| 00:06.46 | *** part/#asterisk DigitalDaz21 (n=digitald@78-105-169-114.zone3.bethere.co.uk) |
| 00:19.21 | raden | whenn i call our 800 # our provider call centric just has a message the customer your trying to reach is unavailable |
| 00:19.32 | raden | call centric web OS says phone is registered |
| 00:19.43 | raden | Name/username Host Dyn Nat ACL Port Status |
| 00:19.44 | raden | 101/101 192.168.1.101 5060 OK (58 ms) |
| 00:19.44 | raden | callcentric/17772445766 204.11.192.37 5080 Unmonitored |
| 00:19.53 | raden | im getting more confussed |
| 00:28.17 | ManxPower | Is your Asterisk server behind NAT? |
| 00:30.04 | raden | yes |
| 00:30.18 | raden | as in a router with NAT ? |
| 00:30.28 | raden | should i put it in DMZ zone ? |
| 00:31.21 | raden | as far as phone setting |
| 00:31.36 | raden | proxy should be pointing at asterisk correct |
| 00:31.48 | ManxPower | turn on qualify=yes or the NAT router is likely to close the NAT translation and the incoming call will fail. |
| 00:31.48 | raden | and registrar should be as well ? |
| 00:32.08 | raden | ManxPower, ok one moment |
| 00:33.21 | raden | 1 SIP registrations. |
| 00:33.21 | raden | [Jul 27 19:32:25] NOTICE[27624]: chan_sip.c:19828 sip_poke_noanswer: Peer 'callcentric' is now UNREACHABLE! Last qualify: 0 |
| 00:33.21 | raden | [Jul 27 19:32:25] NOTICE[27624]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '17772445766' rejected because extension not found. |
| 00:33.21 | raden | [Jul 27 19:32:35] NOTICE[27624]: chan_sip.c:15604 handle_response_peerpoke: Peer 'callcentric' is now Reachable. (56ms / 2000ms) |
| 00:33.52 | ManxPower | looks like you don't have an extension 17772445766 |
| 00:36.58 | raden | this is what callcentric said i needed todo here is my sip.conf file |
| 00:36.59 | raden | http://pastebin.com/m5992df53 |
| 00:37.37 | raden | i have a book , a asterisk pdf download, my phone manual, and manual from callcentric all saying different things :( |
| 00:38.23 | raden | now my phone for the first time ever says NO SERVICE |
| 00:38.39 | ManxPower | I've never seen a provider that was correct in what they told people to set up for Asterisk |
| 00:38.56 | raden | that makes me feel better |
| 00:38.58 | ManxPower | is the 1777 number your DID or the callerid of the calling phone |
| 00:39.19 | raden | thats our callcentric DID also our login |
| 00:39.31 | raden | but there a 888 number attached to the account |
| 00:39.48 | raden | 18664523565 |
| 00:40.28 | ManxPower | And you have an exten => 17772445766,1,whatever in your extensions.conf? |
| 00:40.41 | raden | nope ill paste it on pastebin |
| 00:40.57 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 00:40.59 | ManxPower | Then the error message is correct. |
| 00:41.27 | raden | http://pastebin.com/m2fdc71e5 |
| 00:41.49 | ManxPower | exten s is NOT a wildcard. |
| 00:42.16 | ManxPower | exten s is for when there was NO DESTINATION number received. The destination of the call is 17772445766 so it will never ever match "s" |
| 00:42.37 | raden | i gotcha |
| 00:42.43 | raden | so what would be proper |
| 00:42.49 | ManxPower | I call "s" the (s)tupid extension. As in the calling device is too stupid to send the dialed numbner. |
| 00:42.53 | raden | http://www.callcentric.com/support/device/asterisk |
| 00:42.59 | raden | ^^^ thats what they told me todo |
| 00:43.01 | ManxPower | what is proper is exten => 17772445766 |
| 00:43.10 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 00:43.25 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 00:43.38 | ManxPower | As I said, I never saw a provider that got the docs correct for Asterisk. |
| 00:43.49 | ManxPower | in fact if you set it up like they tell you it is guarnteed to not work. |
| 00:44.37 | raden | ok it got my phone to ring |
| 00:44.45 | raden | but when i pickup the phone keeps ringing |
| 00:44.55 | ManxPower | "s" is used in macros, when immediate=yes, analog FXO ports, and CAS-T1. |
| 00:45.06 | raden | that makes sense |
| 00:45.10 | ManxPower | set canreinvite=no in your sip.conf. can't reinvite when you are behind NAT |
| 00:45.44 | raden | <PROTECTED> |
| 00:47.10 | ManxPower | putting it in the DMZ doesn't really help much. |
| 00:47.11 | raden | on my phone my proxy should be my asterisk server Correct ? |
| 00:47.22 | raden | ok no problem then i added that |
| 00:47.39 | raden | as well as the registrar server ? |
| 00:47.45 | ManxPower | generally yes, phones think of Asterisk as a proxy. Never make the mistake that Asterisk is actually a proxy. It is not. |
| 00:47.57 | ManxPower | But phones are fine thinking it is. |
| 00:47.57 | raden | ok.. |
| 00:53.29 | *** join/#asterisk Kumbang (n=queenbee@rusnas.paume.itb.ac.id) |
| 00:53.43 | raden | here is where im getting hung on phone config |
| 00:53.43 | raden | http://www.voltarclamps.com/sip.png |
| 00:54.52 | ManxPower | other than the backups, set everything to the ip of the asterisk server, make sure you set canreinvote=no in sip.conf |
| 00:55.06 | raden | ok |
| 00:55.26 | ManxPower | leave the ports at 0, that is normally a default to 5060 |
| 00:55.39 | raden | when i dial in right now i can ring the phone but when i pickup the phone it keeps ringing the phone and ringing on the phone im dialing from |
| 00:57.24 | ManxPower | sounds like canreinvite=no is not set |
| 00:57.42 | raden | i did set it in sip.conf |
| 00:57.52 | raden | restarted sip and restarted diaplan since then |
| 00:58.01 | ManxPower | in each [section] ? |
| 00:59.09 | raden | no not in each one let me try that |
| 00:59.41 | ManxPower | I don't specifically remember if it works in [general] or not. |
| 01:00.05 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:01.55 | raden | hmmm doing same thing |
| 01:02.08 | raden | when i dial out on my phone i keep getting a busy signal |
| 01:02.17 | raden | and my phone flashing saying NO SERVICE after i set all that |
| 01:04.22 | raden | im confussed how its ringing through i pickup the line on the phone and it keeps ringing |
| 01:07.35 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:08.55 | ManxPower | make sure you have disallow=all and allow=ulaw in sip.conf |
| 01:09.03 | ManxPower | you can screw with codecs later |
| 01:09.25 | jaytee | but you can't dance with them! |
| 01:10.09 | raden | why would my phone say NO SERVICE ON DISPLAY ? |
| 01:10.20 | raden | also its a off the hook tone i get when i try to dial out after dialing |
| 01:10.48 | jaytee | is it registering to Asterisk? |
