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| 00:21.34 | jdoe | ugh, I'm an idiot, that took longer than it should have... variable inheritance is confusing though. |
| 00:21.50 | jdoe | if you set _VAR, it makes sense to access it as _VAR, not VAR |
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| 00:38.11 | *** join/#asterisk benklop (~bklopfens@little-black-box.vmware.com) |
| 00:39.01 | benklop | hi, i'm having problems using the gtalk channel for incoming calls.. rather new to asterisk.. |
| 00:39.20 | benklop | i keep getting the message [Oct 27 20:37:09] WARNING[14834] pbx.c: Channel 'Gtalk/+13039956970-ddb6' sent into invalid extension 's' in context 'default', but no invalid handler |
| 00:39.44 | benklop | but i can't figure out where context "default" is specified |
| 00:40.00 | thehar | in extensions.conf |
| 00:40.14 | benklop | i don't have a default context in extensions |
| 00:40.17 | benklop | do I need one? |
| 00:41.05 | benklop | i have an inbound context, and that's what the gtalk config file is set up to set as the context for inbound messages |
| 00:41.18 | benklop | er, calls |
| 00:44.04 | benklop | pretty sure I just don't know what i'm doing. how do I route an incoming call to all extensions? |
| 00:47.05 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 00:47.09 | benklop | is there a primer / beginning doc that I could read |
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| 01:21.25 | eternis | hi |
| 01:21.40 | eternis | anyone know why I get this message? --> NOTICE[10299]: chan_sip.c:21917 handle_incoming: Unknown SIP command 'WAKEUP' from '127.0.0.1' |
| 01:23.02 | eternis | that shows up when starting linphone |
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| 01:33.39 | Pio | anyone successfully using gizmo5 for incoming calls on asterisk 1.8? |
| 01:33.44 | Pio | banging my head in to the wall |
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| 01:38.57 | fink | G723 is the lowest bandwidth codec, correct? |
| 01:41.48 | p3nguin | No, GSM is. |
| 01:43.27 | p3nguin | I think, anyway. |
| 01:44.22 | p3nguin | gsm is 13.2 kbps |
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| 01:48.27 | fink | p3nguin: it seems G729 & 723 are lower? |
| 01:49.16 | p3nguin | G.729 is 31.2 kbps. |
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| 01:49.54 | p3nguin | G.723.1 (6.3 Kbps) is 21.9 Kbps |
| 01:50.00 | fink | p3nguin: ah yes |
| 01:50.07 | p3nguin | G.723.1 (5.3 Kbps) is 20.8 Kbps |
| 01:51.58 | fink | G.729.1 is 8-32 kbit/s |
| 01:52.06 | fink | http://en.wikipedia.org/wiki/G.729.1 |
| 01:57.11 | Pio | whats the best way to debug sip registration? |
| 01:57.52 | p3nguin | sip set debug on |
| 01:58.00 | Pio | i have a 'register' line in sip.conf [general], 'sip show registry' shows 0 registrations, doing "sip set debug on" only shows an OPTIONS that is sent to the gizmo service being responded to with a 404 |
| 01:58.51 | p3nguin | I don't know if Asterisk supports G.729.1. |
| 01:59.41 | p3nguin | pio: Show me your register statement. Mask your password so no one steals it. |
| 02:00.36 | p3nguin | Oh... |
| 02:00.39 | Pio | http://pio.longstair.com/misc/sip_debug.txt |
| 02:01.08 | p3nguin | pio: Maybe Gizmo doesn't accept SIP registration. I just checked and I'm not using a register statement for them. |
| 02:01.28 | Pio | isnt a registration necessary to receive incoming calls? |
| 02:01.36 | p3nguin | not necessarily. |
| 02:02.00 | Pio | gizmo has this doc but its old .. http://support.gizmo5.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=191 |
| 02:02.17 | p3nguin | The registration is used to tell a remote system how to contact you. If you configure it statically on that remote system, you wouldn't need to register. |
| 02:02.30 | Pio | hmm |
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| 02:03.07 | p3nguin | Your register syntax looks okay, though. |
| 02:03.42 | Pio | you want to see the sip set debug on output? |
| 02:04.00 | *** join/#asterisk radic (~radic@178.2.223.200) |
| 02:04.26 | Pio | http://pio.longstair.com/misc/sip_debug2.txt |
| 02:05.17 | p3nguin | Assuming you are forwarding all calls to your Asterisk system anyway, just go into the call forwarding tab in your gizmo5 account and mark "Forward all calls" and then mark SIP and add your SIP URI in the box. |
| 02:05.46 | Pio | oh yeah |
| 02:05.55 | Pio | that should work |
| 02:05.58 | p3nguin | Then you don't need to register. |
| 02:06.05 | *** join/#asterisk tengulre (~tengulre@125.71.208.16) |
| 02:06.17 | p3nguin | As long as you have a dynamic DNS host name for your SIP URI, it should always work. |
| 02:06.25 | Pio | i have a static ip i can use |
| 02:06.32 | p3nguin | If you have static, that's even better! |
| 02:06.55 | p3nguin | You certainly don't need registration if you have a static address. |
| 02:07.20 | Pio | huh i commented the register line and it still does the options/404 traffic.. i guess thats from the qualify=yes .. |
| 02:07.59 | p3nguin | Maybe. I wouldn't think they would 404 your OPTIONS packet, though. |
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| 02:17.47 | Pio | p3nguin, that did it, thanks |
| 02:18.43 | tengulre | how to using sed print lines 1000 to 2000 in file a? |
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| 02:21.50 | oss2all | Hello |
| 02:22.26 | oss2all | I'm trying to make a case for asterisk as an IVR over Dialogic's IVR solution |
| 02:22.42 | oss2all | Any one have market share information on the two? |
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| 02:26.12 | [hC] | Any one here familiar with netgear fs728tp switches and multicast traffic (specifically from an aastra phone) |
| 02:26.28 | [hC] | I have configured this before, and it worked fine, and now im getting absolutely butt kiss out of it |
| 02:29.43 | v1s | I have a few different countries in my dialplan and to call them I have the option to dial +countrycode number OR countrycode number I just duplicate the lines is there easier better way to do this? http://pastebin.com/eJJmJc09 |
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| 02:33.33 | *** join/#asterisk eternis (~proba@cpe-67-244-127-222.nyc.res.rr.com) |
| 02:33.35 | eternis | hey |
| 02:34.09 | eternis | I got a problem with linphone showing up as following --> 3000 (Unspecified) D 5060 Unmonitored |
| 02:34.58 | eternis | is there an option in asterisk to trigger a client for authentication? for instance to prompt for a password? |
| 02:35.20 | *** part/#asterisk oss2all (~dcabot@fl-71-54-237-119.dhcp.embarqhsd.net) |
| 02:35.30 | WIMPy | v1s: You culd match +. and goto EXTEN:1 |
| 02:35.47 | eternis | the odd thing is that I can still make phone calls with linphone despite as showing as Unspecified |
| 02:36.07 | *** part/#asterisk fink (~guest@173-133-86-217.pools.spcsdns.net) |
| 02:36.07 | v1s | eternis: try puting qualify=yes in ur sip for that exten? |
| 02:36.22 | WIMPy | Registration is only for calls from Asterisk to the peer. |
| 02:36.40 | eternis | v1s: alright |
| 02:38.44 | v1s | WIMPy: thanks. |
| 02:39.00 | *** join/#asterisk fink_ (~guest@173.158.145.240) |
| 02:39.20 | eternis | odd, doesn't work |
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| 02:39.43 | eternis | I mean linphone completely ignores the password and goes ahead making calls just fine |
| 02:39.53 | *** part/#asterisk fink_ (~guest@173.158.145.240) |
| 02:40.05 | eternis | despite this --> 3000 (Unspecified) D 5060 UNKNOWN |
| 02:40.31 | eternis | also why is showing port 5060? |
| 02:41.41 | WIMPy | eternis: So you have allowed anyone to connect anonymousely? |
| 02:41.49 | WIMPy | Or does the peer actually show up if you place a call? |
| 02:42.19 | eternis | shows up regardless |
| 02:42.22 | WIMPy | Again, that has nothing to do with registration. That's only to tell the server where to find the user. |
| 02:42.31 | eternis | which is the option to allow anonymously? |
| 02:42.43 | WIMPy | allowguests |
| 02:43.05 | WIMPy | allowguest |
| 02:43.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 02:44.47 | eternis | I have it as this --> allowguest=no |
| 02:44.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:46.25 | WIMPy | So theat means that you would only accept authenticated calls. |
| 02:49.48 | eternis | :) --> Registered SIP '3000' at 192.168.7.2 port 5068 NOTICE[20669]: chan_sip.c:18378 handle_response_peerpoke: Peer '3000' is now Reachable. (979ms / 2000ms) |
| 02:49.56 | eternis | that was using freaking command line |
| 02:50.03 | eternis | linphonec |
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| 02:51.26 | eternis | WIMPy: is there, possibly, another option to trigger password prompt? |
| 02:51.49 | WIMPy | What do you mean by promt? |
| 02:52.29 | eternis | ok, ekiga, and empathy have a place to add the password in the preferences. |
| 02:53.09 | eternis | linphone doesn't, apparently when you try to connect will prompt for user name and passwerd |
| 02:53.49 | WIMPy | Sounds interesting. |
| 02:55.55 | eternis | the interesting part is how soft-phones implement stuff in a wildly different ways. |
| 02:56.24 | eternis | now I got three soft-phones working with asterisk :) linphone, ekiga and empathy |
| 02:56.30 | WIMPy | Well, the days of things just working are gone :-( |
| 02:57.15 | eternis | what days were those? |
| 02:58.30 | WIMPy | The ones before voip. |
| 02:58.37 | eternis | ROFL!! |
| 03:00.30 | vinhdizzo | eternis: I see u've been making progress |
| 03:00.41 | vinhdizzo | r u connected to a router? |
| 03:01.22 | eternis | vinhdizzo: most likely |
| 03:01.36 | vinhdizzo | what do u mean most likely? |
| 03:01.48 | eternis | vinhdizzo: thanks, but I think the real challenge will be outbound calls and setting up the inbound stuff |
| 03:02.16 | WIMPy | Actually it already became a little more complicated with competition. But looking back that was absolutely hermless. |
| 03:02.17 | eternis | --> |
| 03:02.28 | eternis | nameserver 192.168.0.1 |
| 03:03.11 | vinhdizzo | eternis: actually i dont think so. read the asterisk book's dialplan section. i think once ur connected, editing extensions.conf is easier. |
| 03:03.31 | vinhdizzo | at least for me...i still haven't figured out sip.conf, the networking part |
| 03:04.11 | eternis | nowadays WHO is not behind a router |
| 03:04.23 | eternis | at least from a home connection |
| 03:05.00 | eternis | WIMPy: just today I read this guy rant --> http://xmppjingle.blogspot.com/2010/10/good-bad-and-ugly.html |
| 03:05.39 | eternis | vinhdizzo: why, are you behind a router? |
| 03:05.49 | vinhdizzo | eternis: yea. |
| 03:06.07 | eternis | well, I was first testing it all out internally |
| 03:06.33 | eternis | so I just call myself from the same computer and talk to myself with two clients |
| 03:06.41 | eternis | a bit odd but... |
| 03:07.08 | WIMPy | never even considered using skype for obvious reasons (some seem to have chosen to overlook for some time). |
| 03:07.29 | vinhdizzo | eternis: so u connect to localhost? |
| 03:07.52 | eternis | nope, to my ethernet nic |
| 03:08.20 | vinhdizzo | eternis: so a real voip phone, not softphone? |
| 03:09.03 | p3nguin | eternis: Having a peer registered has nothing to do with it being able to make calls. |
| 03:09.15 | eternis | WIMPy: in order to do like skype I would have to install each asterisk to each friend and family member right? in order to do the video call and all. |
| 03:09.49 | WIMPy | pardon? |
| 03:10.18 | eternis | make videocalls |
| 03:10.31 | p3nguin | Asterisk is a B2BUA. Why do you need more than one per group of phones? |
| 03:11.13 | WIMPy | For video calls, you need clients capable of that. An Asterisk in the middle might help but technically isn;t neccessary. |
| 03:11.38 | eternis | for example how would a friend let's say in Europe reach me, since the sip address would be my internal ip. |
| 03:12.01 | p3nguin | Put it on the public internet or forward your port like you would for any other service behind the NAT. |
| 03:12.11 | WIMPy | Configure port forwarding on your router. |
| 03:13.16 | eternis | so what would they put in their soft-phone client? |
| 03:13.17 | p3nguin | UDP port 5060 and typically UDP ports 10000-20000 should take care of it. |
| 03:13.29 | p3nguin | They would use your public IP address. |
| 03:13.46 | p3nguin | or a host name that has proper DNS for your public IP address. |
| 03:14.23 | eternis | ohhh! this ?? --> 67.244.127.222 |
| 03:14.41 | p3nguin | That would be your public IP address. |
| 03:15.13 | eternis | so they would have to type sip:friend@67.244.127.222 |
| 03:15.23 | eternis | :\ it's confusing |
| 03:15.25 | p3nguin | If you forward the prescribed ports, connection to the public address will get translated to the internal address of Asterisk. |
| 03:16.10 | p3nguin | Normal people pick up the phone and call the extension number that dials another phone on the system. |
| 03:16.58 | p3nguin | Dialing by SIP URI isn't necessary when the phones are all associated on the same system. |
| 03:17.48 | p3nguin | And if they aren't authenticated to your system and they do dial by SIP URI, your Asterisk will reject the call because you have set allowguest=no. |
| 03:21.18 | eternis | aha |
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| 03:32.25 | *** join/#asterisk MCIML (~ZiKi@static-67-62-120-42.t1.cavtel.net) |
| 03:32.45 | MCIML | Who is at astricon? |
| 03:34.29 | Kobaz | me |
| 03:35.29 | MCIML | did you go to bobby mckeys tonight? |
| 03:36.11 | Kobaz | yeah |
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| 03:37.04 | MCIML | welcome to the after-after party! |
| 03:37.44 | jsgoecke | I am |
| 03:37.47 | jsgoecke | @ Astricon |
| 03:38.38 | MCIML | <PROTECTED> |
| 03:38.53 | jsgoecke | Yes, great conference this year |
| 03:38.55 | jsgoecke | Was at McKeys |
| 03:39.03 | jsgoecke | But the dueling pianos were just too loud |
| 03:39.12 | jsgoecke | Afraid I may have no voice for my talk tomorrow =( |
| 03:39.39 | MCIML | funny... we were just saying how we had such a good time talking to people that we didnt eve notice the pianos |
| 03:39.49 | jsgoecke | Seriously? |
| 03:39.55 | jsgoecke | They were hard not to notice |
| 03:40.14 | Kobaz | #astricon |
| 03:40.25 | Kobaz | yeah, the piano was waaay too loud |
| 03:40.37 | Kobaz | i was thinking of even asking them to turn it down, it was pretty obnoxious |
| 03:41.23 | MCIML | well it was hard to understand what they were saying... |
| 03:42.42 | Kobaz | and the crew that were singing... i think his name is malcom? i think their mic was off |
| 03:49.20 | pabelanger | Yar, too loud |
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| 05:04.41 | sshock | what will I lose if I disable the speex codec? |
| 05:04.48 | drmessano | speex |
| 05:05.31 | sshock | I don't think I'm even using that; how do I know? |
| 05:05.42 | Nugget | "Hey Doc, it hurts when I do this" |
| 05:06.27 | sshock | I think I'm only allowing gsm and ulaw... |
| 05:06.41 | drmessano | Um |
| 05:06.53 | sshock | the reason I bring this up is because apparently speex is locking my /dev/dsp |
| 05:07.14 | drmessano | If you're only allowing those two, then you're only allowing those two |
| 05:07.15 | sshock | which I don't understand because I don't need asterisk to play sound on my linux server |
| 05:07.39 | sshock | ok, goodbye speex.... |
| 05:07.46 | drmessano | Asterisk isn't going to lock /dev/dsp because speex is enabled. I think you've got some wires crossed here |
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| 05:08.34 | drmessano | It sounds like you accidentally the whole thing here |
| 05:09.13 | sshock | ok, just put noload=codec_speex.so, and that solved it |
| 05:09.31 | sshock | well, it beats me, but asterisk was loading codec_speex.so, |
| 05:09.49 | sshock | and that was loading libspeexdsp.so or something |
| 05:17.05 | sshock | so I've got asterisk up and running great, and what happens? no one ever calls me... |
| 05:17.42 | sshock | but actually that's partially a good thing ;) |
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| 05:58.29 | ChannelZ | Want calls? Let me give your number to the Democrats... |
| 05:58.33 | ChannelZ | oh he left |
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| 05:58.39 | ChannelZ | mphm |
