| 00:29.36 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:40.12 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 00:41.14 | *** join/#asterisk cloakable (~cloakable@cpc4-aztw23-2-0-cust856.aztw.cable.virginmedia.com) |
| 00:41.54 | cloakable | How well should a small Asterisk install run on an Atom D525 box? |
| 00:44.11 | [TK]D-Fender | sur |
| 00:44.13 | [TK]D-Fender | e |
| 00:45.38 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 00:47.46 | cloakable | Cool |
| 00:48.18 | cloakable | One FXO to my landline, and a few SIP devices. |
| 00:48.27 | cloakable | Probably voicemail too. |
| 00:55.08 | jaytee | I've got several Asterisk systems running in small offices with 5 to 10 phones and voicemail on DM525 mini-itx board with 32GB SSD and 2GB RAM. They work great |
| 01:08.16 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 01:24.42 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 01:29.31 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:30.01 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:41.17 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 01:48.21 | *** part/#asterisk Steel_Reign (~Steel_Rei@72-28-219-021-dhcp.mia.fl.atlanticbb.net) |
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| 02:07.35 | ChrisInSydney | cloakable: 30 hansets on an atom 1.6GHz hyperthread with 1024 RAM and a conventional disk. G711 passthrough. Voicemail boxes, auto attendants, call twinning Never had an issue |
| 02:08.08 | cloakable | Awesome, thank you :) |
| 02:08.52 | cloakable | This will be doing a maximum of seven handsets (Gigatel DECT base supports six, plus my Android) |
| 02:11.15 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 02:11.16 | ChrisInSydney | "A Piece of Piss" as we might say here in the colony |
| 02:11.56 | ChrisInSydney | so long as you are not transcoding g729/723 or trying to mangle video, you'll be sweet |
| 02:13.34 | cloakable | Awesome |
| 02:14.38 | ChrisInSydney | I still have 25 users accessing a web MySQL / PHP database running on an old Celeron 300A cranked to 450MHz. Been running since late last century when we wrote the app |
| 02:14.52 | ChrisInSydney | mind you thats not voice |
| 02:15.07 | ChrisInSydney | good luck |
| 02:16.39 | cloakable | danke |
| 02:17.24 | *** join/#asterisk adeel (~adeel@72.53.78.136) |
| 02:27.04 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
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| 04:26.09 | *** join/#asterisk [T]ank (Tank@c-174-52-232-140.hsd1.ut.comcast.net) |
| 04:26.55 | [T]ank | When forwarding an inbound call out to a cell phone or something, is there a simple way to send the original callers callerid to the cell phone? |
| 04:35.28 | *** part/#asterisk Bullmoose (~Bullmoose@75-174-79-132.bois.qwest.net) |
| 04:37.49 | [TK]D-Fender | [T]ank: It gets sent unless your peer setup forces one, or your carrier refuses it |
| 04:50.34 | [T]ank | hmm, the carrier uses "fromuser" to authenticate with i think. That value is my ani, which i am assuming is what is causing this, right? |
| 04:51.02 | [TK]D-Fender | Correct. set "sendrpid=yes" and "trustrpid=yes" |
| 04:51.08 | [TK]D-Fender | And retest |
| 04:51.24 | [T]ank | is that in the general section? or just in the user its self? |
| 04:53.09 | [TK]D-Fender | in the peer |
| 04:54.44 | [T]ank | hmmm, now i get the error: Purely numeric hostname (<number i am forwarding to), and not a peer--rejecting! |
| 04:57.26 | p3nguin | Did you set fromuser because they require it or because you thought you needed to set it? |
| 04:57.28 | [T]ank | that was due to a typo... didnt get expected results however... investigating |
| 04:57.39 | [T]ank | yeah the fromuser is required |
| 05:01.19 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
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| 05:58.59 | *** part/#asterisk [T]ank (Tank@c-174-52-232-140.hsd1.ut.comcast.net) |
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| 06:51.22 | KNERD | what is "directmedia" There is NOTHING I can find on it |
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| 07:11.02 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
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| 07:27.58 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:28.00 | schmidts | good morning |
| 07:32.26 | KNERD | bad morning |
| 07:48.52 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:49.53 | wdoekes2 | morning :) |
| 07:54.46 | *** join/#asterisk stix (~stix@193.89.191.209) |
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| 08:10.04 | KNERD | evenin' guvnor |
| 08:13.17 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:13.18 | ChannelZ | Tally ho |
| 08:14.11 | KNERD | evenin' guvnor |
| 08:17.03 | *** join/#asterisk thebomb (~dawiep@cartrackfw02.hosting.co.za) |
| 08:17.13 | *** join/#asterisk black187 (~black187@93-103-22-42.static.t-2.net) |
| 08:17.43 | black187 | does anybody know how to detect Asterisk SIP deadlock - So i can make a script to kill and restart Asterisk? |
| 08:17.43 | dym | ellow |
| 08:22.05 | wdoekes2 | black187: periodically do a sipsak(1) options call on it? use sipsak(1) and/or sipp(1) to test registering and/or dialing |
| 08:22.24 | wdoekes2 | a bit of bash glue on it, and voila |
| 08:23.30 | KNERD | black187: that should be a function ogf the OS |
| 08:23.41 | black187 | Haha - the glue one is worth a shot. Hmm, the sipsak is not a bad idea. |
| 08:23.56 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 08:24.31 | black187 | So a SIP deadlock would not answer to SIP option messagess? |
| 08:24.53 | wdoekes2 | depends on what kind of deadlock |
| 08:25.02 | wdoekes2 | but there is one main loop in asterisk that handles incoming messages |
| 08:25.21 | KNERD | what is "directmedia" There is NOTHING I can find on it |
| 08:25.23 | wdoekes2 | it that thread stalls, you'll detect it |
| 08:25.23 | thebomb | hi, is there anyway to check what queue a user or sip extension belongs to ? |
| 08:25.37 | wdoekes2 | KNERD: whether asterisk relays the RTP stream or not |
| 08:26.04 | black187 | Ok, thanks for the info guys! |
| 08:26.06 | wdoekes2 | KNERD: normally a SIP call is set up with asterisk as the RTP destination for both (all) call legs |
| 08:26.55 | KNERD | oh....thanks. I wonder why I cannot find documentation on this |
| 08:27.02 | wdoekes2 | KNERD: with directmedia=yes, asterisk will attempt to get out of the way (issuing a re-INVITE) |
| 08:27.12 | wdoekes2 | KNERD: originally it was called canreinvite=yes |
| 08:27.27 | KNERD | oh I see |
| 08:27.55 | wdoekes2 | are you looking in the sip.conf.sample ? |
| 08:28.07 | KNERD | no |
| 08:28.10 | wdoekes2 | there are enough sentences devoted to the option |
