asterisk nvictor (n=nvito@avenou.cafe.tg) |
| 11:40.14 | benjk | if you barr them from calling the caribbean in the first place, then you don't have to worry about what kind of numbers there are more or less expensive |
| 11:40.27 | DrCron | admitedly i just want to block premium rate and unregulated rate area codes |
| 11:40.30 | dlynes_laptop | I just block all 1-900, 1-876, and 1-976 calls |
| 11:40.35 | pukkita | thx DrCron! I managed using another way (ExecIf using ${MACRO_CONTEXT}) but using that looks cleaner |
| 11:40.51 | pukkita | so _. mateches always? |
| 11:40.52 | benjk | unless you have a likely need for your users to call the carribean, it is unlikely they have a justified reason to call the carribean |
| 11:41.01 | dlynes_laptop | All of our customers are business customers...I don't think they want their employees calling those numbers, anyways |
| 11:41.10 | DrCron | and if I do, i can always add in a specific pass rule |
| 11:41.20 | DrCron | well, this is for my home use |
| 11:41.42 | dlynes_laptop | DrCron: ah...to keep the teenaged son from calling the sex and party lines ;) |
| 11:41.42 | DrCron | i just dont want someone who gains user level access to be able to call premium numbers |
| 11:42.07 | benjk | check the NANP |
| 11:42.15 | DrCron | no teenage son, i am the son of the family, and heck if i want pr0n, i use cheggit |
| 11:42.16 | benjk | it tells you all about the area codes in use and reserved etc |
| 11:42.58 | DrCron | yhea, i hoped there was something, well, faster then flipping through the full nanp |
| 11:43.09 | pukkita | DrCron how do I keep the call going? it stops there |
| 11:43.10 | benjk | there are some sites |
| 11:43.18 | *** join/#asterisk beyond_ (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:43.21 | benjk | actually, there is some good overview on wikipedia |
| 11:44.03 | DrCron | switch the 1 to n in telsip |
| 11:44.03 | benjk | you probably want to whitelist all the US48, Hawaii, Alaska and Canada, and keep anything else out |
| 11:44.51 | *** part/#asterisk Guest77166 (i=Singh@124.62.150.38) |
| 11:47.03 | nvictor | hi |
| 11:47.10 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 11:47.31 | nvictor | I know it's a little off topic but, how do I get on jabber and chat with someone of which I've got a username?? |
| 11:47.35 | dlynes_laptop | DrCron: try this site: http://www.the-acr.com/codes/cntrycd.htm |
| 11:47.43 | nvictor | I'm under windows xp |
| 11:47.54 | *** join/#asterisk ]Airwolf[ (n=airwolf@89.205.158.36) |
| 11:48.05 | dlynes_laptop | DrCron: or these ones: http://www.1areacodescountrycodes.com/ |
| 11:48.28 | DrCron | i found it in the wiki |
| 11:48.52 | *** join/#asterisk stephane_ (n=stephane@gw.sortilege.net) |
| 11:49.19 | *** join/#asterisk yassinework (n=nnyel@xdsl-87-78-20-65.netcologne.de) |
| 11:49.41 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 11:51.01 | stephane_ | jour |
| 11:51.10 | *** join/#asterisk clive- (n=pirch@dsl-243-94-121.telkomadsl.co.za) |
| 11:51.13 | *** join/#asterisk Skarmeth (n=Skarmeth@201009008073.user.veloxzone.com.br) |
| 11:51.29 | nvictor | salut stephane |
| 11:51.36 | nvictor | aide moi stp |
| 11:52.55 | clive- | hi all, anyone got a few minutes spare to help me figure out why this tdm card with fxo and fxs interfaces doesnt like my configuration ? |
| 11:53.04 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 11:53.28 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:53.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:56.50 | jeremy_g | i want asterisk to load all modules |
| 11:56.53 | jeremy_g | what do i do? |
| 11:57.06 | RoyK | just enable autoload in modules.conf |
| 11:57.10 | jeremy_g | have only autoload=yes in modules.conf under module |
| 11:57.15 | jeremy_g | and remove all noload statements |
| 11:57.32 | FuriousGeorge | hi all |
| 11:58.31 | jeremy_g | hi FuriousGeorge |
| 11:58.42 | *** join/#asterisk evisu (i=hIRC@bzq-88-152-176-54.red.bezeqint.net) |
| 11:59.04 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:03.29 | *** join/#asterisk The_Ball (n=alex@203.27.183.41) |
| 12:04.28 | The_Ball | what does it mean when the TDM wildcard card prints out Freshmaker version: 71 \n 05 != ff Freshmaker failed register test wctdm: probe of 0000:00:0e.0 failed with error -5 |
| 12:04.36 | The_Ball | it lists all registers as ff |
| 12:04.55 | mattfletcher | is it possible to use SendText() to leave a message onscreen which persists? I want to show a particular state to phones at all times |
| 12:05.22 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.26) |
| 12:06.39 | jeremy_g | is there anyone here who uses Progress application |
| 12:06.57 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 12:09.10 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 12:16.32 | DrCron | can you nest brackets? |
| 12:16.57 | clive- | anyone knwo what time digium support wake up? |
| 12:16.57 | DrCron | for matching, ala 2[1[1234]3[14]4[126]] |
| 12:17.27 | benjk | no |
| 12:17.33 | DrCron | oops, forgot commas |
| 12:17.43 | clive- | hi benjk |
| 12:17.58 | benjk | hi |
| 12:18.56 | clive- | whats news from japan? |
| 12:19.19 | benjk | business as usual |
| 12:19.21 | clive- | we chatted ages ago, something about jitterbuffers or something if I recall |
| 12:19.55 | x86 | benjk: you got your voice back ;) |
| 12:20.16 | benjk | yeah, the banlist on the channel got wiped |
| 12:20.38 | x86 | sweet |
| 12:21.28 | clive- | anyone got a few minutes spare to help me figure out why this tdm card with fxo and fxs interfaces doesnt like my configuration ? |
| 12:22.40 | dlynes_laptop | clive-: it's always best to describe what the problem is you've got and provide a pastebin of the log and the config files in question |
| 12:22.44 | DrCron | http://scottstuff.net/blog/articles/2004/09/07/the-sound-of-tcp-screaming-in-pain |
| 12:22.51 | dlynes_laptop | clive-: then ask your question |
| 12:23.09 | *** join/#asterisk beyond (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 12:23.16 | *** join/#asterisk beyond_ (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 12:23.31 | *** join/#asterisk kashmish_ (n=kashmish@m1.ince.net) |
| 12:31.28 | *** join/#asterisk ambriento (n=ambrient@200.192.160.100) |
| 12:32.03 | clive- | dlynes, thanks, ust making the pastebin |
| 12:32.15 | *** part/#asterisk kashmish_ (n=kashmish@m1.ince.net) |
| 12:35.06 | clive- | http://pastebin.ca/269255 |
| 12:36.32 | mattfletcher | Can I use SendText (or anything else) to put a message on the phone's screen, and have it remain after the call that created it has ended? |
| 12:37.02 | *** join/#asterisk tparcina (n=tomo@wr-lama.iskon.hr) |
| 12:38.21 | x86 | mattfletcher: if you figure that out, let me know ;) |
| 12:42.30 | clive- | updated : http://pastebin.ca/269258 |
| 12:42.50 | *** join/#asterisk shellsha1k (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 12:44.29 | *** join/#asterisk xAD (i=doddo@host227-174-dynamic.56-82-r.retail.telecomitalia.it) |
| 12:45.25 | *** join/#asterisk cian_ (n=cian@cian.ws) |
| 12:45.38 | cian_ | \quit |
| 12:45.41 | *** join/#asterisk Kizmet (n=kizmet@ppp167-251-70.static.internode.on.net) |
| 12:45.50 | Kizmet | =O |
| 12:47.48 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 12:50.07 | *** join/#asterisk tparcina_ (n=tomo@33-226.dsl.iskon.hr) |
| 12:57.00 | DrCron | what do sip and iax URI's look like |
| 12:57.13 | DrCron | SIP:user@server? |
| 12:57.42 | *** join/#asterisk M_at (n=matt@dsl092-214-175.atl1.dsl.speakeasy.net) |
| 12:57.45 | yassinework | any suggestions for an asterisk book ? |
| 12:58.10 | DrCron | there is one out there |
| 12:58.12 | Kizmet | www.google.com/search?q=Asterisk+VoIP+Book |
| 12:58.41 | DrCron | and the safari online service has a few |
| 12:58.43 | dlynes_laptop | clive-: so what exactly is the problem? |
| 12:58.58 | dlynes_laptop | clive-: oh...nvm |
| 12:59.05 | *** join/#asterisk Ebola (n=Ebola@host86-134-167-28.range86-134.btcentralplus.com) |
| 12:59.10 | dlynes_laptop | clive-: after you modprobe wcfxo, you need to do a ztcfg -vvvvvvvvvvvvvvvvv |
| 12:59.36 | yassinework | Kizmet, i even know that there is somthing called amazon btw im looking for impressions on differnt book |
| 12:59.39 | dlynes_laptop | clive-: also wcfxo is for x100p cards, not whatever card you're trying to use |
| 13:00.03 | dlynes_laptop | clive-: the tdm400p card needs wctdm, not wcfxo |
| 13:04.55 | DrCron | can someone make an sip call to rszasz@saxon.dhs.org <me |
| 13:05.23 | Kizmet | How do we know it is you =O |
| 13:05.31 | DrCron | you dont |
| 13:05.36 | DrCron | but it is |
| 13:05.38 | M_at | How do you know it is you? |
| 13:06.15 | coppice | nobody else is |
| 13:06.39 | dwmw2 | I'd check my passport,but wenchlet has confiscated it last time I got back into the country :) |
| 13:06.46 | dwmw2 | not allowed it back till January |
| 13:07.00 | *** join/#asterisk asterisk_5432 (n=shady@125.209.112.31) |
| 13:07.04 | asterisk_5432 | hi |
| 13:07.04 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:07.17 | DrCron | so, um could someone ring that address? |
| 13:07.59 | *** join/#asterisk root (n=chatzill@220.225.228.177) |
| 13:08.11 | Kizmet | <PROTECTED> |
| 13:08.14 | coppice | if someone confiscated my passport that would make december a much more tranquil month |
| 13:09.55 | asterisk_baby | hello guys |
| 13:10.12 | asterisk_baby | i was having trouble compiling asterisk on fedora 4 |
| 13:11.12 | asterisk_baby | its a godaddy dedicated server |
| 13:11.42 | asterisk_baby | i followed the step by step instructions on asteriskguru.com for installing asterisk on fedora 4 |
| 13:11.59 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:12.01 | zacky | Kizmet, you need to show the trunk in the extension.comf ex. SIP/trunk/XXXX |
| 13:12.03 | coppice | the only dedication at godaddy is to pissing off hgih bandwidth users :-) |
| 13:12.34 | asterisk_baby | compilation went well but im not able to able run asterisk -rvvv.. not even simple asterisk or safe_asterisk |
| 13:12.49 | asterisk_baby | can you help me out with this problem please.. coppice |
| 13:12.53 | M_at | what error does it give? |
| 13:12.59 | dlynes_laptop | asterisk_baby: pastebin your log of 'asterisk -vvvvvvvvvvvvvvvvg' |
| 13:13.26 | asterisk_baby | bash: asterisk: command not found |
| 13:13.40 | dlynes_laptop | asterisk_baby: ummm....then type in the full path to your asterisk binary |
| 13:13.48 | dlynes_laptop | asterisk_baby: try '/usr/sbin/asterisk -vvvvvvvvvvvvvvvvg' |
| 13:14.17 | asterisk_baby | Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. |
| 13:14.29 | dlynes_laptop | asterisk_baby: i thought you said it wasn't running? |
| 13:14.39 | asterisk_baby | wowww.. amazing |
| 13:14.49 | dlynes_laptop | asterisk_baby: try /usr/sbin/asterisk -r |
| 13:14.57 | dlynes_laptop | asterisk_baby: see if you get an asterisk CLI |
| 13:15.00 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.12) |
| 13:15.04 | asterisk_baby | its working.. thanks dlynes.. thanks so much |
| 13:15.37 | M_at | dlynes_laptop: Sice you fixed that one so quick can you get me a dialtone on a Sangoma A200? ;o) |
| 13:15.40 | asterisk_baby | lol |
| 13:16.00 | dlynes_laptop | M_at: heh....you got logs of your dmesg and that kinda thing? |
| 13:17.03 | dlynes_laptop | M_at: also log of 'wanrouter status'? |
| 13:17.04 | M_at | Yeah - it's detected, ztcfg -vvv is happy, the card has power and provides power to the phone, zaptel is configured, there's just no dialtone or audio - I can make it ring but no audio is heard. Handset checks out on a standalone line. |
| 13:18.20 | M_at | I'm just trashing the config and building a minimal one manually |
| 13:19.40 | *** part/#asterisk darqchild (n=e@206-248-138-220.dsl.teksavvy.com) |
| 13:19.40 | DrCron | hmm, incoming sip calls arent even registering.. kizmet just tried, can anyone else give it a shot? rszasz@saxon.dhs.org |
| 13:19.40 | dlynes_laptop | M_at: Just humor me and pastebin your dmesg |
| 13:19.45 | dlynes_laptop | M_at: i suspect you might have a common issue |
| 13:20.01 | *** join/#asterisk dasenjo_ (n=dasenjo@208.195.215.12) |
| 13:20.18 | M_at | I hope se - 1 minute :) |
| 13:20.54 | dlynes_laptop | M_at: when someone on calls in to the asterisk box with the sangoma card, they don't hear any audio from anything, right? |
| 13:21.03 | dlynes_laptop | M_at: not even background, playback, voicemail, ...? |
| 13:21.09 | M_at | It's an FXS |
| 13:21.19 | M_at | So when I pick up there's no dialtone |