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| 08:10.36 | FlashDeluxe | Hi@everybody! Does anyone knows a gui program which can create dialplans for asterisk? |
| 08:13.54 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 08:16.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
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| 08:53.56 | zplinux | ~seen tzafrir_laptop |
| 08:54.01 | infobot | tzafrir_laptop is currently on #asterisk-dev (37m 16s) #asterisk (37m 16s) #asterisk-bugs (37m 16s), last said: 'At least the link to /asterisk/acf'. |
| 08:56.01 | tzafrir_laptop | ~seen infobot |
| 08:56.02 | infobot | infobot is currently on #utos (9d 8h 57m 21s) #asterisk-doc (9d 8h 57m 21s) ##t42 (9d 8h 57m 21s) #maemo (9d 8h 57m 21s) #fredlug (9d 8h 57m 21s) ##ols (9d 8h 57m 21s) #flyspray (9d 8h 57m 21s) #asterisk-dev (9d 8h 57m 21s) #webos-internals (9d 8h 57m 21s) #opensimpad.org (9d 8h 57m 21s) #asterisk (9d 8h 57m 21s) #byumug (9d 8h 57m 21s) #wowprogramming (9d ... |
| 08:56.11 | *** join/#asterisk spandi (~chatzilla@59.164.188.32) |
| 08:57.40 | zplinux | haya |
| 08:58.02 | tzafrir_laptop | hi |
| 08:58.05 | zplinux | well, how can I help in solving this ,a and allow openvz to run dahdi? |
| 08:58.18 | zplinux | what more info but the paste shoul I supply? |
| 08:58.34 | zplinux | where can I post about it except for the proxmox forum? |
| 08:59.45 | mark22 | is it "normall" for asterisk to crash daily? |
| 08:59.56 | zplinux | NO! |
| 09:01.09 | mark22 | I am still running 1.6.2.11 (so I should probably upgrade) and maybe I should remove lcdial at that time (if I could think of some other auto failover method) |
| 09:01.36 | *** join/#asterisk screenn (~screenn@users-nat.more.com.ua) |
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| 09:13.45 | tzafrir_laptop | mark22, please report such bugs |
| 09:14.08 | tzafrir_laptop | zplinux, can you build any other modules with those kernel headers? |
| 09:14.17 | zplinux | min pls |
| 09:19.48 | *** join/#asterisk garymc (~chatzilla@host81-139-127-32.in-addr.btopenworld.com) |
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| 09:30.29 | zplinux | fixing the audioable aleret for when I get a replay here |
| 09:30.48 | zplinux | well, i didnt try to build any other headers |
| 09:30.54 | zplinux | well, i didnt try to build any other modules |
| 09:31.05 | zplinux | let me try the NIC |
| 09:31.54 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
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| 09:39.13 | cjk | hello, i am playing with function connectedline on SIP protocol. and i dont see any difference in the traffic that I sniff, as if asterisk would ignore the function. any idea? |
| 09:41.48 | *** join/#asterisk joobie (~joobie@CPE-121-219-110-161.lnse2.lon.bigpond.net.au) |
| 09:42.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:42.41 | anny__ | hey |
| 09:42.56 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 09:43.27 | anny__ | can someone plz give me an accurate description of what zaptel and libpri modules because i can't seem to find any on google |
| 09:43.57 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:44.29 | marksaitis | would anybody know why my client device would not accept an incoming call as soon as I set it up to use srtp? It can make outbound srtp call though with no problems. If I would disable srtp, it would then accept incoming call just fine. |
| 09:45.21 | Chainsaw | marksaitis: It has probably decided that it doesn't like the certificate. |
| 09:45.46 | eMBee | good evening |
| 09:46.37 | Chainsaw | eMBee: Hello. |
| 09:46.41 | eMBee | can asterisk detect the extension that was dialed incoming on an analog trunk line? |
| 09:46.57 | Chainsaw | eMBee: If caller ID is in use, yes. |
| 09:47.18 | mark22 | tzafrir_laptop: where should I report it when there isn't any clear information in the logs? |
| 09:47.26 | marksaitis | Chainsaw, u recon? So srtp cares about certificate as well? I thought its only TLS? Well, for example, if certificate is not configured well, clients like eyebeam do say "Certificate Validation Error" or "Certificate Name Mismatch" in most cases and I spent quite a lot of time on that stuff. |
| 09:47.26 | *** join/#asterisk Arnoz (~Arnoz@91.183.59.105) |
| 09:47.46 | Chainsaw | marksaitis: You expect to get clear error messages without digging? |
| 09:48.01 | Chainsaw | marksaitis: The world of VoIP isn't like that I'm afraid. |
| 09:48.06 | eMBee | can you explain the relationship? caller-id gives me number of the user making the call. and it also gives me the extension the user called? (does it give me the whole number the user dialed too?) |
| 09:48.50 | Chainsaw | eMBee: It depends on what caller ID system is in use and how it is configured. |
| 09:49.04 | eMBee | aha |
| 09:49.12 | Chainsaw | eMBee: Analog lines normally carry just voice. If you want extras like that, you have to add them. |
| 09:49.49 | eMBee | or i have to ask the line provider to add them... |
| 09:50.02 | marksaitis | Chainsaw, well its just pissing me off. I found 2 different guides on asterisk18 tls srtp certificate generation and both are different. If I try one of them, eyebeam wont accept the certificate at all, if I try the other one, it would accept it, but I can not make an srtp call from one eyebeam to another.... I tried those guides 20 times each over and over again, checking clearly every letter etc.... same shit |
| 09:50.31 | kaldemar | ~book |
| 09:50.31 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 09:50.34 | kaldemar | anny__: ^^ |
| 09:51.19 | tzafrir_laptop | mark22, is that crash reproducable? |
| 09:51.39 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 09:51.41 | aberrios_ | any polycom users around? |
| 09:51.47 | anny__ | kaldemar: thx |
| 09:52.08 | *** join/#asterisk rocknfra (~rocknfra@firewall1.cattid.uniroma1.it) |
| 09:53.36 | kaldemar | anny__: zaptel was renamed to DAHDI about 2 years ago, so if you plan on installing something, choose it instead. |
| 09:54.14 | mark22 | tzafrir_laptop: it happens everyday between 7 and 8 (UTC) in the morning, nothing special happens (and the logs don't give information that it did crash at all) |
| 09:56.53 | mark22 | at least for the last few days it happens every morning, but before that it didn't happen that often. I've no idea about where to look for a solution (however I also see that when I look shortly after the crash that the RAM on that system is used for around 80%) |
| 09:57.49 | marksaitis | does anybody know a working guide how to create a working certificate for eyebeam? tls srtp |
| 09:59.37 | Chodorenko | marksaitis:http://www.voip-info.org/wiki/view/SIP+TLS |
| 10:00.26 | Chodorenko | marksaitis: i can try its tomorrow night , Work perfect |
| 10:00.33 | *** join/#asterisk Wassim (~IceChat7@193.227.188.210) |
| 10:01.33 | marksaitis | Chodorenko, thats the guide I used to create current certificates. Client accepts it, but as I said, I can make srtp tls call to the pbx, but if I try to make a call from one client to another, it doesnt accept incoming call |
| 10:04.02 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 10:05.56 | marksaitis | if I use TLS only, everything works perfect, but as soon as I tell client to use srtp for voice, incoming call wont work |
| 10:06.43 | Chainsaw | Right, so other then ChanIsAvail, is anything else broken in 1.8.0? |
| 10:08.37 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 10:08.43 | shamelessn00b | hi all |
| 10:08.49 | Wassim | hi |
| 10:09.03 | k-man | is there a soft phone that does sip on the iphone? |
| 10:09.07 | k-man | or for the iphone? |
| 10:09.14 | shamelessn00b | Looking at asterisk SCF |
| 10:09.22 | Wassim | bria |
| 10:09.43 | shamelessn00b | would there be a separate channel on IRC for SCF? |
| 10:10.37 | Wassim | k-man |
| 10:10.40 | tzafrir_laptop | mark22, did you get any core dump from such a crash? |
| 10:10.44 | Wassim | bria from counterpath |
| 10:10.54 | Wassim | same company that does x-lite |
| 10:11.25 | tzafrir_laptop | shamelessn00b, separate channel? Given the history of dundi? I guess no |
| 10:11.45 | k-man | Wassim: is it any good? |
| 10:11.50 | Wassim | yeah |
| 10:11.52 | Wassim | it is |
| 10:12.01 | k-man | ok thanks, ill give it a go |
| 10:12.05 | Wassim | ;) |
| 10:13.33 | marksaitis | regarding self signed certificates, does a client need to have a public key or private key with public key? |
| 10:14.03 | Wassim | i think both, im not sure |
| 10:14.36 | marksaitis | and, does anyone know, if asterisk needs to have tlscertificate= and tlscafile= ? or only one of them? |
| 10:15.05 | tzafrir_laptop | shamelessn00b, and then there is a question of WTF is SCF? |
| 10:15.08 | tzafrir_laptop | http://duckduckgo.com/?q=SCF |
| 10:15.11 | marksaitis | as one guide sugests that it needs to have one of these in sip.conf and another one sugests that it needs to have both |
| 10:15.41 | shamelessn00b | tzafrir_laptop: http://www.digium.com/en/mediacenter/viewpress/Digium-Introduces-New-Open-Source-Project-Asterisk-SCF-to-Simplify-the-Creation-of-Complex-Communications-Systems |
| 10:15.44 | marksaitis | Wassim, im trying to test tls srtp on iphone bria app, a bit painfull |
| 10:15.58 | marksaitis | but other that that, bria for iphone is awesome app |
| 10:16.16 | Wassim | yup |
| 10:16.20 | Wassim | why srtp? |
| 10:16.43 | marksaitis | why not? I need it in my case |
| 10:17.25 | Wassim | ok |
| 10:17.35 | Wassim | here's a question |
| 10:17.59 | Wassim | i need to make direct media call, between 2 extensions regsitred on 2 asterisk, |
| 10:18.05 | Wassim | any clue? |
| 10:18.13 | Wassim | i tried with dundi |
| 10:18.15 | Wassim | and traced |
| 10:18.16 | marksaitis | it all works fine without srto though, or with srtp for calls to pbx like voicemail. But for calls from client to client, it doesnt work. And nobody from this channel knows why |
| 10:19.00 | Wassim | but each talks to his registrar server |
| 10:20.57 | shamelessn00b | I don't know how to explain it but lemme try, I have a caller that dials a specific extension into asterisk which fires up a typical IVR system, now I originate a call from a local channel to extension xyz and call the chanspy() application on that extension and start listening to the call that landed on the IVR extension, now I want to stream this audio over a TCP socket, problem is, call... |
| 10:20.59 | shamelessn00b | ...to chanspy() app is a blocking call |
| 10:22.07 | tzafrir_laptop | shamelessn00b, yeah. I saw that |
| 10:22.11 | shamelessn00b | The only workaround that comes into my mind right now is to modify the monitor() application to make it write channel audio on a TCP socket instead of a file |
| 10:22.25 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:22.31 | tzafrir_laptop | All reports I saw seem to copy basically the same thing |
| 10:22.51 | shamelessn00b | Well, it looks like a pretty interesting project to me |
| 10:22.55 | tzafrir_laptop | shamelessn00b, now, go and look for more information in http://www.asterisk.org/ as it suggests |
| 10:23.06 | shamelessn00b | already did |
| 10:23.11 | tzafrir_laptop | and? |
| 10:23.31 | shamelessn00b | thereshardly anything of interest overthere |
| 10:23.45 | shamelessn00b | The complete Asterisk SCF documentation and development wiki will be available this afternoon (Wednesday, October 27) |
| 10:23.51 | Chainsaw | Can anyone on 1.8.0 try ChanIsAvail for a non-existent SIP peer please, and check the AVAILSTATUS return? |
| 10:24.19 | eMBee | my phone-provider wants me to dial #1234# to activate a feature on the phoneline, how can i make asterisk accept and dial #? do i need to make a new rule with # in the pattern? since X only covers digits? |
| 10:24.35 | shamelessn00b | can you tell me how to cater the above mentioned problem without writing or modifying an existing module |
| 10:25.15 | shamelessn00b | basically I want to be able to stream audio on a channel in a non blocking fashion |
| 10:26.11 | shamelessn00b | I tried jack_hook() but it doesn't work for me |
| 10:26.22 | shamelessn00b | jack() works fine |
| 10:26.29 | shamelessn00b | but its a blocking call,,, |
| 10:28.55 | shamelessn00b | I don't want to block my dialplan execution in the IVR |
| 10:31.17 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 10:32.21 | adnc | hello, i'm using asterisk 1.4 on a debian installation. i saw a iphone voicemail application that needs ARI (Asterisk Recording Interface) unfortunately I can not find much info about it. Is this atool that I can download somewhere? |
| 10:33.29 | cjk | hi, i got callerid updates working on pickup but not on attended transfer on asterisk 1.8. any idea? |
| 10:34.34 | Chainsaw | downgrades Asterisk from 1.8.0 to 1.6.2.13 as ChanIsAvail is beyond repair |
| 10:34.39 | adnc | unfortunately there is no link on http://www.voip-info.org/wiki/view/Asterisk+GUI page |
| 10:35.15 | shamelessn00b | <.< |
| 10:41.00 | fauxalliance | ~#asterisk-gui |
| 10:41.14 | fauxalliance | ~asterisk-gui |
| 10:41.14 | infobot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 10:41.27 | fauxalliance | support^^^ |
| 10:41.52 | adnc | fauxalliance, was this comment for me? |
| 10:42.13 | fauxalliance | yeah, my aim is a little off... it's still early GMT -3.5 |
| 10:42.36 | adnc | fauxalliance, what is the relation between asterisk gui and ARI? |
| 10:43.51 | fauxalliance | ARI is a front end for recording / reviewing recordings... asterisk uses the aforementioned recordings, or the overlying framework does. |
| 10:45.16 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 10:46.01 | fauxalliance | adnc, that, and I presume you are using the asterisk-gui. When the 'professionals' awake, they will most likely point you there... |
| 10:48.39 | *** join/#asterisk Akiraa (~Akira@92.81.182.216) |
| 10:53.55 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:55.01 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 10:58.52 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 11:00.12 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
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| 11:24.54 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 11:31.26 | eject_ck | Hi all, I'm trying to evaluate FAX for asterisk and have problem - http://pastebin.ca/1975414 |
| 11:31.36 | eject_ck | I need your help folks :) |
| 11:31.53 | eject_ck | I'm getting error FAX session '2' failure, reason: 'fax session timed-out' (TIMEOUT) |
| 11:33.18 | *** join/#asterisk alepaes (~quassel@187.23.21.253) |
| 11:37.38 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 11:38.39 | mark22 | < tzafrir_laptop> mark22, did you get any core dump from such a crash? << no (sorry, was away from screen) |
| 11:47.53 | *** join/#asterisk jsgoecke (~Adium@74.11.214.130) |
| 11:51.07 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:53.49 | eject_ck | I'm using free fax for asterisk |
| 11:55.11 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 11:56.51 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 12:07.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:07.37 | *** join/#asterisk efdetonator (~usuario@189.71.160.85) |
| 12:08.20 | efdetonator | Where do I set the audio codec that my agi script will use? |
| 12:09.57 | [TK]D-Fender | efdetonator: "scripts" don't use codecs, CHANNELS DO. that is in your PEER definition |
| 12:10.01 | fauxalliance | efdetonator, on the trunk / channel that you will call tiwh the codec |
| 12:10.11 | fauxalliance | <PROTECTED> |
| 12:10.31 | fauxalliance | o/ |
| 12:14.04 | efdetonator | thanks |
| 12:14.31 | efdetonator | I've set the users to use the speex codec but it seems that zoiper can't use it ;o |
| 12:14.41 | efdetonator | do you guys know a good soft phone? |
| 12:14.42 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:15.34 | zamba | efdetonator: what os? |
| 12:15.41 | efdetonator | linux |
| 12:15.59 | [TK]D-Fender | ~ekiga |
| 12:15.59 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 12:16.17 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 12:16.27 | efdetonator | thanks |
| 12:16.43 | nicola_pav | hello. i have a question about hylafax functionality |
| 12:16.58 | fauxalliance | on a tangent, but shoot nicola_pav |
| 12:17.09 | nicola_pav | in the dial plan under extensions.conf there are variables EMAILADDR and FAXFILE |
| 12:17.19 | nicola_pav | where do those variables get set? |
| 12:17.47 | fauxalliance | extensions.conf? hylafax? |