| 08:28.38 | KNERD | okay |
| 08:28.39 | KNERD | thanks |
| 08:29.20 | KNERD | wdoekes2: however |
| 08:29.51 | KNERD | Someone had me remove ALLOW/DISALLOW and put in "directmedia=yes" |
| 08:30.06 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:30.07 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:33.11 | wdoekes2 | KNERD: and my father-in-law drove me home today |
| 08:33.17 | wdoekes2 | your point? |
| 08:34.31 | KNERD | I am wondering why |
| 08:39.53 | wdoekes2 | KNERD: if you remove the allow/disallow, then every codec is allowed => more chance of the two legs agreeing on a codec |
| 08:40.18 | wdoekes2 | KNERD: if you then attempt directmedia, you'll lose the traffic/processing requirements for that call |
| 08:40.21 | wdoekes2 | KNERD: => profit |
| 08:40.58 | KNERD | thank you kindly |
| 08:43.44 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 08:44.29 | vlt | Hello. My provider told me to dial *21*<target># to activate CFU on my ISDN line. SIP-to-ISDN gateway is an Inalp Patton. When I Dial(SIP/*21*555012345#@patton) I get SIP response 400 "Bad Request" back. Any idea how to solve this? |
| 08:45.19 | *** join/#asterisk iulhk (~iulhk@116.71.181.13) |
| 08:49.21 | KNERD | vlt: maybe first by asking them? |
| 08:51.58 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 08:57.33 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 09:01.43 | wdoekes2 | vlt: SIP is not ISDN, is it? |
| 09:03.08 | wdoekes2 | or do they handle the isdn<->sip for you? |
| 09:03.24 | wdoekes2 | oh wait, perhaps I should read :) |
| 09:04.27 | wdoekes2 | start out by looking at the sip debug of the INVITE (pb it here?) to check if there is anything wrong with the request |
| 09:11.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:23.31 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 09:25.14 | thebomb | hi, is there is anyway to check if a SIP extension is in a certain queue ? |
| 09:25.41 | kaldemar | vlt: ISDN trace from the patton might also be useful. you'd want to know what the gateway tries to dial over ISDN, if it even tries, and the possible response from telco. |
| 09:28.44 | thebomb | nevermind the agent function looks like it can return the extension |
| 09:30.33 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
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| 09:39.21 | kaldemar | thebomb: function QUEUE_MEMBER_LIST |
| 09:39.58 | thebomb | kaldemar, didn't see that one, tx appreciate it |
| 09:48.01 | *** join/#asterisk bipul (75d31984@gateway/web/freenode/ip.117.211.25.132) |
| 09:49.12 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:52.06 | *** part/#asterisk bipul (75d31984@gateway/web/freenode/ip.117.211.25.132) |
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| 10:03.43 | *** join/#asterisk iulhk (~iulhk@119.152.10.205) |
| 10:04.23 | iulhk | using asterisk-10.2.0 is there any way to setup realtime confbrige ? |
| 10:05.58 | kaldemar | what do you mean? getting bridge profile configurations through realtime? |
| 10:06.08 | *** join/#asterisk kieppie (~jaco@ip-58-28-154-35.static-xdsl.xnet.co.nz) |
| 10:06.41 | *** join/#asterisk timahvo1 (~rogue@41.80.23.207) |
| 10:07.16 | kieppie | hi all. I need some help with the "Dialed Number Manipulation Rules", please. I want to substiture the international + prefix with 00. How is that achieved? |
| 10:08.12 | kaldemar | kieppie: depends on how you dial. what does your Dial line look like? |
| 10:08.21 | iulhk | <kaldemar>: yes , do we have any option ? |
| 10:09.28 | kaldemar | iulhk: static realtime is all you have. |
| 10:10.19 | kieppie | hi kaldemar: still new at this, so not entirely sure what you mean. presently my rule for trunk & extention is simply "X." (no prefix or prepend). when I dial from my client, it's usually in the format of "+12 34 567 8901" |
| 10:10.52 | iulhk | do we have any option to record video conferencing? as well can we stream this saved file at some web-browser by using some media player? |
| 10:11.33 | kaldemar | kieppie: what GUI are you using? |
| 10:12.27 | kieppie | freepbx - I'm assuming the configs I'm manipulating here are that of Asterisk itself. If I need to dig in the config, I'm not shy |
| 10:18.13 | kieppie | I've simply tried using "00", "+", "X.", but that's not substituting the "+" with "00" |
| 10:21.49 | kaldemar | you need to do something to cut the + from a variable you're using, generally that's done by putting :1 after a variable name in a reference. e.g. ${EXTEN:1} and then just prepend that with what you want, like 00${EXTEN:1}. but on how to do that in freepbx so it works, ask in #freepbx. |
| 10:23.53 | *** join/#asterisk CommaCrazy (~CCK@87.250.37.26) |
| 10:24.53 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 10:29.24 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:31.28 | kieppie | I found this, thanks: http://www.freepbx.org/support/documentation/howtos/how-to-strip-or-replace-the-character-at-the-beginning-of-a-called-numb |
| 10:31.32 | kieppie | seems to do the trick |
| 11:01.27 | *** part/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 11:03.37 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
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| 11:08.50 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 11:09.02 | Dovid | hi all. is there any way to only send RTP once we get a 200 OK? |
| 11:15.03 | Gugge | Dovid: do you need asterisk to stop sending anything else than RTP after a 200 OK, or do you need it to not send RTP before the 200 OK. and why? |
| 11:15.20 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:26.43 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:36.19 | Dovid | Gugge: I want asterisk to not send any RTP to end point till we get a 200 ok from them |
| 11:36.26 | Dovid | seems to be a bug in Avaya |
| 11:44.16 | iulhk | can we record video calls in asterisk ? |
| 11:49.40 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:54.34 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:55.06 | Dovid | iulhk: Yes you can with MixMonitor |
| 11:59.02 | kaldemar | afaik MixMonitor only utilizes an audio hook, so that's all it can record. |
| 12:03.53 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 12:08.44 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-79-132.bois.qwest.net) |
| 12:11.12 | iulhk | <kaldemar>: so no option for video recording ? |
| 12:12.25 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 12:13.47 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 12:13.59 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:14.10 | phpboy | Hi, how do I go about logging manager data to it's own log file? |