| 01:11.02 | ManxPower | if it can't agree on a codec or picks a codec asterisk does not support you won't get anything. |
| 01:11.34 | raden | [Jul 27 20:10:28] NOTICE[27624]: chan_sip.c:17295 handle_request_invite: Call from '101' to extension '18888264556' rejected because extension not found. |
| 01:11.52 | raden | ttssusa*CLI> sip show peers |
| 01:11.52 | raden | Name/username Host Dyn Nat ACL Port Status |
| 01:11.52 | raden | 101/101 192.168.1.101 5060 OK (56 ms) |
| 01:11.52 | raden | callcentric/17772445766 204.11.192.31 5080 OK (59 ms) |
| 01:12.34 | ManxPower | sip show peers has NOTHING to do with extensions. |
| 01:12.57 | raden | he was asking registering |
| 01:13.38 | ManxPower | whatever context= you have in sip.conf for that device had better have a corrosponding exten line in that context in extensions.conf. |
| 01:13.39 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9ea4a56ed5137034) |
| 01:14.42 | ManxPower | in fact the one thing that links sip.conf with extensions.conf is the context= in sip.conf for that device and the [context] in extensions.conf |
| 01:15.04 | *** join/#asterisk AlmightyOatmeal (n=jamie@66-190-59-171.dhcp.mdsn.wi.charter.com) |
| 01:15.28 | ManxPower | The above statement may be the most important thing you have learned about Asterisk today |
| 01:15.41 | AlmightyOatmeal | i'm trying to configure a polycom soundpoint ip 301 phone and when it connects to my asterisk box it says "no matching peer found" though i added a context for it in my sip.conf |
| 01:15.44 | AlmightyOatmeal | :( |
| 01:16.14 | ManxPower | AlmightyOatmeal: see my polycom config examples: http://www.fnords.org/~eric/polycom-config-examples |
| 01:16.46 | AlmightyOatmeal | ty |
| 01:17.35 | ManxPower | I set up polycoms sip userid as the MAC address + a letter for the line appearance (I register each line seperatly) |
| 01:18.26 | *** join/#asterisk PhunTelTek (n=PhunTelT@cpe-76-188-233-188.neo.res.rr.com) |
| 01:18.35 | AlmightyOatmeal | hmm, then i dont get why asterisk says no matching peer |
| 01:19.11 | AlmightyOatmeal | http://pastebin.ca/1509534 |
| 01:19.17 | AlmightyOatmeal | thats my sip.conf context |
| 01:20.22 | AlmightyOatmeal | and i set my line 1 auth name and password right in the web UI |
| 01:20.45 | ManxPower | and the authname = polycom |
| 01:21.23 | AlmightyOatmeal | of course |
| 01:21.44 | ManxPower | screenshot the polycom config and put it somewhere. |
| 01:22.02 | AlmightyOatmeal | auth user id = polycom, auth password = llamasex, as is in my context |
| 01:22.31 | AlmightyOatmeal | thats a lot of screenshots heh |
| 01:22.38 | ManxPower | no, just the line config screen |
| 01:22.45 | AlmightyOatmeal | ah |
| 01:22.48 | AlmightyOatmeal | 2 sec |
| 01:23.10 | ManxPower | It's been like 2 years since I've used a polycom web config, I always did the config file method |
| 01:23.51 | AlmightyOatmeal | someone just sent me a 301 and i was eager to get it fired up but don't understand why its being such a pain :P |
| 01:24.01 | ManxPower | you need the reg address AND the regid set to the userid |
| 01:24.07 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:24.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:24.31 | ManxPower | regid = id for registration, userid=if for calls (if I recall correctly) |
| 01:24.37 | raden | [Jul 27 20:23:50] NOTICE[27624]: chan_sip.c:17295 handle_request_invite: Call from '101' to extension '18888264556' rejected because extension not found. |
| 01:24.48 | AlmightyOatmeal | there is no regid |
| 01:24.55 | raden | what extension is it not finding exactly ? |
| 01:24.58 | ManxPower | raden: what is the context= line in sip.conf the [101] section? |
| 01:25.41 | raden | context=to-callcentric |
| 01:26.03 | AlmightyOatmeal | ManxPower: http://radioactiverussian.com/polycom.gif |
| 01:26.09 | ManxPower | and what exten => line in the [to-callcentric] section in extensions.conf do you think should match 18888264556? |
| 01:26.27 | ManxPower | address: polycom |
| 01:26.36 | ManxPower | ^^ for Al |
| 01:27.02 | ManxPower | because the only place asterisk will look is in the [to-callcentric] section. |
| 01:27.05 | AlmightyOatmeal | ManxPower: uh, what? |
| 01:27.21 | ManxPower | AlmightyOatmeal: the Address: field in the web config should have the userid in it |
| 01:27.28 | ManxPower | don't ask my why they named it that. |
| 01:27.35 | AlmightyOatmeal | that doesn't make sense |
| 01:27.40 | ManxPower | no, it doesn't. |
| 01:27.42 | AlmightyOatmeal | :P |
| 01:27.46 | raden | [to-callcentric] |
| 01:27.46 | raden | exten => _XX,1,Dial(SIP/${EXTEN}@callcentric) |
| 01:27.51 | raden | i take it thats very wrong ? |
| 01:28.12 | ManxPower | that will match any 2 digit number, and only 2 digit numbers |
| 01:28.38 | ManxPower | _XX. will match numbers of 3 or more digits. "." means "one or more digits" |
| 01:29.40 | raden | so what would be correct |
| 01:30.02 | AlmightyOatmeal | ManxPower: you're wrong |
| 01:30.05 | AlmightyOatmeal | [Jul 27 20:29:33] NOTICE[1427]: chan_sip.c:15948 handle_request_register: Registration from '<sip:7@polycom>' failed for '192.168.1.20' - No matching peer found |
| 01:30.22 | AlmightyOatmeal | the @polycom from the server field? |
| 01:30.38 | ManxPower | *I* would set it as _1NXXNXXXXXX,1,Whatever, but I'm a telecom guy like that |
| 01:30.41 | AlmightyOatmeal | but it should be sip:polycom@192.168.1.50 so that gives me another idea |
| 01:30.57 | AlmightyOatmeal | need to wait for the web UI to come up |
| 01:31.26 | ManxPower | I assume you want "7" to display next to the line key? |
| 01:31.41 | AlmightyOatmeal | no i don't |
| 01:31.45 | AlmightyOatmeal | just "Line 1" is fine |
| 01:31.55 | ManxPower | Label: is what sets what is on the display |
| 01:32.05 | AlmightyOatmeal | oh shit, i changed the wrong address field ManxPower |
| 01:32.08 | AlmightyOatmeal | lol my bad o:) |
| 01:32.16 | ManxPower | looks over his glasses at AlmightyOatmeal |
| 01:32.24 | AlmightyOatmeal | giggles |
| 01:32.36 | eppigy | TRABAJO |
| 01:32.55 | raden | [Jul 27 20:31:57] NOTICE[27624]: chan_sip.c:18600 handle_request_register: Registration from 'sip:101@192.168.1.100' failed for '192.168.1.101' - Peer is not supposed to register |
| 01:33.19 | *** join/#asterisk adr3nalin3 (n=chatzill@204.26.87.226) |
| 01:33.25 | ManxPower | correct. |
| 01:33.43 | ManxPower | Phones should (almost) always be set as type=friend |
| 01:34.11 | raden | it is in extensions |
| 01:34.25 | raden | should i not have a static ip on the phone ? |
| 01:34.28 | ManxPower | chan_sip.c does not even know about extenions.conf |
