| 00:08.41 | *** join/#asterisk bazman (~army@203-217-29-141.perm.iinet.net.au) |
| 00:14.48 | *** join/#asterisk FonJockey (~FonJockey@ma12c36d0.tmodns.net) |
| 00:16.46 | *** join/#asterisk dot123 (~albert@internal.bigair.net.au) |
| 00:20.27 | *** join/#asterisk Maliuta (~nobusines@poltava.lusan.id.au) |
| 00:21.27 | *** join/#asterisk Khronos (~Khronos@108-224-98-39.lightspeed.wepbfl.sbcglobal.net) |
| 00:23.05 | *** join/#asterisk Nova-Man (~Nova-Man@203-217-29-141.perm.iinet.net.au) |
| 00:24.56 | bazman | I am trying to get asterisk 11.2.2 on Fc18 to run out of /etc/astersik-vm. I have updated astectdir=/etc/asterisk-vm however when I run asterisk from the command line "/usr/sbin/asterisk -f -C /etc/asterisk-vm/asterisk.conf" it still wants files in /etc/asterisk. Does anyone know how to move its configuration? |
| 00:25.52 | [TK]D-Fender | pastebin your entire asterisk.conf |
| 00:25.54 | [TK]D-Fender | ~pb |
| 00:25.54 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:25.55 | [TK]D-Fender | ^ |
| 00:26.23 | bazman | Yep have used it before. Sorry |
| 00:29.08 | *** join/#asterisk suneye (~atcmmi@119.139.132.143) |
| 00:32.50 | *** join/#asterisk Maliuta (~nobusines@poltava.lusan.id.au) |
| 00:44.24 | bazman | Found a (!) at the end of [directories]. When deleted all ok. |
| 00:48.31 | [TK]D-Fender | yup.... |
| 00:48.42 | *** join/#asterisk atcmmi (~atcmmi@119.122.152.56) |
| 00:50.58 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 01:04.55 | WIMPy | How do I get dahdi to send early media? I can't find anything obvious in the current version. |
| 01:12.23 | pabelanger | Progress() ? |
| 01:12.50 | WIMPy | Ah, dialplan. Good idea. |
| 01:14.25 | WIMPy | That's it. Thanks. |
| 01:14.37 | WIMPy | Was just sweeping up and down chan_dahdi.conf... |
| 01:16.43 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:20.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.117) |
| 01:22.37 | *** join/#asterisk Khronos (~Khronos@108-224-98-39.lightspeed.wepbfl.sbcglobal.net) |
| 01:28.19 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 01:30.45 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 01:33.23 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 01:36.29 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 02:10.32 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:18.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.121) |
| 02:24.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 02:27.07 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 02:30.59 | *** join/#asterisk Rahail (~Rahail@67.214.121.163) |
| 02:31.09 | Rahail | how ther ehow can I disable the fake ring tone back |
| 02:31.33 | Rahail | ~pastebin |
| 02:31.33 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:31.47 | WIMPy | Don't use option r to Dial. |
| 02:31.56 | Rahail | http://bin.cakephp.org/view/628921936 |
| 02:31.59 | Rahail | this my dialplan |
| 02:32.13 | Rahail | i do not have r in it |
| 02:32.38 | WIMPy | Then it's not coming from Asterisk. |
| 02:32.48 | WIMPy | It's either your phone or your provider. |
| 02:33.05 | Rahail | even i put fake provider ip |
| 02:33.12 | Rahail | i get rink back |
| 02:34.59 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 02:37.17 | *** join/#asterisk FonJocke_ (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 02:47.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:47.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:48.19 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:48.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 02:48.19 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 02:48.19 | *** join/#asterisk asteriskandy (asteriskan@v6.pandora.panicbnc.us) |
| 02:48.19 | *** join/#asterisk 44UAABBGB (~mark@fw.seamans.cc) |
| 02:48.19 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 02:48.19 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 02:49.25 | Rahail | WIMPy every call hit on my server |
| 02:49.26 | Rahail | i get this |
| 02:49.27 | Rahail | res_rtp_asterisk.c:2157 ast_rtp_read: RTP Read too short |
| 02:49.53 | *** join/#asterisk XuW (~xuw@110.77.229.148) |
| 02:52.45 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:52.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 02:52.45 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 02:52.45 | *** join/#asterisk asteriskandy (asteriskan@v6.pandora.panicbnc.us) |
| 02:52.45 | *** join/#asterisk 44UAABBGB (~mark@fw.seamans.cc) |
| 02:52.45 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 02:52.45 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 03:06.24 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 03:09.32 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 03:25.44 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 03:34.32 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 03:38.30 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 03:38.59 | *** join/#asterisk MushroomNZ (~mushroomn@pork.whatbox.ca) |
| 03:58.33 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 04:37.10 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 04:54.08 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 04:55.58 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 05:20.53 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 05:44.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:45.50 | *** join/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 05:57.46 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 05:59.09 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 06:05.32 | *** join/#asterisk zpotoloom (~toomas@tom.data.ee) |
| 06:23.03 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 06:34.50 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:42.09 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 06:55.44 | *** join/#asterisk AllMyFriends (~weowit@c-174-53-130-134.hsd1.mn.comcast.net) |
| 06:55.52 | AllMyFriends | 11-part video tutorial is up. http://www.youtube.com/watch?v=u9DzN1Pu6-Q&list=PLE_de-PBwrTSUMm-Y48aiOOHt_YyT69t0 |
| 06:55.57 | *** part/#asterisk AllMyFriends (~weowit@c-174-53-130-134.hsd1.mn.comcast.net) |
| 06:57.31 | *** join/#asterisk edwin_quijada (macaruchi@190.166.36.156) |
| 06:58.30 | edwin_quijada | Hi! I have a weird problem. I change my server for another and copied all my configurations but now any extension doesnt register all tell me that Registration failed - Wrong Password |
| 06:59.16 | edwin_quijada | I check the same password from the old server and are the same but in the new one doesnt register I check everything and nothing |
| 06:59.24 | edwin_quijada | Anybody has a cluee? |
| 06:59.38 | eirirs | edwin_quijada: you just copied the hashed passwords? |
| 06:59.39 | eirirs | :p |
| 07:00.05 | *** join/#asterisk zamba (marius@flage.org) |
| 07:00.20 | edwin_quijada | hashed password? |
| 07:00.51 | XuW | you have realtime enabled? |
| 07:01.03 | edwin_quijada | My extensions are in a database and I create the sip.conf on fly using #exec |
| 07:01.10 | edwin_quijada | No, no realtime |
| 07:01.34 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 07:02.09 | edwin_quijada | I just copied the same configuration from the old server |
| 07:03.02 | edwin_quijada | I get this |
| 07:03.05 | edwin_quijada | [May 13 02:53:44] NOTICE[2513] chan_sip.c: Registration from '"Ariani Gil" <sip:104@192.168.1.235>' failed for '192.168.1.44:5062' - Wrong password |
| 07:06.13 | *** join/#asterisk Maliuta (~nobusines@poltava.lusan.id.au) |
| 07:06.59 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 07:07.25 | *** join/#asterisk janelleb (~jamii@unaffiliated/janelleb) |
| 07:08.36 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 07:11.12 | janelleb | Does anyone have a link to a simple tutorial on voice menus: i.e. A user phones in and hears a Playback(), then they are prompted to "press 1 for another playback" or to "press 2 for yet another playback" after which * will hangup. |
| 07:12.34 | jacekowski | it's called IVR |
| 07:13.46 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 07:16.58 | v0lZy | janelleb: its straight forward |
| 07:17.42 | v0lZy | janelleb: hold on a min, ill dig it up |
| 07:17.54 | *** join/#asterisk AllMyFriends (~noope@c-174-53-130-134.hsd1.mn.comcast.net) |
| 07:17.55 | AllMyFriends | 11-part video tutorial is up. http://www.youtube.com/watch?v=u9DzN1Pu6-Q&list=PLE_de-PBwrTSUMm-Y48aiOOHt_YyT69t0 |
| 07:17.57 | *** part/#asterisk AllMyFriends (~noope@c-174-53-130-134.hsd1.mn.comcast.net) |
| 07:20.41 | janelleb | jacekowski: yes I know, infact I've done this [0] in my extensions.conf and I've been reading articles/tutorials but I can't get it right. [0] http://pastebin.com/Aja21v9U |
| 07:21.27 | janelleb | v0lZy: thanks, in advance. |
| 07:22.01 | v0lZy | ill paste what i did. |
| 07:23.39 | janelleb | v0lZy: ok |
| 07:25.11 | v0lZy | http://bpaste.net/show/N2QQ2YjQC1UF63SWjVI6/ |
| 07:25.12 | v0lZy | here u go |
| 07:28.03 | v0lZy | janelleb: as you see, the plan is structured through extension mappings |