| 12:21.44 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 12:33.41 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 12:33.44 | qakhan | hi all |
| 12:34.28 | qakhan | i want to setup an ivr which take callerid, name and address. and save then in DB |
| 12:34.41 | qakhan | plz help how can i do this. |
| 12:42.00 | phpboy | the easiest way would be to familiarize yourself with the system command so you can pipe the info to PHP script for example |
| 12:43.12 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 12:44.00 | *** part/#asterisk Bullmoose (~Bullmoose@75-174-79-132.bois.qwest.net) |
| 12:49.31 | qakhan | i want caller speack his name and address and these save in DB |
| 12:51.55 | [TK]D-Fender | SAVE HOW? |
| 12:52.49 | qakhan | i dont know. thats y i am asking here |
| 12:53.35 | [TK]D-Fender | qakhan, Save it in what format? |
| 12:53.52 | qakhan | i want them save in database |
| 12:54.01 | [TK]D-Fender | No, that is WHERE. What FORMAT? |
| 12:54.58 | qakhan | look, caller call in IVR take his name and address and save in database |
| 12:55.08 | [TK]D-Fender | No, that is WHERE. What FORMAT? |
| 12:55.13 | qakhan | i dont want to save in voice format |
| 12:55.44 | [TK]D-Fender | qakhan, there is no voice recognition anywhere near good enough for this to work. It is not going to happen |
| 12:58.14 | qakhan | is there any voice recogniation software which work with * |
| 12:58.40 | [TK]D-Fender | qakhan, All suck. Nothing can handle this kind of request |
| 12:59.01 | [TK]D-Fender | qakhan, It's regrettably a dead-end. Time to pay hemuns to do this cheap for you |
| 12:59.05 | [TK]D-Fender | humans* |
| 12:59.12 | qakhan | hmm |
| 12:59.20 | qakhan | what about sphinx |
| 12:59.23 | [TK]D-Fender | No |
| 12:59.36 | [TK]D-Fender | That has trouble with the alphabet. Forget about "names". |
| 13:00.03 | qakhan | really? |
| 13:00.14 | [TK]D-Fender | qakhan, Game Over. |
| 13:00.18 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 13:00.58 | qakhan | and LumenVox? |
| 13:02.42 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:02.43 | [TK]D-Fender | qakhan, Let me make this completely clear : Forget about the dream of finding a VR that can handle names. Not going to happen. Dead end. Game over. Doesn't exist. |
| 13:03.10 | [TK]D-Fender | qakhan, Accept this and move on to some other productive goal. |
| 13:04.57 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:04.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:05.22 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 13:05.51 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 13:08.48 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:14.47 | Dovid | hi all. is there any way to only send RTP once we get a 200 OK? so if i get a 183 or 180 to send no media? |
| 13:15.46 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kstabftnggqgbwig) |
| 13:15.46 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:17.34 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 13:18.06 | *** join/#asterisk voiper (4426d882@gateway/web/freenode/ip.68.38.216.130) |
| 13:24.23 | *** join/#asterisk OldSmurf (~jens@unaffiliated/oldsmurf) |
| 13:25.06 | schmidts | Dovid i might be wrong but if you get a 183 then you only receive media and didnt sent it. maybe you should try the "r" option of dial, imho this should prevent the playback of early media |
| 13:26.26 | OldSmurf | I am forwarding a call from one Asterisk to another, and adding SIP Header "X-Language". I can see that the header has been set, but I can't seem to be able to retrieve the variable. I am trying: exten => s,n,NoOp("SIP header: ${SIP_HEADER(X-Language)} ") -- What have I missed? |
| 13:28.20 | *** join/#asterisk gavimobile (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 13:29.27 | Dovid | schmidts: Thats not where my issue is. my issue is the pbx that i am calling. |
| 13:30.30 | Dovid | it sends a 180 with SDP. if i send rtp then, it has issues. if i wait for the 200ok and then send the rtp there is no issue |
| 13:31.11 | gavimobile | folks, I need some help making my manual trunk connection. here is my sip.conf and extensions.conf http://pastebin.com/Fw1TDmFZ |
| 13:31.50 | gavimobile | also, im in verbose 5 but I don't see any confirmation of registration |
| 13:34.18 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:36.07 | [TK]D-Fender | gavimobile, Your Dial is very wrong. |
| 13:36.28 | gavimobile | [TK]D-Fender: hi [TK]D-Fender! |
| 13:36.33 | [TK]D-Fender | gavimobile, You've mangled a variable reference and are pointing it to a hostname directly instead of using your peer |
| 13:36.36 | gavimobile | boy am I happy to see you |
| 13:37.01 | gavimobile | [TK]D-Fender: for testing purposes, would this work? exten => _.,1,Dial(SIP/${EXTEN@sip.didlogic.com}) |
| 13:37.08 | [TK]D-Fender | gavimobile, And lack of confirmation of registration is another matter. You should be looking at SIP debug to see what it actualyl happening. |
| 13:37.24 | gavimobile | [TK]D-Fender: oh right, thanks for the reminder |
| 13:37.40 | gavimobile | can't I also use a command like sip show peers for trunks? like sip show trunks |
| 13:37.40 | [TK]D-Fender | gavimobile, No, it will not work. Your syntax is broken. You've inclded things in your variable reference that do not belong. |
| 13:40.01 | [TK]D-Fender | gavimobile, And tried referencing a hostname directly instead of using the peer you created. Go fix all of these and go look at your actual registration attempt |
| 13:40.17 | *** join/#asterisk jkroon (~jkroon@dsl-244-36-150.telkomadsl.co.za) |
| 13:41.00 | gavimobile | [TK]D-Fender: ill try to make sense of what you said and let you know! thanks sir! |
| 13:43.39 | iulhk | using asterisk-10.2.1 how to record every video call ? |
| 13:46.27 | [TK]D-Fender | iulhk, "core show application monitor" |
| 13:48.32 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:50.43 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
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| 14:04.30 | fprior | Hi all, I'm studying "The Definitive Guide", chapter 11 explain Paging. Can me explain an Real Life(TM) situation when I wuold use Paging ? |
| 14:05.02 | *** join/#asterisk Docfxit (~Console1@netblock-75-79-6-10.dslextreme.com) |
| 14:05.35 | [TK]D-Fender | fprior, "John someone is as the front door for you". |
| 14:06.11 | [TK]D-Fender | fprior, "Marketing call parking lot 701" |
| 14:06.23 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:10.13 | fprior | [TK]D-Fender: in this situation, I'm working in a laboratory with another 7 colleagues, and we have only one phone. Is it ? |
| 14:10.13 | *** join/#asterisk zeus (~zeus@gnu/savannah/team/zeus) |