| 12:17.59 | [TK]D-Fender | nicola_pav: Hylafax has NOTHING to do with * dialplan <- |
| 12:18.34 | [TK]D-Fender | nicola_pav: Those variables have no relationship to Hylafax in particular. |
| 12:18.42 | fauxalliance | NONE! |
| 12:18.44 | nicola_pav | ok |
| 12:19.12 | nicola_pav | i send a fax and hylafax works fine, answers the call, saves the fax in in tif format |
| 12:19.15 | fauxalliance | my hylafax uses an ANALOG modem on a POTS line and is FAR AWAY from anything asterisk related.... |
| 12:19.17 | nicola_pav | i need to send to an email |
| 12:19.22 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 12:19.30 | nicola_pav | it seems in the dial plan i have EMAILADDR and FAXFILE |
| 12:19.39 | fauxalliance | nicola_pav, look at avant fax... make sure your email server allows smtp relay from the local net |
| 12:19.41 | nicola_pav | but it seems they r not recognized |
| 12:19.42 | [TK]D-Fender | nicola_pav: * doesn't send e-mails either |
| 12:19.48 | [TK]D-Fender | nicola_pav: Make your own script for that |
| 12:19.57 | fauxalliance | [TK]D-Fender, AVANTFAX |
| 12:20.09 | nicola_pav | if i set manually the EMAILADDR |
| 12:20.14 | nicola_pav | it sends an email |
| 12:20.23 | fauxalliance | hooray for sendmail |
| 12:20.25 | nicola_pav | but does not know what to send since FAXFILE is null |
| 12:20.25 | [TK]D-Fender | nicola_pav: "IT"? WTF is "it"? |
| 12:20.53 | [TK]D-Fender | nicola_pav: Asterisk doesn't send e-mails. There is nothing in * that uses port 25 inherently. |
| 12:20.53 | nicola_pav | in the dial plan under extensions.conf |
| 12:20.53 | fauxalliance | nicola_pav, are you USING ASTERISK OR HYLAFAX? not interchangable |
| 12:21.03 | nicola_pav | if i set the variable EMAILADDR to my email directly |
| 12:21.05 | [TK]D-Fender | nicola_pav: and I see no reason for a channel variable being set by hylafax at all here. |
| 12:21.39 | nicola_pav | ok |
| 12:21.47 | nicola_pav | so its asterisk issue? |
| 12:21.54 | nicola_pav | i mean dial plan? |
| 12:22.03 | fauxalliance | FFS |
| 12:22.21 | [TK]D-Fender | nicola_pav: No, it s a "Who the hell said those variables were meant to ever have anything special in them?" issue |
| 12:22.37 | [TK]D-Fender | nicola_pav: You are talking about 2 variable names as if they were SPECIAL |
| 12:22.50 | fauxalliance | 's ford is broken, perhaps if i tinker with the acura the ford will work? |
| 12:22.57 | nicola_pav | ok, got it |
| 12:23.04 | [TK]D-Fender | nicola_pav: Hylafax natively knows NOTHING about Asterisk at all |
| 12:23.14 | fauxalliance | not even a little about IAXMODEM |
| 12:23.16 | fauxalliance | ! |
| 12:23.38 | [TK]D-Fender | fauxalliance: I see I'm beginning to rub off on you analogically :) |
| 12:24.02 | fauxalliance | [TK]D-Fender, indeed fellow countryman |
| 12:25.29 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:25.38 | *** join/#asterisk asilva (~structz@gandalf.ai.unesp.br) |
| 12:27.44 | nicola_pav | hylafax when receiving fax it will save it somewhere |
| 12:27.47 | nicola_pav | right? |
| 12:27.53 | nicola_pav | in tif mode |
| 12:28.11 | nicola_pav | when i want to send it to an email in my dial plan |
| 12:28.26 | nicola_pav | how can i know which file is so i can send it? |
| 12:28.52 | [TK]D-Fender | nicola_pav: Go read hylafax docs to see what it can offer you. This is not *'s problem |
| 12:29.00 | *** join/#asterisk heffer_ (~felix@fedora/heffer) |
| 12:29.10 | [TK]D-Fender | nicola_pav: And HYLAFAX has its OWN e-mail capabilities |
| 12:33.07 | alepaes | Hello... anyone know why CALLERPRES() does not functions in a QUEUE ? |
| 12:33.25 | [TK]D-Fender | alepaes: Show us |
| 12:33.32 | [TK]D-Fender | ~pb |
| 12:33.32 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 12:33.35 | [TK]D-Fender | ^^^ |
| 12:37.23 | alepaes | [TK]D-Fender: In a Dial, setting the CALLERPRES to unavailable, shows CID as Anonymous in X-lite, for exemple. |
| 12:38.01 | [TK]D-Fender | alepaes: SHOW US |
| 12:38.35 | alepaes | if I set the CALLERPRES before a QUEUE(), the CID goes to the agent without modification. |
| 12:39.03 | fauxalliance | alepaes, pastebin the CALL LOG! |
| 12:40.09 | *** join/#asterisk jpmcallister (~EC06113@200.242.28.231) |
| 12:40.28 | fauxalliance | SetCallerPres(allowed) perhaps.... |
| 12:40.33 | alepaes | one moment, but I don't think that the log helps because CALLERPRES, in my knowledge, affects the SIP header |
| 12:40.52 | fauxalliance | s/call log/sip debug... or even BOTH! |
| 12:41.25 | fauxalliance | SetCallerPres is deprecated. Please use Set(CALLERPRES()=allowed) instead. |
| 12:41.35 | fauxalliance | ^^ *1.6 |
| 12:42.57 | fauxalliance | http://www.spinics.net/lists/asterisk/msg134012.html |
| 12:43.00 | fauxalliance | ^^ bug perhaps? |
| 12:43.41 | alepaes | I sent this message... :) |
| 12:44.47 | alepaes | The only way to not show is define the agents as Local channels |
| 12:45.20 | alepaes | And when QUEUE call them, you set the CALLERPRES and Dial |
| 12:45.34 | alepaes | (or Dial() with the 'N' option) |
| 12:45.44 | fauxalliance | ok, so you have two possible workarounds |
| 12:45.54 | fauxalliance | neither is very encumbered |
| 12:51.43 | alepaes | IMHO, it is a very important behaviour in certains conditions (i.e.: ringall strategy, with the agent picking up the call) |
| 12:51.54 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 12:53.25 | jpmcallister | Gugge: I don't know if you remember the problem I was having connecting 2 PBX via 2 * I talked yestarday. Bu I found a solution: I set jitterbuffer=no in iax.conf and now every call is perfect. I don't have a clue why it worked. |
| 12:55.54 | *** join/#asterisk GlobeTrotterz (~chatzilla@190.219.8.59) |
| 12:56.21 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:56.42 | *** join/#asterisk McBoing (~mcboingbo@mail.hrsg.ca) |
| 12:57.36 | McBoing | I want to test out a new version of Asterisk, using 1.4 right now, trying to setup a test extension on the new Asterisk but use the old Asterisk to forward the call, how can I get relaying setup? |
| 12:59.07 | [TK]D-Fender | McBoing: Go look up "asterisk dual servers" on the WIKI |
| 12:59.09 | [TK]D-Fender | ~wikis |
| 12:59.09 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 12:59.09 | fauxalliance | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers @ McBoing |
| 12:59.19 | alepaes | <PROTECTED> |
| 12:59.35 | McBoing | thanks |
| 13:02.51 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:03.17 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 13:04.29 | Chainsaw | 4.6.7.7.0.2.3.3.7.1.4.4.e164.org. 57 INNAPTR100 10 "u" "E2U+SIP" "!^\\+441733207764$!sip:764@linx.net!" . |
| 13:04.36 | Chainsaw | Asterisk seems very opposed to this. Does it look malformed? |
| 13:04.37 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:05.08 | E-bola | Are the 1.6x addons compatible with asterisk 1.8? |
| 13:05.52 | [TK]D-Fender | E-bola: I'd imagine not |
| 13:06.18 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 13:07.19 | E-bola | Mmm is mp3 then not supported at all in 1.8? |
| 13:09.14 | RypPn | run the script in the contrib dir E-bola , addons are included in the main tarball now |
| 13:09.58 | E-bola | ohh doh |
| 13:10.05 | E-bola | thanks : |
| 13:10.07 | E-bola | :) |
| 13:14.03 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 13:15.58 | eduzimrs | hi, im trying to send a fax using SendFax so, i can´t specify direct the number of the calling fax, so what should i do? |
| 13:17.44 | eduzimrs | create a callfile ? |
| 13:18.42 | *** join/#asterisk jsgoecke (~Adium@12.182.24.2) |
| 13:18.50 | McBoing | Setting up Asterisk as a dual server setup, when both have static IPs and are on same subnet, no need to register correct? But I will need to setup dial plans from sipserverA to sipserverB |
| 13:18.56 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 13:20.00 | jpmcallister | I'm using centos with asterisk16 from digium repositories. Will it be simple to upgrade to asterisk18? |
| 13:26.43 | eduzimrs | anyone can help me? |
| 13:29.29 | McBoing | "myserver" and "password" are user and password info from the manager.conf of SIPserverA in this case? "Dial(SIP/myserver:passwordA@SIPserverA/${EXTEN:1},30,r)" |
| 13:31.15 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 13:32.32 | E-bola | Hmm after upgrading from 1.8 rc2 to final i have weird issues with direct transfers that just fail and the calls are lost |
| 13:34.13 | E-bola | on a snom 320, when u press transfer you dont get a dialtone |
| 13:34.23 | E-bola | If you however press hold first, and then make an attended transfer it works fine.... |
| 13:34.28 | E-bola | Anybody seen something similar to this? |
| 13:34.38 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 13:35.30 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:36.54 | mark22 | what is the best option in a dialplan to use some auto matic failover in case a trunk (peer) is down to use another trunk (peer)? currently I am looking at implementing the macro-safedial listed in http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS (we only use SIP) |
| 13:40.51 | [TK]D-Fender | mark22: Dial them back-to-back |
| 13:41.29 | [TK]D-Fender | mcboin manager.conf has NOTHING to do with DIALING-ing anything |
| 13:41.54 | [TK]D-Fender | McBoing: And you should never put user & pass in your DIAL commands. make a proper SIP peer for them |
| 13:42.51 | mark22 | [TK]D-Fender: what do you mean with "Dial them back-to-back"? |
| 13:43.01 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 13:43.06 | cusco | hm.. hi |
| 13:43.09 | [TK]D-Fender | mark22: Dial the first. Next Priority. Dial the FAILOVER |
| 13:43.44 | cusco | when I press *01 I see on the CLI: Feature Found: queuetransfer1 exten: queuetransfer1 |
| 13:43.49 | cusco | on features.conf I have: queuetransfer1 => *01,callee,Transfer,Local/00001@agents |
| 13:44.22 | cusco | but nothing happens |
| 13:44.23 | cusco | why? |
| 13:44.59 | mark22 | what happens when a call is made and only the person that was called did a hangup? does that prefent the next priority to be used? |
| 13:45.57 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:46.01 | cusco | I think Dial() has a flag to carr on with dialplan mark22 |
| 13:46.02 | t_dot_zilla | hi |
| 13:46.07 | cusco | g if I'm not mistaken |
| 13:46.08 | t_dot_zilla | is everyone at astricon? |
| 13:46.11 | [TK]D-Fender | cusco: because Transfer isn't what you think it is for. It is for using SIP, etc to directly throw the call off your SERVER |
| 13:46.15 | McBoing | [TK]D-Fender: How do I make a proper SIP peer? |
| 13:46.23 | [TK]D-Fender | McBoing: SIP.CONF <------- |
| 13:46.34 | t_dot_zilla | could someone tell me if we are using remote access instead of SIP authentication for calls, could that affect CALL EXECUTION or QoS ? |
| 13:47.04 | cusco | [TK]D-Fender: owch, so the application name in featurs.conf should be blindxfer instead ?!?! |
| 13:47.19 | [TK]D-Fender | cusco: No, there IS NO APPLICATION |
| 13:47.35 | cusco | :( |
| 13:47.50 | cusco | how can I transfer the client do a specified dialplan then? |
| 13:48.05 | [TK]D-Fender | cusco: Go make a program that picks up the channel via AMI and does a Redriect against it |
| 13:48.07 | cusco | can't do it by having a feature, pressing some keys? |
| 13:48.25 | [TK]D-Fender | AMI Redirect <----------- |
| 13:48.32 | cusco | how does pressing some keys is detected by AMI ? |
| 13:50.18 | [TK]D-Fender | cusco: Have your feature call an AGI that does it <- |
| 13:51.25 | mark22 | cusco: now you say it, it looks like there are options for Dial() to do what I want |
| 13:56.26 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-odyowmsrptftxrzb) |
| 13:57.44 | *** join/#asterisk jsgoecke (~Adium@12.182.24.2) |
| 13:58.12 | *** join/#asterisk rocky3 (~rocky@184-15-112-120.dr02.chtn.wv.frontiernet.net) |
| 14:01.43 | *** join/#asterisk pabelanger (~pabelange@12.182.24.2) |
| 14:02.42 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 14:07.43 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 14:13.13 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 14:16.53 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:19.06 | marksaitis | does anybody know a good tls srtp certificate creation guide for asterisk? |
| 14:19.49 | WIMPy | There is a script included to do it. |
| 14:22.40 | pabelanger | <PROTECTED> |
| 14:29.53 | *** join/#asterisk moy_ (~moy@216.149.208.11.ptr.us.xo.net) |
| 14:30.44 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:34.29 | marksaitis | WIMPy, thats script is a peace of **** ;] |
| 14:35.22 | marksaitis | pabelanger, it does create certificates and stuff, but eyebeam always says "Cert Validation Failure" no matter what I do. That guide on voip-info regarding tls works better |
| 14:35.34 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:35.35 | WIMPy | NFI. I tried google. Worked very well for me. |
| 14:35.45 | marksaitis | at least phones do get connected over tls and I can make normal RTP calls from phone to phone |
| 14:35.46 | pabelanger | marksaitis: what OS? |
| 14:35.51 | marksaitis | centos 5.5. |
| 14:36.19 | marksaitis | have you ever tried any of tls srtp capable phones from counterpath? |
| 14:36.48 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 14:37.07 | pabelanger | marksaitis: There was a talk yesterday at astricon explain some softphones are weird cert requirements. I cannot remember if centos was one of the ones. Sounds like you need to configure your phone to accept the self signed cert |
| 14:38.09 | WIMPy | Top FAQ for the last week. |
| 14:38.42 | marksaitis | well, using one guide(at voip-info), works fine and it accepts the cert as I explained. But using that script - impossible :) |
| 14:39.07 | marksaitis | centos 5.5 is my server os runing asterisk. i use softphones from windows |
| 14:39.33 | marksaitis | I believe this cert problem is one of the bigest and most important ones |
| 14:39.39 | pabelanger | I think OSX was the issue, that client did not accept self signed certs |
| 14:40.57 | marksaitis | I think its counterpaths phone taking a piss |
| 14:41.09 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:41.09 | marksaitis | and there are not other softphones like counterpaths... |
| 14:41.13 | marksaitis | all the rest is just crap |
| 14:41.24 | marksaitis | I cant get over this issue for days now |
| 14:41.29 | marksaitis | killing me |
| 14:41.42 | angryuser | marksaitis, tryed qutecom ? |
| 14:41.46 | angryuser | tried* |
| 14:42.05 | marksaitis | I remember trying all of them |
| 14:42.10 | marksaitis | this one as well |
| 14:42.13 | marksaitis | cant remember what happened |
| 14:42.19 | marksaitis | oh |
| 14:42.21 | marksaitis | just remembered |
| 14:42.30 | marksaitis | how to add a custom sip on this one? |
| 14:42.36 | *** join/#asterisk nny (~admin@173.160.86.155) |
| 14:42.43 | angryuser | marksaitis, what do you mean ? |
| 14:42.46 | marksaitis | im just going to install it again |
| 14:42.48 | marksaitis | just a sec |
| 14:42.49 | marksaitis | ;] |
| 14:43.11 | marksaitis | I got myself in to this nightmare |
| 14:43.43 | marksaitis | I should name myself angryuser2 I think |
| 14:44.23 | WIMPy | I never understood, how to get the counterpath things to do /anything/ usefull. |
| 14:44.39 | nny | experiencing an odd issue using 1.4.36 I am seeing 108 Ringing in sip debugs, yet no audio ringing feedback. This is only on certain numbers, using the same sip provider for other numbers and I get ringing. Any help greatly appreciated, kind of scratching my head here |
| 14:45.02 | marksaitis | well, everything is fine with them, except if I enable srtp - its impossible to get an incoming call |
| 14:45.07 | marksaitis | seriously |
| 14:45.38 | WIMPy | Sounds like you're lucku already. |
| 14:45.42 | marksaitis | ok, I am just trying to run cutecom |
| 14:46.16 | marksaitis | WIMPy, well, basically, for tls to run, that asterisk18 cert creation script does not work, only that old guide works for it |
| 14:46.29 | marksaitis | for tls and srtp to work, no guides found |
| 14:47.09 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 14:47.14 | marksaitis | stupid qcutecom, I just downloaded the newest version from the website, installed, launched it, and it instantly said newer version is available please wait for update, and it installed over again, lol |