| 01:34.34 | AlmightyOatmeal | [Jul 27 20:34:23] NOTICE[1427]: chan_sip.c:15948 handle_request_register: Registration from '<sip:polycom@192.168.1.50>' failed for '192.168.1.20' - Username/auth name mismatch |
| 01:34.37 | AlmightyOatmeal | oops |
| 01:34.48 | eppigy | DONDE |
| 01:35.12 | ManxPower | AlmightyOatmeal: looks like it's time to go home. |
| 01:35.20 | AlmightyOatmeal | ManxPower: for you or me? :P |
| 01:35.30 | ManxPower | AlmightyOatmeal: you. |
| 01:35.38 | AlmightyOatmeal | but for real the username and passwords i swear are the same heh, i typed it in like a dozen times |
| 01:35.45 | *** join/#asterisk ArchGT (n=ArchGT@190.149.40.42) |
| 01:36.43 | raden | how do i keep phone from registering ? |
| 01:36.53 | AlmightyOatmeal | why would you want to do that? |
| 01:37.05 | ManxPower | raden: why would you not want a phone to register? |
| 01:37.12 | iewebguy | change the password |
| 01:37.24 | raden | [Jul 27 20:31:57] NOTICE[27624]: chan_sip.c:18600 handle_request_register: Registration from 'sip:101@192.168.1.100' failed for '192.168.1.101' - Peer is not supposed to register |
| 01:37.38 | raden | what does it mean by peer not supposed to register im confussed ? |
| 01:37.53 | ManxPower | why do you have type=peer set in sip.conf? |
| 01:38.08 | raden | thats what the stupid instructions told me todo :( |
| 01:38.25 | ManxPower | raden: your stupid wrong ITSP's instrucitons? |
| 01:38.38 | raden | :( |
| 01:40.10 | AlmightyOatmeal | ManxPower: wtf, is there a special place i need to set the password as well? i changed it to 3 numeric digits and i know the username and password are right |
| 01:40.14 | AlmightyOatmeal | :( |
| 01:40.20 | ManxPower | how many hours have you wasted by following callcentric's instructions? |
| 01:40.45 | ManxPower | AlmightyOatmeal: is there a registration page on the webui? |
| 01:41.06 | raden | books, call centric, google, oreilly book, aastra phone manual, and online version of oreilly book, 3 entire days |
| 01:41.14 | raden | oh and tech support tooo |
| 01:41.24 | AlmightyOatmeal | ManxPower: Callback Mode=Register, thats about it |
| 01:41.31 | ManxPower | they all tell you to set your phones as type=peer? |
| 01:41.46 | raden | type=friend back to busy signals |
| 01:42.00 | ManxPower | raden: that means you have something ELSE wrong. |
| 01:42.11 | ManxPower | phones should (almost) always be set as type=friend. |
| 01:42.31 | ManxPower | AlmightyOatmeal: as I said, I always provision the polycoms from an FTP server |
| 01:42.40 | raden | [101] |
| 01:42.40 | raden | context=to-callcentric |
| 01:42.40 | raden | type=friend |
| 01:42.40 | raden | username=101 |
| 01:42.40 | raden | qualify=yes |
| 01:42.41 | raden | secret=secrets |
| 01:42.43 | raden | host=192.168.1.101 |
| 01:42.45 | raden | canreinvite=no |
| 01:42.49 | Alfio | raden no |
| 01:42.54 | AlmightyOatmeal | ManxPower: then the web UI is not saving the password right or something |
| 01:42.57 | AlmightyOatmeal | :( |
| 01:43.00 | Alfio | ~pastebin |
| 01:43.01 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 01:43.12 | ManxPower | AlmightyOatmeal: chances are I don't remember the name of the field in the web UI |
| 01:43.27 | AlmightyOatmeal | meh |
| 01:43.27 | ManxPower | raden: host=dynamic |
| 01:43.48 | *** join/#asterisk ethicx (n=imperviu@c-98-242-154-14.hsd1.fl.comcast.net) |
| 01:44.09 | raden | set the phone to dhcp then ? |
| 01:44.17 | ManxPower | and disallow=all and allow=ulaw in there just for good measure. |
| 01:44.31 | ManxPower | raden: it maters nothing at all if the phone is a static ip or DHCP. |
| 01:44.46 | ManxPower | all host=dynamic means is "the phone will tell us what the IP is" |
| 01:44.51 | ethicx | If I'm setting up an asterisk box at home with only 1 voip phone do I still need a gateway card such a Digium TDM400P ? |
| 01:45.01 | ManxPower | and since it's usually harder to tell a phone to not register than it is to set host=dynamic...... |
| 01:45.15 | ManxPower | ethicx: do you need to plug any phone lines into the server? |
| 01:45.19 | adr3nalin3 | ethicx: depends how your connecting to the pstn |
| 01:46.51 | *** join/#asterisk supa_disko (n=bleh@iinet.guard.com.au) |
| 01:47.25 | ethicx | I guess not since this card is intended to be used with any analog phones.. right? |
| 01:47.33 | ManxPower | analog phones and analog phone lines |
| 01:47.44 | ManxPower | if you don't need either of those then you don't need the card. |
| 01:48.03 | raden | now busy signals both ways going backwards |
| 01:48.13 | ManxPower | raden: and any messages on the console? |
| 01:48.55 | ethicx | why would someone need to connect analog phone lines to it though? just askin.. |
| 01:48.55 | ManxPower | busy means "failed" It means nothing else. |
| 01:48.56 | ManxPower | ethicx: because they don't want to spend $800/month for a PRI or T-1? |
| 01:48.56 | AlmightyOatmeal | ManxPower: what does this mean? [Jul 27 20:48:20] WARNING[1427]: chan_sip.c:8787 check_auth: username mismatch, have <polycom>, digest has <|a> |
| 01:49.14 | ethicx | =D |
| 01:49.28 | Alfio | AlmightyOatmeal auth problems |
| 01:49.32 | ManxPower | or they are smart enough to realize that the internet is not as reliable as the phone company? |
| 01:49.39 | AlmightyOatmeal | Alfio: i got that much :) |
| 01:49.43 | Alfio | verify the user and secret |
| 01:49.51 | raden | [Jul 27 20:49:07] WARNING[28306]: chan_sip.c:4339 sip_call: No audio format found to offer. Cancelling call to 18888264556 |
| 01:50.01 | AlmightyOatmeal | i did, 20 times now, they are flawless unless the polycom web UI is f0rking up |
| 01:50.34 | raden | ok, When I call in it will ring pickup there is no one there phone im calling from is still ringing, when i call out i get a off the hook sound with that above error message |
| 01:50.34 | AlmightyOatmeal | going to try to config from the phone itself soon |
| 01:50.40 | ManxPower | raden: so you have disallow=all and allow=ulaw in the callcentric and the 101 sections of sip.conf or in [general] of sip.conf? |
| 01:50.51 | raden | yes sir |
| 01:50.57 | raden | double checking right now |
| 01:51.13 | Alfio | maybe he is using g729 |
| 01:51.14 | ManxPower | You did a reload after making the changes? |
| 01:51.23 | Alfio | in the phone |
| 01:51.23 | ManxPower | Alfio: if he is he's totally screwed. |
| 01:51.29 | Alfio | :) |
| 01:51.55 | ManxPower | Alfio: if it was he would get no compatible codecs message |
| 01:52.10 | Alfio | ManxPower k |
| 01:52.12 | ManxPower | looks to me like CALLCENTRIC does not have any codecs set up for the account |