| 13:21.20 | dlynes_laptop | M_at: probably still the same issue |
| 13:21.31 | dlynes_laptop | M_at: no audio getting generated |
| 13:21.37 | M_at | power so local audio echo but nothing * generated |
| 13:23.08 | *** part/#asterisk zacky (n=chatzill@220.225.228.177) |
| 13:23.33 | M_at | It's a biggie http://pastebin.ca/269287 |
| 13:25.05 | dlynes_laptop | M_at: this is an a200u/a200d right? |
| 13:25.33 | M_at | Yup - not the d though |
| 13:25.34 | dlynes_laptop | M_at: or is an a101u, or something? |
| 13:25.48 | dlynes_laptop | M_at: ok, so why do you have it configured as an a101u? |
| 13:25.51 | M_at | the A101 is a Pri interface also in there but that's not connected to anything yet |
| 13:25.56 | dlynes_laptop | ah, ok |
| 13:26.03 | M_at | you're looking at wanpipe1 |
| 13:27.20 | dlynes_laptop | M_at: yeah...now you're using a backplane on it? i.e. you've got daughterboards attached to it? |
| 13:27.38 | M_at | No |
| 13:27.51 | dlynes_laptop | That's how you've got it configured |
| 13:27.54 | dlynes_laptop | module 2 and module 3 |
| 13:28.11 | dlynes_laptop | module 2 is on the first card, module 3 is on a daughterboard connected to the backplane |
| 13:28.16 | M_at | it's a 4 port card, 2 module slots, one populated |
| 13:28.24 | M_at | No backplane |
| 13:28.25 | dlynes_laptop | Yeah, 2 modules, not 3 |
| 13:28.34 | M_at | yeah but they're 2 port modules |
| 13:28.39 | M_at | 2xFXS |
| 13:28.44 | M_at | on one module |
| 13:28.45 | dlynes_laptop | # |
| 13:28.45 | dlynes_laptop | wanpipe1: Module 2: Installed -- Auto FXS! |
| 13:28.45 | dlynes_laptop | # |
| 13:28.45 | dlynes_laptop | wanpipe1: Module 3: Installed -- Auto FXS! |
| 13:28.45 | dlynes_laptop | # |
| 13:29.04 | M_at | Yup - they're packaged together on a single card |
| 13:29.09 | dlynes_laptop | I know that |
| 13:29.12 | dlynes_laptop | I've got several |
| 13:29.26 | dlynes_laptop | All of mine are a200d's |
| 13:29.30 | dlynes_laptop | I also have two a101u's |
| 13:29.50 | M_at | Yup - I've never had any problem with the A100 series |
| 13:29.52 | *** join/#asterisk shellshark (n=x86@74.135.64.209) |
| 13:30.00 | M_at | Got a 102 running faultlessly in the UK |
| 13:30.27 | dlynes_laptop | Maybe because you've got an a101u in that box, the modules are loading kinda weird |
| 13:30.38 | dlynes_laptop | Can you try changing the slot positions for the a101u and the a200u? |
| 13:30.58 | *** join/#asterisk heh_v_water (n=heh_v_wa@70-57-200-16.hlna.qwest.net) |
| 13:31.05 | dlynes_laptop | Then maybe the module numbering might be a little more sane |
| 13:31.10 | M_at | I'll remove the 101 |
| 13:31.19 | dlynes_laptop | ok |
| 13:31.23 | dlynes_laptop | and then pastebin your new dmesg |
| 13:38.43 | *** join/#asterisk xnon_ (i=xnon@200.8.85.221) |
| 13:40.19 | M_at | Here we go: http://pastebin.ca/269294 |
| 13:41.43 | dlynes_laptop | M_at: yeah....still getting detected as module 2 and 3 |
| 13:41.51 | M_at | THey are tho ren't they? |
| 13:41.57 | dlynes_laptop | no |
| 13:42.02 | dlynes_laptop | should be module 1 and 2, not 2 and 3 |
| 13:42.06 | M_at | 0 & 1 are empty, 2 & 3 are one FXS card, 4 & 5 would be on a daughterboard |
| 13:42.32 | dlynes_laptop | M_at: also, it shouldn't even be echoing stuff like that |
| 13:43.11 | dlynes_laptop | Normally you would get something like the following: |
| 13:43.12 | M_at | ecohing how? |
| 13:43.15 | dlynes_laptop | yeah...nvm |
| 13:43.19 | dlynes_laptop | i guess that is correct |
| 13:43.22 | dlynes_laptop | on mine I have: |
| 13:43.24 | dlynes_laptop | wanpipe1: Module 0: Installed -- Auto FXO (FCC mode)! |
| 13:43.25 | dlynes_laptop | wanpipe1: Module 1: Installed -- Auto FXO (FCC mode)! |
| 13:43.25 | dlynes_laptop | wanpipe1: Module 2: Installed -- Auto FXO (FCC mode)! |
| 13:43.25 | dlynes_laptop | wanpipe1: Module 3: Installed -- Auto FXO (FCC mode)! |
| 13:44.00 | dlynes_laptop | M_at: and you don't have echo canceller on there, right? |
| 13:44.08 | M_at | Not as far as I know |
| 13:44.24 | M_at | didn't specify one |
| 13:44.43 | dlynes_laptop | M_at: also, you have an error in your wanrouter setup |
| 13:44.55 | dlynes_laptop | M_at: you haven't specified whether you want to use ulaw or alaw |
| 13:45.54 | dlynes_laptop | M_at: it assumes ulaw for you: zaptel: Span WRTDM/0 didn't specify default law. Assuming mulaw, please fix driver! |
| 13:45.54 | M_at | I have in wancfg but it doesn't seem to be taking |
| 13:46.01 | M_at | Shall blat out the configs |
| 13:46.09 | dlynes_laptop | M_at: did you use the sangoma configuration utility? |
| 13:46.15 | dlynes_laptop | M_at: or did you edit the file manually? |
| 13:46.19 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:46.22 | M_at | Sangoma all the way |
| 13:46.34 | coppice | oh, that's bad :-) |
| 13:46.47 | M_at | coppice: How? |
| 13:46.49 | coppice | that sangoma utlity really sucks |
| 13:46.51 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 13:47.06 | coppice | I only ever seem to get the config right by hand |
| 13:48.14 | coppice | the sangoma stuff is great hardware let down by a crappy install experience |
| 13:48.17 | dlynes_laptop | coppice: works just fine for me |
| 13:48.25 | dlynes_laptop | coppice: but yeah, the install process is horrible |
| 13:48.26 | M_at | And me with a 102 |
| 13:48.37 | ManxPower | coppice: I agree that the Sangoma stuff can be odd to install. |
| 13:48.46 | *** join/#asterisk bitwise (n=bitwise@ipa28.12.tellas.gr) |
| 13:51.49 | pukkita | bye and thanks to everyone |
| 13:51.52 | *** part/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 13:52.07 | M_at | dlyne: Dialtone! |
| 13:53.00 | M_at | I think my wanpipe config may have got a bit screwed |
| 13:53.56 | The_Ball | i have the tdm400 wildcard and compiling zaptel under gentoo i have the option of enabling the watchdog or not. Should i use the watchdog? |
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| 13:58.14 | dlynes_laptop | M_at: cool |
| 13:59.04 | *** join/#asterisk DerPraktikant (n=Tgu@pD95DDBAA.dip.t-dialin.net) |
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| 14:00.01 | DerPraktikant | hi , got a problem with my dialplan , 1 got 2 softphones via SIP which can phone each other , and an zaphfc card in NT mode |
| 14:00.18 | DerPraktikant | at the zap card there is an isdn phone |
| 14:00.43 | DerPraktikant | i can call from the isdn phone to the sip softphones |
| 14:01.02 | DerPraktikant | but not from the sip to the isdn phone |
| 14:01.18 | DerPraktikant | maybe my exten is not true |
| 14:01.40 | DerPraktikant | exten => 4,1,Dial(Zap/1/4028474,60) |
| 14:01.55 | DerPraktikant | did i make something wrong with the syntax? |
| 14:02.12 | RoyK | bristuff? |
| 14:02.15 | DerPraktikant | yes |
| 14:02.31 | DerPraktikant | newest vers |
| 14:02.37 | RoyK | sounds right |
| 14:02.42 | RoyK | or loos |
| 14:02.43 | RoyK | looks |
| 14:03.11 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 14:03.15 | DerPraktikant | i dont understand it , because when i call the softphone the number 4028474 comes |
| 14:03.23 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.12) |
| 14:03.29 | DerPraktikant | so i thought this must be added into the extension |
| 14:03.48 | DerPraktikant | but it dont rings :( |
| 14:04.08 | DerPraktikant | does somebody know why? |
| 14:05.04 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.26) |
| 14:05.06 | DerPraktikant | or maybe u got an reverence syntax which i can modify? |
| 14:06.01 | DerPraktikant | :-( |
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| 14:07.59 | *** part/#asterisk andresmujica (n=andresmu@201.244.244.26) |
| 14:08.32 | DerPraktikant | if i call from the isdn to sip this msg comes : |
| 14:08.33 | DerPraktikant | <PROTECTED> |
| 14:08.33 | DerPraktikant | <PROTECTED> |
| 14:08.33 | DerPraktikant | <PROTECTED> |
| 14:08.33 | DerPraktikant | <PROTECTED> |
| 14:08.33 | DerPraktikant | <PROTECTED> |
| 14:08.35 | DerPraktikant | <PROTECTED> |
| 14:08.37 | DerPraktikant | <PROTECTED> |
| 14:09.06 | DerPraktikant | but from sip to isdn there comes nothing , not even an error |
| 14:13.07 | DerPraktikant | cant nobody help me plz? :/ |
| 14:13.17 | *** join/#asterisk ming_zym (n=zym@124.254.55.160) |
| 14:14.38 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:15.44 | shellshark | DerPraktikant: did you set your verbose level really high? |
| 14:15.56 | DerPraktikant | -vvvc |
| 14:16.03 | shellshark | set it higher |
| 14:16.06 | shellshark | set verbose 99999999999999 |
| 14:16.12 | shellshark | from the CLI |
| 14:16.19 | DerPraktikant | vvvvvc? |
| 14:16.40 | shellshark | that's not what i said |
| 14:16.45 | shellshark | "set verbose 99999999999" |
| 14:16.50 | shellshark | do that from the CLI |
| 14:17.47 | EmleyMoor | My Digium supplier are sending me a new FXO module |
| 14:17.53 | EmleyMoor | :-) |
| 14:17.59 | DerPraktikant | it gives no other msg |
| 14:18.06 | EmleyMoor | I will finally have it working by Christmas |
| 14:18.09 | *** join/#asterisk merbanan (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 14:18.09 | DerPraktikant | i set it to the verbose lvl u said |
| 14:18.25 | shellshark | turn debug on? |
| 14:19.05 | DerPraktikant | yes by the initialtion |
| 14:19.56 | shellshark | "sip debug" |
| 14:22.06 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:22.08 | *** join/#asterisk ming_zy1 (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
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| 14:23.35 | DerPraktikant | ok i cant find something |
| 14:24.49 | *** join/#asterisk aadilismail (n=aadilism@2-237-154-202.wol.net.pk) |
| 14:25.02 | DerPraktikant | maybe i got some error in my syntax exten => 4,1,Dial(Zap/1/4028474,60) |
| 14:25.02 | aadilismail | hi |
| 14:25.41 | shellshark | DerPraktikant: why not just Dial(Zap/4028474|60) ? |
| 14:25.53 | aadilismail | where and how to check a specific call detail record..??"? |
| 14:26.09 | shellshark | aadilismail: depends on the CDR back-end you're using |
| 14:26.19 | DerPraktikant | i will try it |
| 14:26.44 | shellshark | aadilismail: by default, /var/log/asterisk/cdr/* |
| 14:26.44 | shellshark | aadilismail: of course, it's better to set it up to use MySQL |
| 14:27.45 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 14:27.51 | DerPraktikant | call failed : not found |
| 14:28.03 | shellshark | thats... eh... descriptive :p |
| 14:29.15 | DerPraktikant | with the 1 in my expression i defined the group where the zaphfc is in |
| 14:29.37 | DerPraktikant | 2-1 is the channel it used to take |
| 14:30.06 | DerPraktikant | the problem is that the isdn telefon has no synonym like a softphone ;) |
| 14:31.07 | *** join/#asterisk alerios (n=alerios@190.24.97.151) |
| 14:31.13 | DerPraktikant | technik is magic , someway it functions but randomly logical ^^ |
| 14:31.56 | DerPraktikant | did u read the msg asterisk gave me when i call from the isdn phone? |
| 14:32.15 | DerPraktikant | Executing Dial("Zap/2-1", "SIP/platz2") in new stack |
| 14:32.51 | DerPraktikant | the kontext is Zap/2-1 , but if i want to call this with the sip phone it wont function |
| 14:33.29 | DerPraktikant | sry 4 my bad english |
| 14:37.21 | DerPraktikant | well maybe somebody can help me with this problem: Dec 6 15:37:12 NOTICE[5210]: chan_sip.c:5402 sip_reg_timeout: -- Registration for 'platz2@platz2.megafunk.local' timed out, trying again (Attempt #5) |
| 14:37.49 | puzzled | RoyK: in libpri ir says int pri_sr_set_redirecting(struct pri_sr *sr, char *num, int plan, int pres, int reason). Does it make sense to decalre redirect_reason as an int too? |
| 14:39.04 | DerPraktikant | i dont know why it comes , the soft phones tell me the status " registered" but the asterisk gives me all 10 s this error |
| 14:39.40 | DerPraktikant | maybe because i didnt register to an extern sip provider ? i got them registered to my asterisk server |
| 14:39.54 | *** join/#asterisk hoobastoob2 (n=ckwall@63.149.122.93) |
| 14:40.10 | aadilismail | where "var" exist ... in which folder or dir??? |
| 14:40.13 | RoyK | puzzled: where in what file? my patch may be broken |
| 14:41.00 | DerPraktikant | register => platz2:52@platz2.megafunk.local is the string , did i something wrong? qualify is not enabled.. |
| 14:41.12 | puzzled | RoyK: doh, I did not paste the top part of the patch. /me slaps /me with clueby4 |
| 14:43.58 | DerPraktikant | the folder is in root |
| 14:44.24 | DerPraktikant | get into ur terminal and type cd /var |
| 14:44.53 | *** join/#asterisk XR6 (n=reptiliu@CPE-124-177-128-222.qld.bigpond.net.au) |
