| 00:00.34 | TUplink | yup |
| 00:00.47 | TUplink | nat=yes externip is set |
| 00:02.02 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 00:02.35 | k-man | are the Siemens cordless ip phones any good? is there a particular model I should buy? |
| 00:02.40 | *** join/#asterisk TUplink (~hufft@c-24-127-76-4.hsd1.va.comcast.net) |
| 00:02.44 | k-man | is the C470IP any good? |
| 00:02.46 | TUplink | http://pastebin.com/c18DdJ8R |
| 00:02.52 | bougyman | i wouldn't use any of those. |
| 00:03.04 | bougyman | none of the cordless/dect phones i've found are very good. |
| 00:03.10 | k-man | bougyman: oh really? |
| 00:03.12 | bougyman | the longrange bluetooth sets have been best for us. |
| 00:03.24 | WIMPy | k-man: I have issues with them when using more than one mobile. |
| 00:03.30 | bougyman | even that expensive snom m3 or whatever is crap at only 150" in our building. |
| 00:03.36 | WIMPy | Well, mybe also with only one, but I can't say. |
| 00:03.36 | k-man | bougyman: like what brand/model |
| 00:03.51 | k-man | well, its just for home use |
| 00:03.52 | bougyman | we've tried a few. |
| 00:03.54 | bougyman | oh. |
| 00:03.58 | bougyman | wellt hat might be fine, then. |
| 00:04.16 | bougyman | you probably don't get 150" from the base |
| 00:04.34 | k-man | i currently use the ATA in my adsl modem to connect my cordless to asterisk |
| 00:04.40 | k-man | but its less than optimal |
| 00:05.20 | [TK]D-Fender | TUplink: You seemed to miss the point .. ***SPI DEBUG*** |
| 00:05.22 | [TK]D-Fender | SIP |
| 00:05.30 | TUplink | lol..... |
| 00:06.18 | [TK]D-Fender | k-man: Seimens DECT has gotten generally good reviews around here |
| 00:06.27 | [TK]D-Fender | k-man: and is a better idea than WiFi |
| 00:07.45 | WIMPy | Yes, dect is a lot better than wifi. |
| 00:07.46 | [TK]D-Fender | bougyman: 150 INCHES wow.... that's like only 30 longer than the screen on my * server! |
| 00:07.53 | k-man | [TK]D-Fender: yeah, i knew that the idea of wifi was generally considered bad |
| 00:08.17 | k-man | err.... bad with respect to cordless phones that is |
| 00:09.02 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:09.24 | bougyman | [TK]D-Fender: point taken, hanging head in shame. |
| 00:10.28 | TUplink | http://pastebin.com/wpg403Dx |
| 00:11.29 | [TK]D-Fender | bougyman: My warehouse manager uses an Aastra 57i CT w/ DECT handset (actually... he uses the phone as a dumb base ONLY). The range is awesome on that thing in our racking and concrete warehous. Covers the whole building really. |
| 00:11.46 | bougyman | that handset on the 57iCT is the worst. |
| 00:11.55 | bougyman | we went through them like sweet-tarts. |
| 00:12.06 | [TK]D-Fender | TUplink: Contact: <sip:1082258e1@192.168.15.230> <---- from your REGISTER. You did your NAT settings wrong. |
| 00:12.43 | [TK]D-Fender | bougyman: Mine have worked ok, but the batteries don't have a long lifespan |
| 00:12.51 | bougyman | that's what I mean |
| 00:12.52 | [TK]D-Fender | (before trashing) |
| 00:13.00 | [TK]D-Fender | bougyman: Fortunately they are cheap) |
| 00:13.01 | roxdragon | sigh |
| 00:13.03 | bougyman | for call center managers we were getting about 3 weeks out of them. |
| 00:13.10 | bougyman | even cheap that was too much to deal with. |
| 00:13.15 | bougyman | plus lots of fails. |
| 00:13.27 | bougyman | other than just battery, the headset jack was weak on them. |
| 00:13.29 | [TK]D-Fender | roxdragon: You keep blurting out random peoples names and not expressing your problems clearly. |
| 00:14.04 | [TK]D-Fender | bougyman: Never used the headset, but 3 weeks is a sign of a bad production batch or something |
| 00:17.04 | bougyman | happenned with replacements too |
| 00:17.43 | WIMPy | Hmm. Nice. Samsung IADs with 3 a/b and 1 S0 for 10,95. And it supports ATA mode. |
| 00:17.54 | [TK]D-Fender | bougyman: if its from the same place maybe was all the same batch.... |
| 00:18.11 | *** join/#asterisk ^daedalus^ (~daedalus@r190-132-80-74.dialup.mobile.ancel.net.uy) |
| 00:18.24 | ^daedalus^ | hi |
| 00:18.27 | TUplink | anyone see anything wierd with my sip debug? |
| 00:19.02 | ^daedalus^ | http://www.rowetel.com/blog/?page_id=452 |
| 00:19.07 | TUplink | i see a bunch of these Retransmitting #3 (NAT) to 204.155.28.10:5060: |
| 00:19.09 | ^daedalus^ | there is a new gpl voice codec |
| 00:19.27 | ^daedalus^ | 2500 bit/s |
| 00:19.41 | roxdragon | [TK]D-Fender, I decided to implement a video call with Asterisk ... I changed the sip.conf and extensions.conf follow the guidance you need on voip-info but I had no result |
| 00:19.53 | [TK]D-Fender | [20:12]<[TK]D-Fender>TUplink: Contact: <sip:1082258e1@192.168.15.230> <---- from your REGISTER. You did your NAT settings wrong. |
| 00:20.08 | [TK]D-Fender | roxdragon: Show. Us. The. Failed. CALL. |
| 00:21.27 | TUplink | ok... so is it the nat=yes and externip= ? |
| 00:21.35 | TUplink | or should there be more than that? |
| 00:22.01 | TUplink | or is it canreinvite? |
| 00:22.12 | roxdragon | [TK]D-Fender, is not the problem before, but another ... that's been fixed .. Now it is a video call |
| 00:22.12 | [TK]D-Fender | TUplink: I also have little faith you put them in the right PLACE. You should also have set your ITSP peers to nat=NO. |
| 00:22.40 | [TK]D-Fender | roxdragon: "it doesn't work" is a crappy description. Get off your ass and SHOW USE SOMETHING. |
| 00:23.48 | TUplink | ITSP? |
| 00:23.58 | roxdragon | I'll show you the file sip.conf? |
| 00:23.58 | roxdragon | <PROTECTED> |
| 00:24.00 | TUplink | the trunk? |
| 00:24.04 | [TK]D-Fender | TUplink: yes |
| 00:24.40 | roxdragon | hear the audio but not see the video |
| 00:25.12 | [TK]D-Fender | roxdragon: I see no debug. I see no configs. |
| 00:25.21 | TUplink | i have nat=YES in there and then in sip_nat.conf i have nat=yes externip=24.127.76.X |
| 00:25.46 | TUplink | do i need a localnet= ? |
| 00:27.15 | roxdragon | [TK]D-Fender, http://paste.ubuntu.com/499387/ |
| 00:28.28 | [TK]D-Fender | TUplink: What part of "I also have little faith you put them in the right PLACE" wqas not clear? |
| 00:28.54 | [TK]D-Fender | roxdragon: where is the CALL? |
| 00:29.11 | [TK]D-Fender | roxdragon: I also don't see your [general] settings |
| 00:30.01 | roxdragon | I've just pasted the last part that the peer .. Look around the sip.conf file? |
| 00:30.03 | roxdragon | [TK]D-Fender, |
| 00:30.27 | [TK]D-Fender | roxdragon: Where? |
| 00:30.39 | [TK]D-Fender | nvm |
| 00:31.00 | [TK]D-Fender | roxdragon: what do you mean "look around"? Pastbin your [general] section and SHOW US THE CALL |
| 00:33.35 | roxdragon | [TK]D-Fender, http://paste.ubuntu.com/499390/ |
| 00:38.55 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 00:39.36 | TUplink | lol.... ok so where do i put it? |
| 00:39.54 | TUplink | sip_nat.conf would be a good start right? |
| 00:40.08 | roxdragon | [TK]D-Fender, you read? |
| 00:48.23 | [TK]D-Fender | roxdragon: First permanently remove all of the commented out lines. Second, can YOU READ? |
| 00:49.16 | *** join/#asterisk Faithful (~Faithful@180.194.1.34) |
| 00:50.54 | roxdragon | what's the parameter name? |
| 00:50.54 | roxdragon | Second, I asked you if you've already read! |
| 00:51.16 | [TK]D-Fender | roxdragon: read WHAT? |
| 00:51.25 | roxdragon | my sip.conf |
| 00:51.45 | [TK]D-Fender | roxdragon: I did, and told you to remove the 99% of garbage in there |
| 00:51.56 | [TK]D-Fender | roxdragon: And YOU don't seem to have read... |
| 00:52.31 | carrar | reading is so last year |
| 00:55.04 | roxdragon | I am confused ... sorry ... what is the parameter name? |
| 00:55.15 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 00:55.20 | Dovid | hello all |
| 00:55.26 | [TK]D-Fender | roxdragon: What parameter? When did I ever mention a specific parameter? |
| 00:55.51 | roxdragon | <[TK]D-Fender> roxdragon: First permanently remove all of the commented out line |
| 00:56.16 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:56.19 | [TK]D-Fender | roxdragon: ..... all of the lines with ";" in front. |
| 00:56.30 | [TK]D-Fender | roxdragon: Useless filler. |
| 00:57.59 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:58.01 | roxdragon | ok now? |
| 00:58.28 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:59.41 | [TK]D-Fender | roxdragon: And now you've shown me nothing more, and not what I asked for repeatedly and I am out of time. |
| 01:01.35 | roxdragon | i don't speak english, There is a canal in Italian? |
| 01:02.23 | *** join/#asterisk ^daedalus^ (~daedalus@r190-132-81-112.dialup.mobile.ancel.net.uy) |
| 01:03.02 | [TK]D-Fender | [20:31]<[TK]D-Fender>roxdragon: what do you mean "look around"? Pastbin your [general] section and SHOW US THE CALL |
| 01:03.04 | TUplink | do i have to route any ports threw my router for the SipGate? |
| 01:03.14 | [TK]D-Fender | [20:28]<[TK]D-Fender>roxdragon: where is the CALL? |
| 01:03.17 | [TK]D-Fender | TUplink: Yes |
| 01:03.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 01:03.33 | [TK]D-Fender | [20:20]<[TK]D-Fender>roxdragon: Show. Us. The. Failed. CALL. |
| 01:03.56 | [TK]D-Fender | roxdragon: You just don't seem to get it. |
| 01:05.07 | [TK]D-Fender | ~sipnat |
| 01:05.07 | infobot | it has been said that sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:05.11 | [TK]D-Fender | TUplink: ^^^^^ |
| 01:05.39 | roxdragon | [TK]D-Fender, debug call? |
| 01:05.42 | [TK]D-Fender | TUplink: Go to #freepbx and enter "? nat" and "? ports" |
| 01:05.48 | [TK]D-Fender | heads outs |
| 01:11.43 | *** join/#asterisk Ta^3 (~tacvbo@gponr9-203-11-242.iusacell.net) |
| 01:16.12 | TUplink | ***[TK]D-Fender: if i told you sip show channels shows 204.155.28.10 1082258e1 4e49e9ce671f4dd 0x100 (g729) No Rx: INVITE its a natissue right |
| 01:24.15 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 01:32.57 | *** part/#asterisk TUplink (~hufft@c-24-127-76-4.hsd1.va.comcast.net) |
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| 02:24.31 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 02:25.25 | *** join/#asterisk Slingky (~na@modemcable116.187-177-173.mc.videotron.ca) |
| 02:26.08 | Slingky | guys, I have this command in a context, how to modify it so it dial an outside number also ? |
| 02:26.20 | Slingky | exten = s,2,Dial(SIP/52&SIP/55&SIP/63&SIP/51,12) |
| 02:26.28 | Slingky | I mean, at the same time |
| 02:26.38 | Slingky | I tried this: |
| 02:26.50 | WIMPy | The same way you already have four destinations. |
| 02:27.23 | Slingky | exten = s,2,Dial(SIP/52&SIP/55&SIP/63&SIP/51&${Voipms}/5149921123,12) |
| 02:27.31 | Slingky | don't seem to work |
| 02:28.00 | WIMPy | So what is Voipms set to? |
| 02:28.27 | Slingky | Sorry but I'm verry new to asterisk programming, I'm using asterisk 1.6 with freepbx 2.8 |
| 02:28.36 | Slingky | Voipms is the name of one of my trunks |
| 02:29.43 | WIMPy | Maybe you should try #freepbx. |
| 02:30.13 | Slingky | not sure they will help me with asterisk custom contexts :-( |
| 02:30.53 | WIMPy | But they might know what variables are set to what. |
| 02:31.00 | *** join/#asterisk ManxPower (~manxpower@user-24-236-87-78.knology.net) |
| 02:31.26 | WIMPy | Or is "Voipms" the name of the sip peer? |
| 02:31.48 | WIMPy | As in sip.conf that is. |
| 02:32.21 | Slingky | I think it is the trunk name or sip peer, yes |
| 02:33.12 | WIMPy | Trunk name makes no sense, but if it's the peer name, use SIP/Voipms/number. |
| 02:36.45 | Slingky | I tried : |
| 02:36.50 | Slingky | exten = s,1,Dial(SIP/62&SIP/61&SIP/60&SIP/Voipms/5149921123,25) |
| 02:36.58 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 02:37.06 | Slingky | all internal extensions ring, but outside number does not |
| 02:38.35 | Slingky | I forgot to tell you that I have this before my command: |
| 02:38.37 | Slingky | include = default |
| 02:38.44 | Slingky | do I need something else maybe ? |
| 02:40.08 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-21.lv.lv.cox.net) |
| 02:40.43 | WIMPy | Only the freepbx guys can tell you, I guess. |
| 02:41.16 | Besticles | Anyone know why when a call originated from Asterisk Box A, Dials ServerB via SIP and then PlayFile, Server B reports that it played the file, but it didn't actually play it? |
| 02:41.32 | Dovid | does anyone here use mplayer with asterisk ? |
| 03:01.21 | ChannelZ | Besticles: PlayFile? Do you mean Playback? |
| 03:02.15 | Besticles | I'm using Stream File, sorry |
| 03:04.22 | Slingky | WIMPy, thanks for your help |
| 03:23.14 | Dovid | does anyone here use mplayer with asterisk ? |
| 03:30.17 | *** join/#asterisk brunner (~chris@35-137.175-24.bham.res.rr.com) |
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| 03:30.39 | brunner | What do SS7 point codes look like? |
| 03:37.34 | Dovid | brunner: what do you mean ? where |
| 03:37.40 | Dovid | a point code is a point code |
| 03:54.01 | brunner | This is the answer I was looking for: 251-072-004 |
| 03:57.22 | drmessano | OMG THATS MY SOCIAL SECURITY NUMBER |
| 03:57.26 | drmessano | H4X0R |
| 03:58.47 | ChannelZ | Yechh. You can keep your bad credit. |
| 03:59.47 | drmessano | That's ok, my other social security number has great credit and keeps his money in offshore accounts |
| 03:59.51 | drmessano | So have at THAT one |
| 04:02.28 | ChannelZ | Schwwweeet! new guns for ChannelZ! |
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| 06:19.21 | petterl_ | I have an issue with sounds in an server, is there a way to see where asterisk searches for sound files when you get an: "file.c: File astpp-accountnum.gsm does not exist in any format" error? |
| 06:21.18 | ectospasm | petterl_: by default it's /var/lib/asterisk/sounds |
| 06:21.50 | ectospasm | you may need to verify that in /etc/asterisk/asterisk.conf [directories] |
| 06:25.07 | petterl_ | which one of the variables under directories points out the sound dir? astvarlibdir? |
| 06:25.22 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-addxvslvkdffczwx) |
| 06:26.15 | ectospasm | petterl_: yes, astvarlibdir |
| 06:26.26 | ectospasm | petterl_: unless there's one for sounds, I can't remember |
| 06:27.02 | ectospasm | ...doesn't appear to be in the sample file i'm looking at |
| 06:27.40 | ectospasm | petterl_: typically it's because you reference astpp-accountnum in the dialplan, and the system can't find a GSM version |
| 06:27.59 | ectospasm | petterl_: and for some reason it gave up on transcoding |