| 07:28.34 | v0lZy | basically you start the context the way you want, then at a certain point, you invoke WaitExten(seconds) |
| 07:28.36 | janelleb | <PROTECTED> |
| 07:28.42 | v0lZy | after that, you just list extensions and what they wanna do. |
| 07:28.56 | v0lZy | janelleb: yes, of course. |
| 07:29.29 | janelleb | v0lZy: Thans a lot, I'll try this out immediately, then get back to the channel. |
| 07:29.58 | v0lZy | Let me know if you need any help with it. |
| 07:30.22 | janelleb | v0lZy: will do, back in a few. |
| 07:30.55 | v0lZy | janelleb: I should also point out the loop reason. |
| 07:31.21 | v0lZy | janelleb: I did it the way I did it because I was doing this for someone who was using the IVR as a front end for his service |
| 07:32.01 | v0lZy | janelleb: He would get calls always on the same number, but then the person calling could enter their 'PIN' and the system would then interpret that PIN for a certain customer's number and call that number |
| 07:32.26 | v0lZy | janelleb: But he was charged per duration of calls on that number the customer originally reached IVR on. |
| 07:32.45 | v0lZy | janelleb: So I put a loop in their that hangsup after the whole thing cycles 3 times. |
| 07:33.00 | v0lZy | janelleb: so someone wouldnt call him and drain his minutes. |
| 07:34.28 | v0lZy | janelleb: uh, oh, I forgot to paste part of the code |
| 07:34.28 | v0lZy | hold on |
| 07:35.18 | edwin_quijada | I am getting Wrong Password from my extension to register with asterisk |
| 07:35.31 | edwin_quijada | I moved all conf to one server to another |
| 07:35.53 | v0lZy | janelleb: use this: http://bpaste.net/show/LVXTdzUacjCTbqDqoI33/ |
| 07:35.55 | edwin_quijada | and now everything is wrong password any cluees? |
| 07:37.17 | v0lZy | edwin_quijada: sip.conf or whatever protocol u are using holds the passwords |
| 07:37.48 | v0lZy | edwin_quijada: have to reload it |
| 07:38.02 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 07:38.48 | *** join/#asterisk din3sh (2988ff33@gateway/web/freenode/ip.41.136.255.51) |
| 07:39.01 | din3sh | Good Mrning all |
| 07:39.02 | edwin_quijada | I reload it |
| 07:39.09 | edwin_quijada | and nothing |
| 07:39.38 | kaldemar | edwin_quijada: "sip set debug on" |
| 07:40.17 | v0lZy | hi kaldemar |
| 07:40.56 | edwin_quijada | kaldemar: I am seeing all info but... |
| 07:41.36 | kaldemar | edwin_quijada: pastebin it for other to take a look if you can't figure it out. |
| 07:41.40 | kaldemar | v0lZy: hello. |
| 07:41.46 | edwin_quijada | kaldemar:ok |
| 07:42.36 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:42.39 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:44.55 | edwin_quijada | kaldemar: this is something about...http://pastebin.com/MkUZm0kf |
| 07:44.58 | v0lZy | kaldemar: Quick question if I may intrude upon you: For external calls, what route would one take to start 'hunting' around a group of phones when a certain number is called and there is 1 active conversation already going on on that number. Thing is, I have a phone that can do 2 lines, but I want to set it so that when a call is in progress, if the second incoming call is not answered within 3 rings, it starts hunting a group |
| 07:46.49 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:47.30 | kaldemar | v0lZy: timeout in Dial, GROUP functions and device state for starters. |
| 07:47.38 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 07:47.46 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:48.44 | kaldemar | edwin_quijada: check and double check that you have a mathing secret for 127 in asterisk and in the phone. not much more to say. |
| 07:48.45 | *** join/#asterisk Maliuta (~nobusines@poltava.lusan.id.au) |
| 07:48.46 | edwin_quijada | the weird is that in the old server works so fine no problem |
| 07:49.08 | edwin_quijada | in the new 'Wrong passwrod' |
| 07:51.02 | v0lZy | kaldemar: the timeout in dial part and group functions i understand. I'm a bit unclear on the 'if an active call is in progress'... i guess i could grep open channels? |
| 07:51.06 | edwin_quijada | I check one by one any extension |
| 07:51.58 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 07:52.28 | *** join/#asterisk Boardy (~chatzilla@kirakira.xs4all.nl) |
| 07:53.40 | edwin_quijada | kaldemar: what is the secret for 127? |
| 07:54.46 | kaldemar | edwin_quijada: what you configure it to be. |
| 07:55.23 | edwin_quijada | I mean in the pastebin beause I dont see the seccret |
| 07:55.59 | kaldemar | you're not supposed to. |
| 07:56.32 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 07:57.45 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:58.03 | Boardy | I used to send SMSs with smsq, but command to queue messages has been removed (after upgrade Debian to Wheezy). How can I send my SMSs now? |
| 08:04.52 | apb1963_ | watches as asterisk comes to a graceful stop... watching.... watching.... yawning.... watching.... waiting....pulling out hammer.... watching.... watching... |
| 08:09.00 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 08:12.16 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 08:12.42 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 08:14.12 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 08:17.05 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 08:20.05 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 08:21.19 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 08:22.36 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 08:23.42 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 08:24.04 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 08:25.07 | *** join/#asterisk drmessano^ (~nonya@pdpc/supporter/active/drmessano) |
| 08:26.13 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 08:28.40 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 08:31.46 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.smb.curriegrad2004.ca) |
| 08:32.42 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 08:32.44 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 08:33.12 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 08:36.52 | *** join/#asterisk Maliuta (~nobusines@poltava.lusan.id.au) |
| 08:37.33 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 08:43.16 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 08:47.21 | bulkorok | hi |
| 08:47.27 | *** part/#asterisk Boardy (~chatzilla@kirakira.xs4all.nl) |
| 08:50.12 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:52.08 | *** join/#asterisk Maliuta (~nobusines@poltava.lusan.id.au) |
| 08:54.19 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 08:54.21 | *** join/#asterisk admin0 (~admin0@2001:1af8:4100:1128:2173:f039:3361:ad00) |
| 08:54.23 | hrolf | Hi asterisk |
| 08:54.35 | hrolf | How do I check which codec is used by my SIP peer? |
| 08:56.13 | apb1963 | core dumps... get your core dumps here... who wants core dumps?? |
| 08:58.47 | bulkorok | hrolf: if you want "real" data, you should use tcpdump and check your call with wireshark... |
| 08:59.12 | bulkorok | hrolf: if you want to know the possible codecs check "sip show peer YOURPEER" in asterisk-cli |
| 09:12.09 | kaldemar | hrolf: during a call, you'll see the used codec with "core show channel <channel>" |
| 09:13.08 | jacekowski | i've got a problem with DPMA - basically phones are not provisioning in some case correctly - and it seems to be related to asterisk sending packets that are over 1800bytes long |
| 09:13.15 | jacekowski | with MTU set to 1500 |
| 09:18.34 | apb1963 | Here's a funny story... I was driving down ROUTE 1500, and I saw this sign that said <DO NOT FRAGMENT> |
| 09:19.14 | apb1963 | I was in a BIT of a hurry so I ignored it. |
| 09:19.46 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 09:21.20 | apb1963 | that reminds me.... |
| 09:21.46 | apb1963 | jacekowski: Have you checked to see what your router does with large packets? |
| 09:22.08 | *** join/#asterisk FonJockey (~FonJockey@nc-76-5-191-136.dhcp.embarqhsd.net) |
| 09:23.26 | apb1963 | just drank his last cup of coffee |
| 09:23.28 | apb1963 | bedtime! |
| 09:24.21 | *** join/#asterisk Maliuta_ (~nobusines@kiev.lusan.id.au) |
| 09:25.05 | *** join/#asterisk _omer (~omer@116.71.179.37) |
| 09:26.41 | _omer | I want a2billing agi script to spit some logs/debugs on Asterisk CLI when it is executed .... anyhelp please? (I know this room is not for a2billingbut I could not find any place for a2billing help) ... |
| 09:38.35 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 09:39.22 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 09:39.25 | Rico29 | hi all |
| 09:39.54 | Rico29 | il there a way to get per agent queue statistics ? like average hold timer per agent, average talk time per agent, ... |
| 09:42.34 | _omer | I want a2billing agi script to spit some logs/debugs on Asterisk CLI when it is executed .... anyhelp please? (I know this room is not for a2billingbut I could not find any place for a2billing help) ... |
| 09:43.57 | jacekowski | apb1963: drops them |
| 09:44.24 | jacekowski | but why asterisk is sending packets larger than MTU |