| 14:47.21 | WIMPy | I haven't trued srtp yet. Will try when I can upgrade my box to 1.8. |
| 14:48.57 | KavanS | I have an issue with my dialplan...I'm using "goto" for my extensions, and it's messing up my cdr-csv by leaving 's' as my destination - user dials 801 and 801 is: 801,1,Goto(personhere,s,1) |
| 14:49.37 | marksaitis | im trying that cutecom now |
| 14:49.42 | WIMPy | That's the way it works. |
| 14:49.42 | McBoing | [TK]D-Fender: sorry can you help me out some more, I am stuck setting up peers between Asterisknow and Asterisk server |
| 14:49.53 | marksaitis | that shit doesnt even connect |
| 14:50.08 | *** join/#asterisk fors1 (~forsen@pat-tdc.opera.com) |
| 14:50.11 | marksaitis | cant even find tls nor srtp settings for an account |
| 14:50.23 | WIMPy | KavanS: You can put the original extension in some other CDR field. |
| 14:50.58 | pabelanger | marksaitis: Keep an eye for the astricon presentation for SRTP. It should be online in a few days |
| 14:51.19 | fors1 | anyone located in the CET timezone with grandstream phones here? For some reason, my grandstream adjusted the clock from CEST to CET last weekend, while the custom rule specifies it should be done the last sunday in the month (next sunday). |
| 14:51.20 | marksaitis | I dont even think that crappy cutecom has srtp and tls .... |
| 14:51.27 | marksaitis | astricon? |
| 14:51.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:51.50 | KavanS | WIMPy, ok - let me look at our database here |
| 14:51.52 | McBoing | I want to allow our currently live Asterisk 1.4 server to call the new AsteriskNOW server for testing purposes, can someone help please? I do not need the new server to call back to the older PBX (Asterisk 1.4) |
| 14:52.03 | marksaitis | this cutecom phones is the most stupid I evers seen :) |
| 14:52.15 | marksaitis | useless crap, totaly |
| 14:53.38 | KavanS | cutecom? |
| 14:53.39 | marksaitis | I hope this problem with tls srtp will be adressed in astricon |
| 14:53.42 | marksaitis | qcutecom |
| 14:53.48 | marksaitis | crappycom |
| 14:53.52 | marksaitis | however its called |
| 14:53.57 | marksaitis | total nonsense it is |
| 14:54.00 | KavanS | polycom? |
| 14:54.16 | marksaitis | polycom what? |
| 14:54.32 | Naikrovek | useless crap? how DARE you |
| 14:54.38 | Naikrovek | lots of us in here use them and they work GREAT dude |
| 14:54.42 | marksaitis | yes, it is useless crap |
| 14:54.43 | marksaitis | :) |
| 14:54.46 | Naikrovek | if you have a problem with them, and none of us do, i think i see the cause |
| 14:55.16 | Naikrovek | they're not perfect, but they certainly work quite well |
| 14:55.17 | marksaitis | I can repeat it again, qcutecom is useless crap. over |
| 14:55.30 | Naikrovek | and you'll be wrong every time you say that |
| 14:55.34 | Naikrovek | every single time |
| 14:55.48 | Naikrovek | millions and millions of phones laugh at you every time you say taht |
| 14:56.10 | KavanS | is he talking about polycom? |
| 14:56.15 | marksaitis | a good softphone is the one with tls and srtp.... full stop :) |
| 14:56.16 | Naikrovek | yes |
| 14:56.16 | KavanS | man I love these polycom phones we got... |
| 14:56.20 | marksaitis | not polycom |
| 14:56.28 | KavanS | we have rather ;) |
| 14:56.29 | marksaitis | I have nothing bad against polycom |
| 14:56.33 | KavanS | it is too early for perfect english |
| 14:56.35 | Naikrovek | then wtf are you talking about |
| 14:56.49 | marksaitis | qcutecom |
| 14:56.57 | fors1 | http://www.qutecom.org/ |
| 14:56.59 | marksaitis | who said I said smth wrong about polycom :) |
| 14:57.14 | Naikrovek | ah |
| 14:57.18 | Naikrovek | i can't speak for qutecom |
| 14:57.20 | Naikrovek | polycom rules |
| 14:57.29 | marksaitis | I do agree about polycom |
| 14:57.34 | Naikrovek | then we have no issues |
| 14:57.36 | Naikrovek | good day sir |
| 14:57.40 | marksaitis | ;]] |
| 14:57.55 | *** join/#asterisk garymc (~chatzilla@host81-139-127-32.in-addr.btopenworld.com) |
| 14:58.39 | marksaitis | seriously, in this whole channel, has anybody tried tls srtp on any of counterpaths products? |
| 14:58.45 | marksaitis | there must be somebody |
| 14:59.55 | WIMPy | I think you could make your life a lot easier if you tried a real phone. |
| 14:59.58 | fors1 | I tried, but counterpath (x-lite in my scenario) didn't like wildcard certificate, which was the only one I had for my asterisk. |
| 15:01.04 | fors1 | oh. wait, not x-lite. I actually tried with eyebeam. if it matters. |
| 15:01.18 | cusco | ok so I made a sh script using Action: Redirect |
| 15:01.46 | cusco | now how do I call the script from features.conf ? |
| 15:04.45 | *** join/#asterisk jsgoecke (~Adium@12.182.24.2) |
| 15:05.02 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 15:05.32 | ManxPower | It has been a while since I've done any AGI. I'm on 1.4. CALLERID() is a function on 1.4. Would $agi->get_variable("CALLERID(all)") work even though it is a function? |
| 15:06.26 | *** join/#asterisk c4rg (crg@lagoon.freebsd.lublin.pl) |
| 15:07.05 | c4rg | hi, if I have a line in sip.conf like this: register => user:pass@host/extension, and later an user defined - how does asterisk match incoming invite with user definition? |
| 15:07.24 | ManxPower | c4rg, registration has nothing to do with the incoming invite |
| 15:08.03 | ManxPower | all registration does is notify the remote server which dynamic IP is associated with a specific user/pass |
| 15:08.36 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:09.06 | c4rg | so how does it work then? |
| 15:09.14 | ManxPower | registration? |
| 15:09.39 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 15:09.47 | c4rg | matching incoming invite with user definition |
| 15:09.50 | ManxPower | client w/register command -> hey, I'm bob, password 12345 server -> OK. Hot it! |
| 15:10.26 | ManxPower | c4rg, that would match on the [myhappyuserid] section of sip.conf, unless you do not have allowguest=no. If you don't have that then most any call will be accepted |
| 15:10.51 | c4rg | myhappyuserid? |
| 15:11.00 | ManxPower | c4rg, the userid of the incoming call |
| 15:11.20 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 15:13.12 | asterisk-learner | hi |
| 15:13.15 | asterisk-learner | Does anyone knows if we can whisper on a (IAX2) call, while MOH is turned on ? |
| 15:13.39 | marksaitis | fors1, I see. |
| 15:13.44 | McBoing | Can someone help me get Asterisk (Asterisk 1.4) to be able to call another Asterisk server (AsteriskNOW), it only needs to be one way right now for testing, cant seem to get authenticate to work |
| 15:14.47 | c4rg | ManxPower: if invite like this comes in: INVITE sip:testing@a.b.c.d:5060 SIP/2.0 |
| 15:15.04 | *** join/#asterisk mark22 (~mark@unaffiliated/mark21) |
| 15:15.09 | c4rg | it's just routed to defaunt context & extension testing? |
| 15:15.16 | c4rg | default context |
| 15:15.26 | c4rg | or the context is taken from some user's definition? |
| 15:17.57 | *** join/#asterisk jbeitler (~anonymous@www.brc2.com) |
| 15:18.14 | jbeitler | I have a quick question, is anyone online? |
| 15:18.40 | WIMPy | No. The internet has been shut down for today. |
| 15:20.58 | Gianlu | hello everybody. Does anyone have any experience with asterisk-dotnet or asterisk-java? |
| 15:21.02 | *** join/#asterisk not_a_golfer (~si@i6-fw-ha.derwentside.net) |
| 15:21.11 | jbeitler | Okay I am trying to get asterisk 1.7 with GUI going, the problem is when I browse to the IP it tells me to on boot all i get is Oops! Google Chrome could not connect to 172.16.85.138:8080 I have tried it from a number of Browsers and they al say the same thing. It is a base install of AsteriskNow with option 3 (with GUI) |
| 15:21.56 | cusco | how do I cal a ami script from features.conf? |
| 15:22.01 | jbeitler | I can ping the server and do updates from command line on the server as well |
| 15:22.25 | *** join/#asterisk russellb_ (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 15:22.25 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 15:22.37 | Gianlu | jbeitler: is the server listening on port 8080? |
| 15:23.08 | jbeitler | You know I did not think to check.. as it said to use that port |
| 15:24.05 | Gianlu | netstat -ln | grep 8080 (or something like that....) |
| 15:24.44 | espiceland | or look at the 'bindport' setting in /etc/asterisk/http.conf |
| 15:24.59 | jbeitler | it is not.. but why would it want you to connect to that port and then not open it by default? that seems kind of dumb |
| 15:25.39 | ChannelZ | It's more dumb to have your PBX interface sitting out for the world to access |
| 15:27.00 | jbeitler | okay but if the default install says use XXX.XXX.XXX.XXX:8080 and then it is not open then what is the point of telling you to use a port? Second most people do not have the port open and forwarding to 8080 so how is it open to the world? |
| 15:27.39 | Gianlu | hey folks, anyone with some experience in asterisk-dotnet? |
| 15:28.56 | ChannelZ | jbeitler: because you're generally setting up on a LAN |
| 15:29.58 | jbeitler | That was my point |
| 15:29.59 | ChannelZ | there isn't any firewall on by default that I remember and your own network setup is out of their control |
| 15:30.26 | ChannelZ | if the server isn't running then something else has failed for some other reason |
| 15:30.36 | ChannelZ | either way go ask #asterisknow or #freepbx |
| 15:31.12 | ManxPower | c4rg, set allowguest=no and test again |
| 15:31.23 | ManxPower | in sip.conf [general] |
| 15:35.12 | c4rg | ManxPower: how does it work? |
| 15:35.39 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 15:35.59 | marksaitis | does anybody know a working tls srtp softphone? If one exists at all? |
| 15:37.10 | marksaitis | what is the difference between tls (sip) and tls(sips) would anyone know? |
| 15:37.22 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:37.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:38.39 | ManxPower | c4rg, it makes sure you only accept calls from AUTHENTICATED users |
| 15:38.53 | ManxPower | otherwise you will accept calls from just about any IP phone on the planet |
| 15:39.06 | trollasaurus | markaitis, Maybe one uses STARTTLS and the other assumes TLS (like the difference between SMTP and SMTPS) |
| 15:39.34 | ManxPower | c4rg, try it and see what happens. If calling breaks when you add that option then your incoming calls were never authenticated to start with |
| 15:39.35 | marksaitis | trollasaurus ok |
| 15:39.51 | ManxPower | I thought TLS was only for TCP, not UDP? |
| 15:40.25 | WIMPy | ManxPower: Correct |
| 15:40.59 | trollasaurus | You can do SIP over TCP.. I am :-P |
| 15:41.02 | *** join/#asterisk Squeeb (~Debian-ex@host81-149-117-179.in-addr.btopenworld.com) |
| 15:41.17 | Squeeb | I'm trying to set a simple variable to the output of the System Application |
| 15:41.27 | *** join/#asterisk alexshell (~chatzilla@unaffiliated/alexshell) |
| 15:41.30 | Squeeb | but all I'm getting returned is the exit code |
| 15:41.45 | Squeeb | how can I get the output from the command executed by System? |
| 15:41.49 | WIMPy | Squeeb: Use an AGI |
| 15:42.00 | Squeeb | Well, I tried that, but it's not setting the varibale |
| 15:42.22 | Squeeb | I do this in perl: print STDERR 'SET VARIABLE SOMEVAR "blah"'; |
| 15:42.33 | Squeeb | then NoOp("${blah}"); |
| 15:42.33 | c4rg | ManxPower: could you please tell me how an incoming invite is interpreted? ;) |
| 15:42.45 | Squeeb | wait no |
| 15:42.57 | Squeeb | NoOp("${SOMEVAR}"); is what i'm using |
| 15:43.04 | Squeeb | but I get "" returned when NoOp runs. |
| 15:44.13 | *** join/#asterisk dr_ (~duckz@78.96.111.117) |
| 15:45.39 | marksaitis | ok. Can anybody recommend me a good real phone, the smaller the better, RJ45 or USB capable, supporting SRTP and TLS? |
| 15:47.09 | WIMPy | Linksys had a quite small one. Calld 900something IIRC. But no idea what features it has. |
| 15:47.32 | marksaitis | im going to check |
| 15:47.48 | marksaitis | im just realizing that there are no tls srtp capable softphones whatsoever |
| 15:47.53 | marksaitis | they all develop total crap |
| 15:50.19 | *** part/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 15:51.59 | espiceland | Squeeb: Try printing to STDOUT instead of STDERR? |
| 15:53.02 | marksaitis | WIMPy, yeah that phone um entioned, SPA901 is awesome :) |
| 15:53.10 | marksaitis | tls, srtp, cheap, looks good |
| 15:53.13 | cusco | I don't seem to have a file doc/ip-tios.txt |
| 15:53.21 | cusco | is there another name for this file? |
| 15:54.06 | WIMPy | marksaitis: But Linksys has sipura software which is not exactely known to be great. |
| 15:54.16 | cusco | it seems that to set QoS I need to mark sip packets as 0x68 and rtp as 0xB8 |
| 15:55.04 | cusco | can I just set tos_sip=0x68 ? |
| 15:55.21 | WIMPy | 's got a SPA 962, but i like it so much that I don't use it, except for catching dust. |
| 15:56.06 | [TK]D-Fender | McBoing: FreePBX is NOT supported here. |
| 15:59.49 | ManxPower | cusco, incorrect. You need to set the ToS and configure every piece of equipment and every router between you and your gateway to the PSTN (usually ITSP) to recognize that ToS |
| 16:00.04 | Squeeb | espiceland: that worked |
| 16:00.05 | Squeeb | cheers |
| 16:02.57 | *** part/#asterisk ezfox (~ezfox@nat/ibm/x-mlpuzepolwtjnhdw) |
| 16:06.58 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:07.45 | *** join/#asterisk delphiWorld (~Miranda@41.200.4.40) |
| 16:07.48 | delphiWorld | hi |
| 16:07.54 | delphiWorld | after checking out asterisk 1.8 |
| 16:08.00 | delphiWorld | look like is in /usr/local/asterisk |
| 16:08.06 | delphiWorld | how do i install it in standard path? |
| 16:08.19 | delphiWorld | conf files in /usr/local/asterisk/etc/asterisk |
| 16:08.27 | delphiWorld | but i want it to by in /etc |
| 16:09.12 | marksaitis | WIMPy, is there anything great and working well in VoIP world? :) |
| 16:09.39 | jdoe | delphiWorld: that's what ./configure is for. |
| 16:09.45 | marksaitis | yeah |
| 16:09.46 | marksaitis | ;] |
| 16:09.55 | marksaitis | for me it installed normally without using it |
| 16:10.47 | WIMPy | marksaitis: Short answer: No. But it can be made to work quite well. |
| 16:11.14 | delphiWorld | doe wiki is up? |
| 16:11.20 | delphiWorld | wiki.asterisk.org not working |
| 16:11.42 | ManxPower | delphiWorld, define "checking out" is that checking out as in "svn checkout" or as in "checking out the nice ass on that person in front of you" |
| 16:11.45 | marksaitis | ask google dude |
| 16:11.59 | marksaitis | google:wiki asterisk always tell u the truth |
| 16:12.06 | delphiWorld | ManxPower: DUDE |
| 16:12.26 | delphiWorld | marksaitis: lol wiki.asterisk.org that's is but is down |
| 16:14.50 | marksaitis | dude, google knows better |
| 16:14.51 | marksaitis | :) |
| 16:15.29 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 16:15.46 | Scorpio2007 | ERROR[3124] chan_dahdi.c: Unable to get span status: Inappropriate ioctl for device |
| 16:15.49 | Scorpio2007 | any idea? |
| 16:18.09 | marksaitis | Scorpio2007, mabye it the best and easiest place to report problems, but certainly not the best to sort them out. try dev mailing list |
| 16:18.20 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:19.36 | *** join/#asterisk nny (~admin@173.160.86.155) |
| 16:20.10 | *** join/#asterisk kernel_x (~kernel_x@174-140-96-143-dhcp.mia.fl.atlanticbb.net) |
| 16:20.23 | Scorpio2007 | hmm |
| 16:21.11 | eduzimrs | hi, im trying to send a fax using SendFax so, i can´t specify direct the number of the calling fax, so what should i do? |
| 16:21.11 | nny | trying to wrap my head around this ringing issue. So the provider sends the SIP 108 Ringing to asterisk, and asterisk sends it to the phone (Canreinvite=no). I am seeing it in the sip debug for the phone, from console. what am I missing? |
| 16:22.58 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 16:26.04 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
| 16:27.59 | *** part/#asterisk delphiWorld (~Miranda@41.200.4.40) |
| 16:28.48 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 16:30.03 | *** join/#asterisk nny1 (~admin@173.160.86.155) |
| 16:31.21 | nny1 | if anyone has any advice on where I can troubleshoot this issue, I'd be highly appreciative. Willing to paypal someone for their time if needed. |
| 16:32.09 | nny1 | this issue= ringing missing on certain calls from ear piece, sip debug shows 108 ringing yet audible ringing is missing. I can reproduce it with a series of numbers |