| 06:30.35 | petterl_ | but i have one: petter@telelabb:/etc/asterisk$ ls -l /var/lib/asterisk/sounds/astpp-accountnum.gsm |
| 06:30.35 | petterl_ | -rw-rw-r-- 1 www-data asterisk 4884 2010-09-07 19:58 /var/lib/asterisk/sounds/astpp-accountnum.gsm |
| 06:30.55 | petterl_ | is there any other way to verify why asterisk doesnt find the file? |
| 06:37.07 | ectospasm | petterl_: I'd have to see the CLI output of the failure |
| 06:37.11 | ectospasm | pastebin |
| 06:40.23 | *** join/#asterisk Ta^3 (~tacvbo@gponr9-203-11-242.iusacell.net) |
| 06:43.32 | petterl_ | http://pastebin.ca/1948073 |
| 06:43.57 | petterl_ | Im calling an incorrect number i know that but i want the soundfiles to tell me so... |
| 06:44.59 | ectospasm | pastebin.ca seems to be responding slowly |
| 06:45.00 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.28.120) |
| 06:46.10 | ectospasm | petterl_: that's not enough detail. What does the line look like in your dialplan where those files are getting called? |
| 06:46.20 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 06:46.31 | ectospasm | petterl_: you do realize that you need to put them in the dialplan *without* an extension? |
| 06:46.41 | petterl_ | pastebin.com gave 500 error for me =( |
| 06:47.02 | ectospasm | So, have something like Playback(goodbye) not Playback(goodbye.gsm) |
| 06:47.17 | ectospasm | ...because it's gonna try goodbye.gsm.gsm |
| 06:48.47 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
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| 06:52.40 | wdoekes2 | Qwell: excellent. thanks :) |
| 06:53.25 | petterl_ | ectospasm: Ok, its actually from an agi file... i digging down now... thanks |
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| 07:05.40 | Justman | hi guys, i make a simple module, which allow to broadcast device states (here is it http://pastebin.org/1142290), but when i'm unloading module, i receive segfault, what can be wrong in this module ? thanks for any help |
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| 07:11.41 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 07:47.36 | TobSnyder | Hello |
| 07:48.26 | TobSnyder | is there any document that describes which right level is needed for what command when using AMI? |
| 07:50.54 | Justman | doc/manager_1_1.txt for 1.6 |
| 07:51.11 | ChannelZ | manager show commands |
| 07:52.10 | TobSnyder | Justman: I've just read cat /usr/share/doc/asterisk-1.4.33.1/manager.txt but this doesn't cover such information |
| 07:52.26 | TobSnyder | ChannelZ: thanks that seems to have the needed information there |
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| 07:56.22 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:56.24 | schmidts | good morning |
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| 08:27.03 | stix | Does it matter if I write my register-string like this: "register=user:secret@host" or like this: "register => user:secret@host" ? I am using 1.6. |
| 08:27.30 | ectospasm | stix: I think it needs the => but it will not hurt to try = |
| 08:27.37 | ectospasm | I know = works for the dialplan |
| 08:28.06 | stix | I noticed on another system which uses freepbx, it was written by freepbx with only = |
| 08:28.24 | stix | but I have always written them myself with => |
| 08:29.34 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 08:34.58 | ectospasm | stix: yeah, that's the kind of thing that doesn't usually matter |
| 08:35.24 | ectospasm | stix: so it doesn't suprise me that = only does work |
| 08:35.46 | *** join/#asterisk francispereira (~francis@124.153.66.246) |
| 08:37.33 | stix | hmm |
| 08:37.43 | stix | it should be documented somewhere |
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| 08:51.17 | wdoekes2 | stix: indeed, there is no difference |
| 08:51.28 | stix | great :) |
| 08:51.35 | wdoekes2 | I believe it's documented somewhere.. I don't remember where though |
| 08:54.05 | wdoekes2 | in main/config.c: /* Ignore > in => */ |
| 08:54.06 | wdoekes2 | <PROTECTED> |
| 08:54.06 | wdoekes2 | <PROTECTED> |
| 08:54.06 | wdoekes2 | <PROTECTED> |
| 08:54.06 | wdoekes2 | <PROTECTED> |
| 08:54.17 | wdoekes2 | that's not the documentation though ;) |
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| 08:59.09 | wdoekes2 | beware, according to configuration.tex, there is a difference in a couple of files: |
| 08:59.12 | wdoekes2 | In some files, (e.g. mgcp.conf, dahdi.conf and agents.conf), the syntax |
| 08:59.14 | wdoekes2 | is a bit different. In these files the syntax is as follows: |
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| 08:59.35 | shamelessn00b | anyone used DENOISE? |
| 08:59.46 | shamelessn00b | on a live setup? |
| 09:00.29 | wdoekes2 | however, it shouldn't matter for any files that are loadable through the realtime-static method (after all, there is no '=' nor '=>' in the db) |
| 09:00.35 | wdoekes2 | </monologue> |
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| 10:37.08 | EmleyMoor | How many processor cores would be recommended for a sometimes-busy home system? (assume 5 simultaneous calls as a worst case) |
| 10:37.53 | kaldemar | 1 will do |
| 10:40.39 | francispereira | I have a bare basic setup where my sip.conf and extensions.conf look like this http://pastebin.com/K8rDkBta I am unable to log in from my softphone. Help ! newbie here |
| 10:40.56 | EmleyMoor | kaldemar: One running at 2.7GHz adequate? |
| 10:41.07 | EmleyMoor | will be going for 2 anyway |
| 10:42.17 | kaldemar | i used to run 6 people conferences on a 400 MHz PII. you do the math. |
| 10:44.24 | kaldemar | francispereira: define a secret for the peer (1000) and use matching credentials in the phone. |
| 10:45.22 | kaldemar | francispereira: if you still don't have success, enable sip debug and verbosity ("core set verbose 10" and "sip set debug on") in CLI and pastebin a failed registration attempt. |
| 10:45.26 | kaldemar | ~pb |
| 10:45.26 | 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. |
| 10:46.29 | francispereira | let me try that |
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| 10:50.47 | francispereira | kaldemar, what is the port asterisk should be listening on ? |
| 10:51.04 | francispereira | I cant login , and i see no debug output in CLI |
| 10:52.16 | kaldemar | francispereira: 5060 by default |
| 10:52.56 | kaldemar | francispereira: if you see no SIP messages in the cli after enabling debug, asterisk isn't getting any. |
| 10:53.55 | kaldemar | francispereira: are asterisk and the phone in the same LAN? |
| 10:54.21 | francispereira | its on the same maching |
| 10:54.24 | francispereira | my laptop |
| 10:54.37 | francispereira | i am unable to connect to port 5060 |
| 10:54.48 | francispereira | asterisk is running |
| 10:55.00 | francispereira | and iptables is stopped |
| 10:55.11 | francispereira | what could be the problem |
| 10:55.30 | kaldemar | is asterisk listening on the port? run "netstat -nlp" in shell. |
| 10:56.24 | kaldemar | there can be many problems. the phone can be listening to 5060 and prevent asterisk from doing so, you may have misconfigured the phone and so on. |
| 10:57.06 | francispereira | i have killed the phone app |
| 10:57.10 | *** join/#asterisk diegoviola (~diego@190.23.48.144) |
| 10:57.25 | francispereira | i will connect to the server from another machine |
| 10:57.25 | *** part/#asterisk diegoviola (~diego@190.23.48.144) |
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| 10:57.51 | diegoviola | Tayeb: lol |
| 10:57.52 | Tayeb | diegoviola: :P |
| 10:58.04 | francispereira | i can see asterisk on udp port 5060 is that expected ? |
| 10:58.34 | francispereira | upd port ? |
| 10:58.38 | Tayeb | diegoviola: do you use asterisk? |
| 10:58.43 | francispereira | udp |
| 10:58.57 | diegoviola | Tayeb: no, only freeswitch and you? :) |
| 10:59.21 | Tayeb | diegoviola: asterisk sometime but mostly FS. |
| 10:59.33 | diegoviola | i used to use asterisk too |
| 11:00.30 | diegoviola | o |
| 11:00.36 | Tayeb | diegoviola: my problem with asterisk is the dialplan extensions that is ;( ;( ;( ;( :( :( :( :( sadly to understand |
| 11:00.52 | diegoviola | Tayeb: lol |
| 11:02.48 | *** part/#asterisk Tayeb (~Delphi@41.200.4.184) |
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| 11:03.47 | *** join/#asterisk J4zen (~J4zen@a82-95-153-17.adsl.xs4all.nl) |
| 11:04.28 | J4zen | Hi, just a quick question; I currently have a volume of about 20k minutes a month with peak usage of 10 simultaneous calls. How many E1's would i need/ What is the rule of thumb for this? |
| 11:05.18 | shamelessn00b | an E1 is 30 channels |
| 11:05.44 | kaldemar | francispereira: yes, listening to UDP 5060 is expected. |
| 11:06.31 | shamelessn00b | fractional PRI would be good for you |
| 11:06.40 | shamelessn00b | J4zen: |
| 11:07.25 | J4zen | We're expecting an massive increase before the end of this year to approx 100k minutes a month |
| 11:09.54 | shamelessn00b | dunno man we have about 48 E1s and we process in excess of 100K calls / month |
| 11:09.58 | shamelessn00b | calls* |
| 11:10.16 | riscphree | anyone have experience with skype for asterisk? having some problems :( |
| 11:11.11 | J4zen | so you handle about 720 concurrent calls? |
| 11:11.12 | shamelessn00b | peak of around ~850 cuncurrent calls |
| 11:11.24 | shamelessn00b | yesterday |
| 11:11.25 | shamelessn00b | :) |
| 11:11.52 | J4zen | i see heh |
| 11:12.27 | J4zen | well our issue is mainly that we cannot find a reliable party to provide the E1 interconnection for us, hence we feel obligated to do it ourselves; despite the additional cost |
| 11:12.55 | shamelessn00b | we are interconnected with the switches using SS7 |
| 11:13.01 | shamelessn00b | the telco I mean |
| 11:13.11 | shamelessn00b | we aren't using libPRI anymore |
| 11:13.21 | shamelessn00b | or dahdi |
| 11:13.35 | *** join/#asterisk coppice (~chatzilla@m121-203-232-14.smartone-vodafone.com) |
| 11:13.37 | J4zen | yeah i figured with that kind of volume |
| 11:13.49 | J4zen | would you mind if i ask you some questions in private? |
| 11:13.54 | shamelessn00b | sure |
| 11:13.56 | shamelessn00b | ask |
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| 11:26.59 | roxdragon | hola |
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| 11:57.53 | TobSnyder | does ignorepat also emulate dial tone even if all outgoing lines are busy? |
| 11:58.40 | francispereira | kaldemar, Even with core set debug 10 and sip set debug 10 i cant see any debug info in the CLI when a soft phone is connecting. Asterisk is running on udp port 5060 |
| 11:59.17 | bougyman | does it actually connect? |
| 11:59.35 | kaldemar | francispereira: the sip debug command is "sip set debug on" |
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| 12:01.40 | kaldemar | francispereira: and the other command i told you to use is "core set verbose 10", not "core set debug 10". |
| 12:02.12 | francispereira | kaldemar, |
| 12:02.27 | francispereira | my client can now connect with user id 1000 |
| 12:02.29 | francispereira | :) |
| 12:02.39 | francispereira | and i can see the debug output |
| 12:05.19 | *** join/#asterisk _zoom_ (~user@41.218.36.160) |
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| 12:18.53 | francispereira | what would my dial plan be if i want to forward all calls coming to an extension to a media gateway for going out ? |
| 12:18.57 | francispereira | any ideas |
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| 12:26.02 | roxdragon | hi |
| 12:26.26 | roxdragon | exist a software for n95 with codec h236p for videocall? |
| 12:26.33 | crankygeek | I am having some config issues with Asterisk, was wondering if anyone can take a few minutes and give me a hand to check if my setup is correct? |
| 12:28.14 | crankygeek | I am using a SipGate/Google voice setup and when I dial in to my GV number, I get a message telling me an error has occured |
| 12:36.03 | *** part/#asterisk _zoom_ (~user@41.218.36.160) |
| 12:42.00 | francispereira | how do i add a sip trunk in asterisk ? |
| 12:42.21 | francispereira | which file should i be looking at |
| 12:42.47 | *** join/#asterisk jamko (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 12:43.51 | Naikrovek | sip.conf |
| 12:44.33 | Naikrovek | ~book |
| 12:44.33 | 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 |
| 12:44.37 | Naikrovek | francispereira: ^^^^^^ |
| 12:45.52 | leifmadsen | god I can't wait until we've gotten an updated version of that book to refer to... |
| 12:45.56 | leifmadsen | russellb: back to work! :) |
| 12:46.32 | *** join/#asterisk metiu_ (~chatzilla@85-18-228-185.ip.fastwebnet.it) |
| 12:46.56 | Naikrovek | 1.8 stuff in the book? or 1.6 |
| 12:46.59 | drmessano | Does the 2nd Edition even include SIP? Jeez |
| 12:47.24 | Naikrovek | oh you said the CDR stuff whatever it was called would be in there |
| 12:47.29 | Naikrovek | so 1.8 at least partially |
| 12:47.46 | leifmadsen | Naikrovek: the book documenting 1.8 |
| 12:48.00 | leifmadsen | Naikrovek: CEL |
| 12:48.02 | russellb | leifmadsen: O.O |
| 12:48.06 | Naikrovek | that's it |
| 12:48.07 | leifmadsen | Naikrovek: and yes, we're documenting specifically asterisk 1.8 |
| 12:48.10 | leifmadsen | ~cel |
| 12:48.10 | infobot | well, cel is Channel Event Logging |
| 12:48.15 | leifmadsen | russellb: ya, I'm looking at you |
| 12:48.17 | Naikrovek | hells yeah baby |
| 12:48.28 | russellb | leifmadsen: bitch, i'm writing, you're the one talking on IRC |
| 12:48.34 | leifmadsen | russellb: compiling! |
| 12:48.41 | Naikrovek | the ultimate excuse! |
| 12:48.56 | russellb | http://xkcd.com/303/ |
| 12:48.59 | leifmadsen | exactly |
| 12:49.06 | Naikrovek | "why are you assholes shooting nerf guns at the secretary?" "code is compiling!" "oh okay, proceed." |
| 12:49.09 | russellb | write about compiling then |
| 12:49.12 | leifmadsen | although I'm not comfortable with the reference to a "sword fight" in that strip |
| 12:49.12 | schmidts | :D LOL |
| 12:49.55 | metiu_ | hi, is it possible to Page a group of remote numbers, then put the call on hold until I get a DTMF, then play a bell sound to everyone and finally connect the "live" audio? |
| 12:50.20 | russellb | probably |
| 12:50.25 | russellb | with some very clever dialplan |
| 12:50.36 | leifmadsen | sounds like a combination of MeetMe and Originate and some clever..... ya that stuff! |
| 12:51.24 | drmessano | MeetMe is old and busted. ConfBridge is the new hotness |