| 09:44.28 | jacekowski | and why those are not fragmented |
| 09:49.04 | *** join/#asterisk msaraiva (~msaraiva@host81-148-123-180.in-addr.btopenworld.com) |
| 09:51.37 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 09:57.05 | *** join/#asterisk andrewMhiggs (~andy@105-236-70-239.access.mtnbusiness.co.za) |
| 09:57.51 | andrewMhiggs | Hello everyone. |
| 09:59.14 | msaraiva | Is there any way to do a time based conference with ConfBridge without using AMI? There's a note on the Wiki for Asterisk 10, but it's from 2011... |
| 10:02.17 | andrewMhiggs | I need to upgrade a very old machine. It is running Trixbox 2.6 with Asterisk 1.4.22 (yes, I know this is very very old). I am going to switch over to AsteriskNow 3. Is it possible for me to move asteriskcdrdb detail across? I have tried backing the db up and restoring but I assume there must be some db changes I will need to apply? Is there a tutorial (or two as this is a very big jump) I can follow in order to do this? |
| 10:03.44 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 10:05.52 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:07.23 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 10:10.03 | wdoekes | andrewMhiggs: the cdrs rarely change (or rather "shouldn't change") |
| 10:11.07 | wdoekes | I'm pretty sure there are no mandatory db schema changes between 1.4 and recent for the basic cdrs |
| 10:14.00 | andrewMhiggs | Thanks. Perhaps I should clarify as this may make a difference.The CDR's are both on FreePBX machines. Does that make a difference? If I do the straight backup and restore it tells me there are no calls in the db. |
| 10:24.08 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 10:28.02 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 10:28.59 | gavimobile | is it possible for a pots call to be choppy? it sounds like what a voip call would be when there's low bandwidth |
| 10:29.08 | gavimobile | but im using pots |
| 10:29.32 | gavimobile | I have plenty of free memory |
| 10:33.56 | v0lZy | hey guys, can anyone hint as to how to check the number of lines on a phone... I need to do some dialplan logic that triggers when phone has 1 active conversation going, but can still accept an additional call. |
| 10:34.48 | v0lZy | (as in, it can actually have 2 conversations, each on its own line, but i need to redirect incoming calls when a conversation is already in progress... in effect, making it as if it has only a single line... |
| 10:35.45 | v0lZy | i know that in sip.conf there is a busylevel thing |
| 10:36.13 | v0lZy | but i dont think thats what im going for since i do actually want to have the ability to handle 2 calls, im just concerned about incoming calls when 1 call is in progress already. |
| 10:37.27 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 10:38.14 | *** join/#asterisk racho (~racho@46.40.123.204) |
| 10:40.24 | racho | what is the best way to limit the call duration for the whole dialplan? can i Set(TIMEOUT(absolute)=xx) in the [globals] of exten.conf |
| 10:40.25 | andrewMhiggs | v0lZy: I am no expert, but could you not do something like a queue here. This obviously would depend on how many phones you need to set like this. A few might be okay but many more you might prefer a more automatic method. |
| 10:40.29 | kaldemar | v0lZy: there is no way to check the abilities of the phone. use GROUP functins to keep count on how many calls the devices have. |
| 10:43.16 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 10:44.48 | *** join/#asterisk hwt (hwt@curb.thorkildssen.com) |
| 10:45.08 | hwt | hi, is there a way to force asterisk to pass through SIP response codes? |
| 10:45.24 | hwt | on SIP to SIP channels |
| 10:46.16 | v0lZy | andrewMhiggs, kaldemar: I have this specific request .... the user's have an external phone line which takes 2 simultanious incoming calls. However, when the call is already answered on the designated phone, my users would like for the ringing to switch to a different phone so that they dont need to hand signal eachother. If i understand what you are saying, you are suggesting that i bridge the external number to a group, and then use group function |
| 10:47.04 | v0lZy | kaldemar, andrewMhiggs: im confused how exactly asterisk goes about this if it cant figure out that a phone is having a conversation already. |
| 10:47.33 | v0lZy | since the phone can accept 2 lines... though my objective is to have 1 line reserved so that the called person can put calls on hold etc... |
| 10:47.47 | v0lZy | Is this a situation i got myself into because i set busy-level=2 in sip.conf's ? |
| 10:49.34 | kaldemar | v0lZy: a group is not something that can be bridged to anything in asterisk. |
| 10:50.03 | kaldemar | v0lZy: start by reading documentation for the group functions so you understand what they are. |
| 10:50.42 | v0lZy | kaldemar: I apologize on my improper expresssion. by bridged i ment 'point to' |
| 10:52.02 | andrewMhiggs | v0lZy: I agree. I think a ring group is what you are looking for. |
| 10:56.51 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 10:59.02 | v0lZy | andrewMhiggs: I came to that conclusion too. I dont want to ring all the phones at the same time though, but i want to ring them in sequence... but I want this sequence thing starting ONLY when the first peer in the ring group is already having a conversation. |
| 11:02.14 | *** part/#asterisk admin0 (~admin0@2001:1af8:4100:1128:2173:f039:3361:ad00) |
| 11:04.09 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 11:04.36 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 11:05.50 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 11:07.22 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.142) |
| 11:09.39 | racho | can i Set(TIMEOUT(absolute)=xx) in the [globals] of exten.conf to limit call duration of all calls in the dialplan? |
| 11:12.00 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 11:13.22 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 11:14.05 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 11:15.04 | *** join/#asterisk vlad_sta_ (~vlad_star@195.68.167.197) |
| 11:25.01 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 11:25.59 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 11:25.59 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 11:25.59 | *** join/#asterisk zamba (marius@flage.org) |
| 11:25.59 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 11:25.59 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 11:25.59 | *** join/#asterisk asteriskandy (asteriskan@v6.pandora.panicbnc.us) |
| 11:25.59 | *** join/#asterisk 44UAABBGB (~mark@fw.seamans.cc) |
| 11:25.59 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 11:25.59 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 11:28.02 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:30.13 | Greenlight | What's the easiest way to track down which peer the "Retransmission timeout" CLI Warnings are generated from. Annoyingly the message only shows the local IP and the CallID is already gone when I search? |
| 11:31.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:33.03 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 11:36.16 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 11:39.59 | *** join/#asterisk joshuahh (josh@27.32.48.149) |
| 11:41.30 | andrewMhiggs | v0lZy: Set the ring strategy to firstnotonphone. |
| 11:44.06 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 11:44.58 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 11:47.58 | *** join/#asterisk joshuahh (~josh@27.32.48.149) |
| 11:48.05 | joshuahh | hi just confirming, for my setup, my freepbx server is in a remote data centre |
| 11:48.10 | joshuahh | will the extension 4000 (remote) be NAT on/off/never ? |
| 11:49.53 | msaraiva | Is the client behind nat? Is your server behind nat? |
| 11:50.19 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:52.00 | joshuahh | the server is in a datacentre and client is behind a normal adsl2+ modem / router |
| 11:52.45 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:52.47 | ketas | behind nat then? |
| 11:54.10 | joshuahh | yes |
| 11:54.29 | joshuahh | weird thing is, i can call out, i can even call another extension (3000) which is on 3G network (iphone app) and works fine |
| 11:54.46 | joshuahh | but whenever i try to call in to the "4000" extension which is the cisco SPA504G phone, i get service unavailable |
| 11:55.02 | joshuahh | the cisco 504g phone can call voicemail and other extensions perfect |
| 11:56.04 | joshuahh | http://pastebin.com/2vRU5DR6 |
| 11:56.15 | joshuahh | i see that in the debug |
| 11:59.26 | joshuahh | actually nevermind |
| 11:59.32 | joshuahh | asterisk/freepbx was setup correctly |
| 11:59.44 | joshuahh | but i had to change a setting inside the phone provisioning |
| 11:59.45 | joshuahh | thanks |
| 12:02.45 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 12:07.23 | ketas | can't call? |
| 12:07.49 | ketas | ah :) |