| 14:10.30 | zeus | hi all!! |
| 14:10.51 | [TK]D-Fender | fprior, Is it what? |
| 14:11.10 | zeus | I'm looking for some kind of web app to manage only the sip extensions, does any one knows about something like this ? |
| 14:11.21 | fprior | [TK]D-Fender, is correct my assumption ? |
| 14:11.35 | [TK]D-Fender | fprior, What assumption? |
| 14:12.15 | Docfxit | How can I turn off all 911 calls? |
| 14:13.03 | fprior | [TK]D-Fender: paging is used in situations where in a place are present many persons and only one phone. Pagins is used to send voice message to call one specific person. |
| 14:13.16 | [TK]D-Fender | Docfxit, Change your dialplan to not notch them |
| 14:13.37 | Li7h | fprior, it doesn't make sense to page with less than 2 phones |
| 14:13.42 | [TK]D-Fender | fprior, No, paging can call multiple devices simultaneously |
| 14:14.28 | [TK]D-Fender | Li7h, It can make sense. When you want to make sure 1 person goes to grab the phone and not interrupt everybody |
| 14:14.35 | Docfxit | D-Fender I don't usually use terminal mode. How can I edit the dial plan in terminal mode? |
| 14:14.56 | [TK]D-Fender | Docfxit, You aren't telling us what you are using. |
| 14:15.07 | leifmadsen | fprior: paging is useful in a situation where someone might call, for example, a receptionist, ask for someone, the receptionist parks the call on say extension 701, and then pages to the back "shop" to say, "Jimmy, pick up extension 701" at which point Jimmy walks over to some phone nearby, dials 701 and is connected to the caller. |
| 14:15.15 | Li7h | When there's only one phone? Who would be paging? |
| 14:15.42 | [TK]D-Fender | Li7h, Imagine and IVR that does a parkandannounce |
| 14:16.05 | Docfxit | D-Fender I'm using Asterisk in Ubuntu. |
| 14:16.12 | [TK]D-Fender | Li7h, Use your imagination. If you can't find one, consider drugs :) |
| 14:16.28 | Li7h | I think drugs are what shot out my imagination :) |
| 14:16.42 | [TK]D-Fender | Docfxit, you just said "don't usually use terminal mode". Well how did you configure Asterisk in the first place? |
| 14:16.59 | [TK]D-Fender | Li7h, More and/or better. |
| 14:17.07 | leifmadsen | the only useful example I can think of where you would only have 1 phone and having paging, is an inbound caller dials an extension number, which really just parks the call and then does and auto-page that says something like, "Call for extension 20X in parking space 702" |
| 14:17.20 | [TK]D-Fender | leifmadsen, Just said that ;) |
| 14:17.37 | Docfxit | D-Fender Through Double Commander. |
| 14:17.47 | fprior | ok, I understand now. Another doubt: why use Page and not a simple Dial ? If device is auto-answer will respond to Dial too |
| 14:18.05 | [TK]D-Fender | Docfxit, Guess you'd better ask them. We don't support 3rd party GUI's here |
| 14:18.14 | leifmadsen | fprior: no reason you can't do that |
| 14:18.20 | [TK]D-Fender | friFor when there are muliple devices. |
| 14:18.28 | [TK]D-Fender | fprior, For when there are muliple devices. |
| 14:18.35 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 14:19.03 | [TK]D-Fender | fprior, Page is just Dial+ Meetme. It does not inherently trigger any kind of auto-answer |
| 14:19.06 | fprior | [TK]D-Fender: I can use Dial(SIP/phoneA&SIP/phoneB&SIP/phoneC) |
| 14:19.24 | [TK]D-Fender | fprior, You don't get to talk to them all simultaneously. |
| 14:19.34 | Docfxit | D-Fender Double Commander isn't working right now. Please help me with terminal mode. I have an emergency. The police are at the building right now. The 911 service is being fludded with calls. |
| 14:20.17 | [TK]D-Fender | Docfxit, We don't know what that app does for you. Go ask in their channel or prepare to have to drill around and hope they did something even vaguely sane |
| 14:20.29 | kaldemar | isn't double commander just a file manager? |
| 14:20.43 | [TK]D-Fender | kaldemar, Sounds like Midnight Commander... |
| 14:20.43 | Docfxit | kaldemar Yes. |
| 14:21.16 | Docfxit | Double Commander just manages files. |
| 14:21.21 | [TK]D-Fender | ... if it is a file manager... how the hell is that responsible for having configured *? |
| 14:21.26 | kaldemar | Docfxit: so that's not what you use to configure asterisk, right? are you asking what to use to edit configuration files? |
| 14:21.38 | [TK]D-Fender | kaldemar, No, I am asking that |
| 14:21.54 | [TK]D-Fender | kaldemar, And somehow a text editor app was being framed as a GUI interface. |
| 14:22.16 | Docfxit | kaldemar How can I navagate to the dial plan and edit the dial plan. |
| 14:22.30 | [TK]D-Fender | Docfxit, What built it for you in the first place? |
| 14:22.44 | kaldemar | [TK]D-Fender: i see what you've said. |
| 14:23.30 | Docfxit | D-Fender I have no idea. I paid someone to build it. |
| 14:24.13 | [TK]D-Fender | Docfxit, So you have no idea what you've really got or how it's setup.. and now I'm guessing no clue on how to use * at all.... that about right? |
| 14:28.47 | Docfxit | D-Fender Yes. That's about right. I figured out the editor. What file has the outgoing calling rulles? |
| 14:30.14 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 14:30.18 | [TK]D-Fender | Docfxit, extensions.conf |
| 14:30.43 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:30.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:47.45 | fprior | ...silence, Docfxit was arrested ? |
| 14:49.23 | Docfxit | fprior I'm just fine. |
| 14:49.40 | Docfxit | I have removed 911 from our dial plan. |
| 14:50.10 | Docfxit | Now I will get into the logs to find out where that came from |
| 14:50.23 | Docfxit | Thank you for the help. |
| 14:50.33 | fprior | Docfxit, good ! |
| 14:50.44 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
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| 15:31.02 | p3nguin | docfxit: If you find out that you have to pay someone to unbuild it for you, too, I have availability. |
| 15:33.51 | Docfxit | p3nguin What do you mean by unbuild? |
| 15:35.06 | p3nguin | <Docfxit> D-Fender I have no idea. I paid someone to build it. <--- I meant I would fix or undo whatever the other person did that was not to your liking. |
| 15:36.14 | Docfxit | p3nguin Thanks. I'll keep that in mind. |
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| 15:39.59 | *** join/#asterisk Rienzilla (rien@sinas.rename-it.nl) |
| 15:40.33 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 15:41.59 | Rienzilla | Hey everyone. I have an asterisk server for a company, and - amongst other things - it has a calling queue which has members that are on a cellphone. Now the issue is that I want the persons that pick up the call to know both who is calling (by the CID), but also that the call was forwarded from the company pbx instead of placed directly to the cellphone (so the user may say something else when he picks up). How can I do this? For example, can I play a |