| 12:52.15 | metiu_ | the scenario is: off hook, choose "public address" and get the line, then push to talk and announce the start of the PA with a bell... |
| 12:52.25 | leifmadsen | but does it have DTMF support? |
| 12:52.46 | metiu_ | it seems to me that ConfBridge is not working with page, is it? |
| 12:53.57 | Naikrovek | why not prerecord the broadcast, page everyone, then put everyone on hold with a music on hold class that plays the prerecorded announcement |
| 12:54.31 | metiu_ | that's not an option unfortunately, the speaker is live |
| 12:54.37 | leifmadsen | correct, Page() uses MeetMe(), not ConfBridge(), although I think there is a patch or bug open to allow it to work with ConfBridge() |
| 12:54.51 | leifmadsen | does the speaker need to be live? |
| 12:55.05 | metiu_ | yes |
| 12:55.17 | Naikrovek | would be a lot easier if he didn't need to be live |
| 12:55.23 | Naikrovek | but okay |
| 12:55.30 | metiu_ | Page is working, there's only this requirement of playing a bell |
| 12:55.44 | metiu_ | I could feed it to the softphone input, anyway |
| 12:55.54 | Naikrovek | can the person speaking hold a bell up to the phone and ring it? |
| 12:56.04 | schmidts | put dialer into meetme, start a local channel which pages and playes the bell |
| 12:56.16 | schmidts | and put local channel into meetme |
| 12:56.31 | leifmadsen | Naikrovek: lol |
| 12:56.49 | drmessano | Naikrovek, what if someone doesnt have a cow handy? |
| 12:56.59 | Naikrovek | hehe |
| 12:58.20 | metiu_ | can I wait for a DTMF while in a Page, so that I add the "bell player" to the conference? |
| 12:58.29 | metiu_ | or the cow |
| 13:00.38 | schmidts | Page just starts a auto answered call to devices, why not just call them in normal way and if someone answers, do the page and cow thing |
| 13:00.58 | Naikrovek | or just get a PA system |
| 13:01.01 | leifmadsen | I need more cow bell! |
| 13:01.10 | russellb | leifmadsen: cereal bowl |
| 13:01.23 | Naikrovek | or tell everyone to "please join conference 555 for a special announcement from our fearless leader" |
| 13:01.54 | metiu_ | Naikrovek: yeah I could hand it off to some consultant |
| 13:02.04 | Naikrovek | no no |
| 13:02.14 | metiu_ | who would buy a PA system |
| 13:02.27 | Naikrovek | i'm just saying in a round about way that proactively contacting meeting participants is something i've not ever heard of |
| 13:02.43 | metiu_ | no well |
| 13:03.01 | metiu_ | the participants will be auto answering softphones connected to speakers |
| 13:03.07 | Naikrovek | for that reason it's difficult for me to get my head around exactly why you guys want to do this |
| 13:03.13 | Naikrovek | but that's okay |
| 13:03.22 | Naikrovek | there's a lot of stuff i've never seen before |
| 13:03.32 | metiu_ | like a PA system in an hotel |
| 13:03.37 | Naikrovek | ok |
| 13:03.38 | schmidts | leifmadsen something like this? http://www.lucky-mothers.de/pic/foto_seiten/2oo3/kuhglocke.gif |
| 13:03.57 | metiu_ | schmidts: lol |
| 13:04.23 | metiu_ | every floor there's a headset and one or more speakers |
| 13:04.44 | metiu_ | and from any floor you can push a button and page every speaker |
| 13:05.05 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 13:05.08 | metiu_ | only I need the ding sound like the one in the malls |
| 13:05.39 | Naikrovek | hmm |
| 13:06.14 | schmidts | but whats this dtmf thing for? |
| 13:07.50 | metiu_ | oh sorry, I was thinking of dtmf as a way of telling asterisk that the user pushed the ptt button, to make it go on with the bell |
| 13:08.15 | schmidts | you could try the following, start a local chan which does the page and then bridge this to a meetme. for the meetme you just play the bell as greeting message? |
| 13:09.06 | schmidts | Option G would do this |
| 13:11.21 | metiu_ | ok G should be the one |
| 13:11.39 | *** join/#asterisk stevo1664 (c34a9c0b@gateway/web/freenode/ip.195.74.156.11) |
| 13:11.54 | metiu_ | I'll dig the docs on how to start the local chan |
| 13:12.30 | metiu_ | optionally I'll bypass everything and just start the page as soon as the speaker presses the push to talk button the first time |
| 13:12.43 | metiu_ | only the lag will be greater if the conference is not setup |
| 13:12.57 | stevo1664 | i am stuck, I have created an IVR and now I want to direct incoming calls to 19102, which is sent over the SIP trunk to asterisk into the IVR. The trunk is working as I have a Queue on 19101 that works, but I have created an inbound route for 19102 > IVR but I must be missing something |
| 13:13.43 | stevo1664 | I am running latest asterisknow with Freepbx, can anyone assist me please |
| 13:16.00 | TobSnyder | anyone knows why I am getting 45 Method not Allowed? http://pastebin.com/nDEkQitg |
| 13:16.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:16.10 | TobSnyder | SIP Client is 3cx softphone |
| 13:19.06 | *** join/#asterisk GrumpyOldMan (~meanderis@buster.coredial.com) |
| 13:19.43 | *** join/#asterisk stevo1664 (c34a9c0b@gateway/web/freenode/ip.195.74.156.11) |
| 13:19.51 | stevo1664 | sorry I got disconnected |
| 13:19.58 | *** join/#asterisk J4zen (~J4zen@a82-95-153-17.adsl.xs4all.nl) |
| 13:20.36 | J4zen | Can anyone recommend a good read-up/book on SS7 switching combined with Asterisk or VoIP in general? |
| 13:20.58 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:21.19 | metiu_ | can I do stuff while the dialplan is bridging a call? i.e. retrying to connect to some number, playing sounds, ... it seems that the call to Dial or Page is blocking |
| 13:21.46 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 13:21.47 | J4zen | as long as you don't reload the configuration you can change whatever you want |
| 13:22.00 | Katty | :> |
| 13:22.04 | Katty | IT"S FRIDAY!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 13:22.07 | Katty | :>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> |
| 13:22.08 | *** join/#asterisk krash812 (~roni@190.196.71.206) |
| 13:22.14 | J4zen | thee cheers for Katty |
| 13:22.16 | J4zen | three* |
| 13:22.30 | metiu_ | tgif :) |
| 13:22.33 | anonymouz666 | Katty: and what does that mean ? :D |
| 13:22.33 | Katty | does a little dance |
| 13:22.36 | Katty | gets down tonight |
| 13:22.56 | Katty | anonymouz666: it means happy days are coming! |
| 13:23.34 | krash812 | hi everybody, .. a little question, how can i restart the asterisk from the linux shell , without enter in asterisk cli ? |
| 13:23.35 | krash812 | thanks |
| 13:23.48 | stevo1664 | service asterisk restart |
| 13:23.57 | J4zen | asterisk -rx "restart now" or /etc/init.d/asterisk restart or ps aux | grep asterisk, then kill its PID |
| 13:24.11 | J4zen | or that ^^ |
| 13:26.06 | stevo1664 | How do I direct incoming calls over a SIP trunk into an IVR, is that incoming routes? i guess so, I have an inbound route from 19102 >>> IVR but it rings a few times then hangs up. |
| 13:26.54 | J4zen | That's really hard to answer without actually seeing your dialplan. |
| 13:28.20 | anonymouz666 | Katty: the good thing about weekends is the fact that I don't use the computer, at least I try not use |
| 13:28.23 | anonymouz666 | heh |
| 13:28.30 | *** join/#asterisk [cannibalera] (~cannibale@200-180-9-129.fnsce703.dsl.brasiltelecom.net.br) |
| 13:28.37 | stevo1664 | yeah sorry, i am only using asterisk for queuing and IVR, I have a working Queue on 19101, but a queue I can configure with a number, I cant do that with IVR. I am sending 19102 over the trunk |
| 13:29.28 | Katty | anonymouz666: indeed. lots of social stuffs for me too |
| 13:30.19 | stevo1664 | ok i am RTFMing |
| 13:31.03 | J4zen | you you might want to be more specific about your problem, reading up is a good start ;) |
| 13:33.01 | stevo1664 | its true I have no idea, must have lucked out getting my queue working so quickly |
| 13:33.35 | roxdragon | exist a software for n95 with codec h236p for videocall? |
| 13:33.39 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 13:34.20 | schmidts | waves to katty |
| 13:34.27 | drmessano | h236p ? |
| 13:34.37 | drmessano | I doubt it |
| 13:36.06 | *** join/#asterisk Faithful (~Faithful@180.194.0.246) |
| 13:36.41 | Katty | schmidts: hello. |
| 13:41.01 | GrumpyOldMan | should I attempt to get someone else to reproduce an issue I'm having submitting a bug report. happens pretty regularly in production and on both lab servers ? |
| 13:41.08 | *** join/#asterisk Faithful (~Faithful@180.194.0.246) |
| 13:41.13 | GrumpyOldMan | I'm having before submitting. |
| 13:44.48 | Naikrovek | if you can reproduce regularly, just include steps to reproduce. |
| 13:45.00 | Naikrovek | if it happens one in 5 times, say that |
| 13:45.27 | roxdragon | drmessano, 263p |
| 13:46.11 | roxdragon | h263p |
| 13:54.37 | francispereira | How do i register a SIP trunk ?? Someone please help ! |
| 13:56.34 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 13:58.03 | GrumpyOldMan | is the a reference as to what I should set severity to ? I consider it major that there are conditions that chanspy doesn't listen in to a call properly. |
| 13:59.59 | *** join/#asterisk Faithful (~Faithful@180.194.0.246) |
| 14:01.13 | jamko | francispereira: you need to increase your verbosity. |
| 14:02.11 | francispereira | jamko, i have a media gateway where my PRI is terminated |
| 14:02.41 | francispereira | how do i define it in asterisk |
| 14:02.46 | francispereira | and in my dial plan ? |
| 14:04.24 | jamko | http://www.asteriskdocs.org/ |
| 14:05.23 | jamko | download the pdf, and then buy the new one coming out soon. You should get a good start from there. |
| 14:10.29 | *** join/#asterisk oryxtec (~oryxtec@119.152.140.76) |
| 14:10.57 | oryxtec | hello all i have this dial string exten => _X.,n,AGI(a2billing.php|1) |
| 14:11.29 | oryxtec | now i want to add 5 prefix but when dialing any outbound number i want to exclude this 5 digit. |
| 14:11.34 | oryxtec | how can i do this? |
| 14:12.20 | [TK]D-Fender | oryxtec: You aren't passing the number to that AGI and how do we know how it is getting it in the first place? We also don't support a2billing |
| 14:13.23 | oryxtec | i know u dont support a2billing but this is |
| 14:13.37 | oryxtec | regarding asterisk dial plan question |
| 14:13.41 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 14:13.54 | [TK]D-Fender | oryxtec: Again you don't PASS the number to the AGI and we don't know what it USES as the "number" |
| 14:14.11 | oryxtec | exten => _91NXXNXXXXXX,n,AGI(a2billing.php|1) |
| 14:14.13 | oryxtec | humm |
| 14:14.17 | J4zen | Can anyone recommend a good read-up/book on SS7 switching combined with Asterisk or VoIP in general? |
| 14:14.39 | oryxtec | soo i can't exclude 9 prefix |
| 14:22.10 | *** join/#asterisk [cannibalera] (~cannibale@200-180-9-129.fnsce703.dsl.brasiltelecom.net.br) |
| 14:22.29 | *** part/#asterisk [cannibalera] (~cannibale@200-180-9-129.fnsce703.dsl.brasiltelecom.net.br) |
| 14:23.20 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 14:24.40 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:26.25 | Katty | Qwell: http://www.snorgtees.com/-c-31.html?osCsid=d2061ff5b373ea6d77666f60cfd7e3ee |
| 14:29.25 | fullstop | 10:30 and I'm already hungry. |
| 14:32.13 | *** join/#asterisk Faithful (~Faithful@180.194.0.246) |
| 14:33.08 | krash812 | hello everybody , here we go again .. |
| 14:33.20 | krash812 | i got this in my asterisk |
| 14:33.22 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:33.28 | krash812 | exten => 1,1,Queue(from-pstn) |
| 14:33.28 | krash812 | exten => 1,n,Playback(espera) |
| 14:33.42 | krash812 | But the phone rings and dont stop .. |
| 14:33.53 | krash812 | even if i hangup the call .. |
| 14:34.06 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:34.06 | fullstop | I was unaware that people actually spoke esperanto. |
| 14:35.49 | Naikrovek | they certainly do |
| 14:36.49 | anonymouz666 | fullstop: 11:36 |
| 14:36.54 | anonymouz666 | close to 12 |
| 14:37.01 | anonymouz666 | lunch time |
| 14:37.50 | fullstop | Naikrovek: Any idea how many? |
| 14:38.11 | fullstop | anonymouz666: I need to change time zones. |
| 14:38.21 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 14:40.07 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:40.36 | francispereira | My calls go out using a SIP media gateway where PRI lines are terminated. I have created a user with id 1000 with which i can log in and make outbound calls. How do i forward all calls coming to id 1000 be forwarded another number. This is what my current dial plan look like http://pastebin.com/X572rNDh |
| 14:42.20 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 14:42.21 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:42.57 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:43.06 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 14:43.45 | rubberneck | Can I use the Set(CDR()=foo) function after the call has ended? I am trying to find a way to log the ${SIPCALLID} to the CDR of the call. Any suggestions? |
| 14:45.50 | Naikrovek | fullstop: no idea but i know the language is alive and kicking |
| 14:46.23 | Katty | Naikrovek: ohai |
| 14:46.31 | Naikrovek | greetings and salutations :) |
| 14:46.37 | Katty | i can haz hug? |
| 14:46.39 | Katty | now pls |
| 14:46.43 | Naikrovek | hugs Katty. |
| 14:46.47 | Katty | hugs Naikrovek |
| 14:46.50 | Naikrovek | squeezes |
| 14:46.50 | Katty | kthx, all better. |
| 14:46.53 | Naikrovek | okay good |
| 14:47.00 | Katty | tugs on Naikrovek's sleeve |
| 14:47.04 | Katty | Naikrovek: guess what :> |
| 14:47.06 | Naikrovek | what |
| 14:47.09 | fullstop | chicken butt |
| 14:47.12 | Katty | IT"S FRIDAY!!!!!!!!!!!!!!!!!!!!! :>>>>>>>>>>>>>>>>>>>> |
| 14:47.16 | Naikrovek | yes indeed :) |
| 14:47.31 | Naikrovek | did i mention this weekend is team fortress weekend? i guess that's every weekend :) |
| 14:47.41 | Katty | team fortress? |
| 14:47.42 | Katty | is that a game? |
| 14:47.45 | Naikrovek | yes |
| 14:47.47 | Naikrovek | team fortress 2 |
| 14:47.48 | Katty | oh ah :> |
| 14:47.48 | fullstop | TF2 is the best game ever made |
| 14:47.49 | Katty | sounds fun. |
| 14:47.51 | [TK]D-Fender | francispereira: What context does [trunk_media_gateway] point to? Make an exten in THERE to math the number you want to treat differently |
| 14:47.54 | Naikrovek | tf2 is awesome |
| 14:47.54 | Katty | i have laundry tonight. |
| 14:48.00 | Naikrovek | weeeee laundry |