| 12:08.40 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:09.43 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:10.13 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 12:17.00 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 12:17.11 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 12:21.38 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 12:24.38 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 12:31.36 | *** join/#asterisk davlefouAMD (~david@197.15.46.25) |
| 12:33.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:37.25 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 12:37.29 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 12:38.07 | *** join/#asterisk tenspeed705 (~keiths@goatfarm.vianet.ca) |
| 12:38.33 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 12:40.59 | *** join/#asterisk k612 (~K610@cred.epid.ucl.ac.be) |
| 12:43.24 | *** join/#asterisk italoros_ (~italoross@187.60.66.11) |
| 12:44.48 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.143) |
| 12:45.59 | *** join/#asterisk F|ReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:46.02 | *** join/#asterisk thecardsmith (~doug@unaffiliated/protocoldoug) |
| 12:48.41 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 12:52.41 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 12:54.50 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 12:55.53 | *** join/#asterisk vlad_sta_ (~vlad_star@195.68.167.197) |
| 12:56.16 | *** join/#asterisk XuW (~xuw@110.77.229.148) |
| 12:59.39 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kieiusxsaitflwjf) |
| 12:59.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:00.11 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:00.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:01.54 | *** join/#asterisk MushroomNZ (~mushroomn@199.58.87.105) |
| 13:01.54 | *** join/#asterisk janakiram (~janaki@118.102.128.225) |
| 13:01.54 | *** join/#asterisk serafie (~erin@nat/digium/x-vcwmforjtcpzhoyl) |
| 13:02.20 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 13:03.08 | *** join/#asterisk santa0536 (~santa@193.28.87.242) |
| 13:05.02 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 13:07.08 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:07.36 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 13:08.27 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 13:08.40 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:21.53 | *** join/#asterisk JimmyGulp (~james@cpc27-brad20-2-0-cust175.17-1.cable.virginmedia.com) |
| 13:27.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:27.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:29.06 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 13:29.20 | *** join/#asterisk edwin_quijada (macaruchi@190.166.36.156) |
| 13:36.45 | *** join/#asterisk TimeRider (~steve@90.211.239.43) |
| 13:36.58 | *** join/#asterisk joesmoe (~joesmoe@c-76-112-110-95.hsd1.mi.comcast.net) |
| 13:37.36 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 13:42.23 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 13:44.27 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 13:44.27 | *** mode/#asterisk [+o file] by ChanServ |
| 13:44.43 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 13:45.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:45.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:48.27 | *** join/#asterisk davlefouAMD (~david@197.15.46.25) |
| 13:50.24 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:52.43 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 13:53.08 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:53.08 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:53.47 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 13:55.07 | *** join/#asterisk JimmyGulp (~james@cpc27-brad20-2-0-cust175.17-1.cable.virginmedia.com) |
| 13:55.07 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:55.07 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 13:55.07 | *** join/#asterisk zamba (marius@flage.org) |
| 13:55.07 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 13:55.07 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 13:55.07 | *** join/#asterisk asteriskandy (asteriskan@v6.pandora.panicbnc.us) |
| 13:55.07 | *** join/#asterisk 44UAABBGB (~mark@fw.seamans.cc) |
| 13:55.07 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 13:55.07 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 13:55.07 | *** mode/#asterisk [+o malcolmd] by card.freenode.net |
| 13:55.14 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:56.04 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:58.29 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 14:00.26 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 14:03.37 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:05.54 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
| 14:05.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:09.03 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 14:10.29 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 14:17.24 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 14:17.37 | *** join/#asterisk itgrl (~itgrl@12.25.151.58) |
| 14:20.11 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:21.59 | *** join/#asterisk davlefouAMD (~david@197.15.46.25) |
| 14:26.12 | *** join/#asterisk envirocbr (~enviro@unaffiliated/envirocbr) |
| 14:26.57 | envirocbr | Can someone provde a link to setting up asterisk with a POTS line? Someone sent me something like that a long time ago and my lgos don't back so far :( |
| 14:27.15 | envirocbr | Time Warner provided me a pots line with one number, but I need internal extensions and intenral voicemail |
| 14:27.15 | pabelanger | ~book |
| 14:27.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:27.23 | pabelanger | envirocbr: ^ |
| 14:28.48 | envirocbr | Thanks |
| 14:29.12 | *** join/#asterisk XuW (~xuw@110.77.229.148) |
| 14:29.16 | envirocbr | I am going VoIP internally, any recommendations on a good used Cisco VoIP phone? Have 3-4 users, MAX |
| 14:29.24 | envirocbr | Cisco 3750V2 switch |
| 14:29.30 | envirocbr | wth advanced license |
| 14:29.49 | [TK]D-Fender | envirocbr: Cisco phones are usually lower on the recommendation list |
| 14:29.55 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 14:29.58 | [TK]D-Fender | Polycom > All. |
| 14:30.17 | envirocbr | Really? |
| 14:30.18 | [TK]D-Fender | Cisco's are more of a hassle, have licensing issues, etc. |
| 14:30.18 | beardy | envirocbr: 7940, 796*, 799* |
| 14:30.40 | [TK]D-Fender | Cisco's SPA series is friendly are more cost effective. |
| 14:30.48 | [TK]D-Fender | But not the 79XX series |
| 14:31.07 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:31.17 | [TK]D-Fender | envirocbr: Anf for your PST link, depends how many lines, what kind of usage, etc. |
| 14:31.20 | [TK]D-Fender | PSTN* |
| 14:31.23 | envirocbr | $200 a good price for a used Cisoc phone 7940 |
| 14:31.37 | eirirs | no |
| 14:31.50 | envirocbr | [TK]D-Fender: I have one line, not very heave usage |
| 14:31.51 | [TK]D-Fender | Depends where... |
| 14:31.54 | envirocbr | amazon |
| 14:32.22 | envirocbr | I guess I may need to upgrade soon enough to have more than one person on the phone at the same time |
| 14:32.42 | envirocbr | Any experience with Time Warner Cable business class? |
| 14:32.46 | *** join/#asterisk suporte85 (~suporte85@201-95-224-78.dsl.telesp.net.br) |
| 14:33.29 | beardy | Depends where.. but I wouldn't pay more than the equivalent of <$150 |
| 14:33.46 | Greenlight | I'm using the following Dial syntax: SIP/Peer/Dest/ServerIP (eg Dial(SIP/mypeer/0123456789/ip.address.com) ). However, it seems the SIP INVITE being generated contains the FQDN of the peer, rather than the actual IP address used. Would I be correct in thinking this is a bug? |
| 14:34.14 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 14:34.25 | envirocbr | I pay $89 for one line wiht all the features and 10/1 Intenret with a static IP |
| 14:34.41 | [TK]D-Fender | envirocbr: Your POTS delivered via cable modem? |
| 14:34.56 | envirocbr | [TK]D-Fender: Yes, they have their own modem and a pots line comes from that |
| 14:35.04 | ketas | that's not "pots" |
| 14:35.08 | envirocbr | Well, true |
| 14:35.11 | [TK]D-Fender | Greenlight: there is no extra /. Dial(SIP/peer/numbertodial) |
| 14:35.15 | envirocbr | what would you call that? |
| 14:35.39 | Greenlight | [TK]D-Fender: Sure there is, it's just not used very often, it's a valid syntax though, beleive me |
| 14:35.40 | [TK]D-Fender | envirocbr: then you are wasting translations. You'd be better having your server connect directly to them instead |
| 14:35.41 | ketas | depends how they have done it |
| 14:35.45 | ketas | it's probably ip |
| 14:35.54 | [TK]D-Fender | Greenlight: I've never seen anything documenting it. Got a link? |
| 14:36.02 | Greenlight | Yea, two secs |
| 14:36.17 | Greenlight | (Took me a while to find it myself) |
| 14:36.28 | envirocbr | [TK]D-Fender: I guess Asterisk can't be virtual? |
| 14:36.39 | ketas | technically a ata makes your own "pots" but it's weird to call it so |