| 01:52.40 | AlmightyOatmeal | ironic that i was going to use callcentric before choosing broadvoice :P |
| 01:53.05 | ManxPower | Personally I like vitelity.net |
| 01:53.30 | raden | im thinking of switching to them :P |
| 01:53.42 | raden | that was my other choice besides gafachi |
| 01:53.54 | ManxPower | gafachi requires large comitments |
| 01:53.57 | raden | just callcentric 4 fewer hops and latency was 11 ms lower |
| 01:54.05 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 01:54.09 | raden | ok |
| 01:54.22 | raden | i can dial out and have bi-drectional communication |
| 01:54.30 | ManxPower | what did you change? |
| 01:54.43 | raden | <PROTECTED> |
| 01:55.35 | raden | disallow=all |
| 01:55.35 | raden | allow=gsm |
| 01:55.35 | raden | allow=ulaw |
| 01:55.35 | raden | allow=alaw |
| 01:55.46 | ManxPower | I cannot help you if you don't listen to me. |
| 01:56.28 | raden | i been following everything you've been saying im pretty sure |
| 01:56.51 | raden | AlmightyOatmeal, hows broadvoice doing for u ? |
| 01:56.55 | ManxPower | raden: I've said at least twice for you to set disallow=all and allow=ulaw |
| 01:57.18 | raden | i have it set in [callcentric] and [101] |
| 01:57.22 | ManxPower | make sure you have ulaw allowed with your provider. |
| 01:57.52 | raden | i then added allow=gsm allow=alaw and its working for some odd reason toolk them out and its not working |
| 01:58.32 | raden | i have done some messed up things things with networking but this VOIP stuff takes the cake :( |
| 01:58.35 | ManxPower | then you don't have ulaw allowed in your carrier |
| 01:59.13 | raden | ManxPower, ulaw is G.711 right ? |
| 01:59.38 | ManxPower | both ulaw and alaw are G.711 |
| 01:59.52 | ManxPower | G.711u would be ulaw aka PCMU |
| 02:00.11 | *** join/#asterisk s14ck (n=s14ck@190.77.92.227) |
| 02:01.27 | raden | codecs supported |
| 02:01.28 | raden | G.723 6.3Kbps |
| 02:01.28 | raden | G.729A |
| 02:01.28 | raden | G.711 mu-law |
| 02:01.28 | raden | G.711 A-law |
| 02:01.52 | PhunTelTek | nice modern standard. circa 1972 |
| 02:01.56 | voxter | Is there a res_jabber backport for 1.2? |
| 02:04.35 | raden | ok now its working with just allow=ulaw |
| 02:05.25 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 02:05.52 | PhunTelTek | Aren't the codecs supposed to be listed in order of preference? |
| 02:05.53 | raden | i can dial out but not in :( |
| 02:06.38 | PhunTelTek | registered? |
| 02:08.12 | raden | let me rephrase i can dial in get the phone to ring but when pickup phone no one there |
| 02:08.21 | raden | and the other side of phone keeps ringing |
| 02:09.21 | PhunTelTek | when you dial out, can you get audio then? |
| 02:09.58 | raden | yes i can |
| 02:10.05 | raden | i can talk and hear both ways |
| 02:10.22 | PhunTelTek | odd |
| 02:10.51 | PhunTelTek | I've never had that particular problem. |
| 02:11.52 | PhunTelTek | I'm no expert, but i'm thinking context? |
| 02:12.17 | raden | hmmm |
| 02:12.47 | raden | [from-callcentric] |
| 02:12.48 | raden | exten =>17772445766,1,Dial(SIP/101) |
| 02:16.08 | PhunTelTek | dunno. my setup isn't configured exactly right, but it works. |
| 02:17.47 | PhunTelTek | my extensions say context=from-internal and my trunk says context=from-pstn |
| 02:19.55 | PhunTelTek | context = <contextname> : This is the default context and is used when a endpoint has no context property. The context in section of an endpoint is used to route calls from that endpoint to the wanted destination. The context body is located in extensions.conf. |
| 02:20.03 | raden | why cant u buy a paperback copy of Asterisk: The Future of Telephony 2nd edition on amazon ?> |
| 02:20.27 | PhunTelTek | cuz you can print a paperback copy |
| 02:20.50 | raden | all they have for sale is Kindle edition |
| 02:21.03 | raden | i want to order paperback or hardcover just printed grrrr..... |
| 02:21.54 | ManxPower | raden: ora.com should have it |
| 02:22.01 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 02:22.12 | raden | sound quality is freaking crap :( |
| 02:22.19 | raden | thank u |
| 02:22.36 | PhunTelTek | http://www.cyberciti.biz/files/free-linux-books-pdfs/AsteriskTFOT.zip |
| 02:23.17 | raden | i have it in pdf |
| 02:23.24 | raden | i want printed copy |
| 02:24.13 | raden | can someone explain why when i set server to static ip and set DNS addresses browsing is much slower ? |
| 02:27.57 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 02:30.56 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:36.19 | raden | yea it will be here thursday |
| 02:44.12 | raden | well im gone for tonight thanks for all the help guys |
| 02:49.50 | PhunTelTek | yup |
| 02:50.43 | PhunTelTek | Anyone here interface * to radio equipment? |
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| 03:04.10 | *** part/#asterisk MACscr (n=Mark@98.214.100.212) |
| 03:13.44 | Slade- | wow, asterisk will work as an SMS gateway? |
| 03:13.58 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 03:15.23 | [TK]D-Fender | Slade-: * does not receive SMS, and only sends a select interfaces. |
| 03:15.41 | [TK]D-Fender | Slade-: No, it is NOT a viable solution for this |
| 03:15.51 | Slade- | looking for MT only |
| 03:16.19 | Slade- | and really for proof of concept stuff |
| 03:19.05 | Slade- | Fender: is there a decent doc on using it somewhere? or are you suggesting steering clear altogether? |
| 03:21.04 | [TK]D-Fender | Slade-: Look at something else |
| 03:22.39 | Slade- | got a recommendation? |
| 03:23.00 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 03:27.00 | [TK]D-Fender | Slade-: Noep |
| 03:27.35 | Slade- | hmm :/ |
| 03:32.03 | *** join/#asterisk QaDeS (n=mklaus@p4FC7284C.dip0.t-ipconnect.de) |
| 03:34.41 | Slade- | http://www.voip-info.org/wiki/view/SMS+Service+in+the+USA // says all kinds of suck :/ |
| 03:44.58 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 04:40.02 | VooDooNOFX | Is there a way to supress or fix ast_rtp_read: RTP Read too short in my console output? |
| 04:40.21 | VooDooNOFX | Google says it's because the box is a 10-base half-duplex NIC |
| 04:41.21 | VooDooNOFX | these little ATA's only support 10 base half duplex |
| 05:01.47 | *** join/#asterisk webman (n=adamg@52.87.233.220.static.exetel.com.au) |
| 05:12.27 | webman | does anyone know where asterisk will core dump ? |
| 05:12.46 | VooDooNOFX | where it will core dump? |
| 05:12.54 | VooDooNOFX | you mean to which folder the dump will be placed? |
| 05:13.07 | webman | yes, which directory does it dump to when it crashes... |