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| 14:45.51 | XR6 | anyone knows if asterisk can dial out, inform me there is one parked call for me and if i accept it will bridge the parked call to me ? |
| 14:46.08 | XR6 | I've been trying to fingure this out for the last hour |
| 14:46.23 | darkskiez | could script it i suppose |
| 14:46.44 | XR6 | the only way i seen asterisk to dial out is via a .call file |
| 14:46.53 | puzzled | RoyK: what does the alaw PLC fix exactly do? |
| 14:46.58 | darkskiez | yeh, an external script perhaps |
| 14:47.05 | DerPraktikant | yes u should make a script which checks ur parket calls and dial to u if necessary |
| 14:47.08 | XR6 | but it looks like that won't bridge the calls |
| 14:47.12 | darkskiez | invoked via your park action, using originate to inform you. |
| 14:47.41 | aadilismail | if i want to check a single number call detail record then ... its like ... / something??? |
| 14:47.49 | RoyK | puzzled: using alaw with pri means no transcoding, which means no plc |
| 14:48.11 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:48.31 | shellshark | RoyK: i thought pri used ulaw? |
| 14:48.38 | puzzled | or ulaw |
| 14:48.39 | XR6 | so how can triger the script if there is a parked call? |
| 14:48.42 | DerPraktikant | pri can use both |
| 14:48.50 | DerPraktikant | ulaw and alaw |
| 14:48.51 | shellshark | ah |
| 14:49.09 | puzzled | RoyK: why would you want plc with pri which is tdm technology? |
| 14:49.21 | hoobastoob2 | I am having in issue where only one of the 3 members in a queue are ringing. The queue is set up with a ringall strategy. Any ideas? Here is the show queue and the queues.conf entries. http://pastebin.ca/269355 |
| 14:49.32 | RoyK | puzzled: the receiving channel is the one doing dejittering and plc, not the sending channel |
| 14:49.37 | RoyK | don't ask why |
| 14:49.45 | RoyK | erm |
| 14:49.48 | RoyK | anyway |
| 14:49.53 | hoobastoob2 | members are added using add queue member, except for the one member who is static in the queues.conf |
| 14:50.05 | RoyK | receiving a SIP call and terminating the call on Zap, the dejittering and PLC is in Zap |
| 14:50.10 | RoyK | because slav wrote it that way |
| 14:50.11 | puzzled | RoyK: ah right. so the patch makes the jb work between zap & sip channels? |
| 14:50.19 | RoyK | yes |
| 14:50.32 | RoyK | the design is generic |
| 14:50.42 | RoyK | but there's only a sip/zap implementation (in asterisk) |
| 14:50.48 | RoyK | there is in openpbx, though |
| 14:50.59 | puzzled | RoyK: I understand now. thanks. will add the patch to my RPM |
| 14:52.04 | DerPraktikant | RoyK: i only say bristuff |
| 14:52.36 | DerPraktikant | i would breffer anyone of u to use a RedHat distri and A@H , its the simplest way |
| 14:52.59 | XR6 | with .acall how does one invoke an ivr style to the caller, as in Playback(yougotacall-press1-accept-press2-deny) then bridge with parked call, at the moment it dials out but can't bridge the calls |
| 14:53.38 | DerPraktikant | i must use suse 10,1 and it sucks hard |
| 14:54.03 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 14:55.25 | DerPraktikant | what experience u got with the speex codec? in teamspeak i know it at very good , but what u say about it in VoIP ? |
| 14:57.24 | DerPraktikant | well cu |
| 14:57.57 | vader-- | Do any of you guys have a portal system or collaborative system or wiki you use for your IT department to store documents, FAQs, KB, etc? |
| 14:58.20 | darkskiez | vader--: we use tikiwiki |
| 14:58.36 | mattfletcher | Can I use SendText (or anything else) to put a message on the phone's screen, and have it remain after the call that created it has ended? |
| 14:58.37 | darkskiez | I actually store our asterisk config in it, and have scripts that convert that to configuration files |
| 14:58.47 | *** join/#asterisk groogs___ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
| 14:59.22 | vader-- | dark is your site open to the public? |
| 14:59.59 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 15:00.07 | vader-- | dark why tikiwiki and not metawiki? |
| 15:00.14 | vader-- | or wikimedia i mean |
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| 15:00.31 | darkskiez | wikimedia is tooo big and complicated |
| 15:01.00 | darkskiez | vader: no, its vpn/intranet only |
| 15:01.01 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 15:01.20 | vader-- | i setup mediawiki |
| 15:01.38 | vader-- | i don't like it too much |
| 15:02.07 | vader-- | dark any way i could see a few screenshots to get an idea if thats something i would like to implement? |
| 15:02.27 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:02.41 | darkskiez | google |
| 15:02.55 | vader-- | ya im looking more for how other IT departments are using it |
| 15:03.01 | vader-- | i found the tikiwiki site |
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| 15:05.40 | hoobastoob2 | is it possible to make a call go to queue and simultaneously make an extension that is not a member of a queue ring? All at the same time? |
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| 15:17.30 | asd23 | I'm having trouble with dtmfmode. Seems that that in order to get outgoing calls to send dtmf signals to other pbx's I need to set dtmfmode=inband, however then my asterisk server won't recognize incoming calls' dtmf signals in the IVR. Anyone experience this? |
| 15:18.33 | shellshark | what codec? |
| 15:18.37 | asd23 | ulaw |
| 15:19.23 | shellshark | inbound calls are using ulaw also? |
| 15:19.29 | *** join/#asterisk danbrwn (n=danny@216.77.58.40) |
| 15:20.20 | asd23 | Yes, it's all ulaw. I'm using a voip service and it's setup in my sip.conf file in it's own context. If that context's dtmfmode is set to auto or rfc then it works fine, but my outgoing calls suffer. |
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| 15:20.49 | *** part/#asterisk zozo75 (n=cyrilz@dra38-1-82-230-120-110.fbx.proxad.net) |
| 15:20.50 | asd23 | Its a flip flop, can't get incoming and outgoing dtmf to work. |
| 15:23.10 | asd23 | I tried setting dtmfmode=rfc2833 under the general context (for incoming calls) but it gets overridden when I set it to inband under another context. |
| 15:23.30 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 15:23.57 | shellshark | odd |
| 15:24.38 | asd23 | I looked it up and it seems that you can't set the dtmfmode under the general context. But I tried it anyways. |
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| 15:28.23 | asd23 | anyone here knowledable in sip? |
| 15:28.42 | *** join/#asterisk ming_zym (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
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| 15:29.20 | ManxPower | you want rfrc2833 Check the dtmf tone length option to make outgoing DTMF work |
| 15:29.21 | asd23 | anyone here familiar with setting dtmf in asterisk? |
| 15:29.28 | ManxPower | assuming your PSTN connection is via Zap |
| 15:29.44 | asd23 | Everything I'm sending is over a voip connection. |
| 15:29.53 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.198.Dial1.SanJose1.Level3.net) |
| 15:30.04 | ManxPower | asd23: then you must work with your carrier. |
| 15:30.34 | ManxPower | of course you could just use different sip.conf [sections[ for incoming .vs. outgoing like you are supposed to anyway. |
| 15:31.03 | asd23 | ah, ok I was thinking of doing that? How do you setup an incoming context? |
| 15:31.25 | asd23 | How do you tell asterisk to route incoming calls to a specific context in sip? |
| 15:31.34 | *** part/#asterisk hoobastoob2 (n=ckwall@63.149.122.93) |
| 15:34.09 | benjk | context=foobar |
| 15:34.23 | danbrwn | have extension conf setup and sip.conf setup, using sjphone and aastra 480i. how to make both go through asterisk now? |
| 15:35.18 | asd23 | My provider is using his asterisk server to send calls my server over sip. |
| 15:35.44 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 15:37.25 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 15:37.56 | asd23 | benjk: context=foobar send the call to a context in extensions.conf. I need a way to be able to get incoming calls routed to a context in sip.conf. Once in that context, I can then do a context=foobar. |
| 15:39.28 | benjk | sip.conf doesn't handle calls |
| 15:40.07 | ManxPower | asd23: they are not called contexts in sip.conf they are called sections or devices. |
| 15:40.20 | ManxPower | and the [whatever] will match an incoming call if the auth info matches |
| 15:40.32 | benjk | it doesn't matter what you call them, the fact is that you can't send calls into sip.conf |
| 15:41.01 | benjk | you might as well try to send calls into README |
| 15:41.08 | danbrwn | trying to get aastra sip phone and sjphone sip phone working with asterisk server, setup both in sip.conf and extensions.conf. Now what? |
| 15:41.09 | Supaplex | how do you setup more then one isolated registry in sip.conf? I'd like to start different registry each in a different context |
| 15:41.31 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.26) |
| 15:41.40 | ManxPower | Supaplex: register => only notifies the far side what your ip address is. It does nothing else. |
| 15:41.45 | asd23 | well, then how does an incoming sip connection know where to route the call in extensions.conf. It knows by whats setup in a sip section, no? |
| 15:41.56 | ManxPower | danbrwn: start calling |
| 15:42.15 | ManxPower | asd23: the incoming call will have a destination phone number |
| 15:42.25 | benjk | each section for incoming calls should have its own "context=foo" statement |
| 15:42.27 | asd23 | In other words, what tells the remote party (voip service in my case) to send it |
| 15:42.56 | asd23 | In other words, what tells the remote party (voip service in my case) to send it's data to a specific section in my sip.conf? |
| 15:42.56 | ManxPower | asd23: the /extension on your register => can request the far end send the call to a specific extension but the carrier must support it. |
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| 15:43.12 | benjk | then foo in extensions.conf should deal with those calls in the way you want them to be handled |
| 15:43.27 | Supaplex | then why is it when I receive calls from them they go into the context [general] decides? Is there someway to distinguish which provider I've setup and have it start in a specified context? or is it all or nothing? |
| 15:43.29 | ManxPower | asd23: the username / password tells asterisk what sip.conf section to match. Asterisk has no other information about the call |
| 15:43.51 | ManxPower | Supaplex: if the call is going the the [general] context then it is NOT matching a sip.conf section |
| 15:44.16 | Nugget | all your calls are belong to general. |
| 15:44.35 | Supaplex | hehe |
| 15:44.53 | danbrwn | ManxPower: how does the aastra and sjphone know to call into the server when I dial the extension on the phone? |
| 15:45.20 | ManxPower | danbrwn: you configure them to use asterisk as their server. I can't help you with that as it is not an asterisk issue |
| 15:45.32 | Supaplex | so I can put registry in a different section so it'll default to a specific context? I tried that, and it never registered. |
| 15:45.44 | ManxPower | Supaplex: no you cannot. |
| 15:46.09 | ManxPower | Supaplex: Do you know what a register => line does? (no it is not REGISTRY) |
| 15:46.16 | *** join/#asterisk jeebusmobile (n=jeebusmo@cpe-75-80-231-237.dc.res.rr.com) |
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| 15:46.51 | asd23 | Is there a way to get incoming calls to go to another section other than the general section? |
| 15:47.01 | ManxPower | asd23: yes, they will do that by default. |
| 15:47.32 | ManxPower | asd23: IF the incoming call's auth info matches a sip.conf section then the stuff in that [section] will override the stuff in [general] |
| 15:47.37 | asd23 | I ask because I can't set the dtmfmode in the general section. It must be set in another section. |
| 15:47.41 | Supaplex | ManxPower: so register => has nothing to do with sip show registry? |
| 15:48.05 | ManxPower | APPARENTLY either asd23's calls and Supaplex's calls are not coming in with username / secret that matches a sip.conf section. |
| 15:48.25 | ManxPower | Supaplex: It might, but why do we care? |
| 15:48.40 | Supaplex | fantastic |
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| 15:49.21 | ManxPower | This is why I put context=INVALID in sip.conf [general]. This makes sure that if a call comes in that does not match a sip.conf [section] the call is rejected because I do not have a [INVALID] context in extensions.conf |