| 14:36.41 | envirocbr | I'd hate to put yet another server in the rack |
| 14:36.51 | ketas | s/ a / an / |
| 14:36.55 | [TK]D-Fender | envirocbr: It can. |
| 14:37.00 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 14:37.01 | envirocbr | [TK]D-Fender: Cool |
| 14:37.10 | envirocbr | So, what would the easiest thing to do be? |
| 14:37.42 | [TK]D-Fender | Ditch TW's pots service, get a known provider that suits you, get an easy to configure bunch of phones. |
| 14:38.05 | envirocbr | TW offers much better pricing |
| 14:38.11 | Greenlight | [TK]D-Fender: Checkout sip.conf.sample in any recent build |
| 14:38.12 | [TK]D-Fender | envirocbr: Show us. |
| 14:38.29 | [TK]D-Fender | Greenlight: Is there any other documentation you've found? |
| 14:38.36 | Greenlight | [TK]D-Fender: Specifically, the "SIP dial strings" section near the top |
| 14:38.53 | envirocbr | [TK]D-Fender: I got a quote from Centurlink and AT&T, both well over $200 per month |
| 14:38.58 | envirocbr | with 2 year contracts |
| 14:39.06 | envirocbr | TW offered me 1 year, $89 |
| 14:39.16 | Greenlight | [TK]D-Fender: As I say, it's not a syntax that appears to be used too often. It does work, as the SIP invites get sent to the correct IP. |
| 14:39.20 | envirocbr | I can go to a /28 for $5.00 |
| 14:39.26 | envirocbr | everyone else wanted $50+ |
| 14:39.54 | [TK]D-Fender | Greenlight: ; SIP/devicename/extension/IPorHost ----- ; SIP/username@domain//IPorHost |
| 14:39.54 | Greenlight | However, it seems to me that the INVITE is malformed, and I wanted a 2nd opinion |
| 14:40.05 | Greenlight | [TK]D-Fender: Yes |
| 14:40.20 | [TK]D-Fender | Greenlight: #2 looks like a typo, and I'm wondering about the first... the device should spcify the port making a 2nd IP redundant... |
| 14:40.42 | Greenlight | [TK]D-Fender: In my case I use it to do proper SRV |
| 14:40.45 | [TK]D-Fender | Greenlight: I'm wondering if the sample is simply messed up and misleading\ |
| 14:40.58 | envirocbr | So, I would need to setup asterisk to wotk with what I have. TW stated they work with people intalling asterisk |
| 14:41.16 | Greenlight | [TK]D-Fender: It 100% works like that, it uses the "devicename" settings, but sents to "IPorHost" |
| 14:41.40 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:41.55 | [TK]D-Fender | envirocbr: can you confirm the monthly total for the "POTS portion" of that. |
| 14:42.09 | [TK]D-Fender | envirocbr: Not certain for the breakdown you provided |
| 14:42.22 | *** join/#asterisk XuW (~xuw@110.77.229.148) |
| 14:42.35 | envirocbr | [TK]D-Fender: It was significantly more money, that is in the past now |
| 14:42.36 | [TK]D-Fender | envirocbr: And I said ditch them for the VOICE portion, not the internet portion. |
| 14:42.54 | envirocbr | I will eventually get them to do MPLS to our next site in Virginia when it opens |
| 14:43.01 | [TK]D-Fender | envirocbr: Cna you just provide a #$ for that for refernce sake? |
| 14:43.18 | envirocbr | The phone was $40/month |
| 14:43.23 | [TK]D-Fender | This is pretty important as you're about to double-convert a single analog line with no real room for expansion... |
| 14:43.31 | [TK]D-Fender | endthat IS a very high price |
| 14:43.41 | envirocbr | that is beecause I needed features |
| 14:43.42 | [TK]D-Fender | envirocbr: $40 in the US is pretty bad actually... |
| 14:43.47 | elguero | envirocbr: plus a modem rental fee? |
| 14:43.47 | envirocbr | voicemail, hunt group, etc etc |
| 14:43.54 | Greenlight | So, would I be correct in thinking that when dialling with that syntax, the INVITE should show the ACTUAL IP or FQDN it's been sent to? |
| 14:43.55 | [TK]D-Fender | envirocbr: that line doesn't "have features". |
| 14:44.03 | envirocbr | elguero: I negotiated not to pay that |
| 14:44.07 | beardy | Separating your "Internet" service and telephony, completely (with redundancy over "Internet") is adviced. |
| 14:44.25 | [TK]D-Fender | envirocbr: It's a single line, your server should be doing your VM for you and isn't expandable... |
| 14:44.34 | envirocbr | If I remove all the features it becomes peanuts |
| 14:44.52 | *** part/#asterisk andrewMhiggs (~andy@105-236-70-239.access.mtnbusiness.co.za) |
| 14:44.55 | envirocbr | [TK]D-Fender: So, if I were to call my rep, what would I be asking for? |
| 14:45.05 | envirocbr | have them deliver a PRI? |
| 14:45.15 | beardy | If telehpny matters to you. It seems to matter to people. |
| 14:45.34 | envirocbr | Well, yes it does |
| 14:45.39 | [TK]D-Fender | envirocbr: No, perhaps a basic ITSP will do it over your existing link. One sample : http://bandwidth.com/sip-trunking/compare.html |
| 14:45.41 | envirocbr | patients need to call in and we need to call patients |
| 14:46.15 | [TK]D-Fender | http://voip.ms/dids.php |
| 14:46.58 | elguero | envirocbr: seems like your volume must be very light to limit your self to one call at a time... you get more flexibility like [TK]D-Fender is trying to point out with an ITSP |
| 14:47.10 | elguero | if you need more than one call at a time |
| 14:47.29 | elguero | if your volume is light, you are paying too much |
| 14:47.43 | [TK]D-Fender | And if it's heavy... that 's still too much. |
| 14:47.49 | elguero | true |
| 14:47.51 | [TK]D-Fender | And no mention of LD |
| 14:47.57 | [TK]D-Fender | Shopp around |
| 14:48.17 | [TK]D-Fender | Do not make assumptions that pricing is what you think it was... |
| 14:48.21 | [TK]D-Fender | ~itsplist-us |
| 14:48.22 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 14:48.24 | [TK]D-Fender | ^ |
| 14:48.28 | [TK]D-Fender | take a better sampling. |
| 14:48.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:49.24 | envirocbr | I see |
| 14:49.38 | envirocbr | What if the business wants to stick with Time Warner due to political reasons? |
| 14:49.48 | [TK]D-Fender | envirocbr: http://vitelity.net/services_voip/ |
| 14:49.56 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:50.09 | [TK]D-Fender | Flat rate $20/channel |
| 14:50.24 | ketas | what's wrong with isps in us |
| 14:50.49 | [TK]D-Fender | [10:49]envirocbrWhat if the business wants to stick with Time Warner due to political reasons? <- that IS what's wrong then |
| 14:51.30 | [TK]D-Fender | envirocbr: You're paying 40 $ for a single crappy line, no expansion, not even call-waiting. no added value AND have to pay for hardware to plug it into. "Politics" is immensly ignorant. |
| 14:51.43 | [TK]D-Fender | envirocbr: But in that case ... do whatever you want... |
| 14:52.10 | [TK]D-Fender | envirocbr: SPA-3102 to take the line in. |
| 14:52.23 | [TK]D-Fender | envirocbr: But it's a dead-end without any "features" |
| 14:53.14 | [TK]D-Fender | envirocbr: You don't get a PBX to use telco VM. You won't have multiple lines to go out of and you'll find switching internet providers harder as well. You're certainly not doing yourself any favors. |
| 14:53.21 | envirocbr | Arguing why politics is worthless is dead-end in itself. This is the problem with most IT, they bitch about business politics and each time the business punts IT rant to the side and IT gets crapped on |
| 14:53.28 | envirocbr | I am trying to avoid that |
| 14:53.42 | [TK]D-Fender | envirocbr: But you're welcome to it if that's what you want/resigned to do |
| 14:53.47 | envirocbr | [TK]D-Fender: Who said I am going to use their VM? Right now things are temporary |
| 14:53.58 | envirocbr | when we can get an intenal solution, we want to take VM in house |
| 14:54.22 | envirocbr | They're still moving equipment in the building and putting paint on the walls |
| 14:54.23 | [TK]D-Fender | envirocbr: While one person is leaving a VM you aren't placing any calls in/out .... |
| 14:54.39 | [TK]D-Fender | envirocbr: And while you're on a call.. no-one can leave you one calling in either. |
| 14:54.46 | [TK]D-Fender | envirocbr: Just remember what "single channel" really means. |
| 14:54.51 | envirocbr | I am not sure of that as we haven't tested |
| 14:54.56 | envirocbr | This is only less than 1 week old |
| 14:55.08 | envirocbr | Which is why we're wanting to move to an in house solution |
| 14:55.22 | [TK]D-Fender | envirocbr: you have a single POTS coming out of a cable modem. This doesn't require 'testing". It's kind-of a "fact" |
| 14:55.30 | envirocbr | Regardless of carier, I am sure time warner can provide a multi-channel connection |
| 14:55.54 | envirocbr | But as of right now 90% of all communicatino is done via email |
| 14:55.58 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 14:56.00 | envirocbr | few people call in |
| 14:56.03 | envirocbr | for now |
| 14:56.09 | [TK]D-Fender | envirocbr: Umm.. that DOES regard a carrier :) |
| 14:56.10 | envirocbr | That will soon change within an year |
| 14:56.32 | envirocbr | [TK]D-Fender: So, my question is, do you know if Time Warner is a "carrier" |
| 14:56.43 | [TK]D-Fender | envirocbr: they are the boddy phone company! |
| 14:56.46 | [TK]D-Fender | bloody* |
| 14:57.02 | envirocbr | Ok, so youre objection to them is you just don't like them |