| 15:42.05 | *** join/#asterisk stegbth (~stegbth@2001:6f8:9fa:0:221:70ff:fea9:b5e2) |
| 15:42.13 | stegbth | hello everybody |
| 15:42.47 | stegbth | i am running the "latest" version of trixbox (2.8) |
| 15:43.49 | stegbth | does there exist a possibility to change the destination of a incoming route via phone shortdial? |
| 15:44.02 | stegbth | if yes, does there exist an howto? |
| 15:45.32 | p3nguin | rienzilla: You truncated at "can I play a". |
| 15:45.43 | Rienzilla | oh |
| 15:45.47 | Rienzilla | ... For example, can I play a sound file to the call recipient before the call is passed through? |
| 15:45.55 | p3nguin | But the answer is, "Yes, you can play a message." |
| 15:46.19 | Rienzilla | hehe ok how would I do that? |
| 15:47.35 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 15:47.51 | p3nguin | In your Dial, you can use the appropriate option for a macro or a sub, and when the callee picks up, the subroutine will execute. In the subroutine, you'd use Playback() to play a sound file or BackGround() to play a sound file and allow it to be interrupted by DTMF. |
| 15:48.10 | [TK]D-Fender | Never use Background() there... |
| 15:48.11 | p3nguin | That's one way. |
| 15:48.56 | p3nguin | It would be extremely similar to my call screening macro. |
| 15:49.23 | Rienzilla | ok, I can probably make that work |
| 15:49.25 | Rienzilla | thanks |
| 15:49.25 | p3nguin | "You have a caller... press 1 to accept this call, press 2 or hangup to reject it. |
| 15:49.34 | p3nguin | " |
| 15:49.44 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:49.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:50.31 | MarkS- | Hello, is information regarding MeetMe somewhere available? I'm looking for a way to setup the option of conference calls and MeetMe seems to be the best option (but somehow I can't get it to work, it gives errors regarding DAHDI and for as far as I can see everything is correctly configured/installed). Some information from the asterisk cli: http://yourpaste.net/10987/ |
| 15:51.17 | anonymouz666 | p3nguin: sounds useful, will do something like that for my extension |
| 15:52.21 | p3nguin | marks-: Install dahdi and load the dahdi kernel module. |
| 15:52.53 | p3nguin | Dahdi is required for meetme to work. |
| 15:53.23 | p3nguin | Until you can run "dahdi show channels" on the asterisk CLI and see the pseudo channel, it isn't ready for meetme. |
| 15:54.33 | MarkS- | p3nguin: looking at that now, I did already install dahdi kernel modules (using apt-get at debian) |
| 15:54.48 | p3nguin | Is it loaded? lsmod|grep dahdi |
| 15:56.03 | MarkS- | it isn't loaded, looking in to that at the moment |
| 15:59.01 | p3nguin | modprobe dahdi |
| 16:03.42 | MarkS- | the dahdi module isn't available, currently looking in building it from source |
| 16:04.15 | voiper | does anyone know how to send userid and password as part of a SIP request in asterisk 1.4 ? I was able to make it work with 1.6 by sending SIP/exten:password::userid@host. The same is not working in 1.4 |
| 16:04.25 | p3nguin | Someone else ran into this yesterday on Ubuntu. I thought the module should be packaged like all other softwares. |
| 16:06.24 | Docfxit | In Master.csv the times look like GMT. If I change cdr.conf usegmtime=yes to usegmtime=no will it use the local time in the log? |
| 16:06.37 | MarkS- | p3nguin: it should, but somehow it isn't (anymore) |
| 16:09.54 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 16:10.50 | [TK]D-Fender | voiper, Make a proper peer |
| 16:17.20 | voiper | i would like to send the username password dynamically using an agi instead of using the one from peer |
| 16:17.57 | p3nguin | Why would anyone want to use something that isn't well-configured? |
| 16:18.48 | p3nguin | There's a saying I have a tendency to use around my office: Do it right or don't do it at all. |
| 16:25.55 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 16:34.32 | p3nguin | If a call between two asterisk systems seems to stop transmitting audio after 15 minutes, that isn't going to have anything to do with session timers, is it? |
| 16:35.11 | p3nguin | The channel does not go away; it just doesn't have any audio. |
| 16:35.32 | p3nguin | As a matter of fact, the channel NEVER goes away. And then when I try to kill it, it becomes a zombie. |
| 16:36.31 | gavimobile | [TK]D-Fender: I got it connected! |
| 16:36.43 | [TK]D-Fender | gavimobile, Glad to hear. |
| 16:36.48 | gavimobile | [TK]D-Fender: thanks |
| 16:37.10 | *** join/#asterisk hff135_ (~hff135@64-201-203-155.sktn.hsdb.sasknet.sk.ca) |
| 16:37.16 | *** part/#asterisk gavimobile (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 16:37.34 | p3nguin | Maybe it's a problem in 1.8.10.0, because I never had this issue before. |
| 16:38.28 | hff135_ | the asterisk 1.8 sip.conf docs say the choices for 'directmedia' are 'yes', 'nonat', 'update'. so why does 'no' work, and what is that setting? |
| 16:39.36 | p3nguin | The directmedia setting controls if media passes directly between two end points or if it has to pass through asterisk. |
| 16:39.56 | p3nguin | directmedia=no prevents reinvites, keeping asterisk in the media path. |
| 16:40.11 | hff135_ | is directmedia=no same as directmedia=nonat? |
| 16:41.18 | p3nguin | No, but it's similar. nonat allows media to go directly between end points that asterisk determines are within the same network and do not involve nat. If nat is determined to be between the two end points, asterisk will remain in the path. nonat is the setting I prefer. |
| 16:43.19 | p3nguin | In addition to the nonat value for directmedia, I also configure the directmediadeny and directmediapermit settings. |
| 16:50.12 | hff135_ | i'm having a strange issue. we recently upgraded some customers from 1.6.2 to 1.8.9.2. we have canreinvite=no in sip.conf. calls work properly except in the case where a caller calls into a queue and then the queue member answers and doesn an attended transfer off to another employee. often, there is one-way audio (caller can hear employee, employee cannot hear caller) |
| 16:55.22 | hff135_ | if i want to avoid the one-way audio issues entirely, is directmedia=no the best way to go? |
| 16:56.32 | p3nguin | Sometimes. |
| 16:56.53 | p3nguin | The directmedia parameter replaced the older canreinvite parameter. |
| 16:57.32 | *** join/#asterisk wonderworld (~ww@dsdf-4db52b03.pool.mediaWays.net) |
| 16:58.00 | p3nguin | I think in most, in not all, cases, a reinvite through a nat will result in no audio. |