| 16:33.59 | nny | lol |
| 16:34.08 | nny | my dial statement is Dial EXTEN@PROVIDER |
| 16:34.13 | nny | as basic aas possible |
| 16:34.16 | nny | and the issue arises |
| 16:34.23 | nny | eliminates my dial plan :D |
| 16:35.16 | WIMPy | You should ask your provider. |
| 16:35.24 | cusco | ManxPower: we have the pstn gateway, and that is no problem. We have two mikrotik routers geographically far. A VPN between them. In mikrotik docs I'm reading that voip SW/HW must be able to set the DSCP/TOS field in the IP packet |
| 16:35.48 | marksaitis | :D |
| 16:36.10 | nny | WIMPy: yeah that's what I am thinking. I did a debug and they pass the 108 ringing, but maybe it's something else? |
| 16:36.17 | cusco | and according to the docs I would like to set SIP to 0x68 (104) and RTP to 0xB8 (184) |
| 16:40.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 16:41.31 | *** join/#asterisk Tim_Toady (~moi@178.128.36.116.dsl.dyn.forthnet.gr) |
| 16:43.01 | *** join/#asterisk baileyx (~Adium@70-36-142-188.dsl.dynamic.sonic.net) |
| 16:49.41 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:51.48 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 16:55.17 | marksaitis | for a client to accept srtp call, does he need to have a certificate? or a chain file containing private key AND certificate? |
| 16:56.30 | marksaitis | im totaly bored of trying to get this crap working. Voip info page tells loads of phones supporting srtp tls, but surprisingly 90% of them are not being developed/supported anymore and the rest comes with their own problems |
| 16:58.12 | [TK]D-Fender | marksaitis: And which ones have you tried? |
| 16:58.36 | marksaitis | every single one , except 1 or 2 for linux |
| 16:58.45 | [TK]D-Fender | marksaitis: NAMES <- |
| 16:58.49 | marksaitis | ok |
| 16:58.51 | jdoe | surprise, 90% of linux sip clients suck ass. |
| 16:59.01 | marksaitis | Exactly |
| 16:59.30 | WIMPy | Like all the hardware phones? |
| 17:01.52 | marksaitis | I just lost that srtp phones list on voipinfo |
| 17:02.00 | marksaitis | nope, just found |
| 17:02.35 | WIMPy | Well, you know that voip-info has a tendency of being outdated? |
| 17:02.54 | jdoe | or just plain incorrect ;) |
| 17:03.14 | jdoe | marksaitis: I've used bria on my phone, I imagine it does fairly well as a client. X-lite used to have a linux version, Bria may as well. |
| 17:03.23 | jdoe | er, sorry, "as a desktop client" |
| 17:05.39 | [TK]D-Fender | marksaitis: I didn't ask you about some other useless list. I askes whichones YOU TRIED |
| 17:07.19 | marksaitis | ok pjsip - just a lib, privategsm - only mobiles and no srtp, minisip - says its linux only but it has windows bin too, doesnt even launch saying smth expired=crap, twinkle - old unsuported linux only, wengoophone - no settings for srtp nor tls.... such a crap, phonerlite - old windows crap, tried, doesnt work, it doesnt even look like a softphone, a pile of testing crap, counterpath - thats the closest to reality, incoming calls wont work w |
| 17:07.56 | [TK]D-Fender | marksaitis: Which of these CLAIMED to support it in the first place? |
| 17:08.28 | marksaitis | [TK]D-Fender, http://www.voip-info.org/wiki/view/Asterisk+encryption |
| 17:08.31 | marksaitis | have a look for urself |
| 17:08.58 | [TK]D-Fender | marksaitis: The wiki page can fuck itself. I could edit it and say it produces an unlimited supply of cottage cheese for all anyone cares |
| 17:09.02 | [TK]D-Fender | (perhaps the starving) |
| 17:09.13 | [TK]D-Fender | marksaitis: what do the PRODUCT PAGES say? |
| 17:09.32 | marksaitis | If anybody would configure 2 working srtp tls client on windows machines, I would give them my beautifull car, my iphone 4 and my laptop |
| 17:09.59 | marksaitis | Yeah, fuck you wiki page :) |
| 17:10.01 | marksaitis | thats better |
| 17:11.41 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 17:13.21 | fauxalliance | marksaitis, if you make the iphone run android froyo, you got a deal. |
| 17:15.17 | marksaitis | fauxalliance, I suppose u tried all google "android froyo on iphone" results ;] |
| 17:16.44 | fauxalliance | just the UK one |
| 17:16.49 | [TK]D-Fender | marksaitis: Not on YOUR iPhone... he'll need that for "testing", please be sure he has it ASAP :p |
| 17:16.50 | WIMPy | marksaitis: What car? What laptop? And would I have to take the iPhone or could you keep that? |
| 17:16.53 | fauxalliance | personally, i want the experia x10 |
| 17:17.02 | [TK]D-Fender | fauxalliance: No you don't... |
| 17:17.10 | [TK]D-Fender | fauxalliance: Android 1.6 ICK |
| 17:17.23 | [TK]D-Fender | fauxalliance: And there are plenty of better alternatives out there |
| 17:17.38 | [TK]D-Fender | fauxalliance: Smsung Galaxy S FTMFWY |
| 17:17.42 | fauxalliance | [TK]D-Fender, yeah, i almost dropped the droid accidentially on purpose |
| 17:18.25 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 17:18.26 | *** join/#asterisk pabelanger (~pabelange@12.182.24.2) |
| 17:20.17 | marksaitis | WIMPy, mercedes c270, sony vaio, keep iphone 4 ;] |
| 17:20.39 | fauxalliance | has 'champagne' taste, yet only the rootbeer income... |
| 17:20.45 | [TK]D-Fender | WIMPy: Mr. Green in the lobby with a candle-stick |
| 17:20.53 | marksaitis | The ebst phone is iphone 4, why would smbd need damn xperias and crapberys |
| 17:20.53 | marksaitis | ;] |
| 17:20.58 | fauxalliance | should have taken the bonus over the corner office. |
| 17:21.06 | drmessano | iphone? |
| 17:21.10 | drmessano | laughable |
| 17:21.10 | fauxalliance | Hyundai? |
| 17:21.12 | [TK]D-Fender | I DON'T CARE |
| 17:21.18 | *** join/#asterisk ytal (ytal@bzq-218-138-39.cablep.bezeqint.net) |
| 17:21.22 | [TK]D-Fender | pwmed |
| 17:21.28 | drmessano | Every iPhone out has been last months technology |
| 17:21.36 | fauxalliance | s/month/year |
| 17:21.37 | drmessano | err |
| 17:21.38 | drmessano | yeah |
| 17:21.48 | *** join/#asterisk NEEDINGHELP123 (Mordi@v58.sgsvr.com) |
| 17:21.58 | fauxalliance | ^^begging? |
| 17:22.01 | drmessano | The iPhone is nothing but name recognition in a little white box |
| 17:22.22 | NEEDINGHELP123 | hi guys, i am looking for some help regarding H323 in general, i need to create a simple H323 user-agent getter |
| 17:22.44 | drmessano | Get ahold of yourself, iPhone owners.. but don't grip it incorrectly |
| 17:22.51 | [TK]D-Fender | drmessano: Actually... white wn't be an option until NEXT YEAR :p |
| 17:22.56 | marksaitis | drmessano, no. Its way more =) compared to other touchscreen phones, they all look like stone age crap |
| 17:22.56 | NEEDINGHELP123 | i am willing to pay upto $500 for help with this project |
| 17:23.02 | drmessano | NEEDINGHELP123, is this another school project? |
| 17:23.11 | NEEDINGHELP123 | drmessano nope |
| 17:23.14 | NEEDINGHELP123 | this is a personal project |
| 17:23.14 | drmessano | NEEDINGHELP123, another school "contest"? |
| 17:23.15 | marksaitis | :D |
| 17:23.16 | NEEDINGHELP123 | why? |
| 17:23.30 | fauxalliance | curiosity kills many cats around here |
| 17:23.31 | [TK]D-Fender | Becaus ehe will look "coll" and ilke.. stuff... |
| 17:23.33 | NEEDINGHELP123 | drmessano - school - you were right the first time, why? |
| 17:23.40 | ytal | can someone actually read chan_sip? |
| 17:23.56 | [TK]D-Fender | ytal: Yes, now what do you want? |
| 17:24.05 | drmessano | NEEDINGHELP123, you were in here months ago asking us to help you win some competition your teacher had for this class, and we refused |
| 17:24.17 | NEEDINGHELP123 | you refused?????? |
| 17:24.21 | NEEDINGHELP123 | i got the help that in eeded though, |
| 17:24.21 | fauxalliance | ^^cause thats cheating... ;-) |
| 17:24.23 | NEEDINGHELP123 | so maybe YOU refused |
| 17:24.29 | NEEDINGHELP123 | but others no |
| 17:24.29 | marksaitis | NEEDINGHELP123, tell me, what do you need help with :) |
| 17:24.30 | [TK]D-Fender | Its wonderful wehn people want to take credit for other people's work... |
| 17:24.38 | NEEDINGHELP123 | drmessano i think you woke up on a bad day mate |
| 17:24.42 | drmessano | marksaitis, seriously, if you don't realize the iPhone is crap, you deserve to own one |
| 17:24.44 | NEEDINGHELP123 | drmessano so i'm putting you on ignore, thanks alot |
| 17:24.51 | fauxalliance | tell me what you need, I'll tell you how to get along without it. |
| 17:24.55 | drmessano | NEEDINGHELP123, you were just a douche back then too |
| 17:24.57 | NEEDINGHELP123 | marksaitis may I pm you to save bashing? |
| 17:25.14 | *** join/#asterisk jsgoecke (~Adium@12.182.24.2) |
| 17:25.29 | NEEDINGHELP123 | anyway, the project is like this: i need to find a user agent of an h323 device in the simplest way, and i'm willing to pay $500 for help with implementation |
| 17:25.52 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 17:25.52 | drmessano | http://tinyurl.com/35zn8ft <-- Is this the same guy? |
| 17:26.00 | fauxalliance | look at a webserver, it does that often enough |
| 17:26.02 | WIMPy | NEEDINGHELP123: Have you heard about something called google? |
| 17:26.13 | NEEDINGHELP123 | WIMPy yeah, can't quite get the right search term |
| 17:26.50 | fauxalliance | drmessano, your are worse than me... hats off! |
| 17:27.06 | fauxalliance | that would decidedly take a higher priority... ;-) |
| 17:27.23 | drmessano | http://purl.rikers.org/%23asterisk/20100722.html.gz <--- That was his last request.. |
| 17:27.56 | marksaitis | drmessano, tell me a single reason why is it crap? I would say its the best thing a man can have in its pocket! iphone 4 - very good camera for picture shooting, HD recording and with a flash as well and focusing, gps, compass, ipod, wifi internet, office apps, great touchscreen, great games, multitasking, great email client. So to summarize, instead of having a digital camera, a seperate satnav and a seperate mp3 player and a seperate phone |
| 17:27.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:27.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:28.12 | fauxalliance | drmessano, either way a bit of contact dermatitis is nothing to laugh at... at least it's not herpetic |
| 17:28.17 | marksaitis | and, thers even no need for anoying small buttons on the body ;] |
| 17:29.11 | adyn | I can do all that with the Epic 4g |
| 17:29.27 | *** join/#asterisk QbY (~kelvin@96.176.19.11) |
| 17:29.51 | QbY | how can I specify a subnet as a peer's host address? |
| 17:29.52 | drmessano | marksaitis, there's a dozen phones out there that can do more, better than the iPhone for less of an asking price.. they are also not encumbered to AT&T, and they were doing it months before the iPhone, if not longer. The iPhone may do a lot, but it's by no means pioneering, as theres many other phones that blow it away |
| 17:29.53 | marksaitis | well, surely, iphone has more apps and touchscreen is way better thasn epic 4g ;] |
| 17:30.21 | marksaitis | encumbered to At$T ? I dont know even what the hell that ATAT is |
| 17:30.27 | drmessano | marksaitis, but I won't hold that against you, or hold it against you incorrectly |
| 17:30.40 | fauxalliance | marksaitis, any chinese android can do those things... |
| 17:31.01 | marksaitis | iphone is a real pioneer where I live. Its touchscreen is completely uncomparable with anything else. full stop |
| 17:31.11 | fauxalliance | marksaitis, more apps? maybe last year... |
| 17:31.30 | marksaitis | chineese andoird :DDD ppl you make me laugh. Those things are a pile of crap. |
| 17:31.33 | marksaitis | no quality |
| 17:31.38 | marksaitis | ur talking nonsense |
| 17:31.39 | marksaitis | really |
| 17:31.49 | drmessano | marksaitis, if the iPhone was great, then why has Android adoption increased 3000% in the last year? |
| 17:32.02 | fauxalliance | marksaitis, fine, my milestone still kicks any iphones ass. |
| 17:32.05 | drmessano | The iPhone is a pile of crap, with flaws galore |
| 17:32.10 | fauxalliance | marksaitis, FUD! |
| 17:32.12 | adyn | most of this is just opinion anyways. I prefer the Android OS as my personal choice, I know people who choose the iPhone as their personal choice. |
| 17:32.26 | marksaitis | everything has flaws - thats a fact. nobody can judge about that ;] |
| 17:32.30 | fauxalliance | adyn, true... but only one comes out on top |
| 17:32.52 | fauxalliance | marksaitis, but apple doesnt hide nor charge me for mine |
| 17:32.54 | NEEDINGHELP123 | OFFERING $500 FOR THE HELP I NEED .... I NEED TO BE ABLE TO FIND THE " USER-AGENT " OF AN H323 SERVER IN THE MOST SIMPLE WAY THANKS |
| 17:33.08 | marksaitis | I had that chineese clone and iphone 4 in my hand.... if you think they can do the same, ur so wrong |
| 17:33.13 | WIMPy | fauxalliance: Yes. Usually the worst one. Bad design leaves most cash for marketing. |
| 17:33.17 | fauxalliance | NEEDINGHELP123, perhaps you need a transparent bridge to record some traffic... |
| 17:33.41 | drmessano | All caps now? |
| 17:33.51 | [TK]D-Fender | NEEDINGHELP123: Go call the server. Spy on the packets. Lokk at user-agent" in the response from them. |
| 17:33.57 | drmessano | marksaitis, read and sob a little.. http://www.droid-life.com/2010/06/07/comparison-iphone-4-vs-droid-incredible-evo-4g-nexus-one/ |
| 17:34.08 | drmessano | Fact is, the iPhone outperforms nothing |
| 17:34.09 | fauxalliance | WIMPy, this quarter anyways... It takes a special kind of brand loving person to shop for a badge as opposed to features... personally, i take the one that doesnt advertise, they save, i asve |
| 17:34.54 | [TK]D-Fender | iPhone is a perfectly decent phone & platform. They usually start out at the head of the line or neck & neck, and just don't get updated as fast as the competition. |
| 17:34.56 | marksaitis | drmessano, I had HTC and iphone 4 in my hand. When you will try them both, you will understand |
| 17:35.03 | fauxalliance | watched two people fight over one at the teus store.... lawl |
| 17:35.08 | NEEDINGHELP123 | [TK]D-Fender but i need to be able to understand what to send, and what ot look at in the response |
| 17:35.28 | drmessano | marksaitis, I have tried more phones than you can count.. |
| 17:35.31 | NEEDINGHELP123 | fauxalliance what do you mean transparent bridge to record some traffic? sorry |
| 17:35.40 | fauxalliance | NEEDINGHELP123, if you are prepared to be confused, be prepared for a sore bum. |
| 17:35.51 | [TK]D-Fender | iOS is a capable platform in most common respects, but forsaken proper multi-tasking, and is a nasty dictatorship that strives to lock you out of the process at every turn. It is freedom-hostile |
| 17:35.56 | fauxalliance | NEEDINGHELP123, yes, promiscuity is back in ! |
| 17:36.03 | fauxalliance | chokes on a pun |
| 17:36.13 | [TK]D-Fender | [13:35]<NEEDINGHELP123>[TK]D-Fender but i need to be able to understand what to send, and what ot look at in the response <- you've been linked to a meteric ton of docs. Go read them. |
| 17:36.13 | marksaitis | and that list is just a list, doesn tell anything :) |
| 17:36.28 | NEEDINGHELP123 | fauxalliance ok, thanks for your help buddy, i REALLY appreciatey our great insight |
| 17:36.32 | marksaitis | drmessano, you cant say so as you dont know how many I tried :) |
| 17:36.38 | [TK]D-Fender | NEEDINGHELP123: And if you can't learn this then you don't seem to be qualified to even consider taking on this task |
| 17:37.06 | NEEDINGHELP123 | [TK]D-Fender i don't need to read a 'metric tonne of docs' |
| 17:37.17 | NEEDINGHELP123 | [TK]D-Fender i need the answer to my question in a savvy format |
| 17:37.22 | [TK]D-Fender | 20:41.50pabelangerNEEDINGHELP123: https://secure.wikimedia.org/wikipedia/en/wiki/H323 |
| 17:37.24 | [TK]D-Fender | 20:42.33leifmadsenNEEDINGHELP123: there are several links to papers and such here: http://en.wikipedia.org/wiki/H.323#H.323_Network_Signaling |
| 17:37.25 | [TK]D-Fender | 20:42.35pabelangerspecifically H.225.0 |
| 17:37.26 | NEEDINGHELP123 | why would i waste my time trawling through other poeples documents bro? |
| 17:37.27 | [TK]D-Fender | 20:42.54leifmadsensuch as http://hive.packetizer.com/users/packetizer/papers/h323/h323_protocol_overview.pdf |
| 17:37.27 | fauxalliance | NEEDINGHELP123, apparently you do... make that a metric assload of docs |
| 17:37.29 | [TK]D-Fender | NEEDINGHELP123: ^^^^^^^^^^^^ |
| 17:37.36 | [TK]D-Fender | NEEDINGHELP123: You were referred to several SPECIFIC ones earlier |