| 14:57.03 | [TK]D-Fender | envirocbr: of COURSE they are the carrier! it's their magic box spitting out the line :) |
| 14:57.40 | [TK]D-Fender | envirocbr: No, a single pots line with no functionality and double audio-conversion loss that costs you also buying equipment that will waste more time before a call can even be answered... is reason enough |
| 14:57.55 | [TK]D-Fender | envirocbr: It is a shitty technological solution, AND a shitty COST solution |
| 14:57.57 | *** join/#asterisk _mnathani_ (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 14:58.32 | *** join/#asterisk jonniemac315 (~Jonalanni@99-32-51-201.uvs.toldoh.sbcglobal.net) |
| 14:58.35 | envirocbr | [TK]D-Fender: So, moving forward, I just need them to provide me mulitple channels and I should get an Ethernet handoff? |
| 14:59.04 | envirocbr | When I have done deployments with Verizon using MPLS and Avaya we just get an Ethenret handoff |
| 14:59.27 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-208.dia.static.qwest.net) |
| 15:02.15 | envirocbr | My experience is only with the networking side, not the voice stuff |
| 15:02.40 | *** join/#asterisk edwin_quijada (~macaruchi@190.166.164.135) |
| 15:02.43 | edwin_quijada | Hi! |
| 15:02.59 | edwin_quijada | I am having problem trying to register my extension to asterisk |
| 15:03.08 | edwin_quijada | any SIP extension doesnt register |
| 15:03.14 | edwin_quijada | to my asterisk |
| 15:03.18 | *** join/#asterisk ChadArag_ (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 15:06.21 | edwin_quijada | I have tried evrything but nothing happens |
| 15:06.24 | edwin_quijada | any cluees? |
| 15:07.07 | itgrl | we would need to see logs with sip debug enabled for a registration attempt to be able to help. it would also help to see the sip peer configuration. |
| 15:07.50 | [TK]D-Fender | envirocbr: I've given you references to other options for comparison. Just take a serious look at your shorter & longer term needs and see whatthey'll offer you comparatively |
| 15:08.03 | [TK]D-Fender | envirocbr: Just that that single line as-is ... is a pure loss. |
| 15:08.25 | [TK]D-Fender | edwin_quijada: Same as always .... enable SIP debug and actually look at the attempt. |
| 15:09.07 | envirocbr | [TK]D-Fender: So, what question should I ask my rep? -- I want to move from a single line to having a multi-channel conenction to my asterisk solution |
| 15:11.31 | [TK]D-Fender | envirocbr: a direct SIP service. |
| 15:11.36 | [TK]D-Fender | envirocbr: like any other provider |
| 15:12.14 | [TK]D-Fender | envirocbr: Expect it to cost proportionately more. |
| 15:12.48 | envirocbr | Through Time Warner or just in general? |
| 15:14.14 | envirocbr | and these would be delivered over IP using my existing TWC IP connection? |
| 15:16.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-kieiusxsaitflwjf) |
| 15:19.54 | [TK]D-Fender | envirocbr: VoIP. SIP is a VoIP protocol |
| 15:21.33 | [TK]D-Fender | envirocbr: So yes... internet. Packets are packets. |
| 15:21.33 | edwin_quijada | [TK]D-Fender: this is the sip debug http://bpaste.net/show/wQH1eWjGNToR8n7yDSI3/ |
| 15:23.12 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kieiusxsaitflwjf) |
| 15:23.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:25.01 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 15:26.34 | edwin_quijada | http://pastebin.com/wSJWtwa9 |
| 15:26.39 | [TK]D-Fender | edwin_quijada: [May 13 11:18:17] NOTICE[2517]: chan_sip.c:15018 check_auth: Bad authentication received from '"Cibernauta" <sip:777@192.168.1.199>' |
| 15:26.42 | [TK]D-Fender | edwin_quijada: Seems to say it... |
| 15:27.13 | edwin_quijada | the secret is equal |
| 15:27.19 | [TK]D-Fender | edwin_quijada: I'm suspecting you might be running multiple peers on a single IP phone there.... |
| 15:27.22 | [TK]D-Fender | ^ |
| 15:27.57 | edwin_quijada | but just see one IP |
| 15:27.59 | [TK]D-Fender | <--- Transmitting (NAT) to 192.168.1.68:5062 ---> <- that is not the typical port it's use, and they are NOT behind NAT |
| 15:28.12 | edwin_quijada | nop |
| 15:28.13 | [TK]D-Fender | Are you trying to register MULTIPLE accounts on that phone? |
| 15:28.20 | edwin_quijada | I am using debian wheezy |
| 15:28.27 | [TK]D-Fender | I didn't ask about distro... |
| 15:28.38 | edwin_quijada | the weird is that I have a replica from this server and works fine |
| 15:28.47 | [TK]D-Fender | That also does not tell us anything |
| 15:28.51 | [TK]D-Fender | Tell us about that phone |
| 15:29.00 | edwin_quijada | ok |
| 15:29.19 | edwin_quijada | in sip.conf I am using #exec to create all extensions |
| 15:29.34 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 15:29.37 | edwin_quijada | but if I create an extension hard way works |
| 15:29.59 | weinerk | Hi. I run asterisk -rx "sip debug", but after a while sipdebug turns off again. |
| 15:29.59 | weinerk | Am I mistaken or is it supposed to be like that? |
| 15:31.01 | Kobaz | it shouldnt turn off randomly |
| 15:32.01 | [TK]D-Fender | [11:29]edwin_quijadain sip.conf I am using #exec to create all extensions <- there is clearly a difference in how you made them. They are ALL hard. |
| 15:32.11 | [TK]D-Fender | edwin_quijada: Once that exec runs, it IS "hard" |
| 15:32.23 | [TK]D-Fender | edwin_quijada: and you are not SHOWING us anything or answering my previous questions. |
| 15:33.24 | edwin_quijada | which one ? |
| 15:35.39 | envirocbr | [TK]D-Fender: 10/1 is good enough? |
| 15:35.45 | envirocbr | or should I go 50/5 |
| 15:36.20 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 15:36.45 | [TK]D-Fender | envirocbr: calls are 85kbps/direction each max.... |
| 15:37.34 | [TK]D-Fender | edwin_quijada: your CONFIGS. Where is the answer as to how you're using the phone itself? |
| 15:37.53 | edwin_quijada | [TK]D-Fender: this is the http://pastebin.com/wSJWtwa9 |
| 15:38.06 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-208.dia.static.qwest.net) |
| 15:38.17 | edwin_quijada | this is sip.conf |
| 15:40.01 | [TK]D-Fender | .... |
| 15:40.39 | [TK]D-Fender | Ok, this is not worth the effort. Best of luck with it. |
| 15:41.53 | *** join/#asterisk emk (~emk@unaffiliated/emk) |
| 15:43.17 | emk | Hi all: when I initiate an outbound call from Asterisk, what environment variable or channelvariable do I use to get the _recipientID_ ... i.e. the opposite of ${CALLERID} |
| 15:45.02 | weinerk | Kobaz: thanks. I will try to keep an eye. I hope I am just mistaken. |
| 15:47.44 | [TK]D-Fender | emk: that does not make sense... |
| 15:48.14 | [TK]D-Fender | emk: Please clarify what this "id" is supposed to lok like and represent. |
| 15:48.25 | [TK]D-Fender | emk: And perhaps what you expect to do with it... |
| 15:52.36 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 15:54.17 | SuperNull | which version of 1.8 should i be running for 'stable' ? 1.8.15-cert? or the 1.8.21.0 |
| 15:55.54 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 15:55.56 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:56.29 | emk | [TK]D-Fender: I make a call in the CLI, or with a call file, Asterisk shows me the following http://bpaste.net/raw/ezrEcwr2NQVP8BpeA0yE/, I'd like to get that number for use later in the dialpan for doing something with System() |
| 15:56.59 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 15:59.02 | emk | [TK]D-Fender: by "get that number" I mean being able touse the number '+4915111111222' later on with a call to a script that takes the number as an argument... i.e. System(/path/to/script ${THE_VARIABLE_I_AM_ASKING_ABOUT_IN_THIS_IRC_CHAT} |
| 15:59.18 | emk | [TK]D-Fender: I hope the meaning is clear now |
| 15:59.34 | malcolmd | SuperNull: depends on your needs. the "-cert" releases represent what digium provides sla (service level agreement) support on for certain customers. it does not have all of the bug fixes that you'll find in 1.8.21.0 though |
| 16:09.53 | [TK]D-Fender | emk: use SerVar in your call file. you cannot do this when using CLI Originate |
| 16:09.58 | [TK]D-Fender | Setvar <- |
| 16:11.57 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 16:13.06 | SuperNull | thanks malcolmd, i will likely end up using 1.8.21.0 since bugfixes are the primary reason for upgrading from 1.8.6 |
| 16:15.52 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:19.23 | *** join/#asterisk ChadArag_ (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 16:21.12 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:33.04 | envirocbr | [TK]D-Fender: Just latency is what I should be concerned with, correct? |
| 16:33.36 | [TK]D-Fender | envirocbr: and jitter. And packet loss. And most of all ... WEASELS |
| 17:45.53 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:45.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 17:46.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 17:49.13 | *** join/#asterisk envirocbr (~enviro@unaffiliated/envirocbr) |
| 17:49.27 | *** join/#asterisk navaismo (~navaismo@189.241.84.20) |