| 14:48.03 | Katty | at the laundry mat. with my girls. |
| 14:48.04 | fullstop | It's quite the combat-themed hat simulator |
| 14:48.05 | [TK]D-Fender | ~whee |
| 14:48.05 | infobot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 14:48.11 | Naikrovek | hat simulator lol |
| 14:48.15 | Naikrovek | that's about right |
| 14:48.23 | Naikrovek | Katty: http://www.teamfortress.com/ |
| 14:48.24 | Katty | (tispy laundry)++ |
| 14:48.26 | Naikrovek | i think |
| 14:48.33 | Naikrovek | yeah drunk laundry is fun laundry |
| 14:48.44 | fullstop | Although I picked up Borderlands for cheap, so I've let TF2 idle for a bit. |
| 14:48.52 | Naikrovek | blasphemy! |
| 14:48.52 | Katty | Naikrovek: looks like a social thing |
| 14:49.04 | Naikrovek | multiplayer online team deathmatch |
| 14:49.07 | Naikrovek | soo fun |
| 14:49.09 | Katty | oooooooooh, i see. |
| 14:49.28 | fullstop | There are a lot of player classes, so you need a balanced team if you wish to win. |
| 14:49.39 | Katty | i don't have any time for more mmos |
| 14:49.44 | Katty | warcraft hogs so much time as it is. |
| 14:49.44 | fullstop | Each one has different strengths and weaknesses.. |
| 14:49.53 | fullstop | it's not mmo |
| 14:50.06 | Katty | it's still multiplayer tho |
| 14:50.15 | fullstop | not massively. :) |
| 14:50.20 | Katty | i don't even have time for wow on the weekends. |
| 14:50.26 | francispereira | [TK]D-Fender, how are you ? |
| 14:50.27 | [TK]D-Fender | spends a lot of time on RealLifeRockHero |
| 14:50.34 | fullstop | 32 player limit, 24 is typical. |
| 14:50.35 | Katty | i'm lucky if i get 8 hours of sleep before we're off again |
| 14:50.50 | francispereira | i took your advise and have managed to get my gateway going so far |
| 14:51.05 | Katty | Naikrovek: after laundry i'm making pizza. |
| 14:51.14 | Katty | Naikrovek: http://www.pizzamaking.com/forum/index.php?topic=2047.0 <- photos |
| 14:51.32 | Katty | Naikrovek: then it's off to The Library for shots. |
| 14:51.46 | fullstop | That's a lot of toppings |
| 14:51.58 | Katty | fullstop: it's chicago style deep dish |
| 14:52.12 | francispereira | [TK]D-Fender, here is what trunk_media_gateway look like |
| 14:52.13 | francispereira | http://pastebin.com/UsAVpU0R |
| 14:52.22 | francispereira | its a sip interface on the media gateway |
| 14:52.37 | fullstop | We make pizza about once a week or so during the summer, because we have a lot of sauce from the tomatoes we grow. |
| 14:52.50 | Katty | nice. |
| 14:52.51 | [TK]D-Fender | francispereira: context=outgoing_calls_to_gateway <- so put an exten in [outgoing_calls_to_gateway] for what you want to match coming from it |
| 14:52.58 | Katty | fullstop: do you have a specific recipe for pizza sauce? |
| 14:53.53 | francispereira | [TK]D-Fender, i think i got the flow wrong |
| 14:54.04 | francispereira | let me show you my extensions and sip |
| 14:55.45 | fullstop | Katty: We slice the tomatoes up into quarters, and put them in a 9x13 pyrex dish, 1 tablespoon of olive oil, 1 tablespoon of balsamic vinegar and a dash of salt. cook @400 for 40 minutes, drain and put the rest in a blender. |
| 14:55.51 | francispereira | please take a look at this http://pastebin.com/pvicPpUc [TK]D-Fender |
| 14:56.13 | fullstop | It's a lot easier than reducing the tomatoes on the stove for hours. |
| 14:56.37 | [TK]D-Fender | francispereira: bindaddr=0.0.0.0 - bindport=5060 <-- these are only legal under [general], not under peer entries |
| 14:57.11 | [TK]D-Fender | francispereira: context=outgoing_calls_to_gateway <--- and this is actually calls coming FROM your gateway, not TO it |
| 14:57.38 | [TK]D-Fender | francispereira: And you didn't even make that context..... so calls coming from your gateway should fall out. |
| 14:58.27 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 14:59.24 | Katty | fullstop: oooh. that sounds wonderful. |
| 15:01.43 | *** join/#asterisk Faithful (~Faithful@180.194.0.246) |
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| 15:02.17 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:02.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:04.03 | *** join/#asterisk Faithful (~Faithful@180.194.0.246) |
| 15:06.27 | stix | Has anyone used the astmanproxy? |
| 15:07.24 | *** join/#asterisk asilva (~structz@ip253.foar.unesp.br) |
| 15:07.29 | asilva | Hello all |
| 15:08.44 | asilva | Using Asterisk 1.6.2.13 on Debian 5 Lenny 32 bit. I Enabled the atxfer on features.conf, when i press *2 and start dialing the exten, it doens't let me dial more than one digit... any thouths? |
| 15:09.12 | [TK]D-Fender | asilva: Bad context |
| 15:09.42 | asilva | i need to type 4 digits and asterisk only taking 1 |
| 15:09.51 | asilva | it doens't let me type the other 3 |
| 15:11.53 | asilva | is there any configuration on that? |
| 15:11.56 | *** join/#asterisk vader-- (vader@c-71-225-201-226.hsd1.nj.comcast.net) |
| 15:13.27 | [TK]D-Fender | asilva: BAD CONTEXT. Loko at where things point to. |
| 15:14.34 | fullstop | asilva: also, what phone are you using? |
| 15:14.50 | asilva | YEALINK |
| 15:15.21 | asilva | ain't no bad context, asterisk receiving only one digit and my context is 4 digits... i type the first digit and it tries to match into some context and don't even let me type the other 3 |
| 15:15.22 | fullstop | I doubt this is your problem, but polycom phones also have an internal pattern match which may ship the dial string off to * before you are ready. |
| 15:15.50 | fullstop | asilva: check your dialplan and make sure that you are in the context you think you are in. |
| 15:16.01 | asilva | i Type *2 (asterisk says trasnfer and gives me the tone) so i type eg 1234 asterisk CLI gets only 1 and says not found (match context) |
| 15:16.07 | [TK]D-Fender | [11:15]<asilva>ain't no bad context, asterisk receiving only one digit and my context is 4 digits... i type the first digit and it tries to match into some context and don't even let me type the other 3 <- the CONTEXT you are pointing it to is BAD. that is why it is cutting you off |
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| 15:17.06 | *** part/#asterisk ratdeptrai (~dfavro@troy.meta-dynamic.com) |
| 15:17.49 | asilva | http://pastebin.com/tPjafzPu |
| 15:24.43 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.28.120) |
| 15:26.39 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 15:27.13 | *** join/#asterisk jlewisatl (~jlewis@solo.atlantic.net) |
| 15:27.44 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 15:30.10 | jlewisatl | I'm having some issues with moving an AGI from an older system to a 1.6 system using IAX2 to place calls...the AGI is launched from a call file. On the old system, all was well. On the new system, my "exec read ..." starts playing the sound file before the call is answered, and unless the AGI sleeps at the end, the last file played by Playback is truncated by call termination at the end of AGI execution |
| 15:35.56 | *** join/#asterisk b11d` (~no@234-200-29-134.hcc.mnscu.edu) |
| 15:38.53 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-shvyhvirmvfdjkvk) |
| 15:39.41 | jlewisatl | anyone really here? I've never seen the channel this quiet |
| 15:39.51 | b11d` | im here |
| 15:39.59 | b11d` | b11d.. professional idler |
| 15:40.09 | jlewisatl | guess you just failed |
| 15:40.16 | WIMPy | Shhh. We're trying to sleep. |
| 15:40.23 | GrumpyOldMan | not enough caffeine to talk... |
| 15:40.31 | b11d` | mods are asleep.. |
| 15:40.36 | b11d` | 4chan it up |
| 15:41.22 | [TK]D-Fender | O RLY? |
| 15:47.17 | b11d` | :) |
| 15:47.20 | b11d` | ITS A TRAP |
| 16:01.21 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:02.21 | jlewisatl | hmm...I've tested the AGI by moving it to another similarly old system...and it seems the issue with read playing the file before the call is answered was always there...the truncation of the last playback seems new to asterisk 1.6 (vs 1.0...I know...its ancient) |
| 16:10.15 | *** join/#asterisk darkdrgn2k3 (~darkdrgn2@CPE0008f8f39051-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 16:10.38 | darkdrgn2k3 | Hye any one know if the Nortel i2002 has a SIP firmware option? |
| 16:11.14 | jlewisatl | looks like I probably need to write a loop that checks channel status and waits for 6 |
| 16:11.46 | fauxalliance | darkdrgn2k3, I don't believe the i2002/i2004 phones support SIP, only Nortel's proprietary H.323/UNIStim protocol. |
| 16:12.36 | darkdrgn2k3 | fauxalliance: i understand 1.6 has it "built in" |
| 16:13.13 | darkdrgn2k3 | ? |
| 16:13.23 | Naikrovek | [TK]D-Fender: found this a while ago. O RLY http://i.imgur.com/f3ks0.gif |
| 16:13.28 | fauxalliance | darkdrgn2k3, has what built in? |
| 16:13.45 | fauxalliance | ? |
| 16:13.46 | darkdrgn2k3 | UNIStim |
| 16:13.50 | fauxalliance | gross |
| 16:14.48 | darkdrgn2k3 | ? |
| 16:15.02 | fauxalliance | darkdrgn2k3, http://www.voip-info.org/wiki/view/Asterisk+UNISTIM+channels apparently you know more than most. |
| 16:16.06 | darkdrgn2k3 | fauxalliance: nother question,any one know who can repair a i2002? |
| 16:16.21 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:16.22 | aidinb | @b11d haha - got my 'its a trap' tshirt on today |
| 16:16.29 | fauxalliance | if it's broke, DON'T fix it.... |
| 16:17.08 | fauxalliance | darkdrgn2k3, did you get the long term sell your soul to the devil service agreement? |
| 16:17.17 | darkdrgn2k3 | nop |
| 16:17.50 | fauxalliance | SOL |
| 16:17.58 | darkdrgn2k3 | ? |
| 16:18.12 | fauxalliance | JFGI |
| 16:18.32 | darkdrgn2k3 | just wondering if you had any recommendation.. |
| 16:18.45 | darkdrgn2k3 | but i guess ill just point to one randomly on google serps then |
| 16:18.52 | fauxalliance | use supported hardware (that uses open standards) and keep your soul. |
| 16:19.29 | darkdrgn2k3 | fauxalliance but i have a nice 2002 sitting right here... ' |
| 16:19.35 | darkdrgn2k3 | an no one is using it |
| 16:19.38 | jlewisatl | argh...that doesn't help either....channel status = 6 apparently as soon as the iax connection is made...even though it's still ringing out to a pstn number |
| 16:19.45 | fauxalliance | darkdrgn2k3, trade you an avaya, that too doesn't work (properly) |
| 16:19.52 | darkdrgn2k3 | :-P |
| 16:21.59 | breardo | let it be known that henceforth b11d` is now breardo |
| 16:22.15 | Naikrovek | sure thing, beardy |
| 16:22.35 | breardo | :) |
| 16:22.41 | breardo | i do rock a sweet beard |
| 16:23.05 | Naikrovek | guy i went to high school with is an electrician now, saw him the other day after 20 years doing electrical work at my company |
| 16:23.18 | Naikrovek | "epic beard man" i call him now. "kyle" was boring. |
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| 16:36.36 | breardo | epic beard man is the crazy guy from that bus who beat up that black guy who was trying to rob him |
| 16:36.39 | breardo | the name is taken |
| 16:37.57 | Naikrovek | there is more than one epic beard man |
| 16:38.34 | Qwell | no. you've not seen the real epic beard man. he's deserving of the name. |
| 16:38.40 | Naikrovek | i've seen it |
| 16:38.40 | Qwell | it's his. |
| 16:38.43 | breardo | ok... ok.. |
| 16:38.44 | Naikrovek | he's schizophrenic |
| 16:38.47 | breardo | i trust in Qwell :) |
| 16:38.51 | Naikrovek | and he has no beard anymore |
| 16:38.53 | Naikrovek | btw |
| 16:38.56 | Qwell | breardo: I was agreeing with you :p |
| 16:39.00 | breardo | oh :) haha |
| 16:39.09 | Naikrovek | so yaeh |
| 16:39.11 | breardo | well thanks.. i do think he deserves it as well |
| 16:39.29 | Naikrovek | anyway lifehacker is starting to piss me off |
| 16:39.35 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 16:39.39 | Naikrovek | every day "how to multitask better, how to get more done" |
| 16:39.52 | Naikrovek | today: "how to singletask" .... |
| 16:40.05 | Naikrovek | they're like spammers who turn around and sell you anti-spam softwarwe |
| 16:40.08 | Naikrovek | software* |
| 16:40.37 | schmidts | :D |
| 16:40.50 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-182-034.pools.arcor-ip.net) |
| 16:44.58 | *** join/#asterisk d-tech (~d-dtech@72.245.233.107) |
| 16:45.44 | jlewisatl | this is inconceivable...but a bunch of googling suggests this is an unsolved issue...asterisk can't know when the dialed call has actually been answered on the remote end |
| 16:46.57 | Qwell | Depends on the tech used, but yes, in some cases it simply isn't possible. |
| 16:47.59 | jlewisatl | in my case it's most likely going to be sip to global crossing for termination...but for testing, I'm using our PRI |
| 16:48.27 | bmoraca_work | both PRI and SIP are able to know that |
| 16:48.33 | bmoraca_work | analog does not |
| 16:49.03 | bmoraca_work | not sure about h323 because i've never used it |
| 16:49.05 | jlewisatl | in zapata.conf, I have callprogress=yes |
| 16:49.19 | bmoraca_work | what is it you're trying to accomplish? |
| 16:50.01 | darkdrgn2k3 | how can i tell what modules i have installed in asterisk |
| 16:50.01 | jlewisatl | I'm setting up a thing to help verify signups...we give you a code, you give us your phone number...we call you, and ask you to enter the code |
| 16:50.27 | darkdrgn2k3 | module show right? |
| 16:50.39 | jlewisatl | the call is done via a call file which launches an AGI that asks for your code |
| 16:51.25 | jlewisatl | it basically works...except that the exec read digits... starts playing the prompt file as soon as the call has been placed |
| 16:52.57 | bmoraca_work | use the "originate" AMI command instead. you can have asterisk dial the phone and not have the application fire off until it's answered. i thought call files could do the same, but it's been a long time since i've used them |
| 16:53.11 | bmoraca_work | or the "originate" cli command |
| 16:54.11 | jlewisatl | hmm...leme try one more test forcing sip termination first |
| 16:54.53 | bmoraca_work | that won't make a difference if your call file is flawed |
| 16:55.35 | jlewisatl | yeah...same behavior |
| 16:55.57 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-182-034.pools.arcor-ip.net) |
| 16:56.16 | bmoraca_work | "channel originate dahdi/g1/5551234 extension s@my-app" issued from CLI will dial 5551234 through dahdi group 1 and when that call is picked up (as long as dahdi group 1 is not analog) will send the call to the s extension in the my-app context |
| 16:56.21 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 16:56.28 | bmoraca_work | AMI originate is capable of the same |