| 17:37.37 | drmessano | marksaitis, I have had almost every smartphone out there come across my desk at one point or another.. I check them all out before we cut salespeople loose with them.. I am not talking about grabbing one in the store and eyeing it over |
| 17:38.04 | NEEDINGHELP123 | [TK]D-Fender thanks alot, but i just need the packet to ENCODE, and i will decode the received packet, |
| 17:38.06 | [TK]D-Fender | [13:37]<NEEDINGHELP123>why would i waste my time trawling through other poeples documents bro? <- because we're not here to READ them back to you because you're too lazy or incapable of reading them yourself |
| 17:38.10 | NEEDINGHELP123 | i don't need links to full pdf files |
| 17:38.21 | NEEDINGHELP123 | [TK]D-Fender i'm not incapable or lazy friend, just i'm not going to do what others have already done |
| 17:38.29 | NEEDINGHELP123 | i do not believe in 'oh i know' but i cna't tell you 'learn yourself' |
| 17:38.30 | NEEDINGHELP123 | sorry |
| 17:38.35 | drmessano | He wants an app for that |
| 17:38.42 | NEEDINGHELP123 | nope i don't need an app for that drmessano |
| 17:38.45 | NEEDINGHELP123 | i need specific detail |
| 17:38.45 | drmessano | and some cream for that bump |
| 17:38.52 | NEEDINGHELP123 | drmessano i don't need cream for any bump |
| 17:38.53 | marksaitis | drmessano, so you tried HTC and iphone 4 (not talking about older crap), and u can clearly say HTC android is better? |
| 17:38.54 | [TK]D-Fender | NEEDINGHELP123: Maybe noby made a "Single purpose tool to grab the UA from a comminication attemp in an easy way for NEEDINGHELP123" |
| 17:38.54 | marksaitis | ;] |
| 17:38.58 | NEEDINGHELP123 | get some mature cheese |
| 17:39.00 | NEEDINGHELP123 | and come back friend |
| 17:39.01 | [TK]D-Fender | nobody* |
| 17:39.14 | NEEDINGHELP123 | [TK]D-Fender no problem , i will have it made, but i need the packets t osend, the encoding technique |
| 17:39.16 | drmessano | http://tinyurl.com/35zn8ft <-- did it clear up? |
| 17:39.30 | fauxalliance | CONTACT DERMATITIS... .it will go away when he stops playing with it drmessano , not HPV, not herpes... arentt you supposed to be a Doctor or somethin :P |
| 17:39.33 | WIMPy | Maybe not a single one, but a combination of two should do. |
| 17:39.43 | [TK]D-Fender | NEEDINGHELP123: Pleny of libs out there for the packets to send and the full format of the responses. |
| 17:39.52 | NEEDINGHELP123 | don't want naother library |
| 17:39.54 | NEEDINGHELP123 | i want my own library |
| 17:39.55 | NEEDINGHELP123 | that's fine |
| 17:39.58 | NEEDINGHELP123 | i just need the packet |
| 17:40.02 | NEEDINGHELP123 | i'd rather not have to wireshark the packet out |
| 17:40.05 | NEEDINGHELP123 | and then decode it |
| 17:40.08 | NEEDINGHELP123 | and encode it |
| 17:40.09 | NEEDINGHELP123 | no |
| 17:40.11 | [TK]D-Fender | NEEDINGHELP123: You will have to |
| 17:40.12 | drmessano | marksaitis, I am not saying every phone is better than the iPhone.. I am saying the iPhone is FAR from the top of the heap.. and each successive version fails to innovate in any way, unless you go back in time a year |
| 17:40.13 | NEEDINGHELP123 | i'd rather someone helped me for the money i'm offering |
| 17:40.17 | NEEDINGHELP123 | [TK]D-Fender no i won't ) |
| 17:40.19 | NEEDINGHELP123 | ;) |
| 17:40.24 | NEEDINGHELP123 | believe me i won't |
| 17:40.26 | fauxalliance | drmessano, whats up the scale from douche? |
| 17:40.28 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 17:40.30 | [TK]D-Fender | NEEDINGHELP123: Wireshark probably already HAS the ability to make those readable <- |
| 17:40.35 | drmessano | fauxalliance, Bidet? |
| 17:40.36 | NEEDINGHELP123 | yep |
| 17:40.40 | NEEDINGHELP123 | then i need to read how to encode it |
| 17:40.42 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 17:40.44 | NEEDINGHELP123 | and decode it when it comes back |
| 17:40.45 | [TK]D-Fender | NEEDINGHELP123: So go install a client and place a call and spy on it |
| 17:40.47 | WIMPy | NEEDINGHELP123: Ok, append a | grep to wireshark. Do I get the $500 now? |
| 17:40.53 | NEEDINGHELP123 | no, [TK]D-Fender i prefer not to |
| 17:40.56 | marksaitis | drmessano, in ur opinion, which phone is better than iphone 4, and only iphone 4? |
| 17:41.02 | NEEDINGHELP123 | WIMPy yeah you would if you wern't such a pathetic wannabe |
| 17:41.19 | WIMPy | Thought so. |
| 17:41.29 | NEEDINGHELP123 | another one for the ignore list |
| 17:41.35 | drmessano | marksaitis, Droid X, Blackberry Torch |
| 17:41.45 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 17:41.48 | NEEDINGHELP123 | [TK]D-Fender, can you help me settle the problem? i'm going to end up banned from here soon,, it only takes another few stupid GEEKOID remarks from people |
| 17:41.55 | NEEDINGHELP123 | i'm not a fucking neekatron |
| 17:41.56 | NEEDINGHELP123 | they might be |
| 17:41.57 | NEEDINGHELP123 | but i'm not |
| 17:42.02 | WIMPy | NEEDINGHELP123: Wouldn't it be easier to just /part instead of ignoring everyone? |
| 17:42.09 | NEEDINGHELP123 | WIMPy - fuck you |
| 17:42.10 | marksaitis | drmessano, :D stop kidding ppl |
| 17:42.13 | NEEDINGHELP123 | got it? fuck you |
| 17:42.16 | drmessano | lol |
| 17:42.17 | NEEDINGHELP123 | :) |
| 17:42.18 | [TK]D-Fender | NEEDINGHELP123: You going to continue to troll people here badgering for yruo project that has nothing to do with * and put down every other means of getting your answer that is provided and not do any real research on your own? |
| 17:42.19 | russellb | wiki.asterisk.org is now live. |
| 17:42.23 | fauxalliance | NEEDINGHELP123, thats not the way to go about 'hiring' one anyway |
| 17:42.27 | thehar | russellb: that is /topic worthy |
| 17:42.29 | WIMPy | NEEDINGHELP123: Send pics! |
| 17:42.31 | russellb | true statement |
| 17:42.41 | *** join/#asterisk [Prob]CrazyMan (~Prob]Craz@217.7.249.56) |
| 17:42.43 | NEEDINGHELP123 | fauxalliance correct |
| 17:42.52 | drmessano | NEEDINGBAN123 needs a /kick |
| 17:42.52 | NEEDINGHELP123 | fauxalliance but i'm probably not going to be able to hire anyone here anyway |
| 17:42.57 | NEEDINGHELP123 | because everyone has his head u his ass |
| 17:43.01 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Visit the new official Asterisk wiki: wiki.asterisk.org |
| 17:43.12 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 17:43.38 | [TK]D-Fender | Ok, I guess that says it all.... |
| 17:43.38 | marksaitis | russellb, u forgot to add some info about asterisk18 tls srtp compatible softphones :) |
| 17:43.45 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 17:43.47 | drmessano | NEEDINGHELP123, if you recall, this is how your last visit here ended.. badly. Nobody is going to help someone who keeps yelling "FUCK YOU", threatening ignore, etc |
| 17:43.54 | *** mode/#asterisk [+b *!*@v58.sgsvr.com] by [TK]D-Fender |
| 17:43.54 | NEEDINGHELP123 | drmessano ok thanks |
| 17:43.55 | *** kick/#asterisk [NEEDINGHELP123!~chatzilla@216.191.106.163] by [TK]D-Fender (NEEDINGHELP123) |
| 17:43.58 | fauxalliance | W00t |
| 17:44.09 | drmessano | I need to go taunt him in PM.. BRB |
| 17:44.40 | [TK]D-Fender | drmessano: At your leisure |
| 17:44.42 | fauxalliance | Go Danny Go! |
| 17:44.43 | drmessano | Ok, done |
| 17:45.09 | marksaitis | ;]]] |
| 17:45.27 | drmessano | [13:45] <NEEDINGHELP123> your worse than a suciide bomber <-- :( |
| 17:45.35 | drmessano | SUCIIDE! |
| 17:45.39 | *** join/#asterisk dmast_ (~dmast@exchange.newpointe.org) |
| 17:45.43 | fauxalliance | ^^kamakize kompliment? |
| 17:45.43 | ManxPower | drmessano, and he had better not forget it!@ |
| 17:46.10 | marksaitis | :DDD |
| 17:46.22 | drmessano | Yeah |
| 17:46.25 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 17:46.34 | fauxalliance | so, can someone take a look at this rash? |
| 17:46.39 | fauxalliance | LOL! |
| 17:47.18 | drmessano | You should probably use a different nickname when posting to a forum that google indexes, where the topic is "This girl gave me this rash" |
| 17:47.31 | drmessano | Just sayin |
| 17:47.40 | joel_oliveira | hello all. using asterisk 1.4 here and a little bit new to this world of asterisk. my question is, is it possible for to change the way that logs are displaying. I am trying to get the IP in the log of someone who tries to call without being registered with Asterisk |
| 17:47.51 | joel_oliveira | well looks like I entered in the right time of conversation :D |
| 17:47.58 | fauxalliance | drmessano, thats how the internet sorts out those 'in the know' |
| 17:48.34 | [TK]D-Fender | fauxalliance / drmessano : You'd love the PM's I'm getting from him.... |
| 17:48.40 | fauxalliance | ~g foobar |
| 17:48.46 | fauxalliance | no google bot here? |
| 17:49.11 | [TK]D-Fender | fauxalliance: Probably an infobot trigger somewhere.... |
| 17:49.24 | drmessano | http://pastebin.ca/1975701 |
| 17:49.29 | drmessano | ^^^ Theres mine |
| 17:49.33 | fauxalliance | googles [+needinghelp123 +"had been kicked"] |
| 17:49.40 | fauxalliance | s/had/has |
| 17:50.04 | [TK]D-Fender | drmessano: OMG, almost identical to mine! |
| 17:50.07 | McBoingbo | still trying to peer Asterisk to another Asterisk machine, keep getting "Failed to authenticate on INVITE to" |
| 17:50.07 | fauxalliance | roflcakes |
| 17:50.12 | [TK]D-Fender | drmessano: Cut&paste ranting! |
| 17:50.30 | drmessano | NICE |
| 17:50.36 | fauxalliance | thinks about calling his mom. |
| 17:50.49 | drmessano | I want to find out who his teacher is |
| 17:50.55 | drmessano | I joked about it last time he was here |
| 17:51.08 | [TK]D-Fender | drmessano: http://pastebin.com/m9gJZAxz |
| 17:51.14 | WIMPy | didn't get any msgs. Guess I need to practice. :-( |
| 17:51.22 | drmessano | But I would love to call his teacher and tell them he's been on IRC looking to cheat on this assignment and offering $500 for help.. then cussing us out |
| 17:51.27 | fauxalliance | WIMPy, me neither |
| 17:51.37 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 17:51.46 | drmessano | WIMPy, it takes a lot of work to be hated like this. Worked at it for years |
| 17:52.01 | fauxalliance | drmessano, it comes to some of us naturally ;-) |
| 17:52.08 | WIMPy | Maybe I'm better at it in real life :-) |
| 17:53.21 | drmessano | Not only is he stupid, but when he said "that girl", he was very nonspecific.. I have been divorced 11 times. He could at least tell me which one |
| 17:53.43 | drmessano | Maybe he was speaking on behalf of all of them |
| 17:54.37 | [TK]D-Fender | drmessano: So its not that he's wrong... he jsut wasn't specific enough ;) |
| 17:54.40 | fauxalliance | drmessano, i don't believe in divorce, i believe in boating 'accidents' |
| 17:55.11 | McBoingbo | if I have 2 Asterisk servers I want to connect together (only really need one way connection but not sure if that makes a config difference really) using SIP, I do not need to Register them correct? |
| 17:55.47 | [TK]D-Fender | McBoingbo: Correct |
| 17:56.41 | McBoingbo | I am trying to determine what steps are needed with SIP dual Asterisk servers, I only got as far as recieving "Failed to authenticate on INVITE to", need a knudge in the right direction |
| 17:56.52 | drmessano | McBoingbo, not if you have static IPs, configure correctly, and everyone's agreed that everything will turn out fine |
| 17:57.03 | drmessano | </gerryrafferty> |
| 17:57.35 | McBoingbo | so I need to setup trunks to each other then? |
| 17:57.46 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 17:58.06 | McBoingbo | what if I only want serverA to talk to serverB and not the other way around, is it less configuration? |
| 17:58.40 | [TK]D-Fender | McBoingbo: No difference |
| 17:58.43 | voxter | Hey, in SIP, if someone passes me the Privacy: header, specifically Privacy: id, is there a way to honor the fact that this came in on an inbound leg, and retain the header on the outbound leg? or does it require dialplan logic |
| 17:58.53 | [TK]D-Fender | McBoingbo: 1 peer entry on each side. |
| 17:59.03 | [TK]D-Fender | McBoingbo: What you let them do depends on your DIALPLAN |
| 17:59.14 | McBoingbo | so this is my dialplan on ServerA "exten => 404,1,Dial(SIP/404:password@thebe.intranet.local,30)" |
| 17:59.37 | [TK]D-Fender | McBoingbo: No, I told you already MAKE A PEER ENTRY for it <------ |
| 17:59.46 | McBoingbo | I dont understand what that is |
| 17:59.55 | [TK]D-Fender | McBoingbo: sip.conf <----------- |
| 18:00.01 | McBoingbo | sip.conf, yes I know |
| 18:00.03 | [TK]D-Fender | McBoingbo: Like every PHONE and ITSP uses in there |
| 18:00.25 | McBoingbo | ok so the peer in sip.conf has to have some user that is allowed on serverB? |
| 18:00.42 | ANurmi | Can I create multiple OUTBOUNDTRUNK to set up a call hunting situation while dialing out? |
| 18:00.50 | [TK]D-Fender | McBoingbo: yes |
| 18:00.55 | fauxalliance | ANurmi, asterisk can do anything |
| 18:01.00 | [TK]D-Fender | McBoingbo: and dial out that peer entry |
| 18:02.27 | McBoingbo | so SIP/<username>:<password>@<hostname> <--- so username/password are from serverB's Peer info as well as the host is serverB correct? |
| 18:02.54 | fauxalliance | ANurmi, i have seen this for freepbx... http://geekhut.org/2010/02/freepbx-custom-context-module-delegating-outbound-routes/ |
| 18:03.20 | ANurmi | fauxalliance: Thank you I will take a look at it. |
| 18:03.22 | fauxalliance | ANurmi, I.E. the boss gets the best line |
| 18:03.42 | fauxalliance | ANurmi, I get the SIP ITSP... he likes the click that the copper makes or something. |
| 18:03.56 | [TK]D-Fender | McBoingbo: Never ever put a user & pass into extensions.conf <- |
| 18:03.58 | ANurmi | or simply we have 4 lines, and 2 of them are publicly dispersed so always busy |
| 18:04.07 | [TK]D-Fender | McBoingbo: Dial(SIP/peerpointingtotheotherbox/numbertodial) |
| 18:05.10 | ANurmi | I just need when my sales rep goes to call a customer, she isn |
| 18:05.31 | ANurmi | 't waiting on customer service to free the line, and it will just move to the next free line |
| 18:06.16 | fauxalliance | ANurmi, thats inherent.... |
| 18:06.23 | fauxalliance | 'failover' |
| 18:08.38 | McBoingbo | [TK]D-Fender: I dont understand what you mean by peer pointing to the other box, my peer info in sip.conf doesnt have anything remotely close to referencing another host....can you elaborate? |
| 18:10.31 | *** join/#asterisk simonr (~simonr@bas1-toronto05-1176310461.dsl.bell.ca) |
| 18:10.32 | fauxalliance | McBoingbo, perhaps what you named the peer definition will work... |
| 18:10.48 | McBoingbo | [TK]D-Fender: this is my peer info http://pastebin.com/FDg9N67R on serverA to connect to serverB |
| 18:11.34 | Letoric | Hello folks. Is there an easy and reliable way to force people to use a specific line for outbound calls? |
| 18:11.41 | McBoingbo | its a simple entry, nothing in there makes me think, oh thats another server |
| 18:11.51 | fauxalliance | Letoric, yes, contexts |
| 18:12.51 | Letoric | I see where you are going, but that's not quite what I need unless I'm misunderstanding. Let me paint a better picture and see if that's still your suggestion |
| 18:13.25 | McBoingbo | whats the term for info in [blabla] in sip.conf? |
| 18:13.32 | fauxalliance | Letoric, http://www.automated.it/asterisk/lah-3-6-05_2.html |
| 18:13.37 | McBoingbo | still contexts? |
| 18:13.38 | fauxalliance | see if its in there |
| 18:13.42 | Letoric | We have users that have their 'main' line, and then a helpdesk line |
| 18:13.43 | Letoric | ok |
| 18:13.44 | *** join/#asterisk pabelanger (~pabelange@12.182.24.2) |
| 18:13.54 | Letoric | When they push new call and make a call, it defaults to their main line |
| 18:14.03 | Letoric | when they dial the numbers and push speaker, it defaults to their heldpesk line |
| 18:14.09 | Letoric | I don't know why that is, or how to correct it |
| 18:14.28 | Letoric | it's a polycom soundpoint ip 670 |
| 18:14.44 | *** join/#asterisk Lantizia (~lantizia@erebus.seaquake.net) |
| 18:15.00 | Lantizia | hey crazy (or not) idea... I don't suppose T.38 SoftFax programs exist? |
| 18:15.06 | Lantizia | i.e. like a Softphone... but a Fax lol |
| 18:16.13 | fauxalliance | Lantizia, if it does, dont use it |
| 18:16.31 | Lantizia | care to elaborate |
| 18:16.52 | *** join/#asterisk citywok (~Andrew@70.35.113.66) |
| 18:17.08 | fauxalliance | t.38 = monkeys on acid |
| 18:17.32 | fauxalliance | gesticulates appropriately (for a wired monkey) |