| 16:58.47 | p3nguin | You're having one-way audio, which seems to be a problem with the RTP address of an end point. Check your SDP messages on an affected call in sip debug. |
| 17:01.43 | hff135_ | this wasn't a problem on 1.6.2 tho. we have several thousand phones. none of them complained about this problem until we switched them to asterisk 1.8.9.2. the config stayed the same (canreinvite=no). do u know if the directmedia/canreinvite logic changed between the two versions? |
| 17:02.23 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:15.17 | *** join/#asterisk gavimobile (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 17:15.58 | gavimobile | silly question, if I want to add my country code automatically to my dialplan, what would I do? exten => _0NNXXXXXX,1,Answer() |
| 17:15.58 | gavimobile | <PROTECTED> |
| 17:16.25 | gavimobile | as of now, I need to dial my country code, but I don't want to. I want the dial plan to add it for me |
| 17:17.05 | [TK]D-Fender | gavimobile, Then add them directly into your dial |
| 17:17.28 | gavimobile | [TK]D-Fender: I did that with my example, but it doesn't seem to be working |
| 17:18.17 | gavimobile | en example number would be 025005303 but with my new trunk, I need to dial 97225005303 |
| 17:18.17 | p3nguin | What number do you have to dial now on your phone's keypad? What number do you want to dial on your phone's keypad to make the same call with less numbers? |
| 17:18.17 | [TK]D-Fender | gavimobile, Your example isn't a Dial(), and I'm wondering if you're even looking at what you're doing to the exten that was dialed... |
| 17:18.19 | gavimobile | I want to dial the first number p3nguin |
| 17:18.33 | gavimobile | [TK]D-Fender: correct, im testing with saydigits |
| 17:18.34 | p3nguin | 972${EXTEN:1} |
| 17:18.48 | gavimobile | so I can hear what in the variable. |
| 17:19.16 | p3nguin | 972${EXTEN:1} <------ prepend 972 to the dialed number after removing the original first digit. |
| 17:19.43 | gavimobile | p3nguin: nope |
| 17:19.47 | p3nguin | Nope what? |
| 17:20.02 | gavimobile | its still requiring the 972 |
| 17:20.14 | gavimobile | exten => _0NNXXXXXX,1,Answer() |
| 17:20.15 | gavimobile | <PROTECTED> |
| 17:20.16 | p3nguin | That's not my fault. |
| 17:20.17 | [TK]D-Fender | gavimobile, Show us your actual dial |
| 17:20.58 | p3nguin | If you dial something that matches 0NNXXXXXX, 972${EXTEN:1} will turn it into 972 NNXXXXXX. |
| 17:21.19 | gavimobile | [TK]D-Fender: im in cli, asterisk -rvvvvv and I did sip set debug on |
| 17:21.28 | gavimobile | but when I dial I see nothing in the cli |
| 17:21.33 | [TK]D-Fender | gavimobile, I dd not ask for a story |
| 17:21.42 | [TK]D-Fender | gavimobile, where is youre REAL dialplan line? |
| 17:21.52 | gavimobile | [TK]D-Fender: the pastebin? |
| 17:21.54 | [TK]D-Fender | gavimobile, "nope" doesn't help us |
| 17:22.37 | [TK]D-Fender | gavimobile, Show us the actual dialing extension. |
| 17:22.53 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 17:23.21 | gavimobile | im lost |
| 17:23.24 | gavimobile | :-( |
| 17:23.28 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-060.mobile.uci.edu) |
| 17:23.40 | [TK]D-Fender | gavimobile, Show us your actual dialplan |
| 17:23.51 | gavimobile | [TK]D-Fender: from extensions.conf? |
| 17:23.58 | p3nguin | That's where dial plan is. |
| 17:24.01 | [TK]D-Fender | Where else do you have dialplan? |
| 17:24.07 | gavimobile | p3nguin: few |
| 17:24.08 | gavimobile | ok |
| 17:24.12 | gavimobile | phew* |
| 17:24.48 | gavimobile | http://pastebin.com/rgcKHZE1 |
| 17:25.19 | kaldemar | deja vu. |
| 17:25.24 | p3nguin | Dial a number that matches 0NNXXXXXX. |
| 17:25.40 | kaldemar | gavimobile: ${EXTEN:1:8@sip.didlogic.net} <--- something wrong with this. what could it be? |
| 17:26.03 | gavimobile | kaldemar: that is behind a comment |
| 17:26.07 | gavimobile | ; |
| 17:26.13 | p3nguin | It's wrong, though. |
| 17:26.30 | kaldemar | gavimobile: that's one of the three things about it you should change. |
| 17:26.31 | gavimobile | yes its wrong, cause im testing first with playdigits |
| 17:26.42 | gavimobile | kaldemar: thanks for the kint |
| 17:26.43 | gavimobile | hit |
| 17:26.46 | gavimobile | hint* |
| 17:26.47 | p3nguin | Create a peer for didlogic. Then Dial(SIP/didlogic/972${EXTEN:1}) |
| 17:27.36 | gavimobile | p3nguin: hrm |
| 17:28.00 | gavimobile | a peer you say |
| 17:28.03 | gavimobile | :-p |
| 17:28.13 | gavimobile | alright, you guys helped me enough. be back soon! |
| 17:28.13 | p3nguin | Sounds like a new concept to you. |
| 17:28.14 | gavimobile | thanks |
| 17:28.21 | gavimobile | p3nguin: ohhh yes |
| 17:32.25 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 17:34.22 | [TK]D-Fender | gavimobile, I told you all of this repeatedly hours ago |
| 17:35.15 | gavimobile | [TK]D-Fender: :-( sorry |
| 17:38.01 | p3nguin | I have a feeling he'll ask again in another four hours. |
| 17:38.17 | *** join/#asterisk timahvo1 (~rogue@41.80.23.207) |
| 17:38.18 | p3nguin | Even though he has the answer already. Twice. |
| 17:40.36 | [TK]D-Fender | more than |
| 17:49.13 | jaytee | when you type "sip show peers" and you see an N in the NAT column next to a peer is that indicating that asterisk thinks the peer is behind a NAT? |
| 17:49.31 | leifmadsen | jaytee: it indicates you've configured the peer with nat=yes |
| 17:49.41 | jaytee | but I haven't |
| 17:49.45 | leifmadsen | you have somewhere |
| 17:49.53 | leifmadsen | either in general, or with some other option |
| 17:49.57 | jaytee | hmmm, ok will double check |
| 17:50.01 | jaytee | thanks |
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| 18:10.27 | p3nguin | That value doesn't necessarily mean anything about the actual position of the peer. |
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| 18:52.14 | voiper | Hi, how do we force asterisk to take rtp? making both sip peers to canreinvite=no, directrtpsetup=no, nat=yes didn't do the trick. It is asterisk 1.6 |
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| 18:54.24 | p3nguin | directmedia=no keeps asterisk in the media path. |
| 18:55.03 | voiper | thanks p3nguin. I have that set too |
| 18:55.06 | voiper | <PROTECTED> |
| 18:55.35 | voiper | if asterisk is taking rtp we should see the packets by doing rtp set debug on ? |
| 18:55.39 | p3nguin | directmedia replaces canreinvite. |
| 18:59.08 | p3nguin | In other words, don't use both. |