| 16:57.43 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 16:57.51 | bmoraca_work | what's your call file syntax look like? |
| 17:03.26 | jlewisatl | which part? there's not much to it... a Channel: specifying what/how to dial, Callerid:, Application: AGI, Data: /var/lib/asterisk/agi-bin/veridial.agi,code,identifier |
| 17:03.47 | *** join/#asterisk Pelalil (~thomas@jack.nibsc.ac.uk) |
| 17:04.19 | jlewisatl | using channel originate at the CLI and sending it to application agi /var/lib/asterisk/agi-bin/veridial.agi,code,id has the same problem of not waiting for the call to be answered |
| 17:04.27 | fullstop | is ael going anywhere? |
| 17:08.28 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:12.48 | anonymouz666 | ael went to lunch |
| 17:12.50 | bmoraca_work | jlewisatl: the syntax is important because different syntaxes will result in different things happening |
| 17:13.16 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-182-034.pools.arcor-ip.net) |
| 17:14.02 | bmoraca_work | jlewisatl: also, you may consider not sending the call straight to an application, but rather to an extension in a context which then launches that AGI program. lots of reasons for this (answering machine detection, etc) |
| 17:14.15 | jlewisatl | I'm actually trying that now |
| 17:17.09 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 17:19.30 | jlewisatl | [ Context 'veridial' created by 'pbx_config' ] |
| 17:19.30 | jlewisatl | <PROTECTED> |
| 17:19.30 | jlewisatl | <PROTECTED> |
| 17:19.30 | jlewisatl | <PROTECTED> |
| 17:19.44 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 17:19.59 | jlewisatl | sending the call there with channel originate ... code@veridial |
| 17:20.07 | bmoraca_work | use pastebin |
| 17:20.09 | bmoraca_work | ~pastebin |
| 17:20.09 | infobot | [~pastebin] 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. |
| 17:20.11 | ManxPower | Does anyone know if it is possible to execute a dialplan function via manager, like you can execute dialplan applications? |
| 17:20.20 | jlewisatl | I get the same thing...Playback happens and finishes before the call is really answered |
| 17:20.21 | ManxPower | 1.4.x, BTW |
| 17:20.52 | ManxPower | jlewisatl, does the call show answered in the CLI before it is really answered? I assume you are not using analog |
| 17:21.20 | bmoraca_work | CLI output would be helpful, aye |
| 17:21.27 | [TK]D-Fender | jlewisatl: It IS considered answered and you aren't looking at why. |
| 17:22.57 | anonymouz666 | ManxPower: Exec(${FUNCTION()}) helps ? |
| 17:24.07 | *** join/#asterisk LucianoDSD (~chatzilla@201.54.228.178) |
| 17:24.24 | [TK]D-Fender | ManxPower: To do what? |
| 17:26.30 | jlewisatl | ok... http://pastebin.com/YRHdXigv that's from a CLI originated call...I never answered the phone |
| 17:27.18 | bmoraca_work | jlewisatl: show us "dialout-gblxprod" on "work-asterisk" |
| 17:27.57 | [TK]D-Fender | Show us the CALL and the Originate |
| 17:29.29 | bmoraca_work | jlewisatl: if "dialout-gblxprod" answers the call before forwarding it to global crossing, that would cause what you're seeing. |
| 17:31.54 | jlewisatl | it doesn't |
| 17:32.02 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 17:32.08 | jlewisatl | the contexts get a little convoluted...but I'm setting up a paste of them |
| 17:32.45 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:32.45 | *** join/#asterisk [cannibalera] (~cannibale@201-34-131-223.fnsce703.dsl.brasiltelecom.net.br) |
| 17:32.54 | *** part/#asterisk [cannibalera] (~cannibale@201-34-131-223.fnsce703.dsl.brasiltelecom.net.br) |
| 17:33.26 | jlewisatl | ok...I know this is ugly...but it's been in place and working for years http://pastebin.com/cAA7xCSV |
| 17:34.58 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 17:36.16 | bmoraca_work | hmmm...what context is the iax peer "work-asterisk" in on your production system? and, can we get a CLI output of that call going through? |
| 17:36.30 | ManxPower | [TK]D-Fender, auth against voicemail.conf, but I see there is function for that. |
| 17:36.38 | [TK]D-Fender | jlewisatl: exten => _XXXXXXXXXX,3,SetVar(ENCOUNT=0) <--- SetVar does not exist in 1.4+ |
| 17:36.52 | jlewisatl | yeah...that's a 1.0.x system |
| 17:36.54 | [TK]D-Fender | jlewisatl: Where is the CALL, and Originate? |
| 17:37.59 | jlewisatl | [TK]D-Fender: http://pastebin.com/YRHdXigv is the output from the 1.6 system...I can collect output from the 1.0 system actually placing the call |
| 17:38.24 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 17:39.22 | [TK]D-Fender | jlewisatl: "work-asterisk" <- answered you |
| 17:39.33 | jlewisatl | yeah...I actually just noticed that |
| 17:39.45 | jlewisatl | this is a disagreement I've had with my boss for years |
| 17:40.06 | jlewisatl | my testing is probably invalid because I was calling back into our system |
| 17:40.21 | [TK]D-Fender | jlewisatl: What is this "disagree" story? There things that "are", and things that "aren't". |
| 17:40.26 | jlewisatl | he insists on having our PRI all do an answer() in the pri-in context, then route the call |
| 17:40.49 | [TK]D-Fender | jlewisatl: Other system says ANSWERED. That they then pump a call out somewhere else, etc is irrelevant. |
| 17:40.51 | jlewisatl | I think the answer shouldn't be there...it should just try to route the call |
| 17:41.12 | [TK]D-Fender | [13:40]<jlewisatl>he insists on having our PRI all do an answer() in the pri-in context, then route the call <- don't use THAT code then.. or tell him TFB |
| 17:41.29 | [TK]D-Fender | jlewisatl: Use different dial-out code for this peer |
| 17:41.30 | jlewisatl | let me send the test calls to my cell and see what happens |
| 17:41.43 | jlewisatl | different dial-ou code? |
| 17:41.48 | [TK]D-Fender | jlewisatl: Doesn't matter what you call... if it hits that "Answer", you're screwed. |
| 17:41.53 | [TK]D-Fender | jlewisatl: On the 2nd * box |
| 17:42.32 | *** join/#asterisk Pelalil (~thomas@jill.nibsc.ac.uk) |
| 17:42.59 | jlewisatl | yeah...and my stance is doing this http://pastebin.com/wazN4EBP is wrong |
| 17:43.04 | jlewisatl | but he insists on it |
| 17:43.52 | [TK]D-Fender | jlewisatl: Make ANOTHER extension for your call-out to use |
| 17:44.00 | [TK]D-Fender | jlewisatl: Leave that shit of USERS |
| 17:44.02 | [TK]D-Fender | for* |
| 17:44.28 | jlewisatl | its not the call-out...the problem is the incoming context for our PRI (where my DID points) has that Answer in it |
| 17:45.05 | jlewisatl | argh...I want to strangle him |
| 17:45.16 | jlewisatl | sending to my cell phone, it works as you'd expect it to |
| 17:45.40 | [TK]D-Fender | [13:44]<jlewisatl>its not the call-out...the problem is the incoming context for our PRI (where my DID points) has that Answer in it <- Make. Another EXTEN. And point it to THERE instead for this |
| 17:46.19 | jlewisatl | oh...you're saying put an exten for my DID in the pri-in context so my calls can be routed without the Answer ? |
| 17:47.52 | [TK]D-Fender | jlewisatl: Wow, took less than 10 repetitions. Congratulations! |
| 17:48.13 | [TK]D-Fender | grabs a Kewpie doll off the top shelf for jlewisatl |
| 17:48.30 | jlewisatl | well...your saying 'call-out' was confusing me...since the breakage is on the incoming side of our system |
| 17:48.37 | carrar | Pie sounds good |
| 17:48.57 | carrar | pumpkin pie |
| 17:48.57 | WIMPy | Did someone mention f00d? |
| 17:49.09 | carrar | m/f00d |
| 17:49.11 | jlewisatl | am I right that having the system Answer all incoming calls from the PRI before trying to route them is stupid? |
| 17:49.15 | *** join/#asterisk sol (~sol@unaffiliated/sol) |
| 17:49.36 | WIMPy | goes for a clear yes. |
| 17:49.51 | [TK]D-Fender | jlewisatl: In this cae you want to process the call when answered and you are implicitly answering right away. Any otehr questions you'd like to ask for which you already possess the answer? |
| 17:50.55 | jlewisatl | nah...that's probably enough looking stupid for one day |
| 17:51.07 | carrar | oh come on |
| 17:51.08 | jlewisatl | I'll try for more next week |
| 17:51.09 | [TK]D-Fender | jlewisatl: Remember... this is your boss :p |
| 17:51.15 | jlewisatl | unless the weekend is really slow |
| 17:51.39 | [TK]D-Fender | jlewisatl: Next time he goes to shoot himself in the foot... nudge the barrel towards his temple ;) |
| 17:52.06 | jlewisatl | I'd really like to give all our dialplans a complete overhaul...they'd probably provide hours of entertainment if I pastebin'd all of them |
| 17:52.37 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:54.37 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 17:55.06 | carrar | Sounds great |
| 17:55.14 | carrar | Please finish that by MOnday |
| 17:56.26 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 17:56.29 | [TK]D-Fender | jlewisatl: We don't get enough quality humour around here. We expect you back promptly. |
| 17:56.32 | [TK]D-Fender | ENTERTAIN US |
| 17:57.04 | ks3 | Odd issue when using Dial to ring multiple (about 10) phones. Some ring immediately, some ring 2-3 seconds later, and the rest ring 2-3 seconds after the last group. Anyone thoughts? |
| 17:57.18 | *** join/#asterisk oryxtec (~oryxtec@119.152.5.17) |
| 17:57.46 | [TK]D-Fender | ks3: Here's a thought : SHOW US :) |
| 17:58.50 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 18:00.25 | KavanS | pastebin! |
| 18:00.39 | ks3 | I can give the dial command as shown on the console... Dial("SIP/customer-proxy-00000f24", "LOCAL/Donna_000*customer.domain@HUNT_GROUP_USER&LOCAL/Lisa_000*customer.domain@HUNT_GROUP_USER&LOCAL/Julie_000*customer.domain@HUNT_GROUP_USER&LOCAL/Taren_000*customer.domain@HUNT_GROUP_USER&LOCAL/Christy_000*customer.domain@HUNT_GROUP_USER&LOCAL/Matt_000*customer.domain@HUNT_GROUP_USER&LOCAL/Tommy_000*customer.domain@HUNT_GROUP_USER&LOCAL/Brian_000*customer.domain@HUNT_G |
| 18:00.39 | ks3 | ROUP_USER&LOCAL/Storage_000*customer.domain@HUNT_GROUP_USER,,M(rewrite_cdr_dialed)") |
| 18:00.52 | ks3 | I can pastebin part of the config... it's mostly pulling info dynamically using func_odbc |
| 18:01.16 | *** join/#asterisk Faithful (~Faithful@180.194.1.222) |
| 18:03.51 | ks3 | http://fpaste.org/rqhx/ |
| 18:08.13 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:11.49 | *** join/#asterisk oej (~olle@ANice-257-1-62-57.w90-52.abo.wanadoo.fr) |
| 18:12.30 | *** join/#asterisk _UsUrPeR_ (~jsass@fw.acurrus.com) |
| 18:12.51 | ks3 | I've just updated the dialplan so that all SQL queries are done prior to callin the local channels, so it doesn't seem to be an issue with a slow database |
| 18:13.22 | [TK]D-Fender | ks3: Pull the entries then do the dials at once |
| 18:13.42 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:13.59 | [TK]D-Fender | ks3: pastebin the call |
| 18:17.19 | ks3 | [TK]D-Fender, Of course the last 10 or so calls have worked properly... |
| 18:17.19 | jlewisatl | btw...thanks for the pointers |
| 18:17.37 | Katty | just made pizza dough. |
| 18:22.28 | ks3 | http://fpaste.org/iHCF/ - towards the bottom there are three groups of "Called ...." lines. Each appeared on the console 2-3 seconds after the prior. |
| 18:28.36 | jlewisatl | oh...one more thing...at the end of my agi, after doing a playback after which the script ends, should this worry me? |
| 18:28.39 | jlewisatl | [Sep 24 14:27:16] NOTICE[24063]: chan_iax2.c:5094 iax2_read: I should never be called! |
| 18:30.03 | [TK]D-Fender | jlewisatl: Is everything else good? |
| 18:30.08 | jlewisatl | yeah |
| 18:30.10 | [TK]D-Fender | ks3: What ver of *? |
| 18:30.22 | ks3 | [TK]D-Fender, 1.6.2.6 |
| 18:30.38 | jlewisatl | everything seems to be working now...just get that message every time the agi ends |
| 18:30.50 | jlewisatl | using asterisk-1.6.2.13 |
| 18:30.58 | [TK]D-Fender | ks3: 7 releases behind.. I might upgrade.. |
| 18:31.03 | jlewisatl | talking iax2 to 1.0.11.1 |
| 18:31.42 | [TK]D-Fender | jlewisatl: ANTI-FUCKING-DELUVIAN |
| 18:31.57 | fullstop | Is there a way to set a hint for a peer over an IAX2 connection? |
| 18:31.59 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 18:32.17 | fullstop | that is, IAX2/remote-server/2611 ? |
| 18:32.22 | jlewisatl | yeah...we had some issues when we tried upgrading the main systems to 1.2.x, digium couldn't figure them out (paid for support) so we just never upgraded |
| 18:32.34 | ks3 | [TK]D-Fender, Worth a shot. I'll have to try over the weekend. |
| 18:33.21 | ks3 | [TK]D-Fender, I had originally thought it may be related to the OpenSIPS proxy sitting between Asterisk and the phones, but packet captures showed that the proxy was receiving the invites at different times. |
| 18:33.24 | jlewisatl | according to *-users archive, it's nothing to worry about...but that message is like 6 years old |
| 18:34.26 | jlewisatl | when we tried upgrading to some early 1.2 release, we'd get badly garbled audio on certain types of calls |
| 18:37.20 | [TK]D-Fender | jlewisatl: * 1.8 branch is about to be released.... |
| 18:37.38 | jlewisatl | I saw that when I downloaded the code for this 1.6 box I just setup |
| 18:38.02 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 18:39.17 | jlewisatl | the only issue we have with 1.0 that would be strong encouragement to upgrade would be if whatever causes this to happen on PRI every so often had been identified and fixed |
| 18:39.20 | jlewisatl | Sep 23 10:55:28 WARNING[12120]: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. |
| 18:39.29 | *** join/#asterisk deonv (~adium@41.218.75.152) |
| 18:39.34 | jlewisatl | when it starts doing that, we can't accept incoming calls anymore |
| 18:40.16 | jlewisatl | I ended up writing something to watch the message log, and restart astersik when it sees those...and notify the agents to log back in to handle the queues |
| 18:42.40 | bmoraca_work | 1.0 is way old |
| 18:42.53 | jlewisatl | so are the systems running it |
| 18:42.57 | bmoraca_work | are you using Fonality or something? :P |
| 18:43.10 | Deeewayne | FONcore! |
| 18:43.23 | jlewisatl | never heard of it |
| 18:43.36 | jlewisatl | we've just been using * for a rather long time |
| 18:43.59 | bmoraca_work | it would probably be rather worth it to upgrade to 1.6 |
| 18:44.05 | bmoraca_work | lots of cool new features, etc |
| 18:44.11 | fullstop | wait a few weeks and go to 1.8. :P |
| 18:44.47 | jlewisatl | our dialplan would probably need a lot of massaging...and we'd have to re-figure out how the queues work |