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| 15:50.35 | ManxPower | This has saved me hundreds of hours in trying to figure out sip issues. |
| 15:50.49 | Supaplex | so far I have it ending up in a recording for tracing/debugging information, but it hasn't done much yet. http://rafb.net/paste/results/A3IkmB85.html |
| 15:51.11 | *** join/#asterisk henrique_ (n=henrique@201-43-227-125.dsl.telesp.net.br) |
| 15:51.11 | asd23 | here is what I have |
| 15:51.11 | asd23 | [authentication] |
| 15:51.11 | asd23 | auth = value1:value2@ip address |
| 15:51.15 | mattfletcher | Can anyone explain to me how I should use the 'd' option for the Dial() command. I'm hoping to use it to allow callers to access a hidden menu of options |
| 15:51.18 | converx | ? |
| 15:51.31 | ManxPower | asd23: wrong |
| 15:51.44 | asd23 | You mean to tell me that I need a sip section named after something in my auth? |
| 15:52.01 | asd23 | whats wrong? |
| 15:52.02 | ManxPower | asd23: of course! |
| 15:52.12 | danbrwn | what is the port number asterisk is using for clients |
| 15:52.20 | ManxPower | asd23: there is no such option called auth= |
| 15:52.34 | asd23 | hmmm... |
| 15:52.36 | asd23 | ok |
| 15:52.38 | ManxPower | [yourusername] |
| 15:52.43 | ManxPower | username=yourusername |
| 15:52.50 | ManxPower | type=friend |
| 15:52.55 | ManxPower | secret=yourpassword |
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| 15:53.01 | ManxPower | context=extensionsconfcontext |
| 15:53.40 | ManxPower | in [general] you would put your register => yourusername:yourpassword@your.sip.provider.com |
| 15:53.43 | asd23 | so what goes in the [authentication] section? |
| 15:54.00 | ManxPower | asd23: there is no such thing as an [authentication] section. |
| 15:54.08 | ManxPower | Where do you get this crap? The Wiki? |
| 15:54.19 | ManxPower | unless your username is "authentication" |
| 15:55.01 | asd23 | Thats what I get for copy and paste eh? |
| 15:55.34 | asd23 | So the only reserved section name is general? |
| 15:55.54 | ManxPower | asd23: the only reserved section name in sip.conf is [general] |
| 15:56.11 | ManxPower | unless you are using something like 1.4. I don't know how it is done in 1.4 since that has not even been released yet. |
| 15:56.21 | asd23 | well, then don't I feel stupid. :) |
| 15:56.51 | asd23 | I'm using 1.2.13 |
| 15:57.23 | Supaplex | my default config had an empty authentication section to |
| 15:57.31 | ManxPower | in [general] you put your default options as well as your register => lines |
| 15:57.51 | ManxPower | Supaplex: there is an [authentication] section in sip.conf.sample? |
| 15:58.14 | Supaplex | must be a debian thing. *shrug* |
| 15:58.15 | asd23 | Manx: If I don't have a section named after my username then the incoming calls get routed to general, I'm assuming. Correct? |
| 15:58.16 | ManxPower | Or are you doing something really stupid like trying to use freepbx/trixbos configs? |
| 15:58.23 | ManxPower | asd23: correct. |
| 15:58.28 | Supaplex | heck no |
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| 15:58.41 | asd23 | Lightbulb |
| 15:58.52 | ManxPower | asd23: you didn't read The Book did you? |
| 15:58.54 | ManxPower | ~book |
| 15:58.55 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:59.08 | asd23 | I'm reading O'Reilly |
| 15:59.35 | asd23 | But O'rielly's book is more for breadth and less on depth. |
| 15:59.41 | ManxPower | *grumble* I guess I should start packing for my trip. |
| 16:01.36 | asd23 | Manx: The only thing I should have under general then is the register statement, right? |
| 16:01.50 | ManxPower | asd23: that is a good start |
| 16:01.54 | asd23 | ok, cool |
| 16:05.16 | asd23 | manx: I tried it, but my IVR won't accept dtmf signals, I tried dtmfmode=rfc2833 and auto. |
| 16:06.51 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 16:08.27 | heh_v_water | so if I set musiconhold as well as announce on a queue should it play the music and every once in a while announce their status? |
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| 16:29.53 | b11d | hello lads |
| 16:30.17 | b11d | I've got a problem.. two Polycom 501's cant talk to each other.. they can call each other just fine, but once both ends pick up, it goes silent. |
| 16:30.29 | b11d | I can successfully call each of those 501's from other 501's and it works.. |
| 16:30.38 | b11d | just those two cannot speak to one another.. |
| 16:30.40 | b11d | it doesnt make any sense |
| 16:30.46 | b11d | rtp seems to be fine.. sip seems to be fine.. |
| 16:34.24 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 16:34.34 | TripleFFFF | anyone know if stun setting sin xlite can casue drop calls ? |
| 16:34.41 | TripleFFFF | sees m 1.2.13 and lxlite are nasty |
| 16:40.00 | shellshark | is there english in there somewhere? |
| 16:41.58 | b11d | heh |
| 16:42.33 | *** join/#asterisk newbie^^ (n=DSM@90.153.134.253) |
| 16:42.54 | shellshark | b11d: are you provisioning all of the phones? or using their web-based config interfaces? |
| 16:43.18 | synthetiq | im trying to access a meetme confernce room , it used to work before, but know when i try to access it i constantly get message that zap/pseudo hangs up...any idea for the cause? |
| 16:43.45 | b11d | im provisioning them from an ftp server |
| 16:43.52 | b11d | it is successfully pulling the configs and uploading log files |
| 16:43.56 | b11d | it is = they are |
| 16:44.19 | ManxPower | TripleFFFF: I can imagine that it could cause dropped calls. Don't use STUN |
| 16:44.36 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 16:44.39 | newbie^^ | do i have to wait for a turn something? |
| 16:44.48 | hmmhesays | why not use stun |
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| 16:47.37 | newbie^^ | i'm new to Asterisk but my callcentric insisted that i install it if i wanted to use multiple channels.. anyway everything is working fine.. what i want to ask about is the following.. if i have a phone book that contains let's say 100 entery and i want to give each entery a time limit for calling into my PBX like a monthly plan of 15 hours.. is that possible? |
| 16:48.24 | M_at | Is there any easy way to see exactly what MOH file is being played to a call? |
| 16:49.17 | shellshark | M_at: ps aux | grep mpg123 ? |
| 16:49.17 | M_at | I hope not - I deleted all the MP3 files :) |
| 16:50.20 | shellshark | then how is it able to play anything at all? ;) |
| 16:50.43 | M_at | Because I have installed the native versions in A, u, G722, G726 and Wav |
| 16:51.13 | M_at | Want to make sure the u law version is played on this call |
| 16:51.22 | M_at | g72 on the g726 calls etc |
| 16:51.27 | shellshark | yuck |
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| 16:51.45 | M_at | why yuck? |
| 16:51.52 | shellshark | you should just keep them wav's and let asterisk transcode them for you |
| 16:51.57 | shellshark | or use icecast :) |
| 16:52.19 | M_at | should? Or "the wasy way is to" |
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| 16:57.43 | hmmhesays | can read take multiple file names to playback? |
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| 16:59.37 | b11d | my problem was that I had to specify canreinvite=no for the two polycom's having the issue.. |
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| 17:01.44 | CunningPike | M_at: We are using native MOH with ulaw files - works fine |
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| 17:02.15 | *** join/#asterisk BSD_Tech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 17:02.22 | newbie^^ | i'm new to Asterisk but my callcentric insisted that i install it if i wanted to use multiple channels.. anyway everything is working fine.. what i want to ask about is the following.. if i have a phone book that contains let's say 100 entery and i want to give each entery a time limit for calling into my PBX like a monthly plan of 15 hours.. is that possible? |
| 17:02.46 | b11d | you want to allow each number only 15 hours of "PBX" time a month? |
| 17:03.02 | b11d | or.. they can use that number for up to 15 hours each month? |
| 17:03.53 | b11d | ok.. so.. you want help.. and then you dont respond? |
| 17:03.57 | b11d | nice.. |
| 17:04.06 | b11d | ttyl |
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| 17:05.13 | newbie^^ | sorry i'm back |
| 17:05.17 | newbie^^ | and... you're gone |
| 17:05.50 | b11d|bbl | nah |
| 17:05.51 | b11d|bbl | im still here |
| 17:05.59 | b11d|bbl | just I might be in-and-out |
| 17:06.09 | newbie^^ | ok .. |
| 17:06.31 | newbie^^ | and the answer is yes.. each number would get 15 hours per month.. |
| 17:07.11 | b11d|bbl | well.. you'd want to start by looking at the available billing packages.. |
| 17:07.12 | b11d|bbl | http://www.asteriskbilling.com/asterisk/software.htm |
| 17:07.17 | b11d|bbl | that might get you started |
| 17:07.28 | newbie^^ | will go and have a look |
| 17:07.29 | b11d|bbl | I dont know how they integrate, but I know its possible to do what you're asking.. other people do it.. |
| 17:08.09 | TripleFFFF | 10ManxPower: 01TripleFFFF: I can imagine that it could cause dropped calls. Don't use STUN |
| 17:08.12 | TripleFFFF | hmm yeah ? |
| 17:08.40 | TripleFFFF | what config then i dont see NOT USE |
| 17:08.54 | TripleFFFF | http://whistler.counterpath.net/images/stun15a.PNG |
| 17:08.57 | TripleFFFF | that the menu |
| 17:09.08 | b11d|bbl | haha |
| 17:09.11 | b11d|bbl | you rule TripleFFFF |
| 17:09.30 | TripleFFFF | what |
| 17:09.33 | TripleFFFF | ;) |
| 17:09.33 | TripleFFFF | b11 |
| 17:09.42 | b11d|bbl | hehe.. nothing.. |
| 17:09.59 | TripleFFFF | oh..whistler..hmm |
| 17:11.26 | mattfletcher | What is the easiest way to run a PHP from a call? AGI, mod_php or curl? |
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| 17:16.59 | mattfletcher | I have a trivial PHP script which returns a name from our bespoke customer DB when passed a number. What is the best way to integrate this into my dialplan? |
| 17:18.16 | b11d|bbl | AGI as I understand it.. |
| 17:18.24 | b11d|bbl | but I dont know that for sure, nor do I know if thats the best solution |
| 17:18.45 | TripleFFFF | FAST AGI with |
| 17:19.01 | TripleFFFF | set_time_limit(0.2); |
| 17:19.20 | TripleFFFF | this wait you can limit time it gets.. if whatever reason the tcp url is hung |
| 17:26.41 | *** join/#asterisk SoftIce (i=admin@infinity.security.web.za) |
| 17:26.58 | SoftIce | hi, to get isdn stuff working on asterisk can i just add the modules or do I have to install bristuff? |
| 17:27.10 | SoftIce | or how would one of you sugest to get an isdn modem working? |
| 17:27.23 | CunningPike | Has anyone experienced problems with Polycoms IP4000 failing to get their config files? We are using vsftpd and our setup works fine for 100+ 501s, but the IP4000 gives curl timeouts when trying to download its config files and sp.ld |
| 17:27.48 | CunningPike | This is a replacement phone, so it seems to be something specific to the model - maybe a bootblock issue? |
| 17:28.19 | XIN01OZ | I need a good friend/mentor/business partner that knows business ethics and asterisk that is good at making money flow.. heh - really though if somebody has a moment that might be able to help me, please pm me. I think I might have alot going... |
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| 17:31.06 | XIN01OZ | I could really use the help. |
| 17:31.18 | SoftIce | same here ;) |
| 17:31.25 | SoftIce | what is the best way to install an isdn card for asterisk? |
| 17:32.06 | XIN01OZ | hmmhesays: is it not through ZAP? |
| 17:33.17 | *** part/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:34.18 | XIN01OZ | with a name as SoftIce- that seems like a simple question |
| 17:34.35 | DrCron | is there a nice place to put voip addresses in outlook? |
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| 17:35.12 | SoftIce | XIN01OZ i'm new to asterisk and asking wha the best way around is, using a bri patch or to install briasteisk base with the allready customisation to zaptel, etc. |
| 17:37.40 | XIN01OZ | I think you have to compile Zaptel with the drivers configured |
| 17:38.15 | SoftIce | isdn ? |
| 17:38.22 | *** join/#asterisk waverly360 (n=waverly@209.12.249.243) |
| 17:38.41 | XIN01OZ | not sure- have not used isdn in asterisk |
| 17:38.47 | SoftIce | err, never midn.. i'll work it out. |
| 17:39.00 | SoftIce | i have configured this before with bristuff for 1.2 |