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| 19:04.12 | devdvd | hi all, using asterisk 1.6.2.9. Is there any way in this version (or even in later versions) to specify the order asterisk looks for queue members. For example, right now ordering queue members the way i want is a complete pain. say for example i have Bob(Local/1@phones)>Sally(Local/2@phones)>John(Local/3@phones)>Jim(Local/4@phones) in queue support in that order. Right now as you know, asterisk will order it bob>sally>john>jim b |
| 19:04.12 | devdvd | ecause it orders the select by the interface. What i want to do is know if theres a way to change that behavior. Idealy id like to be able to add a new field called priority and sort on that but id even be willing to sort on membername. The ultimate goal is to be able to move the order people receive calls easily without having to modify what extension they are at. so say i bob has the day off and i want Jim to receive calls befo |
| 19:04.13 | devdvd | re anyone else just for today. right now I have to modify jim's extension to make sure the query sees it first. Is there a better way? |
| 19:05.20 | devdvd | i am using realtime (mysql) for the queue members |
| 19:06.01 | p3nguin | devdvd: In 1.8, you can use the linear strategy to use them in the order they are listed in the config file. |
| 19:06.46 | Chainsaw | And I can say firsthand that upgrading from 1.6.2 to 1.8 is not particularly disruptive. |
| 19:07.13 | p3nguin | Even from 1.4 to 1.8 is pretty easy. |
| 19:07.35 | Chainsaw | p3nguin: 1.2 to 1.6.0 was a bit of a pain though. |
| 19:08.43 | MarkS- | FYI: earlier today I mentioned an issue with getting dahdi working, with the debian package I get the error listed at http://permalink.gmane.org/gmane.linux.debian.devel.bugs.general/915825 and now I'm building it from source |
| 19:09.39 | jaytee | I have 4 peers that are on the internal LAN (single subnet) and a peer definition for Flowroute in my sip.conf. I have nat=yes for Flowroute and I don't have nat= set anywhere else in my sip.conf yet the 4 peers on the LAN show as N under the nat column when I type sip show peers. |
| 19:12.48 | p3nguin | Doesn't matter. |
| 19:13.10 | p3nguin | Having nat=yes set generally does not magically turn local peers into natted peers. |
| 19:13.36 | p3nguin | And also, flowrouter is NOT behind nat, so that's not a good setting. |
| 19:14.44 | p3nguin | If you support any peers behind NAT, set nat=yes in the general section and don't set it anywhere else in the entire file. If you do not support ANY peers behind a NAT and Asterisk is also not behind a NAT, then you can set nat=no in general section and don't set it anywhere else in the file. |
| 19:15.04 | p3nguin | jaytee: That's for you. |
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| 19:16.59 | hff135_ | devdvd: we're using RT queue members |
| 19:17.07 | hff135_ | there is no way to order the queue members |
| 19:18.35 | devdvd | p3nguin, i need to be able to use the database though. Because i need to allow people t pause and unpause themselves (they aren't using locally connected phones)...its just extensions linked to outside phone numbers |
| 19:18.47 | devdvd | hff135_, thats what im finding, which is why i came to ask the experts :) |
| 19:19.31 | hff135_ | i think we got around this by name the local channels specially |
| 19:19.32 | hff135_ | jon (Local/000__jon@queue_calling/n) (realtime) (paused) (Not in use) has taken no calls yet |
| 19:19.32 | hff135_ | sally (Local/001__sally@queue_calling/n) (realtime) (Unavailable) has taken no calls yet |
| 19:19.32 | hff135_ | rick (Local/008__rick@queue_calling/n) (realtime) (Unavailable) has taken no calls yet |
| 19:21.56 | devdvd | but then how do you deal with changes? that pretty much looks like what im doing now |
| 19:22.56 | voiper | p3nguin i tried all the options i still don't see rtp going through asterisk. |
| 19:23.38 | jaytee | p3nguin, thank you for the info. I thought I needed nat=yes for flowroute as there is a firewall between my Asterisk system and we're using NAT to redirect SIP and RTP. |
| 19:24.22 | p3nguin | That isn't what the nat parameter is for. |
| 19:24.28 | hff135_ | devdvd: when we encountered this problem, our channels were Local/jon@queue_calling, etc |
| 19:24.32 | jaytee | I've been using pretty much the same config for over a year now on several systems but for some reason this one want's to show all the peers on the same lan as N |
| 19:24.48 | hff135_ | to fix the problem, we had to introduce the numbering at the beginning |
| 19:24.53 | p3nguin | It's fine. It doesn't matter if they show N or not. |
| 19:24.56 | hff135_ | Local/xxx__jon@queue_calling |
| 19:25.00 | devdvd | right, mine are 2XXX |
| 19:25.04 | p3nguin | It will not magically make them behind a NAT. |
| 19:25.05 | leifmadsen | jaytee: did you recently upgrade? |
| 19:25.16 | devdvd | so Local/2800, 2801 etc |
| 19:25.18 | p3nguin | Stop worrying over something that isn't a problem. |
| 19:25.22 | gavimobile | wow, my mistake was sooo silly |
| 19:25.29 | devdvd | so it sounds to me that we're doing the same thing |
| 19:25.51 | p3nguin | Current asterisk version defaults to force_rport if available, or yes if the previous is not available. |
| 19:25.55 | jaytee | p3nguin, yeah at least it's registering ok. I can call in but where I was getting two way audio a week ago they must have changed something on the firewall because now I get retransmit on critical packet errors. |
| 19:26.10 | p3nguin | If you do not have any devices behind a NAT, set nat=no in the general section and move on. |
| 19:27.47 | p3nguin | If your asterisk is behind a NAT, which you indicated it is, you can set nat=yes in the general section and don't worry about the fact that you don't have any phones behind other NATs. |
| 19:28.15 | p3nguin | The phones which are not behind other NATs will not magically get a NAT server installed between them and asterisk. |
| 19:28.25 | p3nguin | Therefore, the N is okay. |
| 19:28.48 | p3nguin | Correctly set your localnet value and the rest should be history. |
| 19:28.56 | hff135_ | devdvd: are those numbers (2800, 2801) the extensions numbers? |
| 19:32.35 | devdvd | yea, in my dialplan i have a context called queue_external and in that i have stuff like 2800,1,Dial(SIP/${OUTTRUNK}/5555555555) then in my database queue members table i have Local/2800@queue_external as the interface name |
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| 20:03.40 | hff135_ | i'm having a problem with one-way audio. we recently upgraded from 1.6.2 to 1.8.9. all phones behind NAT. pstn_caller calls into queue. problem does not occur all the time. phone1 answers and then does attended transfer to phone2. phone2 cannot hear pstn_caller but pstn_caller can hear phone2 |
| 20:03.48 | hff135_ | one-way audio occurs only when pstn_caller calls phone1 via queue. if pstn_caller calls phone1 directly, problem does not occur. |