| 18:17.40 | voxter | Any of you know of the right way to preserve Privacy headers on an incoming sip call for an outgoing sip call? |
| 18:18.03 | Pio | hm i have it set up so when i get incoming calls with a specific caller ID, it does a ParkedCall() to pick up a certain parked call number.. when the incoming call is from one sip peer, it works.. from another sip peer, it i got a "Spawn extension ... exited non-zero' .. what differences between the peers could cause this? for the record, i havent verified that the peer that fails to pick up the parked call has proper inbound/outbound audio.. i only know for |
| 18:18.03 | Pio | <PROTECTED> |
| 18:18.38 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 18:20.12 | McBoingbo | Can someone just walk me through some of the steps to allow asterisk to call another asterisk box, all the telephony terms confuse me so I am getting lost, please |
| 18:21.55 | ManxPower | which asterisk.conf option is used to specify where to dump core? |
| 18:22.54 | [TK]D-Fender | McBoingbo: I justr showed you a dial line. Now go make a proper peer. |
| 18:23.32 | McBoingbo | I dont understand how a peer on serverA referes to ServerB is what I am trying to tell you |
| 18:23.50 | McBoingbo | everywhere I am reading it shows username:password in dialplan, so its confusing |
| 18:27.37 | [TK]D-Fender | McBoingbo: Look at ANY provider entry out there period. |
| 18:27.51 | [TK]D-Fender | McBoingbo: host, username, secret, codecs, etc. |
| 18:28.07 | [TK]D-Fender | McBoingbo: Look at your PHONE entries FFS. Its all the SAME. |
| 18:28.11 | [TK]D-Fender | McBoingbo: SIP is SIP |
| 18:28.57 | McBoingbo | wow |
| 18:29.24 | joel_oliveira | Having some trouble on getting IPs on the logs. Can do it for register events but not on invite events. Does anyone has a way to put them there? |
| 18:29.36 | Pio | anyone know of a test tool where you can put in a sip address and it'll generate a call to you, preferably with an echo test? |
| 18:29.37 | joel_oliveira | it seems like I am having the same problem as this guy http://forums.digium.com/viewtopic.php?f=1&t=74947&p=147355 |
| 18:29.38 | McBoingbo | you obviously dont understand where my confusion is and thats where your frustration lies |
| 18:29.48 | McBoingbo | guess I will have to keep poking around |
| 18:30.26 | joel_oliveira | Pio: will this help: http://sipp.sourceforge.net/ ? |
| 18:31.01 | Pio | joel_oliveira, yeah that looks like it might be just what i need, i'll try it thanks |
| 18:31.10 | joel_oliveira | Pio: no problem |
| 18:32.08 | [TK]D-Fender | McBoingbo: A phone is a peer like any other. Go look what your PHONES fill in. |
| 18:35.44 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 18:38.35 | *** join/#asterisk GameGamer43|Mac (~GameGamer@12.182.24.2) |
| 18:39.16 | McBoingbo | [TK]D-Fender: Yes I heard and understand that, but I tried to pastebin one of my sip.conf phone contexts and it has nothing that would lead me to believe its to connect to another peer, like shouldnt there be host info pointing to serverB from serverA's sip.conf peers? |
| 18:40.20 | nny | what is the proper way to compile asterisk with cdr_adapative_odbc.so? |
| 18:40.36 | [TK]D-Fender | McBoingbo: Yes, there should. HOST=ip.of.other.box |
| 18:40.52 | McBoingbo | my pastebin of serverA's phone/peer [404] shouls call out to serverB |
| 18:41.11 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:41.36 | [TK]D-Fender | McBoingbo: host=dynamic <--- you put this. Unless that other box registered to this one you won't be able to contact them |
| 18:41.38 | McBoingbo | [TK]D-Fender: so if I add host=serverB's IP for context [404] on serverA it should connect? (provided 404 exists on serverB I guess) |
| 18:41.44 | *** join/#asterisk simonr (~simonr@bas1-toronto05-1176310461.dsl.bell.ca) |
| 18:42.00 | [TK]D-Fender | McBoingbo: Well.. at least it will have somewhere to CALL to. Being accepted is another matter |
| 18:43.15 | McBoingbo | [TK]D-Fender: ok so the peer [404] on serverA should have username/password/host from serverB, thats how the remote call is made, right? |
| 18:44.13 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 18:45.57 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 18:46.26 | [TK]D-Fender | McBoingbo: yes |
| 18:46.45 | McBoingbo | hey new error "Forbidden - wrong password on authentication for INVITE to" |
| 18:46.52 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 18:47.23 | Letoric | fauxalliance I didn't see anything in that link that covered the difficulty I'm having |
| 18:47.34 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 18:47.55 | ANurmi | Where do I change dialing time out for making outbound calls? |
| 18:48.16 | Letoric | The issue I'm having, which I would still appreciate help with, is that when somebody dials the number, and THEN presses speaker, on the phone, it runs through the lines and uses their helpdesk line. If they dial in other ways, it uses the correct line. |
| 18:49.52 | [TK]D-Fender | ANurmi: What "timeout"? |
| 18:50.26 | ANurmi | when i have the receiver off the hook before i can get all the digits in for a long distance call it is bumping to congestion. |
| 18:50.56 | ANurmi | but if I am to dial and then press send on the settop, it will dial out fine. |
| 18:50.58 | ManxPower | ANurmi, that is TOTALLY configured in your SIP phone. |
| 18:51.07 | ANurmi | ok |
| 18:51.17 | ManxPower | remember the call doesn't even get to asterisk until the phone thinks you are done dialing and sends all the digits at once. This is how SIP works. |
| 18:51.38 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 18:51.43 | ANurmi | ManxPower: Thanks. |
| 18:52.33 | *** join/#asterisk trapa (~trapa@d207-81-179-49.bchsia.telus.net) |
| 18:52.41 | nny | anyone know how to compile the adaptive cdr odbc module for 1.4? |
| 18:52.56 | trapa | Does anyone know where i should go to get info on how to reset a password on a Digium switchvox aa60 |
| 18:53.15 | espiceland | nny: ./configure && make && make install |
| 18:53.56 | nny | espiceland: heh so it is included. Maybe need to tell asterisk to load the module? |
| 18:54.07 | espiceland | No, it is not included in Asterisk 1.4 |
| 18:54.16 | espiceland | it's located in a separate subversion repository |
| 18:54.43 | nny | espiceland: oh ok this? http://svn.digium.com/svn/asterisk/branches/1.6.0/cdr/cdr_adaptive_odbc.c |
| 18:55.00 | espiceland | That's for Asterisk 1.6.0. Use: http://svn.digium.com/svn/asterisk-addons/branches/1.4/ |
| 18:55.09 | espiceland | Oh hold up |
| 18:55.16 | *** join/#asterisk delroy (~delroy@tba.usask.ca) |
| 18:55.30 | espiceland | was thinking cdr_odbc.c |
| 18:55.39 | espiceland | have you tried "make menuselect" ? |
| 18:56.09 | delroy | Any opinions as to the best analog TDM cards to use? Brand? |
| 18:56.20 | nny | espiceland: I can one sec |
| 18:56.22 | delroy | Rhino, Sangoma? |
| 18:56.22 | McBoingbo | one more favor, can someone look at this and tell me why I am still getting " Forbidden - wrong password on authentication for INVITE to" here are my details http://pastebin.com/mpKEkCtQ |
| 18:57.52 | nny | espiceland: make menuselect gives me some options, looking for the proper category |
| 18:58.08 | espiceland | delroy: I don't know much about them, but Digium has analog TDM cards: http://www.digium.com/en/products/analog/ |
| 18:59.34 | *** join/#asterisk Guest64399 (~saint42@c-76-98-53-209.hsd1.pa.comcast.net) |
| 18:59.39 | nny | espiceland: http://pastebin.com/u5cPsRJG only opitons I see for cdr |
| 18:59.44 | delroy | need 8 analog line ports and thinking of using PCIe TDM cards and DHADI with OSLEC |
| 18:59.58 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:00.13 | mmlj4 | at some point, used channelbanks begin to look attractive |
| 19:00.27 | nny | https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=9751 |
| 19:00.43 | nny | I see a link there to grab the module, not sure if it's the right one |
| 19:00.44 | bn-7bc | I have a strange problem astrisk 1.8.0 willnotbind to ipv6 Pastibin for relavant sip.conf: http://pastebin.com/a9vtRTM8 |
| 19:01.38 | *** join/#asterisk eugeneoden (~goden@99-62-173-58.lightspeed.austtx.sbcglobal.net) |
| 19:01.52 | bn-7bc | can anyone take a look? |
| 19:02.01 | *** part/#asterisk baileyx (~Adium@70-36-142-188.dsl.dynamic.sonic.net) |
| 19:02.07 | nny | hmm lmadsen has a comment that it's backported to community trunk here https://issues.asterisk.org/view.php?id=13333 will try that |
| 19:03.25 | espiceland | nny: It's in menuselect in 1.6.0. |
| 19:03.59 | nny | espiceland: this is 1.4 |
| 19:03.59 | fauxalliance | Letoric, there was a whole paragraph on custom contexts.... think outside the box |
| 19:04.11 | espiceland | Yep. |
| 19:04.16 | *** join/#asterisk rossand (~aross@207.219.49.68) |
| 19:04.45 | nny | espiceland: what category is it under in 1.6 menuselect? |
| 19:04.56 | espiceland | nny: may need to add a line to menuselect-tree in 1.4 |
| 19:04.56 | *** part/#asterisk rossand (~aross@207.219.49.68) |
| 19:05.01 | espiceland | it's under Call Detail Recording |
| 19:05.17 | nny | espiceland: found a link to the backported module in bug tracker but it's dead :\ |
| 19:05.18 | nny | http://svncommunity.digium.com/view/tilghman/branches/1.4/ |
| 19:05.25 | nny | maybe url change? |
| 19:06.04 | espiceland | I dunno. |
| 19:06.31 | nny | hmm |
| 19:06.44 | [TK]D-Fender | McBoingbo: Get rid of the permit/deny |
| 19:06.53 | Pio | http://pio.longstair.com/misc/extensions.conf mmm i got the google voice ringback down to an art form now |
| 19:07.12 | nny | does anyone have a linkk to 1.4 svncommunity ? I am trying to get a backport of cdr_adaptive_odbc.so for 1.4 |
| 19:07.33 | *** join/#asterisk Steveandlisa (~chatzilla@59.164.188.32) |
| 19:07.34 | [TK]D-Fender | McBoingbo: The receiver can be "host=dynamic" as you are looking to do this "one-way" |
| 19:07.40 | McBoingbo | [TK]D-Fender: I dont think you can remove it, so I tried putting in the serverA IP and still fails with auth fail |
| 19:07.52 | [TK]D-Fender | McBoingbo: And you currently have them pointing to the SAME IP which clealry would be bad in ANY scenario |
| 19:08.17 | [TK]D-Fender | McBoingbo: and on the receiver change the type to "user" from "peer" |
| 19:11.08 | nny | maybe svncommunity is down. Just my luck |
| 19:12.36 | *** join/#asterisk simonr (~simonr@bas1-toronto05-1176310461.dsl.bell.ca) |
| 19:13.12 | McBoingbo | [TK]D-Fender: they have the same IP because they should no? ServerA [404] context is ServerB's info, as you said host=IP of serverB, on ServerB 404 is an extension, so naturally host= local ip no? local ip on serverB and serverA 404 host is serverB so they are the same, what did I miss? |
| 19:13.48 | McBoingbo | [TK]D-Fender: what about context= on each side? |
| 19:19.19 | *** join/#asterisk Wassim (~IceChat7@89.108.165.110) |
| 19:21.37 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 19:23.29 | [TK]D-Fender | McBoingbo: can 2 people have the same PHONE NUMBER? No.... A isn't supposed to point to ITSELF |
| 19:23.49 | [TK]D-Fender | (rhetorical for the rest of you) |
| 19:24.04 | [TK]D-Fender | McBoingbo: Fix as I have suggested |
| 19:25.31 | McBoingbo | [TK]D-Fender: I did everything as you have suggested, I think one of the problems here is that I am using 404 as a peer when it is an extension, which you stated is essentially a peer |
| 19:26.32 | [TK]D-Fender | McBoingbo: Doesn't really matter. show me your current layout |
| 19:26.39 | SaiSoma | hey guys, this patch was blocked from 1.6.2, but it looks like it might be included at some point? https://issues.asterisk.org/view.php?id=8824 |
| 19:28.12 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 19:28.47 | *** join/#asterisk QbY (~kelvin@96.176.19.11) |
| 19:29.53 | QbY | if i want to declare a peer/friend but it can be any ip address from 10.0.0.1 - 10.0.0.30 or 10.0.0.0/27.... is it possible? or do i have to make one for each |
| 19:30.44 | McBoingbo | [TK]D-Fender: http://pastebin.com/EEWRC2nY |
| 19:34.27 | ManxPower | QbY, see permit=/deny= Have you looked at the sample sip.conf that comes with Asterisk? |
| 19:35.01 | QbY | i was under the impression that was for general |
| 19:35.04 | QbY | not for a peer |
| 19:35.09 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 19:35.47 | QbY | well darn |
| 19:37.23 | [TK]D-Fender | McBoingbo: What is your error? |
| 19:37.24 | QbY | so i'd set the host=dynamic and then deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.255.224 |
| 19:40.01 | nny | [TK]D-Fender: The provider that I was having ringing issues with says they were passing data "as is" and have changed it to force 180 ringing, yet still no ring. I have tried it with another provider and it works fine. |
| 19:40.08 | nny | [TK]D-Fender: The customer's group was set to "pass as is", which means that if got a 183 (session progress) instead of a 180 (ringing) we would send a 183 to the customer. We changed the configuration on the group to force 180 for terminating SIP and TDM responses <-- their reponse |
| 19:40.29 | ManxPower | nny, does the call get answered in your dialplan at any point? If so I have an idea that may help. |
| 19:40.48 | nny | ManxPower: been testing directly with a Dial statement |
| 19:41.01 | [TK]D-Fender | ManxPower: No, we're specifically AVOIDING "r" |
| 19:41.18 | nny | ManxPower: literally exten => _X.,1,Dial(SIP/${EXTEN}@provider,40,) |
| 19:41.26 | nny | (just to test) |
| 19:41.31 | [TK]D-Fender | ManxPower: He wants proper supervision so it doesn't fuck with his billing, etc |
| 19:42.10 | nny | They're testing further now |
| 19:43.08 | nny | just odd that vitelity has no issue, yet this one does heh. Not sure what I am suppose to do |
| 19:45.03 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 19:55.20 | *** join/#asterisk abel408 (429863dc@gateway/web/freenode/ip.66.152.99.220) |
| 19:57.00 | *** part/#asterisk benklop (~bklopfens@little-black-box.vmware.com) |
| 19:57.16 | abel408 | Hey everyone. I'm having a rangback issue. Incomming sip calls do not hear any ringing (just silence) when calling into my asterisk system. I have tried playing around with progressinband with no luck. The only thing I see strange is the order in which sip send "180 ringing" and "183 Session Progress" messages. The 183 messages gets sent before the 180 ringing message. |
| 19:57.48 | abel408 | ringback* |
| 19:58.16 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 19:59.14 | SaiSoma | hey guys, i need called party id as referenced here: https://issues.asterisk.org/view.php?id=8824 it looks like it should be available, including looking here: https://reviewboard.asterisk.org/r/201/ |
| 19:59.27 | SaiSoma | but it doesn't seem to be in 1.6.2.8 (my current version) |
| 20:04.25 | ManxPower | I don't think it is available in 1.6.x. |
| 20:04.40 | SaiSoma | is it in 1.8 perhaps? |
| 20:04.51 | ManxPower | it is in 1.8 AFIK |
| 20:05.00 | SaiSoma | excellent. i'l install on test box to see |
| 20:05.02 | SaiSoma | thans |
| 20:05.05 | SaiSoma | thanks* |
| 20:05.05 | *** join/#asterisk jkroon (~jkroon@dsl-241-243-218.telkomadsl.co.za) |
| 20:05.06 | ManxPower | notice the Asterisk version on that link you posted |
| 20:05.23 | ManxPower | then notice the comments |
| 20:05.26 | SaiSoma | i know, older. but I couldn't find anything that referenced it that was newer:( |
| 20:05.58 | nny | [TK]D-Fender: did a dump of the interface in tcpdump, if I see the 180 ringing, is there something in asterisk that could not be interpreting it? |
| 20:09.20 | nny | gah odd |
| 20:09.26 | nny | yeah I get 180 Ringing in stream |
| 20:09.35 | nny | stream/ tcpdump |
| 20:10.07 | nny | seems the only time it plays is when the ringing is in the media channel upstream. If the provider sends the 180 Ringing with no media in stream, asterisk ignores it |
| 20:13.55 | nny | http://i.imgur.com/dodwg.png |
| 20:14.02 | nny | ^^^ thats a screen cap of the dump |
| 20:14.53 | nny | http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband' |
| 20:14.56 | nny | http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband |
| 20:15.00 | nny | does this apply? |
| 20:15.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:17.37 | bn-7bc | dose anoine kbow iftrere is an issue with cebtos 5.5 the prevents Asterisk 1.8.0 to bind to ipv6, i can not get Asterisk to listen fom port 5060 (ipv6) ipv4 worjs fine, and yes I have working ipv6 on the box? |