| 17:39.06 | SoftIce | this is version 1.4 of asterisk this person is using |
| 17:39.53 | XIN01OZ | Sorry Im not the one to help you on that situation |
| 17:40.21 | SoftIce | its fine, i remeber using chan capi ages ago, but i think its all moved to bristuff now |
| 17:42.30 | mattfletcher | tripleffff: where will i find docs about fast agi? |
| 17:42.47 | TripleFFFF | google is the main doc |
| 17:42.51 | TripleFFFF | as far as i know |
| 17:43.01 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 17:43.02 | TripleFFFF | google is my main encyclopedia |
| 17:43.04 | XIN01OZ | Anyone in here do Asterisk consulting? |
| 17:43.10 | zoa | yes |
| 17:43.13 | zoa | me |
| 17:43.25 | zoa | What do you need ? |
| 17:43.49 | XIN01OZ | ah, nice. what basis do you generally perform consultations? |
| 17:43.51 | SoftIce | well i need for started can asterisk 1.4 support isdn modems |
| 17:43.58 | SoftIce | or do i have to install bri asterisk? |
| 17:44.07 | SoftIce | ;) |
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| 17:53.43 | PupenoR | Hi. |
| 17:53.58 | PupenoR | How do I compile 1.4.0beta3 with debuging symbols (-ggdb) ? |
| 17:54.52 | *** join/#asterisk zamsler (i=zamsler@12.161.149.28) |
| 17:56.59 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 17:57.24 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 17:59.53 | *** join/#asterisk cmdln (n=cmdln@203.39.cm.sunflower.com) |
| 18:00.27 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.246.telnor.net) |
| 18:02.52 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:06.39 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:09.24 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:09.36 | mattfletcher | how on earth do u pass an argument to an agi script? i've tried everything i can think of! |
| 18:10.15 | Nugget | http://www.google.com/search?q=pass%20argument%20to%20asterisk%20agi%20script |
| 18:11.09 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 18:11.09 | mattfletcher | doh, pipe not comma, grr! thanks |
| 18:13.14 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@8.10.2.50) |
| 18:14.03 | *** join/#asterisk TheCops (n=henri@207.164.28.98) |
| 18:15.01 | TheCops | Hi, I have a Bell Canada voicemail and I want to map the MWI of my Polycom (Passing trought Asterisk) for this Voicemail. Someone know how? |
| 18:15.56 | Qwell[] | TheCops: I don't know if zaptel detects the stuttertone from the telco |
| 18:16.03 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 18:16.19 | Qwell[] | afaik, it only does it when the line goes off hook |
| 18:16.37 | Qwell[] | there may be an option in one of the configs to check periodically though. I'd take a look in the zaptel sample configs |
| 18:16.49 | TheCops | thanks |
| 18:19.22 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 18:21.00 | te_lo_meto_mami | grrr I am so frustrated I am a newbie at asterisk and I HAVE read voip-info, but with no success on a stupid little issue. During Background message I am trying to allow users to interact with asterisk and when they are hitting the #5 it does not playback the 5 , but goes on to transfer the call |
| 18:21.03 | te_lo_meto_mami | can someone help |
| 18:21.09 | te_lo_meto_mami | I have a small dialplan |
| 18:21.12 | te_lo_meto_mami | nothing fancy |
| 18:21.14 | te_lo_meto_mami | just learning |
| 18:22.09 | TheCops | Qwell, this is normal to see asterisk taking the line after 2 or 3 ring? |
| 18:22.27 | TheCops | Qwell, I know it is waiting the caller ID, but 3 ring is very long. |
| 18:23.20 | te_lo_meto_mami | Can someone help me out wit this |
| 18:23.21 | te_lo_meto_mami | <PROTECTED> |
| 18:23.21 | te_lo_meto_mami | <PROTECTED> |
| 18:23.22 | te_lo_meto_mami | exten => s,2,Background(jorges-recording) |
| 18:23.22 | te_lo_meto_mami | exten => 1,1,Playback(digits/5) |
| 18:23.23 | te_lo_meto_mami | exten => 1,2,Goto(damn,s,1) |
| 18:23.27 | Qwell[] | ~paste |
| 18:23.30 | jbot | i heard paste is http://rafb.net/paste/ |
| 18:23.30 | te_lo_meto_mami | [damn] |
| 18:23.31 | te_lo_meto_mami | exten => s,1,Answer() |
| 18:23.31 | te_lo_meto_mami | exten => s,2,Dial(SIP/pimp) |
| 18:23.53 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:24.08 | *** mode/#asterisk [-b g0tw00d!*@*] by Qwell[] |
| 18:25.45 | te_lo_meto_mami | <<<<sighs, aight guys thanks 4 the help :-( |
| 18:25.46 | tzafrir_laptop | ~pb |
| 18:25.49 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:26.08 | mcab | CunningPike: re the IP4000 - is it running the same BootROM as the 501s? (and stupid question - it is trying to get to the server you think it is, right? :-) ) |
| 18:26.13 | tzafrir_laptop | Qwell[], shorter, and with instructions |
| 18:26.32 | Qwell[] | tzafrir_laptop: yeah, people keep changing them |
| 18:27.02 | CunningPike | mcab: Yes to both - xferlog on the server shows the ftp attempts...... |
| 18:27.18 | tzafrir_laptop | te_lo_meto_mami, first-off, please avoid foul language here |
| 18:27.39 | Qwell[] | tzafrir_laptop: foul language? His context name was "damn" :p |
| 18:27.44 | te_lo_meto_mami | language? |
| 18:27.51 | te_lo_meto_mami | ahh |
| 18:27.52 | te_lo_meto_mami | sorry |
| 18:28.00 | te_lo_meto_mami | context not me typing at someone |
| 18:28.03 | te_lo_meto_mami | but my bad |
| 18:28.04 | Qwell[] | I don't consider that foul, to be honest |
| 18:28.17 | te_lo_meto_mami | i dont know asterisk room rules |
| 18:28.21 | te_lo_meto_mami | like i said i am new |
| 18:28.36 | te_lo_meto_mami | next time ill just rename before pasting |
| 18:28.43 | tzafrir_laptop | I don't really mind that myself. The point is that when people see such words they tend to become more emotianal and think about their meaning rather than about your question |
| 18:29.03 | tzafrir_laptop | A similar reason to why you should avoid typos |
| 18:29.05 | *** join/#asterisk _-Jon-_ (n=jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 18:29.12 | _-Jon-_ | Hey all |
| 18:29.15 | te_lo_meto_mami | can u guys look pastthe word and help me out please |
| 18:29.35 | tzafrir_laptop | anyway, you have not defined a problem |
| 18:29.47 | te_lo_meto_mami | i will repeat problem |
| 18:29.54 | _-Jon-_ | I'm having a sligh problem I'm hoping someone can assist with.. Basically when someone calls my number, it answers, then dials my SIP device. However it doesn't ring to the calling party |
| 18:29.56 | tzafrir_laptop | How do you see that something is wrong? |
| 18:30.15 | te_lo_meto_mami | well when i dial phone asterisk answers and plays recording no problem |
| 18:30.24 | te_lo_meto_mami | but i want caller to hit 5 |
| 18:30.26 | mcab | CunningPike: and a 501 in the same location works fine? Do you have an ethereal capture of what the IP4000 is doing? |
| 18:30.48 | te_lo_meto_mami | than when they hit 5 i want asterisk to say 5 and Goto context defined |
| 18:31.08 | te_lo_meto_mami | if i hit 1 it says 5 |
| 18:31.30 | CunningPike | mcab: Yes - it does. And I'm setting up for a pcap as we speak........ |
| 18:31.30 | te_lo_meto_mami | weird |
| 18:31.36 | *** join/#asterisk Katty (n=Administ@68-119-251-157.dhcp.cpgr.mo.charter.com) |
| 18:31.43 | mcab | CunningPike: :-) |
| 18:31.47 | Katty | morning |
| 18:31.51 | tzafrir_laptop | where? I don't see any extension 5 |
| 18:32.35 | tzafrir_laptop | the Playback is in the extension of 1 |
| 18:32.40 | *** join/#asterisk spyder5150 (n=scott@hera.copi-rite.com) |
| 18:32.53 | te_lo_meto_mami | in dial plan i need to setup a variable than i guess is what you are saying like extension 1020 = 5? |
| 18:33.23 | te_lo_meto_mami | i have a sip peer called SIP/pimp |
| 18:33.28 | te_lo_meto_mami | excuse the name |
| 18:33.31 | tzafrir_laptop | In the context where the caller is, you need to have something in the lines of: |
| 18:33.53 | tzafrir_laptop | exten => 5,1,DoSomething |
| 18:34.07 | tzafrir_laptop | e.g: |
| 18:34.26 | tzafrir_laptop | exten => 5,1,Dial(SIP/11) |
| 18:34.42 | tzafrir_laptop | Please take a look at the sample extensions.conf |
| 18:34.44 | Katty | spyder5150: hi. |
| 18:34.57 | spyder5150 | hi |
| 18:35.12 | *** join/#asterisk Lunatic (n=RodeO@88.242.198.255) |
| 18:35.19 | te_lo_meto_mami | i understand , 5 is the name of the extension defined in digits/5? |
| 18:35.21 | Katty | spyder5150: you don't recognize me, do you :P |
| 18:36.03 | te_lo_meto_mami | thanks laptop, appreciate it |
| 18:37.01 | _-Jon-_ | Any one have any idea why Dial(SIP/200,25,r) wouldn't produce a ringing sound? |
| 18:39.13 | Qwell[] | _-Jon-_: get rid of the r |
| 18:39.56 | tzafrir_laptop | digits/5 is an arbitrary file name |
| 18:40.07 | hmmhesays | r should produce ringing no matter what is happening though |
| 18:40.24 | Qwell[] | hmmhesays: it causes very bizarre things... |
| 18:40.49 | Katty | hmmhesays: are you causing trouble again? |
| 18:41.34 | hmmhesays | Katty: always |
| 18:42.31 | _-Jon-_ | Qwell, but then how is the caller to hear anything? |
| 18:42.51 | Qwell[] | _-Jon-_: the remote side is supposed to take care of it |
| 18:43.21 | _-Jon-_ | Qwell, my provider doesn't seem to |
| 18:43.34 | Qwell[] | Then you have bigger problems... |
| 18:43.49 | Katty | hmmhesays: i setup my second asterisk box yesterday :> |
| 18:43.50 | _-Jon-_ | I have s,1,Answer then s,n,Dial(SIP/200,25) |
| 18:43.53 | Katty | hmmhesays: all by myself :>>> |
| 18:44.02 | _-Jon-_ | If my * box is answering, how is my provider to know to provide a ring? |
| 18:44.08 | Qwell[] | Don't answer :) |
| 18:44.36 | _-Jon-_ | Nice, it worked :) |
| 18:44.52 | hmmhesays | Katty: cool, what for? |
| 18:45.22 | Katty | hmmhesays: just demoing purposes. |
| 18:45.50 | hmmhesays | cool |
| 18:45.51 | Katty | hmmhesays: it's not gonna be used for real. |
| 18:49.36 | te_lo_meto_mami | so laptop for asterisk to say 5 i must create a gsm sound file and put in /sounds directory and than it will play 5 once the 5 key is hit? |
| 18:50.33 | awannabe | is the only way to set callerid different for phone call (outgoing caller id) is by a if statement? |
| 18:51.56 | Katty | hmmhesays: what kinda neat stuff can you do with asterisk and sql? |
| 18:52.12 | Katty | hmmhesays: besides dumping the log into it. |
| 18:52.16 | hmmhesays | Katty: what do you want to do? |
| 18:52.22 | Katty | hmmhesays: i dunno. |
| 18:52.28 | Katty | hmmhesays: just looking for something fun and new. |
| 18:52.30 | hmmhesays | store your configs? cdr's? |
| 18:52.43 | hmmhesays | i use cmd mysql to pull values out of a database sometimes |
| 18:52.47 | Katty | yeah, we're gonna store the csv thingy in there... |
| 18:52.56 | Katty | hmmhesays: but i really dunno what else is possible. |
| 18:52.57 | hmmhesays | and I wrote a pre-paid dialplan based on cmd mysql |
| 18:53.01 | Katty | hmmhesays: or even slightly useful. |
| 18:53.17 | hmmhesays | the cdrs are especially if you use areski's asterisk stat page |
| 18:53.27 | Katty | okay, dumb qustion...i'm gonna bite. cdr? |
| 18:53.36 | hmmhesays | call detail record |
| 18:53.42 | *** join/#asterisk Ebola (n=Ebola@host86-134-167-28.range86-134.btcentralplus.com) |
| 18:53.51 | Katty | oh. |
| 18:54.02 | Katty | that's just the same thing as the master.csv then, just in a different format, etc. |
| 18:54.11 | hmmhesays | yes |
| 18:54.16 | Katty | whaaat about.... |
| 18:54.20 | Katty | like callerid stuff |
| 18:54.33 | hmmhesays | put it in a database with a nice web page front end and they become more readable |
| 18:54.37 | Katty | make a database of numbers, and whatever callerid you want it to read |
| 18:54.47 | hmmhesays | you could |
| 18:55.09 | Katty | that might be kinda neat to have if you want to put notes into the database. |
| 18:55.17 | Katty | and then you can do a netsend message.. |
| 18:55.24 | Katty | and i dunno if you can get a link into the net send message or not |
| 18:55.27 | hmmhesays | based on what? |
| 18:55.30 | Katty | callerid |
| 18:55.55 | Katty | john doe calls, net sent $computer message with who it is and a link, person clicks link, brings up database inflimation with notes fields... |
| 18:55.55 | hmmhesays | i don't know if you'd need sql to do that |
| 18:55.59 | *** join/#asterisk TheBearded1 (n=criggs@203.39.cm.sunflower.com) |
| 18:56.17 | Katty | it's still somethin you could do with sql. |
| 18:56.23 | spyda | reverse number lookup! that'd be awesome |
| 18:56.31 | TheBearded1 | anybody here that can help a dumb linux guy that knows zilch about pots? |
| 18:56.34 | hmmhesays | a *nix system call to a windoze webserver could accomplish that |
| 18:56.34 | Katty | and you could pull up the other table and get latest call information or somethin |