| 20:03.54 | hff135_ | problem does not occur for blind transfer. it also does not occur for attended transfer where phone1 transfers the call without waiting for phone2 to answer. it only occurs when phone1 attended transfers to phone2 and waits for phone2 to answer. |
| 20:04.26 | hff135_ | i have compared the REFER from the one-way audio case to the two-way audio case. there doesn't seem to be much difference. |
| 20:04.52 | hff135_ | i've been working on this for days but i'm not making much progress. can someone point me in the right direction? |
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| 20:13.01 | MarkS- | p3nguin: thanks for helping earlier today, it works at this moment (investigating other nice options to offer to clients using asterisk) |
| 20:16.32 | [TK]D-Fender | checkout time, later all |
| 20:21.56 | hff135_ | anyone? |
| 20:34.38 | woleium | I'm trying to play an announce to all active, on a call queue members when a new call enters the queue. |
| 20:34.49 | urvg4 | hi all,having voice breaks during calls.Any way to fix this? |
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| 20:35.17 | woleium | I've looked through the docs, but I'm a bit confused. Where should I be looking? |
| 20:35.28 | Docfxit | Someone got into our switch remotely and figured out how to dial out. One of the numbers they were trying was 91122455202180. Our dial plan dialed out to 911. There were many numbers they tried. They all started with 911. I'm looking at the master.csv. How can I see how they got into the system? |
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| 20:36.21 | pabelanger | Docfxit: review syslogs |
| 20:36.33 | pabelanger | and setup a firewall and not put your asterisk server on the public web |
| 20:36.43 | woleium | Docfxit: it's tricky, and they amy have installed a backdoor. You should reinstall. :-( |
| 20:37.26 | woleium | in the meantime change all your passwords, install fail2ban and hook it into asterisk |
| 20:37.41 | Docfxit | pabelanger Where can I find the syslogs? |
| 20:37.53 | woleium | what distro are you using? |
| 20:37.56 | pabelanger | /var/logs? |
| 20:40.19 | kaldemar | Docfxit: do you have some actual information on the "got into our switch" part? as in do you know that your switch system is compromised or did someone just make calls? |
| 20:41.25 | kaldemar | it i possible to leave a system open for unauthenticated calls, so that can happen without anyone actually breaking in to the box. |
| 20:44.44 | hff135_ | i have directmedia=no. but i'm getting one-way audio on transferred calls. these are not re-invites. they are REFERs. what would cause this? |
| 20:50.44 | Docfxit | kaldemar No one was in the building. The police showed up at the door this morning saying someone from this address was fludding the 911 system with phone calls. |
| 20:51.28 | kaldemar | Docfxit: by breaking in i don't mean physically. |
| 20:52.27 | Docfxit | kaldemar How ever it was done, I need to figure out how to stop it. And fast. |
| 20:57.58 | Docfxit | pabelanger I have one file called syslog that only shows after I rebooted the machine. I have syslog.0 that shows today after 7am. The next syslog.1.gz is from March 7. The problem happened today at 5:25am. |
| 20:58.53 | pabelanger | Docfxit: this is not asterisk specific but OS support. Your best to seek support from your OS (eg: Ubuntu / Fedora) |
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| 20:59.51 | hff135_ | did anyone see my messages from earlier? |
| 21:00.26 | Docfxit | If this came in from the internet how can I record an IP address in Asterisk for each call? |
| 21:02.26 | kaldemar | Docfxit: there is no one correct answer to any part of this. but look at /var/log/asterisk/cdr-csv/Master.csv if it exists in your system, it is the default place where call data records are saved in. |
| 21:04.30 | Docfxit | kaldemar I'm looking at that file. I see all the calls. There is nothing in collum A or B. The numbers start in collum C which I think is the number it's calling. I'd like to record the number it came from. |
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| 21:05.54 | dubberl | I have a question about NoOp(). |
| 21:06.35 | Docfxit | If I recorded caller ID from blocked numbers maybe I would see that info. |
| 21:07.05 | cusco | dubberl: just ask |
| 21:07.10 | dubberl | We have a .agi that is in perl tha runs and it has $AGI->noop("something") so we can debug. But when I set verbose to at least 3 I don't see anything when the script runs. Am I missing something? |
| 21:07.24 | cusco | no |
| 21:08.30 | Docfxit | I have to go now. I'll be back. |
| 21:08.36 | dubberl | So it should work? |
| 21:08.50 | dubberl | I connect in via asterisk -rvvv |
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| 21:20.51 | dubberl | Any ideas? |
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| 21:38.39 | urvg4 | having voice breaks during calls.Any way to fix this? |
| 21:39.48 | urvg4 | using a 100Mps link from the data center where * is hosted |
| 21:39.59 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 21:40.18 | urvg4 | but my peak traffic been around 290KB/s |
| 21:40.21 | Steel_Reign | can anyone tell me why i am seeing this error |
| 21:40.28 | Steel_Reign | chan_sip.c:25545 check_rtp_timeout: Disconnecting call 'SIP/miahav305-0000003d' for lack of RTP activity in 31 seconds |
| 21:41.49 | Steel_Reign | i can make from from the remote server to my local server and it works fine but i get no audio and this error trying to call the remote server |
| 21:42.05 | Steel_Reign | make calls* |
| 21:42.35 | kaldemar | ~sipnat |
| 21:42.35 | 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
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| 22:17.22 | dubberl | urvg4: what is your latenancy at? |
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| 22:18.15 | urvg4 | how do I measure that? |
| 22:18.27 | dubberl | Just run a ping. |
| 22:18.40 | dubberl | get the MS you have. |
| 22:18.52 | dubberl | If its really laging then that may be an issue. |
| 22:18.56 | dubberl | Or a dealy in talk. |
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| 22:19.50 | urvg4 | ok |
| 22:19.52 | urvg4 | one sec |
| 22:23.57 | urvg4 | dubberl:seems iptables is blocking pings |
| 22:24.21 | urvg4 | even localhost did not respond |
| 22:25.16 | dubberl | Hmm I always leave ICMP packets on helps to troubleshoot. |
| 22:25.19 | dubberl | But thats just me. |
| 22:30.23 | urvg4 | do I disable the iptables? |
| 22:30.53 | dubberl | I don't think that would be good if its doing NAt and such. |
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