| 05:13.41 | VooDooNOFX | check your /etc/profile for a ulimit line |
| 05:13.51 | VooDooNOFX | it may not be saving them at all |
| 05:14.10 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 05:14.11 | webman | I set a ulimit -c unlimited from the shell before starting asterisk |
| 05:14.48 | webman | basically, it will crash reliably whenever I try to transfer a call... |
| 05:15.16 | webman | so I'm trying to work out the needed information to find out why it is crashing |
| 05:15.40 | webman | so I'm trying to work out the needed information to find out why it is crashing |
| 05:15.41 | VooDooNOFX | checkout http://aplawrence.com/Linux/limit_core_files.html for info on manually controlling where they dump to |
| 05:15.54 | webman | sorry wrong window |
| 05:15.54 | VooDooNOFX | or you can check the services script for the location |
| 05:29.21 | *** join/#asterisk ethicx (n=imperviu@c-98-242-154-14.hsd1.fl.comcast.net) |
| 05:30.45 | ethicx | Im confused ... if a T1/PRI line is in place ..who provides DID numbers for the extensions ? the t1 provider? |
| 05:36.15 | webman | yay: Segmentation fault (core dumped) |
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| 05:51.45 | webman | hmm, I'm getting one way audio on a sip client, but yet I can see RTP packets being sent/received on asterisk and also on the sip client.... has anyone got any ideas how to debug further ? |
| 05:56.24 | TrentCreek | SIP SET DEBUG? |
| 05:57.15 | TrentCreek | what audio is it? |
| 05:58.15 | TrentCreek | Who know about MeucciSolutions? |
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| 06:01.58 | webman | sip debug doesn't appear to have any problems... they seem to negotiate gsm quite happily .... |
| 06:02.21 | TrentCreek | maybe a irewal issue |
| 06:02.24 | TrentCreek | firewall |
| 06:02.56 | TrentCreek | crap..I keep getting calls from a MeucciSolutions and going to a non-existant extension |
| 06:02.58 | webman | the packets are sent/received on the asterisk box and the client device.... (tested with tcpdump on both ends using a softphone |
| 06:03.35 | TrentCreek | then possible a NAT issue |
| 06:05.11 | webman | but the packets are already on the client (ie, through the nat)... and there is only nat on the client side, not on the server, and this problem happens for all remote sip endpoints (which were working previously on asterisk 1.4 and asterisk 1.2... last night I upgraded to 1.6.0.11 and it stopped working |
| 06:06.21 | TrentCreek | oops..there is your problem right there..1.6 is still not suggested for production boxes |
| 06:06.53 | webman | huh... I thought 1.6.0 was 'stable' and 1.6.1 and a.6.2 were development ? |
| 06:07.46 | TrentCreek | I hav enot heard any such thing..Everywhere I go, I read do not use those |
| 06:08.12 | webman | and the reason for the upgrade from 1.4 to 1.6 was to solve the horribly high CPU usage when more than around 30-40 channels in-use... |
| 06:09.49 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 06:10.14 | TrentCreek | That does not sound right, unless you are doing DA/AD |
| 06:10.27 | TrentCreek | Is it an old CPU? |
| 06:10.37 | TrentCreek | low memory? |
| 06:11.28 | webman | dual cpu opteron with 2G RAM (mostly un-used, not swapping at all) doing g729 codec translation, and echo cancel, but nothing else.... basically |
| 06:12.03 | TrentCreek | 64 bit? |
| 06:12.22 | TrentCreek | OS |
| 06:12.32 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 06:12.37 | TrentCreek | and compiled Asterisk? |
| 06:12.38 | webman | yes, amd64 with debian stable |
| 06:12.56 | drmessano | 1.6 is more than suggested for production |
| 06:12.56 | webman | self compiled asterisk, and also tried the debian package version... |
| 06:13.36 | TrentCreek | it could be that echo cancel overloading. Software doing echo cancel most of the time sucks and uses too much CPU |
| 06:13.59 | TrentCreek | you with that setup should be able to handle maybe 1000 cals |
| 06:14.00 | drmessano | and 1.6.1.x AND 1.6.0.x are current/stable branches |
| 06:14.05 | drmessano | 1.6.2.x is beta |
| 06:14.15 | webman | I did reduce echocancel from 128 to 64, but there was no difference either... |
| 06:14.26 | TrentCreek | I am still hearig stay away from 1.6 for production |
| 06:14.44 | drmessano | heh |
| 06:15.16 | TrentCreek | I dont need head ache so I stay happy with 1.4 |
| 06:15.17 | drmessano | I think you're making shit up like you always do, TrentCreek.. Folks in here have been pushing 1.6 for production for months.. its far more stable than 1.4 |
| 06:15.33 | TrentCreek | no,. just reading things off sites |
| 06:15.43 | drmessano | Links? |
| 06:16.00 | webman | trentcreek: well, I would have been happy with 1.4 if it was working properly.... :( |
| 06:16.19 | TrentCreek | let me see if I cna find one..it's ben a month since I last read |
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| 06:20.24 | TrentCreek | Now here is a political answer ""I recommend the version that has the features you require, and that passes the testing you have done prior to rolling it out"." |
| 06:21.21 | *** join/#asterisk mintos (n=mvaliyav@209.132.188.254) |
| 06:24.22 | TrentCreek | Asterisk 1.6.1 do have modifications to support this, but unfortunately, it does cause problems with actual Aastra 2.2.1 xml scripts. So we'll need to wait certainly about one year more before before to be able to use * 1.6 in production. |
| 06:24.45 | TrentCreek | July 17th, 2009 http://www.freepbx.org/forum/freepbx/users/status-of-freepbx-2-6 |
| 06:25.12 | TrentCreek | you wre saying> |
| 06:27.24 | TrentCreek | Because of the new release policy, we won't see a |
| 06:27.26 | TrentCreek | stable SIP channel with TCP/TLS in the 1.6 generation for quite some |
| 06:27.28 | TrentCreek | time, since coming releases will contain other large changes in |
| 06:27.30 | TrentCreek | addition to the TCP/TLS code. |
| 06:27.32 | TrentCreek | |
| 06:27.33 | TrentCreek | http://lists.elastix.org/pipermail/developers/2008-October/000019.html |
| 06:28.18 | drmessano | TCP/TLS |
| 06:28.22 | drmessano | A NEW FEATURE |
| 06:28.39 | TrentCreek | I am just throwing in examples you wanted |
| 06:28.40 | webman | TrentCreek: so not using TCP/TLS and not using aastra xml scripts (both new features) means what for 1.6 ? |
| 06:28.40 | drmessano | and its actually coming along well.. |
| 06:28.43 | drmessano | Another challenge we get with ToIP, is that we do not have caller id update after attended transfers. This is a difficult situation, because Aastra phones do support this since more than one year now, but asterisk 1.4 and FreePBX do not support it. I would be happy to see some efforts in this direction. |