| 18:56.45 | Katty | yeah, i've done a net send popup from linux before |
| 18:56.53 | Katty | that's how we used to tell our receiptionist to answr the phone |
| 18:57.08 | hmmhesays | you could send a gtalk message |
| 18:57.15 | TheBearded1 | looking to setup an asterisk box but know nothing about what is required on the pots side |
| 18:57.19 | Katty | i think our receptionist is too dumb for that ;) |
| 18:57.29 | Katty | yahoo, maybe...she already uses that |
| 18:57.40 | hmmhesays | if you could find a linux commmand line yahoo client |
| 18:57.47 | hmmhesays | TheBearded1: nothing if you don't want |
| 18:57.48 | Katty | ! |
| 18:57.56 | Katty | oh wow, i never thought about it that way |
| 18:58.02 | Katty | just do a sytem command and spam somebody |
| 18:58.20 | Katty | hmmhesays: you're a genious! |
| 18:58.47 | TheBearded1 | we want a business single 1-800 number, allowing ~6 simultaneous calls inbound or outbound |
| 18:59.00 | TheBearded1 | we don't know if we need DID + 6 SIP accounts |
| 18:59.06 | TheBearded1 | or one SIP account with 6 channels |
| 18:59.10 | TheBearded1 | or what |
| 18:59.17 | hmmhesays | depends on the company you go with |
| 18:59.30 | hmmhesays | where you buy your 800 number from |
| 18:59.37 | Katty | hmmhesays: arg, you make me wanna go to work now |
| 18:59.40 | Katty | hmmhesays: damn you! |
| 18:59.44 | *** join/#asterisk tsurk0 (n=tsurko@80.72.68.86) |
| 18:59.50 | hmmhesays | haha |
| 18:59.55 | hmmhesays | do you still work at the same place? |
| 18:59.59 | TheBearded1 | we already have an 800 number, would it need transfered? |
| 19:00.05 | Katty | hmmhesays: oh yeah (= |
| 19:00.13 | hmmhesays | where does the 800 number go to? |
| 19:00.16 | Katty | hmmhesays: i'm just..uh...sick today. yeah. |
| 19:00.28 | TheBearded1 | this pbx system in question is for our new office we're moving to |
| 19:00.30 | hmmhesays | ahh, I ditched the 8-5 back in august |
| 19:00.34 | TheBearded1 | 800 number goes to our current office |
| 19:00.45 | hmmhesays | analog lines coming in or a t1? |
| 19:00.50 | TheBearded1 | ???? |
| 19:00.53 | TheBearded1 | we didn't set up this |
| 19:00.59 | TheBearded1 | we know it works and we get phone calls |
| 19:01.01 | *** join/#asterisk enjay- (n=yea@209.181.130.105) |
| 19:01.03 | hmmhesays | in the new office then |
| 19:01.13 | TheBearded1 | nothing, trying to figure it out |
| 19:01.14 | hmmhesays | analog lines or a t1 coming in |
| 19:01.19 | TheBearded1 | T1 for our net |
| 19:01.25 | TheBearded1 | for voice, we don't know, that's why we're here |
| 19:01.27 | hmmhesays | ok, what about for the phones |
| 19:01.30 | hmmhesays | um |
| 19:01.32 | hmmhesays | its up to you |
| 19:01.41 | TheBearded1 | maybe partial T1 for phone, but we're open to voip going outbound |
| 19:01.41 | hmmhesays | if you're moving to a new office with nothing in it |
| 19:01.51 | hmmhesays | how many incoming channels? |
| 19:01.56 | hmmhesays | total |
| 19:02.03 | TheBearded1 | we want 12 channes |
| 19:02.09 | TheBearded1 | want* 12 channels* |
| 19:02.12 | enjay- | What file format and sample rate is best for asterisk recordings? |
| 19:02.14 | hmmhesays | then you'd probably end up with an incoming t1 |
| 19:02.20 | hmmhesays | get a t1 card |
| 19:02.24 | enjay- | as far as playing them via asterisk to an end-user. |
| 19:02.25 | hmmhesays | either digium or sangoma |
| 19:02.39 | TheBearded1 | okay, just for the hell of it what would be required to implement the same with voip outbound? |
| 19:02.40 | hmmhesays | both make t1 cards |
| 19:02.54 | hmmhesays | a good ip connection |
| 19:03.00 | TheBearded1 | k |
| 19:03.02 | hmmhesays | and an outbound account with someone |
| 19:03.33 | te_lo_meto_mami | got it going thanks laptop |
| 19:03.34 | te_lo_meto_mami | [mainmenu] |
| 19:03.34 | te_lo_meto_mami | exten => s,1,Answer() |
| 19:03.34 | te_lo_meto_mami | exten => s,n,Background(jorges-recording) |
| 19:03.34 | te_lo_meto_mami | exten => s,n,WaitExten |
| 19:03.35 | te_lo_meto_mami | exten => s,n,Playback(digits/5) |
| 19:03.37 | te_lo_meto_mami | exten => i,1,Playback(invalid) |
| 19:03.39 | te_lo_meto_mami | exten => 5,1,Dial(SIP/pimp) |
| 19:03.44 | hmmhesays | pastebin |
| 19:03.47 | hmmhesays | ~pastebin |
| 19:03.58 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:04.00 | TheBearded1 | okay, so we hook up the t1 cable into our t1 card, how does that translate into a telephone number |
| 19:04.09 | TheBearded1 | how does that enable us to get calls just by plugging in a t1 |
| 19:04.16 | hmmhesays | no |
| 19:04.25 | awannabe | can you set outgoing callerid over a Zap channel on a per phone/extension basis? I have one extension/login that needs to have a different callerid set |
| 19:04.31 | hmmhesays | you would have to order a voice t1 from your telephone carrier |
| 19:04.32 | TheBearded1 | do we need to setup something with, say at&t, if att provides the partial t carrier do they have to know we're using it for voice |
| 19:04.41 | hmmhesays | yes |
| 19:04.48 | hmmhesays | because their the ones the provision your t1 |
| 19:04.54 | hmmhesays | s/the/that |
| 19:05.14 | TheBearded1 | so att/sprint would basically hook their end of the t1 into their telephone network |
| 19:05.17 | TheBearded1 | and our end goes to ours |
| 19:05.25 | hmmhesays | something like that |
| 19:05.34 | TheBearded1 | ridiculously oversimplified ofcourse |
| 19:05.39 | TheBearded1 | so DID |
| 19:05.58 | TheBearded1 | does it forward calls from the ordered number to the local number provided by att/sprint? |
| 19:06.10 | TheBearded1 | or is the DID number the one attached to the T1 connection? |
| 19:06.22 | hmmhesays | both could be the scenario |
| 19:06.26 | TheBearded1 | do we have to have a local number? |
| 19:06.40 | hmmhesays | probably not |
| 19:07.46 | Katty | hmmhesays: i ever tell you about my lil xmms/paging project on asterisk? |
| 19:08.07 | TheBearded1 | so we get a local number t1 line from att/sprint, then we go to a DID provider? |
| 19:08.11 | hmmhesays | xmms? |
| 19:08.14 | enjay- | What recording format and sample rate is best for asterisk playback? |
| 19:08.15 | Katty | hmmhesays: yup. |
| 19:08.17 | hmmhesays | x multimedia sstem? |
| 19:08.22 | Katty | hmmhesays: two servers...public ssh keys... |
| 19:08.35 | Katty | hmmhesays: one server plays xmms over speakers all day long. |
| 19:08.38 | TheBearded1 | okay we saw plans on DID sites for phone numbers, but they listed channels |
| 19:08.45 | TheBearded1 | are their channel requirements on a DID account? |
| 19:09.02 | Katty | hmmhesays: the asterisk box sshes over to the second one, uses xmms-shell to kickt he volume down...and then asterisk connects to the line in part of the second box...and it's unmuted, so it just plays whatever it hears... |
| 19:09.22 | Katty | hmmhesays: then it dumps the audio file it recorded to the regular output...when then routes through the second box...and out the speakers |
| 19:09.25 | hmmhesays | Katty: cool |
| 19:09.30 | Katty | hmmhesays: kicks the volume back up...and kills the ssh session |
| 19:09.38 | hmmhesays | TheBearded1: you should call your carrier |
| 19:10.09 | hmmhesays | they'll be able to be more specific than any of us |
| 19:10.44 | TheBearded1 | would att/sprint be able to provide us the DID service as well? |
| 19:10.52 | hmmhesays | ugh I have to take my zinc pill now |
| 19:10.55 | Katty | hmmhesays: i guess...in theory... |
| 19:11.09 | Katty | hmmhesays: i could make the speakers say whoever's calling on based on callerid and wav files ;) |
| 19:11.11 | hmmhesays | oh, I thought you alread set this up? |
| 19:11.22 | TheBearded1 | me? no |
| 19:11.29 | TheBearded1 | we are trying to plan a system out |
| 19:11.33 | hmmhesays | the last comment was directed at katt |
| 19:11.38 | hmmhesays | *katty |
| 19:11.41 | TheBearded1 | and are unsure of what to buy |
| 19:11.56 | hmmhesays | TheBearded1: i suggested that you call the carrier that will be providing service to the new site |
| 19:12.10 | TheBearded1 | i thought you were asking me if i had built the system |
| 19:12.28 | TheBearded1 | we're gonna go ahead and call att/sprint to finish this up |
| 19:12.40 | TheBearded1 | you did clarify some of the cluelessness we had here though |
| 19:13.50 | enjay- | What recording format and sample rate is best for asterisk playback? |
| 19:13.52 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 19:14.53 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 19:16.31 | hmmhesays | enjay: whatever format your calling party is using for a codec |
| 19:17.51 | TheBearded1 | thanks for the consult guys |
| 19:17.53 | *** part/#asterisk TheBearded1 (n=criggs@203.39.cm.sunflower.com) |
| 19:18.09 | hmmhesays | he can pay me in hookers and blow |
| 19:18.49 | awannabe | thats a nice payment :) |
| 19:19.25 | *** join/#asterisk Delta239 (i=Delta_Of@201.226.130.55) |
| 19:19.36 | *** join/#asterisk hohum (n=dcorbe@host-12-195-58-235.iad1.interceltelecoms.net) |
| 19:20.06 | Delta239 | hello... i have a problem.. my softphone wont connect to my asterisk server... if i go to the CLI it doesnt even shows up that is trying to connect |
| 19:20.22 | hmmhesays | what softphone |
| 19:20.31 | Delta239 | eyebeam sip softphone |
| 19:20.38 | hmmhesays | Katty: so your project wants to play the calling party on a remote box? |
| 19:20.51 | hmmhesays | check your settings |
| 19:21.01 | Delta239 | everything is fine |
| 19:21.21 | hmmhesays | obviously not |
| 19:21.26 | Delta239 | i have done this many times... adding and removing extensions |
| 19:21.36 | Delta239 | if i go to sip show peers |
| 19:21.50 | hmmhesays | what is your verbosity level? |
| 19:22.01 | Delta239 | 16 |
| 19:23.32 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:23.50 | Delta239 | on my softphone says registration error 408 request time out |
| 19:24.07 | Delta239 | on the options on the domain i have the correct ip |
| 19:24.47 | Delta239 | im using astrisk 1.2.12.1 |
| 19:25.02 | hmmhesays | set get out your packet sniffer and check where it is sending the registration request |
| 19:26.40 | Delta239 | how |
| 19:29.14 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-215-87.cablep.bezeqint.net) |
| 19:29.17 | *** join/#asterisk philippel (n=p_lindhe@c-24-16-243-129.hsd1.wa.comcast.net) |
| 19:29.37 | philippel | anyone know off hand what the size limit is for an astdb variable (string length)? |
| 19:33.07 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
| 19:33.12 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:33.52 | *** join/#asterisk mspiceland (i=mike@nat/digium/x-caa6932e7bfed694) |
| 19:33.56 | *** part/#asterisk mspiceland (i=mike@nat/digium/x-caa6932e7bfed694) |
| 19:34.49 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:39.22 | stephane_ | jour |
| 19:39.29 | stephane_ | soir |
| 19:41.05 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 19:41.57 | BlepsoaF | hello all, does anyone use automon? It seems when I set Ww on the dial command it will block sending of all DTMF digits. IE if you call an automated bank teller line, you wont be able to do anything....anyone experience this? |
| 19:42.06 | *** join/#asterisk mithraen (n=mithraen@87.228.121.245) |
| 19:42.53 | SheriF_SpacE | what is automon BlepsoaF ? |
| 19:43.16 | BlepsoaF | call recording on the fly when entering a digit on the phone, such as *1 |
| 19:43.27 | BlepsoaF | via features.conf |
| 19:43.47 | SheriF_SpacE | ah i c |
| 19:43.50 | SheriF_SpacE | works fine with me no problem |
| 19:44.02 | BlepsoaF | you use that feature? |
| 19:44.12 | SheriF_SpacE | but never tired to do it then try to use DTMF yes |
| 19:44.25 | BlepsoaF | hmm strange |
| 19:44.31 | SheriF_SpacE | i have all IN - OUT calls recored using monitor and teh sip internlay on demand wiht wW option |
| 19:44.44 | SheriF_SpacE | but i never tired to use DTMF after i start to record |
| 19:44.52 | SheriF_SpacE | let me try with my voicemail |
| 19:44.52 | BlepsoaF | this is even before |
| 19:44.54 | EmleyMoor | Why would a softphone be taking ages to register? |
| 19:45.22 | BlepsoaF | it could be because I made changes to features.conf to enable it, but didnt restart asterisk, I just did a reload res_features |
| 19:46.22 | *** join/#asterisk jakehow (n=jake@66.246.95.2) |
| 19:46.39 | SheriF_SpacE | EmleyMoor: enable full logger asterisk |
| 19:46.51 | SheriF_SpacE | and set verbose 9999 in * CLI and see why |
| 19:46.59 | SheriF_SpacE | i think it might be firewall droping packages ? |