| 18:45.02 | fullstop | with 1.0, you are hopelessly outdated. With 1.8, you will be bleeding edge and will have just as hard a time finding examples and documentation that you have with 1.0. |
| 18:45.35 | bmoraca_work | queues are a LOT easier in 1.6 than any other version |
| 18:46.43 | jlewisatl | a couple of our servers are 1.4.x...but the biggest ones that do the most stuff are 1.0 |
| 18:46.59 | fullstop | I'm still trying to figure out why exten => 2611,hint,IAX2/pbx-remote/2611 appears to show the state of IAX2/pbx-remote.. |
| 18:48.26 | bmoraca_work | i didn't know you could check state of a device across a trunk...then again, I don't do much with IAX2 and I haven't touched 1.8 yet |
| 18:49.34 | fullstop | bmoraca_work: I'm using 1.6 for this, and perhaps you can not check the state of a device across a trunk. |
| 18:49.48 | fullstop | I am vigorously trying to, though. |
| 18:49.59 | *** join/#asterisk [cannibalera] (~cannibale@200-180-11-100.fnsce703.dsl.brasiltelecom.net.br) |
| 18:50.06 | fullstop | related: http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg29166.html |
| 18:51.26 | bmoraca_work | are you trying to do BLF with a remote phone or are you just trying to determine busy/not busy? |
| 18:52.58 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 18:56.12 | fullstop | busy / not busy |
| 18:56.39 | fullstop | I don't care about BLF at all in this situation. I just don't want a remote agent getting 2 calls at the same time. |
| 18:56.55 | bmoraca_work | do all calls to those extensions for which you need to track busy/not busy go over that IAX trunk? |
| 18:56.56 | fullstop | perhaps there is a better way -- I'm new at this. |
| 18:57.14 | bmoraca_work | if they do, you can use local extensions and the concept of "groups" |
| 18:58.01 | WIMPy | There seems to have existed something that propagates device states via AMI, but I never managed to find any information on it. |
| 18:58.16 | fullstop | There is an asterisk server at each location. We are trying to avoid having calls come back to the HQ for calls in the same office. |
| 18:58.44 | fullstop | WIMPy: I saw something w/XMPP as well.. |
| 18:59.02 | bmoraca_work | do you want to restrict an agent to only one call or only one queue call? |
| 18:59.09 | WIMPy | "esel" |
| 18:59.22 | fullstop | one queue call |
| 18:59.27 | bmoraca_work | by which, i mean that if the agent is on a queue call, he can still get calls placed directly to his phone |
| 18:59.31 | WIMPy | But the esel.conf is the only trace of it I could ever find. |
| 18:59.40 | fullstop | bmoraca_work: yes, that is my goal. |
| 19:00.07 | bmoraca_work | ok, are all the queues originating from the one pbx, or are there queues to which that agent belongs on both PBXes? |
| 19:00.20 | fullstop | although, if it went straight to VM when they were on a queue call, that would be acceptable. |
| 19:00.32 | fullstop | all queues are originating from one pbx. |
| 19:01.05 | fullstop | The remote PBX is somewhat dumb -- it handles calls for the people there, and sends everything else back to the central server. |
| 19:01.24 | fullstop | inbound calls from the outside world come through the central server |
| 19:01.26 | bmoraca_work | ok, then when you add the agent to the queue, you'll want to add him as a local channel...when that local channel is dialed, you'll use GROUP_COUNT to determine whether that user is already on a queue call |
| 19:01.58 | bmoraca_work | http://www.voip-info.org/wiki/view/Asterisk+func+group |
| 19:02.07 | fullstop | Thanks. I'll look into it. |
| 19:02.19 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:02.50 | *** join/#asterisk krash812 (~roni@190.196.71.206) |
| 19:02.54 | bmoraca_work | it's kind of obscure, but it works when you get the hang of it |
| 19:03.12 | fullstop | I've been through this doc before, but nothing jumped out at me that said "USE THIS" :) |
| 19:04.05 | krash812 | hi everybody .. i need somebody just to tell me if i m wrong .. i m doing this . |
| 19:04.09 | krash812 | exten => 2,1,Queue(from-pstn2,20,r) |
| 19:04.09 | krash812 | exten => 2,n,Playback(espera) |
| 19:04.10 | roxdragon | exist a software for n95 with codec h263p for videocall? |
| 19:04.48 | bmoraca_work | what you'll do is when the queue dials the agent (which is added as like Local/2611@foreign-agents), add the channel to the group for that agent (Set(GROUP()=2611), then do a GROUP_COUNT to see if it's over 1, return busy if yes, dial agent if no |
| 19:04.59 | roxdragon | asterisk 1.4.26 support the moh format .mp3?? |
| 19:05.15 | Qwell | ~mp3 |
| 19:05.15 | infobot | [mp3] (MPEG-1 layer 3) This is a compression standard for music. It enables you to fit over 100 full length songs on a single CD with almost no loss of quality. You can find MP3 players and MP3 files on the Web--you just have to look. The music industry is unhappy about MP3 files being swapped around and has shut down some sites that distribute them.. MIME type - audio/mpeg |
| 19:05.21 | oryxtec | hi all |
| 19:05.21 | Qwell | stupid bot |
| 19:05.23 | Qwell | ~asterisk-addons |
| 19:05.37 | bmoraca_work | lol |
| 19:05.56 | *** join/#asterisk f|shie (~chatzilla@119.153.146.243) |
| 19:05.57 | [TK]D-Fender | roxdragon: Asterisk-Addons. Separate package that includes MP3 playback support |
| 19:06.01 | bmoraca_work | roxdragon: if you have compiled and installed asterisk-addons, then you can use mp3 natively in MOH |
| 19:06.18 | oryxtec | <PROTECTED> |
| 19:07.05 | oryxtec | <PROTECTED> |
| 19:07.05 | oryxtec | i m looking to get hardware with these specs http://www.serverloft.com/dedizierte-server/server-details.php?products=2 please tell me know how is this system. how many concurrent calls it can support plus their recording? |
| 19:07.11 | bmoraca_work | oryxtec: such requirements are above the capabilities of a virtualized system |
| 19:07.28 | Naikrovek | oryxtec: recording calls is not really a load on the server, and you won't have any bandwidth issues, you'll need about a t1 worth of bandwidth and a small quad core machine probably |
| 19:07.30 | fullstop | oryxtec: you can get SIP lines which deliver calls pre-encoded as g.729a |
| 19:07.30 | bmoraca_work | if it's dedicated, you'll be OK with that system |
| 19:07.46 | [TK]D-Fender | ~asterisk-addons |
| 19:07.46 | infobot | [asterisk-addons] Asterisk branches prior to 1.8 have a separate add-ons tarball taht adds MP3 playback support, MySQL apps, alternate H.323 channel drivers, and more. Highly recommended to install this... |
| 19:07.48 | fullstop | then you would be just recording and not trans-coding |
| 19:07.56 | roxdragon | [TK]D-Fender, How to check if I have addons installed? |
| 19:08.19 | [TK]D-Fender | roxdragon: Did you CHOOSE it specifically? How do you NOT know what you have? |
| 19:08.20 | oryxtec | soo this system is good |
| 19:08.23 | oryxtec | correct |
| 19:08.59 | fullstop | [TK]D-Fender: probably installed from a package. |
| 19:09.09 | oryxtec | all calls will be from my sip provider |
| 19:09.12 | [TK]D-Fender | oryxtec: Way more than enough |
| 19:09.17 | oryxtec | kool |
| 19:09.24 | fullstop | oryxtec: coming in as 729? |
| 19:09.32 | oryxtec | yes |
| 19:09.38 | roxdragon | [TK]D-Fender, I remember installing them ... I'm sure 99% |
| 19:09.40 | fullstop | overkill then. :-D |
| 19:09.53 | [TK]D-Fender | roxdragon: go see if the .so 's are installed |
| 19:10.15 | roxdragon | what's directory? |
| 19:10.17 | oryxtec | fullstop: soo you are saying i should not use g729? |
| 19:10.21 | oryxtec | in this server? |
| 19:10.28 | fullstop | No, not saying that at all. |
| 19:10.33 | [TK]D-Fender | oryxtec: Do you NEED to use it? |
| 19:10.43 | oryxtec | yes... |
| 19:10.49 | oryxtec | i need to use g729. |
| 19:10.50 | [TK]D-Fender | oryxtec: Then so be it |
| 19:10.51 | fullstop | I'm just saying that your system will not be encoding the 729 data |
| 19:11.05 | fullstop | so it will probably sit idle most of the time except for disk i/o. |
| 19:11.26 | oryxtec | why this hardware will not encode g729 data? |
| 19:12.06 | fullstop | oryxtec: Maybe it was a translation problem. What format is your SIP provider presenting your calls as? |
| 19:12.13 | roxdragon | [TK]D-Fender, what is the directory where I check? |
| 19:12.31 | oryxtec | they offer g729 and g711 |
| 19:12.51 | fullstop | If the calls come in as g729, then the recordings will just be writing the data stream to disk. |
| 19:13.13 | Qwell | fullstop: the assumption is that nothing else will actually be done with that leg of the call. |
| 19:13.13 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.wlan.noare-1.holmedal.net) |
| 19:13.19 | Qwell | which is almost certainly false. |
| 19:13.47 | [TK]D-Fender | roxdragon: just pastebin "core show modules" |
| 19:14.04 | oryxtec | or can i do this.. i will connect all calls to my provider with g711 codec.. and from my center to my hosted server all calls in g729 codec.. |
| 19:14.15 | oryxtec | will this make any differnce? |
| 19:14.38 | fullstop | Qwell: It really depends on what he is doing. |
| 19:14.52 | Qwell | He was rather clear with what he was doing. Call center. |
| 19:15.10 | Qwell | Which means there are 2 (or more) legs to each call. |
| 19:15.26 | *** join/#asterisk ahowlader (~Adnan@119.30.38.50) |
| 19:15.33 | fullstop | And if the agents have 729? |
| 19:15.53 | oryxtec | yes the agent has g729 phone |
| 19:16.02 | roxdragon | ~paste |
| 19:16.02 | infobot | extra, extra, read all about it, paste is http://pastebin.org/ or http://bin.cakephp.org/ or http://pastebin.ca/ |
| 19:16.13 | Qwell | How do you plan on recording them? Combined audio? |
| 19:16.20 | Qwell | individual files for in/out? |
| 19:16.42 | roxdragon | [TK]D-Fender, http://pastebin.org/1153172 |
| 19:16.42 | *** join/#asterisk drenda (~bond@host151-184-dynamic.0-79-r.retail.telecomitalia.it) |
| 19:16.47 | oryxtec | i m planning to record them using freepbx |
| 19:16.51 | drenda | Hi guys |
| 19:16.57 | fullstop | Obviously to combine them, you would need a license -- but depending on what you want with them, you may wish to combine them on an as-needed basis. |
| 19:17.05 | *** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:17.11 | [TK]D-Fender | roxdragon: "show modules" then |
| 19:17.39 | oryxtec | fullstop: license for wht? |
| 19:17.44 | oryxtec | u mean for g729? |
| 19:18.02 | Maliuta | <PROTECTED> |
| 19:18.02 | fullstop | oryxtec: sorry, 729 license. |
| 19:18.10 | *** join/#asterisk mpe (~mpe@x1-6-00-1e-2a-2a-b3-a2.k758.webspeed.dk) |
| 19:18.27 | roxdragon | [TK]D-Fender, http://pastebin.org/1153196 |
| 19:18.31 | fullstop | just to encode the two audio streams |
| 19:18.49 | oryxtec | i have downloaded g729 from this website http://asterisk.hosting.lv/ |
| 19:18.55 | oryxtec | and this is free.. |
| 19:18.57 | Qwell | This conversation is done. |
| 19:18.59 | Qwell | period. |
| 19:19.03 | Maliuta | Digium is nice enough to sell a channel licence for $10US, which is about $0.50 in any other currency :) |
| 19:19.04 | Qwell | oryxtec: You cannot use that codec legally. |
| 19:19.09 | fullstop | oryxtec: don't bring that up |
| 19:19.24 | [TK]D-Fender | roxdragon: format_mp3.so MP3 format [Any rate but 8000hz mono i |
| 19:19.25 | Maliuta | oryxtec: an illegal |
| 19:19.31 | Qwell | wonder if I can get Tim to have infobot to autoban on that URL. |
| 19:19.56 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 19:19.59 | oryxtec | humm |
| 19:20.03 | [TK]D-Fender | Qwell: Lots of ways of doing that |
| 19:20.12 | telnettech | ~nat |
| 19:20.13 | infobot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:20.24 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:20.31 | oryxtec | let me get in touch with digium |
| 19:20.34 | telnettech | tk: where is the link to setup nat with asterisk |
| 19:20.40 | oryxtec | to get this licnce |
| 19:20.46 | Maliuta | although _that_ URL was handy before there was a digium codec for 1.6 |
| 19:20.59 | Maliuta | oryxtec: it's on the website |
| 19:21.01 | Qwell | there have always been modules for 1.6. |
| 19:21.14 | Maliuta | Qwell: not when I first installed it |
| 19:21.19 | Qwell | Then you installed a beta. |
| 19:21.21 | roxdragon | [TK]D-Fender, ok :D, upload MP3 on var/lib/asterisk/moh ??? |
| 19:21.49 | telnettech | or anybody that knows....... where is the link that helps setup NAT with Asterisk |
| 19:22.04 | Qwell | ~sipnat |
| 19:22.04 | infobot | [sipnat] Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:22.05 | oryxtec | one thing more.. is there any quality differnce between codec which digium offers and codec which is free on that website? |
| 19:22.13 | telnettech | thanks Qwell |
| 19:22.20 | Maliuta | Qwell: It was just about the time of release, I kept watching for codecs because I had noticed some chans had borked |
| 19:22.26 | Qwell | oryxtec: When I said done, I meant done. Period. |
| 19:22.43 | oryxtec | ok thanks |
| 19:23.35 | Qwell | Maliuta: well, we always make them available when we release |
| 19:23.40 | Qwell | for 1.8, they're available now even |
| 19:23.43 | Maliuta | also thanks #deity that he has backups of his licence file and it's tied to a Gb card that can move machines |
| 19:24.38 | Maliuta | Qwell: I just remember not being able to fine one when 1.6 hit the debian repo ... although it may have been a 1.6 vs 1.6.2 issue |
| 19:24.49 | Qwell | oh, well, Debian |
| 19:25.06 | fullstop | Qwell is about to turn into the Incredible Hulk |
| 19:25.14 | Qwell | fullstop: no, that was last night. |
| 19:25.19 | fullstop | haha |
| 19:25.23 | Maliuta | Qwell: as opposed to??? baby Debian^W^WUbuntu |
| 19:25.33 | Qwell | I mean, wtf, how hard is it for somebody to listen, when I say not to stab the dragon in the face? |
| 19:25.34 | Qwell | really. |
| 19:25.36 | Qwell | </rage> |
| 19:25.38 | bougyman | Qwell: what about paying royalties for those as the guy who has the page suggests? |
| 19:25.42 | bougyman | not possible? |
| 19:25.46 | *** join/#asterisk oej (~olle@ANice-257-1-62-57.w90-52.abo.wanadoo.fr) |
| 19:26.15 | Qwell | bougyman: First off, it'd be about a million dollars for them to talk to you, plus you'd still be violating Intels copyright. |
| 19:26.27 | Maliuta | First thing I did before installing * was buy 2 channel licences for g729 |
| 19:26.41 | bougyman | Qwell: you mean patent? |
| 19:26.45 | Qwell | no |