| 17:49.53 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:49.53 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:50.22 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 17:52.17 | *** join/#asterisk rotten777 (~matthewby@cloud1.byrdits.com) |
| 17:52.35 | rotten777 | is anyone awake? |
| 17:52.50 | [TK]D-Fender | rotten777: Possibly |
| 17:52.55 | rotten777 | struggling to figure out why one of my DID's passes audio and the other does not… two different providers :\ |
| 17:53.19 | rotten777 | ah good haha |
| 17:53.48 | msaraiva | Now that is weird |
| 17:53.49 | rotten777 | i'm a bit of an idiot when it comes to asterisk |
| 17:53.50 | msaraiva | asterisk: catalog.c:219: mysql_table_status_i_s: Assertion `to - buff < sizeof(buff)' failed. |
| 17:53.55 | msaraiva | SIP Realtime |
| 17:53.57 | rotten777 | but networking and such i'm ok |
| 17:54.36 | rotten777 | i have sip set debug on |
| 17:54.51 | rotten777 | iptables off, incoming and outgoing calls work to this DID and ring to my extension but audio won't pass. |
| 17:55.03 | rotten777 | another DID rings my extension and audio does pass on that |
| 17:55.15 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 17:55.21 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 17:55.56 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 17:58.52 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 17:58.52 | *** join/#asterisk asteriskandy (asteriskan@v6.pandora.panicbnc.us) |
| 17:58.52 | *** join/#asterisk 44UAABBGB (~mark@fw.seamans.cc) |
| 17:58.52 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 17:58.52 | *** join/#asterisk zamba (marius@flage.org) |
| 17:58.52 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 17:58.52 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:58.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 17:58.52 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 17:59.20 | [TK]D-Fender | rotten777: pastebin your trunk settings masking only the secret, along with your call attempt |
| 17:59.26 | [TK]D-Fender | (peer) |
| 17:59.28 | [TK]D-Fender | !pb |
| 18:00.33 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 18:00.44 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 18:00.56 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 18:02.33 | rotten777 | ok give me a second |
| 18:04.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.226) |
| 18:04.47 | [TK]D-Fender | ~pb |
| 18:04.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:04.49 | msaraiva | mysql-connector bug, maybe? |
| 18:04.49 | [TK]D-Fender | ^ |
| 18:05.05 | msaraiva | weird thing is that i have the same setup on another server |
| 18:05.22 | msaraiva | same packages versions for mysql and mysql-connector-odbc |
| 18:05.28 | msaraiva | And i don't get this error. |
| 18:06.14 | rotten777 | http://pastebin.com/U3kvVTF8 |
| 18:06.21 | rotten777 | sip.conf, do you need more? |
| 18:08.09 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:10.11 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:11.45 | msaraiva | Ok, just so everyone know: mysql-connector currently on EC2 is broken (Amazon Linux AMI) |
| 18:12.00 | msaraiva | Use this one: http://mirror.cogentco.com/pub/mysql/Connector-ODBC/5.2/mysql-connector-odbc-5.2.2-1.el6.x86_64.rpm |
| 18:14.59 | *** join/#asterisk Khronos (~Khronos@204.13.1.113) |
| 18:17.04 | SuperNull | anyone use the builtin stun server? .. is it deployment grade(couple hundred users) or small office only ? |
| 18:17.14 | rotten777 | [TK]D-Fender: Any luck with the sip.conf or did you need something else? |
| 18:17.54 | [TK]D-Fender | rotten777: ;nat=yes <- this should be YES (uncommented for [general] |
| 18:18.20 | [TK]D-Fender | rotten777: and "canreinvite" has been replaced with "directmedia" in 1.6+ set this under [general] and ALL your peers accordingly. |
| 18:18.56 | [TK]D-Fender | rotten777: [flowroute] <- this ishould be nat=no |
| 18:19.04 | [TK]D-Fender | [voipms] <- them too |
| 18:19.37 | msaraiva | nat = yes is deprecated |
| 18:19.45 | msaraiva | Use force_rport,comedia |
| 18:21.47 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 18:23.05 | rotten777 | msaraiva both force_rport,comedia? or one or the other |
| 18:23.30 | rotten777 | pbx is behind nat, 6000 is hardware phone on same subnet, both DID's on public servers |
| 18:24.11 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 18:24.19 | rotten777 | one of the DID's rings, answers, and there's audio… the other rings, answers, no audio |
| 18:25.36 | msaraiva | Both |
| 18:25.43 | msaraiva | The whole string |
| 18:26.17 | rotten777 | ok so that on the general as well as my DID providers? |
| 18:26.55 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.226) |
| 18:27.26 | msaraiva | You can either set it on general, or on each peer. |
| 18:27.47 | msaraiva | If you have peers that are only on the internal network, you may set these to nat = no. |
| 18:28.57 | [TK]D-Fender | rotten777: yes |
| 18:29.03 | rotten777 | so i changed to nat=force_rport,comedia and now I have this error |
| 18:29.04 | rotten777 | [May 13 14:28:30] WARNING[2990]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead |
| 18:29.08 | rotten777 | when reloading |
| 18:29.23 | [TK]D-Fender | rotten777: if your other calls are hitting [general] then the NAT setting will cause problems. Make sure you have peers to match them |
| 18:30.33 | rotten777 | i'm still experiencing the same symptoms. both DID's take calls only 1 passes audio |
| 18:32.04 | [TK]D-Fender | rotten777: I asked you to show the call... do this with your now supposedly updated and applied configs |
| 18:32.32 | rotten777 | sip set debug on ??? |
| 18:34.15 | rotten777 | i don't know how to show you the call |
| 18:35.23 | rotten777 | externip was wrong |
| 18:35.29 | rotten777 | firewall issues |
| 18:35.33 | rotten777 | i need a beer..... |
| 18:38.26 | [TK]D-Fender | No ... too great a risk that alcohol was already to blame for this ;) |
| 18:41.05 | rotten777 | haha true enough |
| 18:47.59 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:47.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:52.19 | _abc_ | does anyone here use chan_sccp with asterisk? |
| 18:52.48 | _abc_ | i got some 6921 phones which work fine but do not ring. I assume it is a protocol issue (sccp protocol version). Any ideas what number to set? |
| 18:54.51 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 18:55.48 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.181) |
| 19:00.40 | [TK]D-Fender | _abc_: You haven't confirmed clearly which ones are working, which ones aren't, provided the debug of the ones that failed, or confirmed what version of * you are running. You'll want to change that.... |
| 19:01.29 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:01.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:01.57 | _abc_ | [TK]D-Fender: true... I was just trying to find someone who is on rx... |
| 19:02.50 | _abc_ | [TK]D-Fender: rephrase: two of two tested 6921s work fine but do not ring. The ring lights flash and one can pick up and talk when they flash but nothing is heard. It is not a volume issue. No ring tone file is loaded at all. |
| 19:03.02 | _abc_ | The device protocol is set to default auto in the sccp.conf |
| 19:03.22 | _abc_ | I did not dig deeper yet. I assume the protocol must be set to something for these devices, other than auto. |
| 19:03.31 | _abc_ | I use cnah_sccp 3.6.x |
| 19:03.53 | _abc_ | More info can be obtained tomorrow. What do I need to know, exactly? |
| 19:04.05 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.129) |
| 19:04.09 | _abc_ | Simplest would be for someone with 6921s on schan_sccp to say config. |
| 19:04.11 | _abc_ | thanks |
| 19:04.11 | [TK]D-Fender | I'd recommend coming back with all of that including the Cisco configs... |
| 19:04.28 | _abc_ | the cisco configs? You mean the *.xml.cnf? |
| 19:04.39 | _abc_ | It is bare, defines call manager and port only. |
| 19:05.41 | [TK]D-Fender | _abc_: Come back with all of it to show. Then those with more experience will have something to go on. |
| 19:05.44 | _abc_ | rephrase2: above 'two of two tested 6921s work fine but do not ring. The ring lights flash and one can pick up and talk when they flash but nothing is heard.' <- should be: sound is heard both ways, but RINGING is never heard |
| 19:05.52 | _abc_ | [TK]D-Fender: okay |
| 19:06.17 | _abc_ | [TK]D-Fender: I'll try and wait a little longer maybe some wabbit with 6921s will quip |
| 19:06.32 | _abc_ | I also mailed to the sccp user support list |
| 19:06.40 | _abc_ | with a pertinent question |
| 19:07.58 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 19:09.57 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.181) |
| 19:12.44 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 19:13.37 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 19:14.59 | jeffspeff | why is this ---> exten=s,1,NoOp(${CALLERID(num)} called for ${CALLERID(rdnis)}) resulting in this ---> -- Executing [s@DID-DETECTION:1] NoOp("SIP/INTELEPEER-0000bdf7", "5551231234 called for ") in new stack ? |