| 19:47.30 | SheriF_SpacE | BlepsoaF: u mean ur asterisk ignoures ur DTMF ? |
| 19:47.33 | jakehow | anyone got an idea what could cause intermittent DTMF issues on outgoing calls? |
| 19:47.46 | jakehow | (touchtone prompts dont recognize tones) |
| 19:47.47 | EmleyMoor | Hmm... started working again OOB |
| 19:48.41 | *** join/#asterisk tim27 (i=tim27@97-70.dr.cgocable.ca) |
| 19:51.13 | tim27 | my sip provider... is babytel.ca, i have a account with multiple DID, they invite my server with INVITE 18198502500@192.168.1.101 and send the DID info in the TO: FIELD as TO: 18198502523@192.168.1.101 ... i called them and they say they allways send DID info in the TO: field and i have to program my server to be able to link the TO: field to the inbound DID |
| 19:51.28 | EmleyMoor | One of the others will be adapted to have two sockets when the system goes live |
| 19:51.44 | BlepsoaF | SheriF_SpacE: Dunno, my cell phone voice mail for example doesnt receive anything |
| 19:51.44 | EmleyMoor | Anyone on FWD care to give me a call? 794933 |
| 19:52.34 | SheriF_SpacE | EmleyMoor: i think mine is not activated anymore |
| 19:52.36 | SheriF_SpacE | let me try |
| 19:53.47 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
| 19:54.16 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-215-87.cablep.bezeqint.net) |
| 19:54.43 | awannabe | is there any good examples how to use the if function, i dont see any comments on it really |
| 19:55.19 | SheriF_SpacE | awannabe: voip-info and there is nice examples there about call status and so on |
| 19:55.30 | SheriF_SpacE | EmleyMoor: give me sometime i need to fix a dialplan for Fwd |
| 19:56.09 | awannabe | SheriF_SpacE: ok ill look again, there is nothing under the actaul wiki page for fucntion if, heh |
| 19:56.18 | *** join/#asterisk jsolares (n=jsolares@206.113.229.70) |
| 19:58.18 | EmleyMoor | Is there any way to drive the voicemail using a rotary phone? |
| 19:58.28 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:58.46 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:59.53 | Qwell[] | EmleyMoor: sure, if whatever you connect it to understands rotary |
| 20:00.17 | EmleyMoor | It seems to be the voicemail itself that doesn't understand |
| 20:00.54 | SheriF_SpacE | EmleyMoor: ok will try |
| 20:01.57 | tim27 | any asterisk consultant here can help me i can pay by paypal ... |
| 20:03.18 | *** join/#asterisk shellsha2k (n=x86@74.135.64.209) |
| 20:03.26 | SheriF_SpacE | EmleyMoor: sorry my connection not great |
| 20:03.46 | EmleyMoor | Never mind - proves it's working anyhow |
| 20:03.57 | EmleyMoor | That was my rotary phone, btw |
| 20:03.59 | SheriF_SpacE | yes |
| 20:04.11 | SheriF_SpacE | how come rotary ? |
| 20:04.33 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 20:04.41 | EmleyMoor | I just happen to have one and I like the way it rings |
| 20:05.54 | SheriF_SpacE | EmleyMoor: sorry i didn't get it i'm from egypt :-) so most of USA talk is out of my small brain :D |
| 20:05.56 | EmleyMoor | They were still quite common here until about 10 years ago |
| 20:06.08 | EmleyMoor | I'm English |
| 20:06.50 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 20:07.24 | SheriF_SpacE | okay nice to meet u :) |
| 20:07.33 | SheriF_SpacE | what kind of 11 line u'll use EmleyMoor ? |
| 20:08.06 | EmleyMoor | 1 Zap and a 10-number DDI range from a VoIP provider |
| 20:08.19 | SheriF_SpacE | DID ? |
| 20:08.35 | SheriF_SpacE | DIDs incoming right >? |
| 20:08.59 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 20:10.01 | EmleyMoor | Well, yes |
| 20:10.40 | SheriF_SpacE | thats easy |
| 20:10.42 | *** join/#asterisk SwK[Work] (n=SwK@br0.asteriasgi.com) |
| 20:11.15 | EmleyMoor | Yes - been waiting for a working FXO port before I go too far - should be here within 2 days |
| 20:11.39 | *** join/#asterisk shellsha3k (n=x86@74.135.64.209) |
| 20:11.42 | *** join/#asterisk uwe (n=uwe@213.6.13.67) |
| 20:12.00 | EmleyMoor | Getting the 10-number range so that selected people can bypass the IVR |
| 20:12.08 | *** join/#asterisk AuPix (n=root@adsl-04-85.abel.net.uk) |
| 20:13.11 | awannabe | what is the point of exten =>1234/XXXXXXXXXX,1,Dial(SIP/bla,20,tr) what does the /after the xtension do? |
| 20:14.26 | *** join/#asterisk ManxPower (n=manxpowe@16.sub-75-202-111.myvzw.com) |
| 20:16.50 | SheriF_SpacE | EmleyMoor: working ? u got a damage one before ? |
| 20:17.16 | EmleyMoor | Yes - it went off-hook on the first ring and stayed there for good |
| 20:17.24 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-142-96.dsl.tul2ok.sbcglobal.net) |
| 20:17.47 | SheriF_SpacE | EmleyMoor: digium ? |
| 20:17.52 | EmleyMoor | Yes |
| 20:19.05 | SheriF_SpacE | very strang |
| 20:20.04 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 20:22.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:22.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:23.33 | awannabe | can anyone help me get callerid to work over a zap channel, but have it different depends on what extension it orginated from? also willing to pay for help to! |
| 20:26.31 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-115-171.vic.westnet.com.au) |
| 20:31.00 | SheriF_SpacE | good night guys |
| 20:31.30 | SheriF_SpacE | awannabe: i have caller ID over zaptel card |
| 20:32.31 | awannabe | i do to, i just gotta change it per phone |
| 20:32.38 | SheriF_SpacE | what u mean ? |
| 20:33.30 | awannabe | like i have extension 200, and when it makes a outgoing call it needs a diff outgoing callerid |
| 20:38.41 | *** join/#asterisk Soul (n=Soul@87-196-111-65.net.novis.pt) |
| 20:40.03 | hmmhesays | so use an if statement to change the callerid |
| 20:40.15 | EmleyMoor | awannabe: You mean you want to change the outgoing caller ID over the Zap channel? |
| 20:40.15 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 20:40.18 | awannabe | hmmhesays: yeah thats what im trying, i guess im just confused |
| 20:40.28 | fiber0pti | Has anyone else had problems with getting AsteriskNOW working? |
| 20:40.30 | awannabe | EmleyMoor: correct, but just when a certain extension dial it! |
| 20:40.44 | EmleyMoor | awannabe: I very much doubt that is possible |
| 20:40.45 | fiber0pti | I'm getting a bunched of "undefined" errors in the GUI.. can't even make a new extensions |
| 20:40.49 | awannabe | like if extension 200000 dials out, caller id is set to 1234565555 |
| 20:41.34 | awannabe | EmleyMoor: well the PRI can handle is, just cant figure out the login in *. |
| 20:41.36 | hmmhesays | extension 2000000 can't dial out |
| 20:41.44 | hmmhesays | user 200000000 can dial out |
| 20:42.11 | awannabe | what do you mean? |
| 20:42.30 | hmmhesays | exactly what I said |
| 20:42.40 | awannabe | well sorry, user |
| 20:42.52 | awannabe | im not thinking good today! |
| 20:43.18 | awannabe | confused how this if statement works |
| 20:43.59 | hmmhesays | exten => s,1,Set(foo=${IF($[ ${x} = 7]?tval:fval)}) |
| 20:44.01 | hmmhesays | like that |
| 20:45.01 | hmmhesays | exten => s,1,Set(foo=${IF($[ ${CALLIERID(num)} = 2000]?1800BOOBIES:${CALLERID(num)})}) |
| 20:45.09 | hmmhesays | whoops forget that one |
| 20:45.20 | hmmhesays | exten => s,1,Set(CALLERID(num)=${IF($[ ${CALLIERID(num)} = 2000]?1800BOOBIES:${CALLERID(num)})}) |
| 20:45.23 | hmmhesays | there that is better |
| 20:45.24 | *** join/#asterisk Kyler (n=chatzill@74.132.227.26) |
| 20:45.47 | hmmhesays | in that case if the callerid that came in was 2000 it would be reset to 1800BOOBIES otherwise it would stay the same |
| 20:45.56 | Kyler | I recently started getting "SIP/2.0 484 Address Incomplete" when trying to place calls through Telesthetic. |
| 20:46.45 | Kyler | I'd been using "17657421212" but I've tried "+17657421212" and "7657421212" with no luck. |
| 20:46.55 | hmmhesays | awannabe: hows that? |
| 20:47.09 | awannabe | i think so! just gotta try it! |
| 20:47.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:47.32 | hmmhesays | ok i'm going to play battlefront II now |
| 20:48.01 | *** join/#asterisk ToTo (n=ToTo@host108-163-dynamic.2-87-r.retail.telecomitalia.it) |
| 20:48.13 | awannabe | hmmhesays: thanks for the help, ill mess with that! |
| 20:49.15 | hmmhesays | awannabe: np it'll work |
| 20:49.22 | *** join/#asterisk jsolares (n=jsolares@206.113.229.70) |
| 20:49.49 | russellb | hmmhesays: that kind of Set() line is why people should use AEL :-p |
| 20:50.06 | hmmhesays | not if he is only doing it for one callerid |
| 20:50.17 | hmmhesays | then changing your whole dp over to use ael would be a pain in the ass |
| 20:50.30 | hmmhesays | imo |
| 20:50.31 | awannabe | yeah i want the easiet! lol |
| 20:50.40 | russellb | i have this evil side of me that really enjoys seeing complicated lines like that, ha |
| 20:50.48 | file | O.o |
| 20:50.56 | hmmhesays | that one isn't bad at all |
| 20:51.05 | hmmhesays | something with LEN in it ARGH |
| 20:51.10 | russellb | lol |
| 20:51.10 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-132-114.dsl.tul2ok.sbcglobal.net) |
| 20:51.21 | hmmhesays | or multiple comparisions |
| 20:51.37 | hmmhesays | anyhoo, i'm about to take this rebel scum base |
| 20:51.46 | awannabe | hmmhesays: yeah it works, thanks alot! |
| 20:51.51 | hmmhesays | awannabe: np |
| 20:51.58 | *** join/#asterisk henrique (n=henrique@201-43-33-140.dsl.telesp.net.br) |
| 20:52.10 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-132-114.dsl.tul2ok.sbcglobal.net) |
| 20:53.18 | awannabe | if for some reason thats typed wrong, it wont crash * on a realod, right? |
| 20:54.01 | *** join/#asterisk ternaryworks (i=ternaryw@dynamic-216-211-45-248.tbaytel.net) |
| 20:54.12 | ternaryworks | Hi all! |
| 20:55.13 | awannabe | hmmhesays: now except if its doest match, it only passes the callerid number, not name to |
| 20:55.38 | awannabe | oops |
| 20:55.40 | awannabe | i see! lol |
| 20:56.36 | ternaryworks | I'm trying to execute some basic commands via a PHP AGI script, anybody have any experience with this? It's probably en easy thing I'm overlooking |
| 20:58.10 | *** part/#asterisk philippel (n=p_lindhe@c-24-16-243-129.hsd1.wa.comcast.net) |
| 20:58.56 | Delta239 | hey quick question.. how to load the ztdummy?? |
| 20:59.37 | Nugget | that's a quick question but not a quick answer. |
| 20:59.59 | Nugget | perhaps you'd be better off taking a look at the docs, giving it the old college try, and then coming to the channel if you have a specific problem? |
| 21:00.10 | Qwell[] | not the new college try? |
| 21:00.34 | Nugget | I don't trust that "new math" stuff. |
| 21:00.41 | russellb | short answer. 1) Install Zaptel. 2) as root, modprobe ztdummy |
| 21:02.05 | Delta239 | thanks |
| 21:02.58 | ternaryworks | Well, if anyone knows anything, I'm running a PHP script via AGI and it'n not passing in or out the information, though I know the PHP script is executing as it leaves a log. |
| 21:03.38 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-215-87.cablep.bezeqint.net) |
| 21:05.14 | vooduhal | Question all. If I have two SIP phones on a call between them (alaw) and they are set to reinvite=no, how much bandwidth would be used by the pbx for this call? |
| 21:07.26 | vooduhal | And can you guys see any reason to use reinvite=no when all of the phones are on the same network segment? Would have any effect on echo that we may be having? |
| 21:07.53 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 21:10.46 | *** join/#asterisk vIR_uS (n=I@p54B16A48.dip.t-dialin.net) |
| 21:11.58 | vIR_uS | hi. I have a short question. I googled for about 2 hours but I could not find a simple hint. Is it possible to send/receive chat messages over IAX? If it's possible to send SMS, it should be so, or? Which softphone supports this? |
| 21:12.51 | russellb | well... the protocol supports TEXT frames |
| 21:13.01 | russellb | but i'm not aware of any softphones that have good support for using it |
| 21:14.31 | vIR_uS | hmmm |
| 21:15.08 | vIR_uS | I would not care if they didnt have good support. I'd be glad to find one with any support. |
| 21:15.21 | vIR_uS | any text support at all |
| 21:16.16 | vIR_uS | Would a switch to SIP make anything better? |
| 21:16.57 | Qwell[] | seems like it'd be fairly straight-forward to add it to an existing IAX2 softphone |
| 21:17.31 | Qwell[] | it would already support different frame types, so adding one more would be simple, then...a textbox for reading/writing text :p |
| 21:17.51 | *** join/#asterisk Dr-Linux|home (n=nono@202.59.73.131) |
| 21:17.56 | Qwell[] | (okay, maybe not THAT easy, but still) |