| 06:28.43 | drmessano | Asterisk 1.6.1 do have modifications to support this, but unfortunately, it does cause problems with actual Aastra 2.2.1 xml scripts. So we'll need to wait certainly about one year more before before to be able to use * 1.6 in production. |
| 06:29.03 | drmessano | and thats your forum post, including your one line you chose to paste.. ignoring the rest |
| 06:29.15 | drmessano | webman: Very little |
| 06:29.44 | drmessano | TrentCreek just likes to troll and pretend he knows what he's talking about.. many of us are on 1.6 and it's working well |
| 06:30.08 | drmessano | I could find dozens of posts relating to shortcomings of 1.4.. what does that prove? |
| 06:31.03 | TrentCreek | I am not talking about :shortcomings" |
| 06:31.24 | TrentCreek | now yoy are throwing in red herring |
| 06:31.46 | drmessano | Pointing out some year old post about TCP/TLS is trolling |
| 06:32.07 | TrentCreek | yeah...WTFE |
| 06:32.30 | TrentCreek | yeah talking about verson 1.6.1 is "trolling" |
| 06:32.31 | drmessano | Got any real data... have you actually USED 1.6 lately? have you run it in production or heavy lab use? |
| 06:32.34 | drmessano | No, so STFU |
| 06:32.42 | TrentCreek | it seems to me only one starting trolling/arguments is you |
| 06:33.57 | drmessano | By asking you to support your trolls about 1.6 being not recommended for production, etc? |
| 06:34.36 | *** join/#asterisk xrmx__ (n=rm@host129-254-dynamic.2-87-r.retail.telecomitalia.it) |
| 06:34.37 | drmessano | It's called fact checking.. maybe you've heard of it |
| 06:39.59 | webman | ok, in the meantime, could anyone help me debug why my audio is not getting from the sip phone to asterisk and out onto the pri ? |
| 06:42.46 | webman | hmmm, calling sip <-> sip works with two way audio.... and iax to pri works as well.... |
| 06:43.06 | webman | seems to just be sip to pri has one way audio... strange :( |
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| 06:50.17 | TrentCreek | webman: you may have to wait for Fender to return and have copies of your results and settings ready in pastebin |
| 06:54.53 | webman | TrentCreek, how much debugging would be needed ? sip debug, rtp debug ? on the asterisk server... what about the client, tcpdump ? |
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| 07:09.57 | Balistic | Hi, I have a problem where none of the phones on the network show up when I issue a 'sip show peers', this happens every now and then. Also when this occurs I notice for each phone a line similar to 'Really destroying SIP dialog '061d0bc02354aa0312a12bbf4b302890@192.168.0.11' Method: OPTIONS' every 60secs or so. The phones are pingable on the network. |
| 07:10.53 | TrentCreek | webman: sip debug and your settings |
| 07:12.08 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:12.22 | webman | hmmm, is 92 of "channel.c: Avoiding initial deadlock for channel '0x7fc8e8040220'" in a row (in one second) is not good is it ? |
| 07:13.32 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 07:14.38 | TrentCreek | webman: dont lnow about that..better try google |
| 07:15.32 | TrentCreek | Hope this is a lesson to always get an extra box to test before putting it to production, especially if it is used in business |
| 07:16.12 | webman | TrentCreek: well, I did suggest that, but they figured we should go ahead anyway.... |
| 07:16.31 | TrentCreek | so they get to pay you overtime now? |
| 07:16.46 | webman | TrentCreek: unfortunately ... no :( |
| 07:17.10 | TrentCreek | well then I guess at quitting time..."see ya" |
| 07:17.56 | VooDooNOFX | CYA dude. save those emails stating "we want it anyway" |
| 07:17.57 | VooDooNOFX | :D |
| 07:18.18 | webman | haha, I was working on this until 5am, and started again at 11am... still going and it is 5:20pm ... |
| 07:18.50 | TrentCreek | i would work regular hours |
| 07:19.38 | VooDooNOFX | webman: what version of * are you using? |
| 07:19.43 | TrentCreek | simce they do not want to payyou overtime, it must not be very important |
| 07:19.56 | webman | VooDooNOFX: 1.6.0.11 |
| 07:20.28 | VooDooNOFX | why not deploy the stable 1.4 branch instead to appease them, then upgrade later to 1.6? |
| 07:21.18 | webman | VooDooNOFX: because with 1.4.26 we had massive CPU utilisation (ie, big spike) whenever we hit more than 30-40 calls (60-80 channels) |
| 07:21.31 | TrentCreek | oh, but drmessano says 1.6.1 IS stable |
| 07:21.57 | VooDooNOFX | webman: what codecs are you using? |
| 07:22.09 | webman | VooDooNOFX: At this rate the fallback option is to revert to 1.2 which was working for years quite nicely, though in the last couple weeks we had three crashes.... |
| 07:22.12 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:22.35 | webman | VooDooNOFX: using g729 as codec for all channels (from pri on TE410p to either sip/iax2) |
| 07:27.36 | TrentCreek | webman: are they going to give you an extra box this time? |
| 07:28.19 | *** join/#asterisk superc (n=superc@ip-77-25-43-194.web.vodafone.de) |
| 07:29.25 | webman | TrentCreek: well, not yet... almost ordered one yesterday to try and resolve the high CPU... but personally I think there was/is something wrong, so I don't know that faster CPU will solve it, or just make it go more wrong faster... |
| 07:29.37 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 07:29.58 | webman | which is why I'm trying 1.6.0.11 now... to see if whatever the problem was is resolved in this version.... but now I have another problem.... (one way audio) |
| 07:30.22 | TrentCreek | why not just start over and in stall PBX in a flahs..you can be back up within an hour |
| 07:30.46 | webman | the box is remote.... |
| 07:31.01 | TrentCreek | yes..so is PBX in a flash |
| 07:31.02 | *** join/#asterisk Da-Geek (n=Da-Geek@ip-217-204-53-131.easynet.co.uk) |
| 07:31.14 | TrentCreek | ther eis a way to install it remotely |
| 07:31.21 | Da-Geek | Hi All.... |
| 07:32.12 | Da-Geek | is looking for any testers for an rpm that installs the NV-Fax modules for astrisk... anyone interested ? |
| 07:32.58 | webman | TrentCreek: what will a re-install solve? A new/clean OS, but that isn't going to cause/solve one way audio, a new/clean install of asterisk where I will need to restore the old configs anyway? maybe the new/clean binaries, but I just don't see how it will help |
| 07:33.02 | TrentCreek | in this day of people trying to get free call soff my box..no thanks |