| 19:15.15 | jeffspeff | am i not getting the rdnis value correctly or is rdnis not being sent? |
| 19:18.56 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 19:20.51 | *** join/#asterisk Guest50437 (~joako@opensuse/member/joak0) |
| 19:25.25 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 19:31.04 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 19:36.08 | *** part/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 19:36.18 | *** join/#asterisk tallest_red (~CNZ@ip98-169-197-57.dc.dc.cox.net) |
| 19:44.35 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 19:45.34 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 19:50.27 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 19:51.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.181) |
| 19:55.34 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 19:58.37 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:00.33 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:00.33 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:00.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.181) |
| 20:00.59 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 20:02.59 | *** join/#asterisk gomez1 (~kvirc@bzq-79-176-225-144.red.bezeqint.net) |
| 20:06.47 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 20:08.39 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:10.10 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 20:11.53 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 20:11.53 | *** join/#asterisk 44UAABBGB (~mark@fw.seamans.cc) |
| 20:11.53 | *** join/#asterisk asteriskandy (asteriskan@v6.pandora.panicbnc.us) |
| 20:11.53 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:12.55 | jeffspeff | why is this ---> exten=s,1,NoOp(${CALLERID(num)} called for ${CALLERID(rdnis)}) resulting in this ---> -- Executing [s@DID-DETECTION:1] NoOp("SIP/INTELEPEER-0000bdf7", "5551231234 called for ") in new stack ? |
| 20:12.57 | jeffspeff | am i not getting the rdnis value correctly or is rdnis not being sent? |
| 20:13.54 | WIMPy | The same reason you end up at s I suppose. |
| 20:15.16 | jeffspeff | WIMPy, the number comes in to a certain DID in CONTEXT-A and then does a goto(did-detection,s,1) where it's supposed to show the rdnis |
| 20:16.27 | WIMPy | Try to debug the channel to find out if you receive anything at all. |
| 20:20.05 | [TK]D-Fender | [16:12]jeffspeffam i not getting the rdnis value correctly or is rdnis not being sent? <- I would recommend actually looking at the call. |
| 20:21.29 | jeffspeff | thanks WIMPy and TK. I was thinking that using transfer() on one box to redirect the call to another DID on another box would result in box-a sending the rdnis to box-b; but it doesn't. |
| 20:24.55 | WIMPy | You need to set it in your dialplan. |
| 20:28.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 20:28.38 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 20:28.38 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 20:28.38 | *** join/#asterisk zamba (marius@flage.org) |
| 20:42.52 | *** join/#asterisk LeLutin (~lelutin@vps-1036922-2706.manage.myhosting.com) |
| 20:43.09 | LeLutin | hello |
| 20:45.16 | LeLutin | sound is choppy when I make a call from the server in the office and our voip provider. how can I find what is causing those sound issues? |
| 20:45.51 | tzafrir_laptop | netsplits? |
| 20:46.30 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:46.45 | LeLutin | tzafrir_laptop: you mean that latency could cause the issue? |
| 20:47.37 | tzafrir_laptop | I was kidding. But generally it's not the latency. Rather: the jitter (differing latency) |
| 20:47.57 | tzafrir_laptop | or simpler: dropped packets |
| 20:48.11 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 20:48.13 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 20:48.15 | tzafrir_laptop | what type of connection is that? SIP? IAX? |
| 20:48.22 | tzafrir_laptop | (SIP: UDP?) |
| 20:48.28 | danfromuk | Hi, is ActionID required in AMI? |
| 20:49.02 | tzafrir_laptop | danfromuk, no |
| 20:52.20 | danfromuk | Perfect. Thanks for confirming. |
| 20:52.43 | LeLutin | tzafrir_laptop: sip from phones to asterisk, then iax from asterisk to provider |
| 20:53.02 | LeLutin | the jitterbuffer is enabled for iax. let me see if it is to sip too |
| 20:53.53 | LeLutin | apparently no |
| 20:55.09 | LeLutin | ah, actually I asked asterisk with "sip show settings" instead of just trusting the config files. and the jitterbuffer is enabled for sip too |
| 21:00.50 | danfromuk | I'm converting from realtime to static conf files. Is there a limit on the size of a dialplan? |
| 21:03.33 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 21:03.34 | jeffspeff | danfromuk, I wouldn't imagine so. i'd think the limit was in memory to load the DP |
| 21:03.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:03.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:15.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:19.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.102) |
| 21:32.50 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 21:45.50 | *** join/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 21:48.21 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 21:48.55 | j-fish | I'm using my router's vpn feature,i gave it the start ip and end ip,could that cause problems if it tries to use one of the phones/computer's ips ? |
| 21:50.21 | beardy | Without knowing exactly what you mean, yes. |
| 21:50.49 | beardy | Use one range for DHCP and another for your vpn. |
| 21:51.00 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 21:52.36 | beardy | Use different subnets for telephony and the rest. |
| 21:56.09 | *** join/#asterisk cedr (cedr@unaffiliated/cedr) |
| 22:01.28 | j-fish | since i've started using it,random computers losing internet connection thought it might have something to do with it |
| 22:03.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:05.29 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 22:08.10 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:08.10 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:08.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:09.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-txgruwpymfjafdfn) |
| 22:09.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:12.07 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 22:24.23 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-rmlkufxqwlbqvbbe) |
| 22:26.33 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 22:30.32 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 22:34.05 | *** join/#asterisk natschil (~nathanael@stgt-5f708b5d.pool.mediaWays.net) |
| 22:38.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-txgruwpymfjafdfn) |
| 22:48.54 | *** join/#asterisk deviantlinux (~deviant@unaffiliated/deviantlinux) |
| 22:49.58 | deviantlinux | Question on the websockets feature of asterisk 11 - can I use this to receive realtime manager events (call initiated, hung up, etc)? If so, is there a proper method I should send in the URL? Right now I am just using ws://<ip>:8088/ws |
| 22:53.40 | *** join/#asterisk JustinAiken (~Adium@74.118.180.13) |
| 22:54.21 | JustinAiken | I'm trying to setup confbridge to place an admin and a regular user in a room; I want the admin to hear the confbridge-join and confbridge-leave sounds when the other user comes in an out, but not the other user |
| 22:54.42 | JustinAiken | i've set quiet=yes for the regular user, and quiet=no for the admin, |
| 22:54.48 | JustinAiken | but the admin still can't hear the sounds |
| 22:54.56 | JustinAiken | ooh, and my spelling is actually right in the .conf :p |
| 22:55.54 | JustinAiken | any ideas? |
| 22:58.53 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 23:04.34 | *** join/#asterisk voar (~voar@97.101.195.94) |
| 23:08.56 | *** join/#asterisk Robotman321 (~brad@50-194-126-10-static.hfc.comcastbusiness.net) |
| 23:19.07 | *** join/#asterisk Robotman321 (~brad@c-98-228-171-21.hsd1.il.comcast.net) |
| 23:21.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.143) |
| 23:21.33 | *** join/#asterisk Khronos (~Khronos@c-76-110-123-39.hsd1.fl.comcast.net) |
| 23:36.40 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 23:36.43 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 23:43.25 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 23:44.29 | danfromuk | Hi. I want to test g729 passthrough is working before asking the client to try it out. The simplest way is to set up g729 with an echo test. Is there a free way to use g729 temporarily for testing purposes if its not going to be resold? |
| 23:45.55 | danfromuk | Alternatively, how quickly are g729 orders processed by digium? |
| 23:46.31 | navaismo | no, there is no "free" way, orders are very quick just 10 usd if you want to test |
| 23:53.34 | *** join/#asterisk joesmoe (~joesmoe@c-76-112-110-95.hsd1.mi.comcast.net) |
| 23:56.32 | voar | Has anyone ever found a fix for realtime not routing by caller id? IE, the conf equivilent of exten => _1NXXXXXXXXX/_321NXXXXXX,1,Dial(SIP/peer,30) ? |