| 21:18.16 | vIR_uS | I'll download the IAX spec and kiax sources. Let's give it a try... but not before next weekend :-( |
| 21:18.18 | Dr-Linux|home | hey Qwell[] :) |
| 21:18.37 | Dr-Linux|home | Qwell[]: what about Sergio new chan_sccp version? or patch for old one? :) |
| 21:18.44 | Qwell[] | nothing yet |
| 21:18.55 | vIR_uS | that could be faster than waiting for anyone else to implement that |
| 21:19.43 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 21:22.56 | *** join/#asterisk backblue (n=moo@87-196-32-242.net.novis.pt) |
| 21:23.26 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.18) |
| 21:23.35 | TripleFFFF | any way to sip reload one peer only ? |
| 21:23.39 | *** join/#asterisk TheCopsss (n=henri@207.164.28.98) |
| 21:23.41 | TripleFFFF | or does sip reload drop active calls ? |
| 21:23.49 | Qwell[] | sip reload doesn't affect calls |
| 21:24.13 | TripleFFFF | hmm wont it all reset register to now().. then i get hammered with them ? |
| 21:24.40 | TripleFFFF | and any idea on how to USER1 -> AST1 -> AST2 -> USER2 |
| 21:24.44 | TripleFFFF | i get 407's |
| 21:24.46 | awannabe | exten => _XX,2,Set(CALLERID(ALL)=${IF($[ ${CALLERID(num)} = 400]?Foo BAR<77777777>:Set(CALLERID(all)=Other Call<111111111>))} |
| 21:24.54 | awannabe | anyone know why that command hates me and wont work? |
| 21:26.59 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 21:27.31 | *** join/#asterisk dr0ne (n=fn@S01060016b6b541d2.va.shawcable.net) |
| 21:29.25 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 21:29.49 | elriah | Hey guys, I've been looking (voip-info, google) and can't find the column layout for Master.csv, anyone have a link? |
| 21:35.00 | jsolares | elriah: look at cdr/cdr_csv.c in the asterisk sources |
| 21:38.14 | elriah | heh... docs are now in the source code? lol, thanks jsolares ... |
| 21:38.59 | *** join/#asterisk hads_ (n=hads@mail.nice.net.nz) |
| 21:39.16 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:39.20 | *** join/#asterisk xnon_ (n=xnon@200.8.84.40) |
| 21:40.23 | jsolares | well it's documented at the beginning of that file, and since that's the file that writes it out... you might want to add to the voip-info site with that info |
| 21:42.57 | BSDTech | anyone here have the polycom 2.0.3 firmware ? |
| 21:43.55 | *** part/#asterisk Mother_ (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 21:50.47 | sahafeez | strange issue. i transfered a call to the general line 4201, which goes to 4201 voice mail if not picked up. the msg light is on for 4 phones now for that one voicemail, 4202,04,03 |
| 21:52.30 | danbrwn | registration failed for 192.168....50 [DansLaptop], type=friend, host=192.168...50,secret=1234,dtmfmode=rfc2833,mailbox=101,context=sip,callerid=" |
| 21:52.52 | danbrwn | "DansLaptop" <101> |
| 21:54.02 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
| 21:56.45 | fiber0pti | Has anyone tried AsteriskNOW? |
| 21:57.07 | Nugget | I'll try it LATER. |
| 21:57.39 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 21:58.40 | mog | i tried it earlier |
| 21:59.24 | Nugget | when it was AsteriskTHEN? |
| 22:02.13 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-69.dynamic.qsc.de) |
| 22:02.20 | fiber0pti | I can't get it to work |
| 22:02.32 | fiber0pti | GUI comes up but there's a bunch of "undefined" errors everywhere |
| 22:04.18 | Juggie | sahafeez, all your sip peers share the same mailbox, 101 for MWI |
| 22:04.38 | *** join/#asterisk Deeewayne (n=dwayne@adsl-070-145-146-225.sip.mgm.bellsouth.net) |
| 22:04.54 | sahafeez | yup. figured that out a few mins after i posted. made a ton of changes and did not think it thru. thanks for you help! |
| 22:12.39 | *** join/#asterisk remmo (n=chatzill@202.172.106.161) |
| 22:12.54 | *** part/#asterisk remmo (n=chatzill@202.172.106.161) |
| 22:16.24 | hmmhesays | haha vagina bolton |
| 22:17.19 | sahafeez | strange one. ever since i updated to 1.2.13 i see this on the console even tho the call transfers Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.119.59 |
| 22:17.48 | *** join/#asterisk ac3c (n=chatzill@wuser144-shapiro.umnet.umich.edu) |
| 22:21.02 | sahafeez | are there any wireless sip phones for a home. something that i do not have to plug into an ethernet but can use 802.11 |
| 22:21.44 | Juggie | yes. |
| 22:21.50 | Juggie | none of them are 'amazing' |
| 22:21.54 | Juggie | but there are a few decent ones. |
| 22:22.06 | DrCron | and they cost a bundle |
| 22:22.34 | DrCron | i would realy like to see a decent sip or iax -> dect gateway |
| 22:22.59 | *** join/#asterisk Lennart (n=Lennart@LONDON14-1168104322.sdsl.bell.ca) |
| 22:23.12 | sahafeez | Juggie: an module you can point me @ |
| 22:23.24 | *** join/#asterisk AuPix (n=root@adsl-04-85.abel.net.uk) |
| 22:23.29 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:24.54 | reza_ | what does this mean : chan_iax2.c:6907 socket_read: Rejected connect attempt from 204.11.194.34, who was trying to reach 's@' |
| 22:25.40 | DrCron | reza_, someone from 204.11.194.34 tried to call in to extension s@ |
| 22:26.52 | *** join/#asterisk Nate_ (n=Nate@208.52.141.138) |
| 22:29.41 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 22:34.22 | sahafeez | anyone ever use the DPH-541 (Dlink) sip phones |
| 22:34.49 | vIR_uS | why does iax2 only allow html instead of xml? |
| 22:35.14 | mog | ? |
| 22:35.27 | mog | html frames are supported through out asterisk |
| 22:35.28 | Corydon-w | vIR_uS: what makes you think it only allows html? |
| 22:35.29 | mog | xml arent |
| 22:35.41 | vIR_uS | http://www.ietf.org/internet-drafts/draft-guy-iax-02.txt |
| 22:35.42 | mog | send_html is a supported frame type |
| 22:35.51 | Corydon-w | You're certainly welcome to embed xml in the frame type |
| 22:36.07 | Corydon-w | although some things may not interoperate |
| 22:36.27 | vIR_uS | if xml was allowed, html would also be ok, or? |
| 22:36.51 | Corydon-w | The intent of the html was to send web-viewable content |
| 22:36.57 | mog | yes |
| 22:37.03 | mog | gnophone supported it |
| 22:37.21 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 22:37.30 | Corydon-w | vIR_uS: no, because html is not strictly xml-compliant |
| 22:38.21 | Corydon-w | vIR_uS: what precisely are you trying to do that you cannot? |
| 22:39.28 | vIR_uS | i was (crazily) thinking of using iax for a whole peer2peer-groupware-solution. an i thought there would be much more possibilities with xml. |
| 22:39.56 | vIR_uS | or for filesharing |
| 22:40.16 | vIR_uS | but this would need a solution to transport binary data |
| 22:40.16 | Corydon-w | UDP is not a good protocol for filesharing |
| 22:40.53 | Corydon-w | IAX2 is built to be realtime, not built for ensuring that all packets transmitted are received |
| 22:41.11 | vIR_uS | oic |
| 22:41.15 | Corydon-w | Therefore, it's pretty well useless for file sharing |
| 22:41.46 | Corydon-w | unless you don't mind having files with 1500byte holes in the center |
| 22:42.32 | *** join/#asterisk Omer (i=Omer@202.133.79.19) |
| 22:43.12 | vIR_uS | i've gone through iaxclient and kiax. iaxclient supports text frames. |
| 22:43.30 | vIR_uS | as kiax makes use of it, it shouldn't be that hard to implement that |
| 22:43.36 | Corydon-w | Text frames are intended for use as instant messages |
| 22:43.50 | vIR_uS | but no softphone supports it |
| 22:44.00 | Corydon-w | except gnophone |
| 22:44.03 | vIR_uS | this is what i asked some minutes ago |
| 22:44.35 | Corydon-w | The IAX protocol was developed around gnophone, as a multimedia endpoint |
| 22:44.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:44.41 | DrCron | do any of you know if there is a dect gateway for iax? |
| 22:45.27 | Omer | i can not hear anything when i connect to my office * machine |
| 22:45.33 | Omer | neither music on hold |
| 22:48.56 | hads | Heh, donut files. |
| 22:54.58 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 22:55.31 | *** join/#asterisk tim27dr (n=tim27@97-70.dr.cgocable.ca) |
| 22:56.41 | tim27dr | any consultant here to resolve a incoming DID problem from a sip account provider i can pay by paypal |
| 22:59.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:59.36 | spyda | disconnect |
| 22:59.41 | spyda | exit |
| 22:59.49 | spyda | wow, I suck. sorry |
| 23:00.01 | puzzled | hi |
| 23:00.16 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:00.28 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 23:01.04 | *** join/#asterisk TheCops (n=henri@207.164.28.98) |
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| 23:03.19 | tim27dr | my proglem is that my sip provider say than cant pass their DID info trought the INVITE field from the sip header... they always pass the principal DID of the account in the invite field ie INVITE 18198502500@192.168.1.101 and they say i can get the did info from the TO: field... ie TO: 18188502523@192.168.1.101 ... any way of doing that |
| 23:03.41 | Juggie | have you tried looking @ all the variables of a channel |
| 23:03.43 | Juggie | when the call is up |
| 23:03.46 | Juggie | its probally in there somewhere |
| 23:12.15 | BSDTech | any one know is asterisk does channel wisper |
| 23:12.34 | BSDTech | where I can whisper to the exten but not the person he is talking to |
| 23:12.44 | *** join/#asterisk Chris-NB (n=chris@62.99.152.178) |
| 23:14.44 | *** join/#asterisk blick (n=gallaghe@71.12.199.187) |
| 23:15.59 | *** part/#asterisk blick (n=gallaghe@71.12.199.187) |
| 23:16.01 | *** join/#asterisk blick (n=gallaghe@71.12.199.187) |
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| 23:26.36 | vIR_uS | i'll go to bed, good n8 all |
| 23:28.02 | EmleyMoor | n8? Are you Hornsey? <g> |
| 23:28.54 | *** part/#asterisk Kerry_G_JR (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
| 23:29.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 23:31.23 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:38.55 | EmleyMoor | Is there a way to write a "ringback test" in *? |
| 23:39.19 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-149-57-198.mia.bellsouth.net) |
| 23:39.43 | EmleyMoor | (by that I mean a number is dialed on a channel, a reorder tone is given and the channel is rung back when it becomes free - the reorder tone being played again when answered) |
| 23:43.31 | [TK]D-Fender | EmleyMoor : Sort of. You could use 1 script to get the "call-back" #'s, and then generate a call file for it starting with chan-local. Then if the real target succeeds a ChanIsAvail, it could generate a "real" call file, or just patch a double-ended local channel based one. |
| 23:43.34 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:44.15 | [TK]D-Fender | EmleyMoor : Of course you'd have to poll constantly (or on manager event). Ugly for sure |
| 23:44.46 | aptura | looks like the editor body of Cnet has been found. |
| 23:44.49 | EmleyMoor | Hmmm... |
| 23:45.16 | EmleyMoor | Still, I seem to have three lines here now all capable of making 3-wire phones ring :-) |
| 23:45.38 | [TK]D-Fender | aptura : That verdict has been flip-flopping on Digg for the alst while. Nothing conclusive on the final condition last I checked |
| 23:46.03 | aptura | [TK]D-Fender what are you talking about ? |
| 23:46.33 | JT | editor body of cnet? |
| 23:46.37 | aptura | I was talking about the body of james kim, editor of CNET was found in the Oregon forest. Its been all over the news. |
| 23:46.47 | hads | Aparently he was missing. |
| 23:46.58 | JT | you mean the body of the cnet editor then |
| 23:47.02 | aptura | his body was just found about a hour ago. |
| 23:47.09 | JT | right, perhaps in the US it's all over the news |
| 23:47.21 | hads | Yeah. |
| 23:48.07 | aptura | I have SAR training and one of the things you should not try to do is leave your vehicle. It is almost impossible to spot a person from the air in a sar mission. Vehicles are alot easier to see from the air. |
| 23:48.17 | hads | OK |
| 23:48.33 | aptura | Also bring emergency food and gear incase your stuck off the betten path. |
| 23:48.55 | JT | and NBC suits |
| 23:48.59 | JT | just in case |
| 23:49.14 | aptura | I have my land cruiser and hamradio gear so If this ever happened to me it would not be to hard to contact someone. |
| 23:49.24 | hads | So.. the situation in Fiji is dissapointing. |
| 23:49.45 | JT | not as disappointing as the siutation in iraq though |
| 23:49.58 | aptura | IRAQ is a failuer. |
| 23:50.02 | aptura | Failure :) |
| 23:50.42 | [TK]D-Fender | aptura : Not after GWB redefines the word "success" ;) |
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