| 07:33.20 | webman | TrentCreek: mind you, at this rate I just might get that desperate and try anyway :) |
| 07:33.45 | TrentCreek | PBX in a Flash does it all..It's the OS and a web interface..and backup of the whole setup |
| 07:33.57 | TrentCreek | and with a Operator panel |
| 07:34.44 | TrentCreek | the backups can be automated. |
| 07:35.01 | TrentCreek | it is also already setup with basic system security |
| 07:35.12 | TrentCreek | asterisk 12. and 1.4 is available for it |
| 07:35.36 | webman | TrentCreek: I don't think that would suit my environment... we don't use fop, web interface, etc.. just asterisk + asterisk-addons (mysql) + mysql server... nothing extra |
| 07:35.57 | TrentCreek | well it would be easier for admin |
| 07:36.04 | Da-Geek | webman, I always thought one way audio was normally a firewall issues/ or mismatch of NAT settings on the handset config and the Asterisk server... |
| 07:36.11 | TrentCreek | and comes with 3rd party addons for extra nice goodies |
| 07:36.55 | webman | Da Geek: thanks, me too, but I have checked both of those, I can see the rtp moving in both directions on the asterisk box... |
| 07:37.10 | TrentCreek | You can also bring up CDR in a flash |
| 07:37.24 | TrentCreek | it has a LOT of features |
| 07:38.56 | webman | we would probably use 2 x SSD in RAID1 if we do get a new machine... but I don't like those pre-packaged configs.... they are great if you know nothing and want to do exactly the same as what they allow you to do.... otherwise they are a pain... (my opinion) |
| 07:40.56 | TrentCreek | well it does allow you to see what is going on with the machine in real time |
| 07:41.27 | webman | so it would allow me to see why I have one way audio? |
| 07:42.41 | TrentCreek | yeah..using the CLI ;-) |
| 07:42.56 | TrentCreek | Damn..who knows those bastards at MeucciSolutions/ |
| 07:43.25 | webman | so what is different with their CLI compared to my current one ? |
| 07:43.34 | TrentCreek | nothing |
| 07:43.40 | webman | sorry, I might be just a little frustrated right now :) |
| 07:44.20 | kaldemar | start with SIP debug of the call and pastebin it. |
| 07:45.11 | webman | yes, I'm cutting out all the other sip debug crap so I can do that... 5000 lines of debug is what I started with.... so far I've kept 260 and got 2500 lines to go.... |
| 07:46.21 | TrentCreek | webman: set verbose to Zero (0) then turn on sip debug |
| 07:46.22 | *** part/#asterisk Balistic (n=gert@196.31.150.107) |
| 07:46.55 | webman | it's mostly the sip debug output (98% of it anyway...) |
| 07:47.05 | TrentCreek | Damn..who knows those bastards at MeucciSolutions |
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| 07:55.24 | VooDooNOFX | Night all |
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| 07:59.51 | webman | sip debug is at http://www.pastebin.ca/1509754 if anyone can provide hints on why I get one way audio it would be very helpful (audio from the sip client to asterisk is not getting out the pri to the 2nd party). |
| 08:00.38 | webman | will be back later tonight to continue... thanks all |
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| 08:13.18 | jra__ | if I have a alfanumerical mailbox-name a.doe@default it works fine, but j.doe@default gets stripped to .doe@default because it also expects options there or something, is there a way to prevent this from happening (besides not using characters in the mailbox name)? |
| 08:14.36 | kaldemar | elaborate |
| 08:16.33 | tzafrir_laptop | jra__, so give an explicit options parameter |
| 08:17.35 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
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| 08:30.15 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 08:31.00 | dwery | anyone knows how to remotely force a reboot on the C470IP? They keep behaving strangely from time to time |
| 08:35.15 | jra__ | tzafrir_laptop, ah, that indeed skips the parsing of the prepended options, thanks |
| 08:36.17 | *** join/#asterisk Psychobilly (n=moi@adsl117-205.kln.forthnet.gr) |
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| 08:51.00 | *** join/#asterisk mallchin (n=mallchin@mail.dataproservices.co.uk) |
| 08:51.22 | mallchin | hey guys, I am getting this error, could anyone point me in the right direction please? |
| 08:51.39 | mallchin | == Everyone is busy/congested at this time (1:0/0/1) |
| 08:51.51 | mallchin | was working yesterday |
| 08:53.28 | jeremy_g | hello |
| 08:54.08 | jeremy_g | to support 1000 users, how many sip concurrent sip channel shud my * box be able to create |
| 08:55.12 | jeremy_g | e.g. i heard kpflemming saying in a seminar, to support 6000 users, our asterisk cluster should be able to support 1000 simulateneous sip channels |
| 08:55.30 | jeremy_g | how do you calculate this? i think it has sth to do with the erlang capacity |
| 08:55.32 | tzafrir_laptop | mallchin, (1:0/0/1) - it's not busy not congested. |
| 08:55.37 | tzafrir_laptop | it's "unavailable" |
| 08:55.51 | tzafrir_laptop | What was the Dial command? |
| 08:57.25 | mallchin | tzafrir_laptop: I get 1010 too |
| 08:57.25 | mallchin | exten => 400,1,Dial(SIP/202|20) |
| 08:57.46 | mallchin | tzafrir_laptop: it works if I dial ext 202 directly, but not from this context |
| 08:59.03 | tzafrir_laptop | I'm not sure what "unavailable" means in chan_sip |
| 08:59.31 | mallchin | from what I have read it seems to occur when a box looses dns |
| 08:59.47 | mallchin | however I am able to ring the extension from another context |
| 09:11.31 | dwery | how comes that queue show xxx always shows the members as not in use? |
| 09:14.35 | cusco__ | to show the ones that are, core show channels |
| 09:15.43 | dwery | cusco__: yes, but that's a different issue :D |
| 09:15.59 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 09:16.04 | dwery | I would like the queue app not to send SIP messages to busy members |
| 09:18.06 | *** join/#asterisk joobie (n=joobie@124-168-112-63.dyn.iinet.net.au) |
| 09:22.00 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 09:23.32 | dwery | https://issues.asterisk.org/view.php?id=14316 so Not in use happens if you don't have call-limit, but call-limit is deprecated. Am I missing something? |
| 09:23.50 | Stese | hey all, is there a alternative way of checking the features between 1.6.0 and 1.6.1 trunks other than the CHANGES file? |
| 09:27.45 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 09:37.22 | *** join/#asterisk Sajam (n=chatzill@212.40.146.245) |