| 19:26.46 | bougyman | that's what I meant, pay the patent holders. |
| 19:26.50 | Qwell | I mean copyright. |
| 19:26.51 | *** join/#asterisk oej (~olle@ANice-257-1-62-57.w90-52.abo.wanadoo.fr) |
| 19:27.02 | Qwell | *they* didn't write the code they're using. Intel did. |
| 19:27.44 | bmoraca | why doesn't Intel DMCA their website? |
| 19:27.48 | bougyman | it says it's based on code by Daniel Pocock, with refs. |
| 19:27.59 | bougyman | <PROTECTED> |
| 19:28.00 | Qwell | bougyman: the Asterisk module code maybe |
| 19:28.02 | bmoraca | i'm just curious |
| 19:28.08 | bougyman | no, the codec code. |
| 19:28.47 | bougyman | Intel originally offered a sample G.729 and G.723.1 implementation for Linux. |
| 19:28.47 | bougyman | Daniel Pocock ( daniel at readytechnology.co.uk ) converted Intel's code into an Asterisk module. |
| 19:28.53 | bougyman | Arkadi Shishlov and others supplied compiler optimizations, and bug fixes. |
| 19:28.58 | bougyman | that's what pocock claims, anyway. |
| 19:29.01 | Qwell | Like I said - Intels code. |
| 19:29.16 | drenda | Hi guys, someone can help me with a problem in setting caller id? |
| 19:30.54 | bougyman | Qwell: it'd be $180 + a per-licence fee for the codec to use his stuff. |
| 19:31.00 | bougyman | taht would be through intel. |
| 19:31.05 | Qwell | for the *code* |
| 19:31.06 | bougyman | he's got the links for it on his site. |
| 19:31.09 | Qwell | not for the patent :) |
| 19:31.16 | bougyman | no, $180 to be able to distribute the code in my own product |
| 19:31.21 | bougyman | the codec is per-channel. |
| 19:31.29 | fullstop | So my 6 year old daughter saw the 80s Incredible Hulk movie on TV the other day. We only have broadcast TV, so it was on the Retro TV channel. Her description was hilarious: "He got bit by a snake and got angry.. and then he got REALLY BIG." |
| 19:32.00 | bmoraca | bougyman: no, the $180 is for the IPP libraries, not the codec code |
| 19:32.13 | Maliuta | fullstop: was that the '80s porno piss-take? |
| 19:32.19 | Katty | wow. |
| 19:32.30 | Katty | i have reached a new height for boredom |
| 19:32.31 | bougyman | right, the ipp libraries (paid to intel), his open source code, and the individual codec licenses. |
| 19:32.37 | bougyman | it can be used legally. |
| 19:32.39 | Maliuta | waves a squirrel at Katty |
| 19:32.44 | Qwell | bougyman: except, it still can't be. |
| 19:32.51 | Qwell | because then you would be violating the GPL |
| 19:33.00 | Katty | i'll violate your gpl in a minute |
| 19:33.02 | bmoraca | bougyman: no, that doesn't cover the patent on the codec...only the libraries required to make that implementation of the codec work. |
| 19:33.06 | Qwell | Katty: hot |
| 19:33.15 | Maliuta | Katty: feel free :P |
| 19:33.16 | [TK]D-Fender | is skeered |
| 19:33.18 | bougyman | bmoraca: correct, you have to pay for the codec itself per channel. |
| 19:34.03 | Qwell | bougyman: anyways, the point is - it isn't to be discussed here. |
| 19:34.04 | bmoraca | so why not let Digium pay the $180 and pay Digium $10 per channel and just use their codec implementation? |
| 19:34.08 | fullstop | Maliuta: It was this one: http://en.wikipedia.org/wiki/The_Incredible_Hulk_Returns |
| 19:34.10 | Katty | Qwell: so i made pizza dough. |
| 19:34.18 | Qwell | Katty: oh? |
| 19:34.23 | fullstop | She needs pepperoni. |
| 19:34.31 | Katty | Qwell: yes'r. would you like to see the picture of the supposed finished recipe? |
| 19:34.32 | Maliuta | fullstop: you are at risk of becoming the butt of the joke :) |
| 19:34.36 | bougyman | bmoraca: because I don't use 729 on asterisk. |
| 19:34.36 | Qwell | Katty: sure |
| 19:34.36 | fullstop | hahaha |
| 19:34.39 | bougyman | well, at all, really. |
| 19:34.43 | fullstop | Maliuta: :-P |
| 19:34.55 | Katty | Qwell: http://www.pizzamaking.com/forum/index.php?topic=2047.0 <- |
| 19:34.57 | Maliuta | fullstop: perky butt though it may be :P |
| 19:35.33 | drenda | Hi guys, someone can help me with a problem in setting caller id? Thanks :) |
| 19:35.43 | fullstop | Anyway, the point of the story was that this is how I will forever remember Qwell. |
| 19:35.51 | Katty | Qwell: starting the pizza around 8ish tonight |
| 19:35.58 | Qwell | fullstop: I'm not even mad. |
| 19:36.02 | Qwell | bougyman has seen me mad. |
| 19:36.05 | Katty | WELL I"M MAD |
| 19:36.12 | Katty | or so i claim. |
| 19:36.21 | fullstop | Qwell: Yeah.. David Banner said the same thing. |
| 19:37.11 | bougyman | i'm gonna keep asking this once a month or so til I find something worthy. |
| 19:37.15 | bougyman | soft phone for windows? |
| 19:37.17 | Maliuta | Eric Banner? the .au comedian? |
| 19:37.22 | bmoraca | LOL @ Max number of channels: 50,000 |
| 19:37.23 | bmoraca | <PROTECTED> |
| 19:37.23 | bmoraca | <PROTECTED> |
| 19:37.24 | Katty | bougyman: zoiper. |
| 19:37.45 | bougyman | we've been through the counterpath stuff, 3cx, sjphone, a couple others. |
| 19:37.47 | fullstop | another vote for zoiper |
| 19:37.52 | fullstop | it even has iax |
| 19:37.53 | bougyman | i couldn't get zoiper to do crap against asterisk. |
| 19:37.56 | Maliuta | and the winner is .... zoiper |
| 19:38.00 | citywok | zoiper works great for me |
| 19:38.00 | bougyman | but we're all freeswitcho n those servers now, i'll give it another shot. |
| 19:38.07 | fullstop | under linux, I was not very successful with zoiper. |
| 19:38.08 | citywok | against * 1.6.2 |
| 19:38.12 | fullstop | win32 was fine. |
| 19:38.14 | Maliuta | bougyman: then you didn't try |
| 19:38.22 | Katty | now that's not fair. |
| 19:38.24 | Katty | maybe he DID try. |
| 19:38.33 | bougyman | Maliuta: we did, but we didn't spend too much time on it. |
| 19:38.35 | Katty | maybe he just needs to try again. |
| 19:38.36 | fullstop | my microphone must not be supported fully in linux. |
| 19:38.37 | bougyman | it failed, sjphone worked. |
| 19:38.38 | bougyman | sjphone won. |
| 19:38.46 | Maliuta | the only zoiper issue I have is that some of their stuff isn't available for Debian unstable |
| 19:38.54 | bmoraca | i've had zoiper working just fine...even across a NAT without STUN... |
| 19:39.13 | Katty | Qwell: make it 5. |
| 19:39.19 | bougyman | sjphone seems the best with dtmf. |
| 19:39.23 | Maliuta | Other than that it registers and receives/sends calls fine, even over SIP+NAT |
| 19:39.23 | citywok | zoiper works for me from the philippines with the server in redmond wa |
| 19:39.26 | Katty | Qwell: use your fancy time bending abilities. |
| 19:39.44 | citywok | we use zoiper web actually, it's pretty slick. our call center agents open a webpage to get to their phone. |
| 19:39.52 | bougyman | i'll give zoiper another shot though. |
| 19:39.53 | Katty | Qwell: or i will use my fancy time bending abilities. |
| 19:39.59 | Katty | Qwell: mine just take about 2 hours to work. |
| 19:40.00 | bougyman | does it do conference and transfer? |
| 19:40.07 | citywok | not the free version afaik |
| 19:40.12 | telnettech | anybody know how to see sip packets on an ACME packet SBC? |
| 19:40.14 | Maliuta | Katty: why not use your Jedi mind tricks? |
| 19:40.16 | citywok | transfer yes, but not conference anyways. |
| 19:40.21 | Maliuta | Katty: or is that just for pool? |
| 19:40.23 | bougyman | citywok: free is not a requirement. |
| 19:40.24 | bmoraca | i'm sure you can do DTMF trasnfers :P |
| 19:40.35 | bougyman | we almost paid $33/copy for bria til we noticed it doesn't work on win2k |
| 19:40.44 | bougyman | we still have probably 200 workstations on win2k |
| 19:40.47 | citywok | win2k? wtf? |
| 19:41.01 | citywok | thats like 11 years old |
| 19:41.13 | bougyman | yes. |
| 19:41.25 | Katty | Maliuta: i have all sorts of jedi mind tricks, what are you talkin about |
| 19:41.59 | Katty | Maliuta: it's called the sly grin! |
| 19:41.59 | bmoraca | i have a collegue installing a VB6 app on Server 2008 R2 as we speak...some customers use old shit |
| 19:42.17 | bougyman | our database is PICK |
| 19:42.18 | Maliuta | "<Katty> Qwell: or i will use my fancy time bending abilities." or are these not the time bending skills I am looking for? |
| 19:42.22 | bougyman | that's circa 1968. |
| 19:42.44 | Katty | Maliuta: no. i think you want to go home and rethink your life. |
| 19:42.59 | carrar | Go for a week long hike |
| 19:43.02 | Katty | Maliuta: i also happen to be wearing a star wars shirt today. |
| 19:43.03 | bougyman | I chuckle at these new NoSQL implementations thinking: why didn't you just use PICK or MUMPS? |
| 19:43.20 | bougyman | those two have the unique classification of being both pre-relational databases and post-relational, go figure. |
| 19:43.31 | Maliuta | Katty: at home already after a night out. Also don't want to re-think my life until I can figure out the romantic side at the moment |
| 19:43.46 | Katty | Maliuta: oh i see :< |
| 19:43.57 | carrar | When stop trying to figure it out then you will be on the right track |
| 19:44.15 | carrar | waste of time otherwise |
| 19:44.16 | Maliuta | Katty: the only "tour" shirt I own is the "Star Trek: the 5 year mission" one |
| 19:45.11 | Maliuta | carrar: I just got burnt by my first real relationship since being diagnosed and treated for cancer almost 7years ago, not quite sure how to react to the new interest |
| 19:45.12 | Katty | Maliuta: post gifs. |
| 19:45.37 | carrar | accept it and life your life as you normally would |
| 19:45.42 | carrar | live |
| 19:45.53 | Katty | Maliuta: how long was your relationship? |
| 19:46.16 | Maliuta | Katty: it's almost 6am and I have to get up in a few hours to sort a barbeque, will take photos of the shirt tomorrow |
| 19:46.28 | Katty | Maliuta: cheers (= |
| 19:46.47 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:46.52 | Maliuta | Katty: with the immediate ex? about a year, and was the first time I had someone move in with me |
| 19:47.05 | Maliuta | and the 7 year old kid didn't help |
| 19:47.06 | carrar | Living in SIN!! |
| 19:47.26 | Maliuta | I live in sin when I'm on my own anyway :P |
| 19:47.36 | Maliuta | _is_ sin |
| 19:47.55 | Katty | hmm. |
| 19:48.07 | Katty | i'd take a couple months off and just not see anyone |
| 19:48.23 | Katty | well you know what i mean |
| 19:48.55 | Katty | Maliuta: i also just recently went through that. |
| 19:49.04 | Katty | Maliuta: but it was 3.5ish years |
| 19:49.17 | Katty | Maliuta: took some adjusting |
| 19:49.36 | Maliuta | Katty: well it's been over a couple of months since it ended and this new girl just-kinda-sorta showed up in my social scene and it's only just started looking relationshippy |
| 19:50.27 | Maliuta | Katty: the major head fuck is that it was my first real relationship in 6 years, and in the end she broke it off with me and was with someone new straight away |
| 19:50.30 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 19:51.19 | fullstop | Maliuta: congrats on being cancer free. |
| 19:51.23 | Katty | Maliuta: i am sorry :< |
| 19:51.27 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 19:51.55 | Maliuta | fullstop: Long term remission, we don't use the four letter C word in the cancer care community :) |
| 19:52.09 | fullstop | My father in law is going under the knife in a week to hopefully finish off pancreatic cancer. |
| 19:52.37 | Maliuta | Katty: it is what it is, I'm just trying to deal with it like I have to deal with the rest of my life |
| 19:53.07 | Katty | Maliuta: only way to do it |
| 19:53.10 | Maliuta | fullstop: ouch, I am glad I had a liquid tumor (leukaemia, blood cell based cancer) |
| 19:53.21 | fullstop | While he was under chemo, he was bit by a mosquito. I watched the mosquito to see if it would just fall from the sky. It did not. |
| 19:53.28 | drenda | Hi guys, someone can help me with a problem in setting caller id? Thanks :) |
| 19:53.36 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:53.51 | fullstop | He's done amazingly well, but radiation was very hard on him. |
| 19:53.55 | fullstop | drenda: just ask |
| 19:54.16 | drenda | tranks fullstop |
| 19:54.17 | Maliuta | fullstop: I puked from a 5th story balcony all the time while having chemo, the plants below still lived |
| 19:54.43 | fullstop | Maliuta: Yeah, but this was blood straight from the source. |
| 19:55.01 | Maliuta | radiation is tough, I only had total body irradiation before my Bone Marrow Transplant |
| 19:55.24 | drenda | I've Asterisk 1.6.2.14-rc1 and I'm making some tests with a dialplan very simple as this: |
| 19:55.26 | fullstop | He had radiation for over a month |
| 19:55.27 | drenda | exten => s,1,Answer() |
| 19:55.27 | drenda | exten => s,n,Set(CALLERID(all)="Tizio Caio"<0426123456>) |
| 19:55.28 | drenda | exten => s,n,Dial(SIP/1) |
| 19:56.12 | drenda | the problem is that on my telephon Linksys SPA942 I can see ONLY the name and as number I see the number of the telephone, in this case 1 |
| 19:56.32 | Maliuta | fullstop: TBI is a higher dose, although a shorted period (if the don't so the BMT after 3 days of TBI you die) |
| 19:56.51 | drenda | an not the number that I set: 0426123456 |
| 19:57.13 | fullstop | Maliuta: You know what's scary about that? That they know the limit is 3 days. Someone found out the hard way. |
| 19:57.47 | roxdragon | hi |
| 19:57.52 | Katty | this is a scary topic :< |
| 19:58.22 | krash812 | hi everybody .. i need somebody just to tell me if i m wrong .. i m doing this . |
| 19:58.31 | krash812 | <PROTECTED> |
| 19:58.31 | krash812 | <krash812> exten => 2,n,Playback(espera) |
| 19:58.41 | Maliuta | fullstop: it was called the Manhattan Project |
| 19:58.49 | Maliuta | Katty: this is my life |
| 19:59.02 | Maliuta | Katty: without the dramas of relationships :) |
| 19:59.25 | krash812 | i something like the queue never stop to ring ... |
| 19:59.37 | krash812 | the original line is this |
| 19:59.42 | krash812 | exten => 2,1,Queue(from-pstn2) |
| 20:00.18 | Maliuta | anyhows, I'm off to bed at the best I get 6 hours sleep before I need to be actively doing stuff |
| 20:00.52 | Katty | Maliuta: you're stronger than i am. |
| 20:01.45 | *** join/#asterisk maximCH (~maximCH@adsl-ecom-4-c15-p038.vtx.ch) |
| 20:02.55 | Maliuta | Katty: no, you just have to deal with some shit |
| 20:10.55 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 20:12.25 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 20:15.40 | krash812 | damn .. i cannot explain in english .. XD |
| 20:15.53 | krash812 | somebody that speak spanish ? |