| 20:17.59 | ManxPower | nny, I never set progressinband |
| 20:20.42 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
| 20:20.52 | aiksa[LV] | Hi everyone, long time no seen |
| 20:21.28 | aiksa[LV] | I was wondering is there a max. length for KEY:VALUE line? |
| 20:21.46 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:23.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:23.24 | nny | ManxPower: yeah, but there is no ringing in the media stream from the sip provider. What/who is suppose to generate the ring? The phone? |
| 20:23.43 | aiksa[LV] | because somehow DATA i pass to origante method gets stripped at the 85th char |
| 20:24.02 | nny | ManxPower: hmm "Polycom Phones require a non-standard setting for progressinband" |
| 20:25.39 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.62) |
| 20:25.54 | mandragor | how can I see which extensions are in a queue? |
| 20:26.08 | aiksa[LV] | queue show? |
| 20:27.41 | mandragor | that's from the asterisk CLI? |
| 20:27.49 | mandragor | can I do it from an agi script? |
| 20:28.34 | aiksa[LV] | not sure about the agi, but you can most certainly do it through the AMI |
| 20:29.00 | aiksa[LV] | AGI relates to a specific call, AMI to the PBX as whole |
| 20:30.16 | mandragor | in an AGI script would be it possible to see how many extensions are on a queue then? |
| 20:30.45 | aiksa[LV] | what are you using as your AGI scripting language? |
| 20:31.15 | mandragor | ruby |
| 20:31.17 | mandragor | adhearsion |
| 20:31.58 | aiksa[LV] | I would do something like shell("/usr/sbin/asterisk -rx 'queue show MyQueue | grep 'pattern for waiting extensions' | wc -l ") |
| 20:32.20 | aiksa[LV] | I would do something like shell("/usr/sbin/asterisk -rx 'queue show MyQueue' | grep 'pattern for waiting extensions' | wc -l ") |
| 20:32.27 | mandragor | I see |
| 20:32.28 | mandragor | thanks |
| 20:32.29 | aiksa[LV] | you`ll figure out from here? |
| 20:34.03 | mandragor | I think so |
| 20:34.07 | aiksa[LV] | I would however connect to AMI and get data from there |
| 20:34.23 | aiksa[LV] | but I guess thats just a personal prefernce |
| 20:34.35 | mandragor | hmm that might be better because I will not be running in the asterisk machine |
| 20:36.01 | aiksa[LV] | damn!!! it do seem that AMI is cutting the "Data:" key value at 80 signs, when performing originate. what a mess... |
| 20:36.08 | aiksa[LV] | anyone else seen this before? |
| 20:39.13 | nny | so in my dump the provider is sending a 183 w/SDP |
| 20:39.15 | nny | is this valid? |
| 20:40.47 | nny | http://www.mail-archive.com/users@lists.opensips.org/msg03267.html |
| 20:40.48 | nny | ^^^^ |
| 20:40.57 | nny | 180 sdpless followed by a 183 w/SDP |
| 20:46.03 | nny | actually here is the other half of the conversation (provider to peer) sorry, didn't catch the one sided nature of the screen cap http://i.imgur.com/g2KWC.png + http://i.imgur.com/dodwg.png |
| 20:46.04 | bn-7bc | ok did some more testing and I can get Asterisk to bind to either ipv6 or ipb4 bot bot both at the sane time am I truing to do the impossible here? |
| 20:46.10 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:48.40 | *** join/#asterisk SLCarrijo (SLCarrijo@200.174.143.18) |
| 20:51.38 | *** part/#asterisk SLCarrijo (SLCarrijo@200.174.143.18) |
| 20:52.27 | *** join/#asterisk SLCarrijo (SLCarrijo@200.174.143.18) |
| 20:55.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:59.36 | aiksa[LV] | oh.crap this might have some serious consequences and explains those ghosting issues i have had over past year |
| 20:59.39 | aiksa[LV] | :) |
| 21:05.32 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn210.78-98-252.t-com.sk) |
| 21:05.57 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 21:06.04 | *** join/#asterisk kn0x (~pinochle@74.63.68.42) |
| 21:06.07 | kn0x | Whats a good name for SIP digest-auth users (ie IP-authenticated users are Trunks)... I don't want to call them user[s| agents] because technically so are trunks |
| 21:10.56 | aiksa[LV] | dynamicTrunks? |
| 21:12.32 | aiksa[LV] | as in host=dynamic |
| 21:14.32 | nny | ugh |
| 21:14.38 | nny | this ringing issue is driving me batty |
| 21:15.12 | nny | http://i.imgur.com/g2KWC.png http://i.imgur.com/dodwg.png can someone please look at those tcpdumps and tell me if anything looks out of place? From what i can see, the 180 ringing is being sent, yet no ring is being generated |
| 21:18.57 | aiksa[LV] | tzafrir: hi |
| 21:19.48 | nny | willing to paypal anyone who can assist me with this issue. |
| 21:20.44 | Letoric | nny, if you are willing to pay, I would suggest calling a consultant on it. You'll get more direct attention |
| 21:21.18 | nny | Letoric: true. Just confused. I think this is a bug, but can't verify |
| 21:22.00 | *** join/#asterisk moltar_net (~Roman@CPE940c6dac4ffb-CM001ade8cc08e.cpe.net.cable.rogers.com) |
| 21:22.01 | nny | Letoric: I mean, the SIP dialog is there, yet no media is being sent during the 180 and 183 phase of the call. 200 works |
| 21:22.10 | moltar_net | Anyone here uses Mitel phones? |
| 21:23.46 | Letoric | nny: I'm not the guy to help with that issue, sorry. Was just making the suggestion of a consultant because you implied willingness to pay |
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| 21:27.02 | *** part/#asterisk [cannibalera] (~cannibale@201-35-230-227.fnsce703.dsl.brasiltelecom.net.br) |
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| 21:31.30 | aiksa[LV] | nny - neither can i. |
| 21:32.30 | aiksa[LV] | i have seen something a bit similar to this in cases when after progress signal operators woould send inband messages regarding the network status, but cant recall head or tail of what i did back then to make it work |
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| 21:34.05 | mandragor | how can I check if a user is available to answer the phone? |
| 21:34.22 | Letoric | ask them? |
| 21:34.23 | Letoric | ;p |
| 21:34.30 | Letoric | you can use hints |
| 21:34.33 | nny | aiksa[LV]: thanks, yeah I think this is the provider doing something in a way that asterisk doesn't understand, yet I can't figure out what the issue is |
| 21:34.48 | mandragor | Letoric, what kind of hints? |
| 21:35.22 | aiksa[LV] | :)) |
| 21:35.33 | aiksa[LV] | DevState in other words! |
| 21:35.45 | mandragor | I am using fonality and users login when they start working, and log off when they leave. Is this a fonality thing only? |
| 21:36.05 | Letoric | not familiar with fonality, sorry |
| 21:36.15 | aiksa[LV] | same over here. |
| 21:36.36 | Letoric | you can use hints in asterisk to establish whether a user is on the phone or not |
| 21:36.45 | aiksa[LV] | Letoric: call and ask them :) circular reference |
| 21:36.59 | Letoric | whether they are sitting at their desk....don't think a phone system is going to tell you that unless you have them log into a queue |
| 21:37.34 | nny | ahh! |
| 21:37.35 | nny | http://forums.digium.com/viewtopic.php?t=4911 |
| 21:37.36 | aiksa[LV] | hate to point @ voip-info, but, here: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate |
| 21:37.39 | nny | maybe?!!! |
| 21:37.45 | nny | i have SDP w/ 183 |
| 21:37.49 | nny | showing up in this case |
| 21:39.12 | aiksa[LV] | nny - hmm from description seems to relate to the case I was talking about |
| 21:39.43 | aiksa[LV] | mandragor: take a look at that voip-info link. perhaps that is what you are looking for |
| 21:39.50 | WIMPy | If that IS the situation, I'd call it a bug. |
| 21:40.18 | nny | WIMPy: I see the provider sending a 180 w no SDP |
| 21:40.28 | nny | and I see astrerisk responding with 183 SDP |
| 21:40.31 | nny | er |
| 21:40.33 | mandragor | thanks |
| 21:40.43 | nny | asterisk sending 183 w/ sdp to the peer |
| 21:41.07 | WIMPy | But you received 183 w/o sdp? |
| 21:41.26 | tzafrir | aiksa[LV], hi |
| 21:42.02 | aiksa[LV] | tzafrir: hi, i wa swondering if you have seen this somewhere and could nudge me in the correct direction. |
| 21:42.05 | nny | I recieved no 183 on the last call, but I believe the provider changed the setting to force 180 for testing |
| 21:42.14 | nny | no 183 from the provider** |
| 21:42.20 | nny | let me call and have them change it |
| 21:42.39 | aiksa[LV] | I have stumbled across a fact that AMI mesages are cut short at 85th char. is that by design? |
| 21:42.54 | WIMPy | So you receive inly 180 and out comes 183 and 180? |
| 21:43.21 | nny | WIMPy: http://i.imgur.com/g2KWC.png http://i.imgur.com/dodwg.png |
| 21:43.25 | aiksa[LV] | no messages, but lines in messages to be precise |
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| 21:44.03 | nny | WIMPy: I receive a 180 ringing, but asterisk sends a 180 and 183 w/sdp to the phone |
| 21:45.01 | nny | WIMPy: 10.0.0.5 and 64.194.212.250 are the asterisk box, 10.10.212.37 is the provider |
| 21:45.04 | WIMPy | And the 183 even comes after the 180. That does not look very sensible to me. |
| 21:45.23 | nny | WIMPy: the problem is no ring back unless generated on the media stream |
| 21:45.32 | nny | WIMPy: should I file a bug report? |
| 21:45.47 | nny | WIMPy: not sure what asterisk *should* be doing in this scenario |
| 21:46.30 | WIMPy | Well it seems to be responsible for the lac of the ring-back, so I'd call it a bug, yes. |
| 21:47.07 | WIMPy | And additionally I'm not sure if a 183 would be allowed after a 180, but I'm not a SIP expert. |
| 21:47.26 | WIMPy | No matter if with or w/o sdp. |
| 21:50.27 | nny | heh is this major or minor.. |
| 21:50.38 | nny | kind of debilitating to the system right now |
| 21:51.03 | tzafrir | aiksa[LV], what version of Asterisk (AMI messages cut short) |
| 21:51.43 | WIMPy | isn't there a 'normal'? It's not causing real malfunction, but it's surely annoying to users. |
| 21:52.39 | aiksa[LV] | <PROTECTED> |
| 21:53.33 | aiksa[LV] | to be more precise [Data] key in originate action. |
| 21:54.36 | nny | WIMPy: yeah heh |
| 21:54.39 | aiksa[LV] | tzafrir: sorry, forgot to HL you. |
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| 21:54.55 | nny | WIMPy: either fire or hey it's too warm. Need a smoke setting |
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| 21:57.45 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 21:58.40 | adnc | my debian installation suggests me to upgrade my 1.4.21 version of asterisk to 1.6.2 can I do this safely or do I need to consider hanges on config files? |
| 21:59.42 | drmessano | adnc, you need to read all upgrade notes and see if that applies to you |
| 22:00.07 | drmessano | You may need to fixup your settings/dialplan or you may not |
| 22:00.34 | Kobaz | sooooo, how was everyone's astricon |
| 22:00.41 | tzafrir | aiksa[LV], IIRC it was fixed in some later version. Not sure if it was left in 1.4 due to compatibility considerations or not |
| 22:00.53 | aiksa[LV] | ok. thanks |
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| 22:01.19 | aiksa[LV] | tzafrir: quick google serach didnt gave anything worth interest |
| 22:01.40 | aiksa[LV] | so excuse me for bugging you, but just wanted to find out. |
| 22:01.44 | tzafrir | adnc, almost no config file has changed by name (zapata.conf->chan_dahdi.conf . Sadly this is not handled by scripts) |
| 22:01.50 | mick_laptop | hi can someone tell me the current status of video support and encryption in asterisk? Googling gives me results from 2005 and video docs: http://www.asterisk.org/doxygen/trunk/AstVideo.html gives me a 403 |
| 22:02.06 | aiksa[LV] | as a matter of fact: this now explains a lot of issues i have been having in last year :)) |
| 22:02.15 | tzafrir | adnc, dpkg will probably nag you for config files that have changed |
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| 22:11.09 | adnc | mhh |
| 22:18.49 | timholum | I am looking at asterisk, it looks as if sms is in it but does anyone know how to tie it in so it works with verizon sprint ext... |
| 22:18.51 | timholum | ? |
| 22:19.23 | aiksa[LV] | timholum: you will be better of if you will use dedicate SMS service |
| 22:20.11 | timholum | aiksa[LV]: do you know of any good one's, I need to intigrate sms into a php application |
| 22:20.18 | aiksa[LV] | kannel |
| 22:21.08 | aiksa[LV] | timholum: although i am not sure how it works in u.s. |
| 22:21.39 | adnc | kannel is very nice, you can use most of the available sms protocols |
| 22:22.40 | timholum | I have attempted to set up kannel befor, but never successfully got it to send a text to my phone, or receave one |
| 22:23.19 | adnc | it can handle mo and mt messages |
| 22:24.11 | timholum | adnc: what is a mo or mt message? sorry I am fairly new to sms stuff ( as you can probably tell ) |
| 22:24.38 | adnc | mobile oriented and mobie terminated |
| 22:25.32 | aiksa[LV] | or you can go the easy way: and just find a company offering SMS Gw service |
| 22:25.33 | timholum | adnc: do you know of a good tutorial that could help me set it up? |
| 22:26.11 | aiksa[LV] | again - cant speak about u.s. bet in Europe the prices are in par, if not lower than standart SMS rates |
| 22:27.00 | aiksa[LV] | you will free yourself from the hassle of modem messages, PDUs and all that other stuff. |
| 22:28.19 | timholum | ahh kannel requires a modem, I think i will just search for a sms carrier ( our office is entirely voip ) |
| 22:29.06 | aiksa[LV] | timholum: oh, you were thinking of sending SMSes over SIP or the like. |
| 22:29.27 | aiksa[LV] | just find sms gw service provider |
| 22:29.33 | aiksa[LV] | you will be better of that way |
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| 22:32.45 | adnc | timholum, kannel can work with modem or sms gateways, it is quite flexible. there are plenty of tutorials out there. i'm sure one will fit your needs. unfortunately i do not know one by hard |
| 22:34.02 | timholum | ok thanks, I was expecting ( more hopping I guess ) that it would just send a message over the internet like email to the phone's, but I can see why that would not work :) |
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| 22:40.53 | adnc | i don't know if I should do the upgrade from 1.4 to 1.6, also I did read that 1.8 was announced. are there much differences to the configs? |
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| 22:45.15 | aiksa[LV] | adnc - some things are basically the same, some are totally different :) |
| 22:45.19 | aiksa[LV] | Read Upgrade.txt |
| 22:45.38 | adnc | aiksa[LV], in comparison to 1.8 or 1.6? |
| 22:46.20 | adnc | i use sip, iax have more or less a basic dialplan no analog or digital cards |
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| 22:50.03 | aiksa[LV] | i meant 1.4 to 1.6 |
| 22:50.16 | aiksa[LV] | not sure about 1.8 |
| 22:50.55 | aiksa[LV] | adnc sounds like you will not be affected too much. But DONT take my word for it, for your own sake read that file first. |
| 22:52.18 | adnc | yeahh, i'll if i find it ;) |
| 22:52.54 | adnc | aiksa[LV], do you know what the asterisk recording interface is? |
| 22:56.07 | aiksa[LV] | what? |
| 22:57.38 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 22:58.30 | aiksa[LV] | what do you mean by recording interface, audio records of phone conversations? CDR records? |
| 23:01.15 | adnc | aiksa[LV], well thats what i didn't understand aswell. there is an iphone app that says it needs ARI asterisk recordng interface, there is also a small text in voip wiki without link |
| 23:07.13 | aiksa[LV] | apparently this: http://www.venturevoip.com/news.php?rssid=942 |
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| 23:08.17 | aiksa[LV] | here are some forum posts: http://www.elastixconnection.com/index.php?option=com_fireboard&func=view&catid=12&id=25&Itemid=67 |
| 23:08.28 | aiksa[LV] | that thing look dead |
| 23:10.24 | adnc | yes it seems to be |
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| 23:32.02 | vinhdizzo | Just for the record, I resolved the call Google Voice #, asterisk process call but user getting to gv voicemail, by adding exten => s,1,Answer() |
| 23:32.02 | vinhdizzo | exten => s,n,Wait(2) |
| 23:32.02 | vinhdizzo | exten => s,n,SendDTMF(1) |
| 23:32.33 | vinhdizzo | then have your auto-attendant (dialplan extensions.conf) do whatever u want |
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| 23:51.42 | Kobaz | hmm |
| 23:51.47 | Kobaz | crash in ast_func_read() |
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