| 20:16.32 | krash812 | i m tryng to explain the problem in english , but i cannot |
| 20:17.43 | [TK]D-Fender | krash812: "core show application queue" |
| 20:21.18 | krash812 | the queues is working .. i dont have problem , is something about the config .. if i put exten => 1,1,Dial(SIP/1000,20,r) works , but if i put exten => 1,1,Queue(from-pstn3) and i call , even if i hangup , that continue to ring ... i need to study english ... is really hard to me explain somethings about asterisk .. |
| 20:24.14 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 20:24.37 | *** join/#asterisk [cannibalera] (~cannibale@201-35-229-5.fnsce703.dsl.brasiltelecom.net.br) |
| 20:25.05 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:25.46 | [TK]D-Fender | [15:58]<krash812> exten => 2,1,Queue(from-pstn2,20,r) <-- who told you that Queue() takes teh same options as DIAL? Go read teh INSTRUCTIONS for Queue() |
| 20:26.18 | [TK]D-Fender | checkout time, back in a bit. |
| 20:27.33 | n3hxs | [TK]D-Fender if an * box is configured with 2 pri cards, and a call is set to do a follow me back out the second PRI, what kind of audio delay will I see, if any or if noticable? |
| 20:31.05 | malcolmd | not much; your users won't complain about it. |
| 20:31.22 | krash812 | i read .... originaly i got something like this exten => 3,1,Queue(queue-name) |
| 20:31.22 | krash812 | exten => 3,2,Playback(goodbye) |
| 20:31.22 | krash812 | exten => 3,3,Hangup |
| 20:35.56 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:42.36 | *** join/#asterisk fracBlend (~fracBlend@unaffiliated/fracblend) |
| 20:43.03 | fracBlend | hi, Anyone have any experience or reference websites for flashing 7941s to SIP? |
| 20:43.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:49.41 | Katty | stares at the clock |
| 20:51.26 | *** join/#asterisk DoDaT69 (~DoDaT69@173.160.86.155) |
| 20:52.07 | bmoraca | fracBlend: are you getting an error or something? |
| 20:52.28 | fracBlend | Naw, I just haven't done this since 79x1's existed |
| 20:52.46 | fracBlend | since before, rather |
| 20:53.07 | fracBlend | So, I'm looking for a more step by step guide with the newer firmware bundles |
| 20:53.20 | fracBlend | Since there's no POS files for these |
| 20:55.08 | bmoraca | the process is pretty much the same as with 7940s, except that they look at XMLDefault.cnf.xml to find what firmware version to load instead of OS79XX.TXT or SIPDefault.cnf |
| 20:55.21 | fracBlend | Ah |
| 20:55.47 | fracBlend | So I just need to find a template for that new xml? |
| 20:55.56 | bmoraca | their firmware is also, of course, significantly different |
| 20:56.20 | *** join/#asterisk csnook (~chris@pool-74-104-211-128.bstnma.east.verizon.net) |
| 20:56.25 | bmoraca | that file is relatively simple...and it's the same whether you're using SIP (with 3rd party) or SCCP (with CME) |
| 20:57.02 | fracBlend | Great, know where i can get an example? |
| 20:57.08 | fracBlend | or just use the googles? |
| 20:57.12 | bmoraca | cisco.com or on the wiki |
| 20:57.14 | bmoraca | ~wiki |
| 20:57.34 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 20:57.38 | bmoraca | voip-info.org, that is |
| 20:58.03 | fracBlend | super |
| 20:58.27 | fracBlend | anyone know if the latest and greatest sip software still works with 3rd party systems? |
| 20:59.00 | bmoraca | i'm not a huge fan of 7941s (too many quirks), but the last version I used was 8-0-2SR1S |
| 20:59.19 | bmoraca | i tried several later than that, but they didn't work very well at all with Asterisk |
| 20:59.20 | fracBlend | I grabbed 9-0-3 |
| 20:59.32 | fracBlend | bogus |
| 20:59.35 | bmoraca | 8.0.2 SR1s, from what I recall, was relatively stable |
| 21:01.25 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 21:04.18 | fracBlend | thx |
| 21:14.27 | *** join/#asterisk TimeRider (~steve@5ac7b3bb.bb.sky.com) |
| 21:37.30 | Katty | stares at clock |
| 21:37.31 | Katty | sighs. |
| 21:37.38 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 21:37.39 | Katty | today is taking /forevers/ |
| 21:38.09 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 21:42.28 | carrar | Hit FF |
| 21:44.15 | Katty | ff? |
| 21:44.18 | Katty | what is this |
| 21:44.27 | mmattice | fast forward? |
| 21:44.34 | Katty | oh :< |
| 21:44.39 | mmattice | yeah |
| 21:44.40 | Katty | i seem to have missplaced my remote |
| 21:44.50 | Katty | but i did repaint my toenails. |
| 21:44.53 | Katty | that killed about 5 minutes. |
| 21:56.07 | carrar | PICS!! |
| 21:57.15 | bmoraca | god damn idiot "techs" enabling write caching on a server that doesn't have a battery backup |
| 22:01.07 | ectospasm | heh, I had one job where the UPS just... didn't... work.... |
| 22:01.16 | ectospasm | ...and the documentation was lousy |
| 22:01.27 | ectospasm | I won't buy tripp-lite because of it. |
| 22:01.39 | Katty | carrar: http://farm5.static.flickr.com/4107/5021597200_7cf6e8ecbf_b.jpg <- |
| 22:02.03 | Katty | and now i am FREE :> |
| 22:02.04 | Katty | poofs |
| 22:05.44 | *** join/#asterisk diegomad (~mad@190.147.221.78) |
| 22:07.19 | bmoraca | ectospasm: well, it's worse because it's an exchange box...it's ALWAYS writing to disk...when it goes down hard (we have shitty power in this building), it has problems. and now it won't come back up. and of course i get to try and fix it! woo! |
| 22:20.20 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 22:24.50 | carrar | Katty, yeah thats HOT++ :) |
| 22:25.39 | carrar | The half screen of icons and that PC, not so hot |
| 22:31.17 | roxdragon | exten => s,1,Dial(SIP/401&SIP/402&SIP/403) |
| 22:31.32 | roxdragon | What's parametr "s"? |
| 22:33.21 | *** join/#asterisk fprior (~chatzilla@200.43.69.102) |
| 22:33.33 | fprior | hi, is this a place for make a question regard "how apply a patch from reviewboard.asterisk.org" ? or better #asterisk-bug ? |
| 22:36.51 | [TK]D-Fender | roxdragon: that is the EXTENSION. |
| 22:37.04 | [TK]D-Fender | ~book |
| 22:37.04 | 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 |
| 22:37.07 | [TK]D-Fender | ^^^^^^^ |
| 22:37.30 | roxdragon | i don't remember the significat "s" |
| 22:38.47 | [TK]D-Fender | ~stdextens |
| 22:38.47 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 22:39.10 | [TK]D-Fender | roxdragon: Asterisk Standard Extensions. GO READ |
| 22:40.22 | *** join/#asterisk Bloudermilk (~Bloudermi@dsl081-234-075.lax1.dsl.speakeasy.net) |
| 22:40.31 | Bloudermilk | Afternoon everyone |
| 22:40.37 | Besticles | I'm setting up festival. It says to restart festival, how do I do that? |
| 22:40.43 | Besticles | AFternoon Blouder. |
| 22:41.05 | Bloudermilk | I'm hoping someone can provide me some assistance on an issue I'm having with meet_me |
| 22:41.19 | roxdragon | [TK]D-Fender, |
| 22:41.23 | roxdragon | http://paste.ubuntu.com/500009/ |
| 22:41.54 | Bloudermilk | I've just started messing around with the Ices (http://bit.ly/bcCu98) application |
| 22:41.54 | roxdragon | What exactly does this add? |
| 22:42.10 | Bloudermilk | My goal is to have a conference of 5-6 people talking |
| 22:42.11 | carrar | scrolls back |
| 22:42.14 | carrar | and laughs |
| 22:42.32 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup2) |
| 22:42.38 | Bloudermilk | And one channel connected which is Ices, and streaming |
| 22:42.39 | carrar | Whats with all these "extensions" anyways |
| 22:42.44 | carrar | I hear so much talk of it |
| 22:42.59 | Bloudermilk | The problem is that both meet_me and ices are blocking |
| 22:43.17 | carrar | I'm here because there is a "asterisk" on my keyboard |
| 22:44.06 | Bloudermilk | Does anyone have experience with meet_me and executing a dialplan (or even better, AGI) once connected to a conference? |
| 22:45.43 | [TK]D-Fender | roxdragon: "Add"? Pardon? |
| 22:45.48 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 22:46.04 | [TK]D-Fender | roxdragon: That exten dials a bunch of devices indefinitely |
| 22:49.42 | fprior | Hi, I'm spent my afternoon on reviewboard.asterisk.org, can anyone help me ? |
| 22:51.14 | carrar | I'm spent too |
| 22:51.37 | [TK]D-Fender | bl"blocking"? In what way? |
| 22:51.46 | [TK]D-Fender | cashes carrar in. |
| 22:51.53 | carrar | heh |
| 22:52.31 | Bloudermilk | D-Fender, in that I can no longer execute anything in the application until the channel as exited the conference |
| 22:52.40 | Bloudermilk | And therefore cannot call Ices() |
| 22:53.13 | carrar | Such a plethora of implications in this channel |
| 22:53.43 | fprior | carrar: are you a volunteer for my problem ?.... |
| 22:54.00 | [TK]D-Fender | Bloudermilk: You would Originate a separate call to a LOCAL CHANNEL. One end would conect to Ices, the other would dump into the MeetMe. |
| 22:54.10 | carrar | who knows, you haven't said what your problem is |
| 22:54.26 | carrar | for I know your dog ran away |
| 22:54.27 | [TK]D-Fender | blThis has to be separate from a call from a "phone like device" |
| 22:54.41 | carrar | and you are looking on online forums o find it |
| 22:54.48 | Bloudermilk | Fender, I see |
| 22:55.09 | carrar | I CAN'T KNOW EVERYTHING |
| 22:55.09 | Bloudermilk | Fender: I'm not entirely sure how to implement what you're suggesting, but I suppose I'll give it a shot |
| 22:55.26 | [TK]D-Fender | hordes his pre-release copy of res_omniscience.so |
| 22:55.31 | carrar | However TK can |
| 22:55.39 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:55.45 | [TK]D-Fender | Bloudermilk: "core show application Originate" |
| 22:56.37 | Bloudermilk | Fender: I'm familiar with Originate, I'm just not sure what the dialplan would look like |
| 22:58.08 | Besticles | Question about Festival. Do I have to leave a terminal window open executing festival --server to run in server mode, or is it always running? |
| 23:01.52 | [TK]D-Fender | Bloudermilk: One end would call Ices, the other would call meetme. |
| 23:02.10 | Bloudermilk | D-Fender, Gonna give it a go |
| 23:02.51 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 23:04.06 | fprior | for example, I'm trying to patch from reviewboard.asterisk.org/r/301/, but I dont't know (for newbie on patch) which source version I must download |
| 23:05.21 | *** join/#asterisk mattvv (~matt@dsl081-234-075.lax1.dsl.speakeasy.net) |
| 23:07.57 | [TK]D-Fender | fprior: What are you trying to do it on right now? |
| 23:10.41 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 23:12.29 | [TK]D-Fender | fprior: First that patch is over a YEAR old. |
| 23:12.41 | fprior | [TK]D-Fender: my problem is on how working a revision system. for ex: a diff file from reviewboard.asterisk.org is on Revision 204194. What can I do if my source version is older os newer ? |
| 23:13.04 | [TK]D-Fender | fprior: Next"/branches/1.4/main/pbx.c (Diff revision 1)" should ahve beena clue |
| 23:13.18 | [TK]D-Fender | fprior: And given its age it is already probably merged in |
| 23:14.53 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 23:15.42 | *** join/#asterisk jetlag (~jetlag@pool-173-61-240-74.cmdnnj.east.verizon.net) |
| 23:19.10 | fprior | [TK]D-Fender: my server has Asterisk 1.4.22. you are say me to download from actual branch. and now ? What recompile ? All modules or only one ? |
| 23:19.52 | [TK]D-Fender | fprior: You are FOURTEEN releases behind. For security reasons alone you should be doing a n upgrade ASAP |
| 23:20.02 | carrar | HURRY |
| 23:21.43 | carrar | Jianhuaq Qian from China is about to haX0r your box |
| 23:21.54 | carrar | from the Zhejiang Province Network |
| 23:23.04 | carrar | denied udp 124.160.91.69(5060) -> SE.NS.OR.ED(5060), 1 packet |
| 23:24.30 | fprior | [TK]D-Fender: my predecesor install trixbox. If I update his asterisk I will have problem with that ? |
| 23:25.51 | *** join/#asterisk Ta^3 (~tacvbo@gponr9-203-11-242.iusacell.net) |
| 23:26.39 | [TK]D-Fender | fprior: go ask in THEIR forums what the proper upgrade path is. They are running forked versions of everything. |
| 23:26.48 | [TK]D-Fender | fprior: And are not supported here |
| 23:28.13 | roxdragon | [TK]D-Fender, this extensions is a incoming call?? exten => s,1,Dial(SIP/401&SIP/402&SIP/403) |
| 23:28.37 | carrar | ~book |
| 23:28.37 | 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 |
| 23:29.03 | roxdragon | carrar, i am italian don't speak english eheh |
| 23:29.12 | carrar | thats funny |
| 23:29.14 | carrar | read the book |
| 23:29.22 | fprior | [TK]D-Fender: my predecesor install trixbox. If I update his asterisk I will have problem with that ? |
| 23:29.56 | carrar | fprior, not it you delete it first |
| 23:30.46 | carrar | Now is the time to start A NEW |
| 23:30.50 | fprior | [TK]D-Fender: I know. Last question: in general, which is the better maintaining program on a asterisk box ? For example: every month check last release, download source and compile ? |
| 23:31.15 | fprior | roxdragon: anch'io |
| 23:31.45 | carrar | http://lists.digium.com/mailman/listinfo/ |
| 23:31.57 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 23:31.59 | roxdragon | fprior, ecco |
| 23:32.16 | roxdragon | conosci quell'estensione? |
| 23:32.34 | carrar | fprior: Asterisk.org community currently includes over 60,000 members. Join today |
| 23:36.13 | fprior | carrar: thanks. but how manage installation update ? |
| 23:36.47 | carrar | Did you not read what [TK]D-Fender said in response to your question? |
| 23:37.25 | carrar | It also applies to your last question FYI |
| 23:38.15 | carrar | I would suggest you migrate off what you are running |
| 23:38.22 | carrar | and install Asterisk from SOURCE |
| 23:39.04 | carrar | Als if you have not already read "The book", please do so at this time. |
| 23:39.11 | carrar | ~book |
| 23:39.12 | 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 |
| 23:39.56 | carrar | Then, once you have read the book, come on back and ask away!! |
| 23:40.31 | fprior | carrar: I understand that, and I know here is not a place for trixbox questions. |
| 23:40.39 | carrar | excellent |
| 23:40.43 | carrar | Have a GREAT DAY! |
| 23:40.54 | carrar | and be sure to take along a towel |
| 23:41.27 | fprior | carrar: I collect your information, thanks, really |
| 23:41.38 | carrar | so does google |
| 23:43.24 | carrar | http://ibot.rikers.org/%23asterisk/20100924.html.gz (check tomorrow) |
| 23:58.16 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 23:59.43 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup2) |