| 09:37.52 | Sajam | guys, what is the best linux for installing asterisk. I installed asterisk on debian and the window is corrupted now. |
| 09:38.29 | Sajam | the window manager is corrupted after installing asterisk. |
| 09:38.45 | kaldemar | Sajam: the distribution you're most comfortable with. |
| 09:38.48 | scoof | Sajam: hardly an asterisk or debian issue |
| 09:39.20 | *** join/#asterisk aurax (n=aurax@bzq-179-76-199.static.bezeqint.net) |
| 09:40.09 | Sajam | scoof: it's a debian issue, but the cause is asterisk, since after installing it I cannot logon the window manager. That's why i am asking what the best OS for asterisk that work with. |
| 09:40.38 | scoof | Sajam: I still very much doubt it. Can you reproduce this on a fresh install? |
| 09:41.01 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:41.15 | kaldemar | Sajam: did you install from the debian package? |
| 09:41.42 | Sajam | scoof: i already did, it's the second time now. First one fresh install on the boot and not on virtual machine and same result (After logon, no desktop appears) |
| 09:42.05 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:42.05 | Sajam | kaldemar: i installed it via, apt-get install asterisk |
| 09:42.52 | kaldemar | that's a debian issue then. plain asterisk doesn't touch window manager or user settings upon install. |
| 09:43.47 | Sajam | kaldemar: ok then, i don't want to reinstall asterisk on debian, i want to try other OS, what do you recommend?? |
| 09:43.51 | scoof | kaldemar: dpkg won't touch other package files |
| 09:44.12 | scoof | kaldemar: unless asked by the user |
| 09:44.20 | scoof | Sajam: which debian release? |
| 09:44.45 | kaldemar | scoof: i know it's not supposed to. but this hardly has anything to do with plain asterisk, as you too said earlier. |
| 09:44.45 | Sajam | lenny, 4.0 |
| 09:45.08 | kaldemar | Sajam: use whatever distribution you know best. |
| 09:46.02 | Sajam | Kaldemar: do you recommend Ubuntu or Fedora for asterisk, or same |
| 09:46.35 | scoof | Sajam: Lenny is Debian 5 |
| 09:46.50 | kaldemar | i haven't used fedora myself so no comments on that. |
| 09:47.07 | kaldemar | besides, i always install asterisk from source. |
| 09:47.31 | Stese | Sajam > have you read the Book? |
| 09:47.34 | Stese | ~book |
| 09:47.35 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 09:48.32 | Sajam | Kaldemar: This is what i want, i want to install it directly from source, but there's a lot of dependencies in installing it, that's why i tried apt-get. |
| 09:49.42 | Sajam | is there's a manual or something for installing asterisk form source. and what are the dependencies? |
| 09:49.45 | Stese | Sajam > the closest to what you require, i beleive is the book |
| 09:49.53 | Stese | ~book |
| 09:49.54 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 09:50.42 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/x-6f454dfc7590016e) |
| 09:52.12 | Sajam | Stese: thanks for the info. I want to ask if asterisk support video over IP? |
| 09:55.47 | kaldemar | yes, but not video conferencing. |
| 09:57.42 | Sajam | Kaldemar: you mean, it supports peer to peer video call using softphonel |
| 09:59.47 | Psychobilly | Sajam if u are using debian or somehting debian based u can just run apt-get build-deb asterisk and it will install all u need to compile from source |
| 10:00.14 | mallchin | == Everyone is busy/congested at this time (1:0/0/1) |
| 10:00.29 | mallchin | anyone point me in the right direction for that error please? setup seems sane |
| 10:01.09 | Sajam | Psychobilly: thanks, now after run the command i need to continue with ./configure and ........ ? |
| 10:01.32 | Psychobilly | after that u get the asterisk source and compile it as usual |
| 10:02.10 | mallchin | Sajam: depends how you want to configure it |
| 10:03.39 | Sajam | mallchin: Ok, to prevent asking too dumb questions :P i will read the whole book first. I understand from your question that's there's more than one configuration for asterisk. does it include in the book? |
| 10:04.01 | kaldemar | Sajam: yes |
| 10:04.01 | Psychobilly | Sajam ahy dont u just use the binary packages from your distro? |
| 10:04.13 | Psychobilly | its easier to start using asterisk this way |
| 10:04.24 | Psychobilly | s/ahy/why |
| 10:05.43 | Sajam | Psychobilly: OK great, what i am trying to do is start developing asterisk, so that's why i asked if it supports video over IP. i am a developer on MS windows and i have tasks to start a development on asterisk. |
| 10:06.15 | Psychobilly | ok then |
| 10:06.24 | Psychobilly | the source is there, have phun :> |
| 10:07.21 | Sajam | Psychobilly: thanks |
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| 10:17.45 | tzafrir_laptop | Sajam, the answer is generally "basically, yes" |
| 10:18.04 | tzafrir_laptop | but AFAIK there are some gotchas |
| 10:18.22 | tzafrir_laptop | And I really don't know much about this |
| 10:18.57 | Sajam | tzafrir_laptop: the answer is yes about what?? |
| 10:19.12 | tzafrir_laptop | oh, I thought you asked about support for video |
| 10:19.41 | tzafrir_laptop | ah, but I see you already got that answer :-) |
| 10:35.00 | scoof | Sajam: FYI, I just installed a Lenny with gnome/gdm and asterisk, and the window manager works fine |
| 10:35.56 | Sajam | scoof: did you restart the OS? |
| 10:36.02 | scoof | Sajam: yes |
| 10:36.46 | tzafrir_laptop | scoof, err... and how is a WM related to Asterisk? |
| 10:37.04 | tzafrir_laptop | Though naturally Lenny is a fine OS :-) |
| 10:37.17 | scoof | tzafrir_laptop: no idea. Just trying to verify what Sajam said |
| 10:37.41 | scoof | tzafrir_laptop: since it was pretty easy to boot up a kvm with Lenny and test it |
| 10:38.19 | Sajam | scoof: i tried it on debian 4.0 |
| 10:38.23 | tzafrir_laptop | scoof, here's something for you to play with: |
| 10:38.50 | tzafrir_laptop | http://updates.xorcom.com/iso/live-2.0.0-beta.img |
| 10:38.53 | scoof | Sajam: are you sure? You said Lenny 4.0 |
| 10:39.13 | Sajam | scoof: yes, 4.0 |
| 10:39.17 | scoof | tzafrir_laptop: why would I? |
| 10:39.24 | tzafrir_laptop | 4.0 is Etch. Lenny is 5.0 |
| 10:39.48 | tzafrir_laptop | Don't use Etch unless you have a very good reason ("you know what you're doing") |
| 10:40.11 | Sajam | scoof: one question, did you use apt-get install asterisk , or something else |
| 10:40.14 | scoof | Sajam: why not lenny? |
| 10:40.20 | scoof | Sajam: yes, I used apt-get |
| 10:40.44 | tzafrir_laptop | the image I pointed to you is not something you can install, but under kvm: just kvm -hda live-2.0.0-beta.img |
| 10:40.56 | Sajam | scoof: ok then, i will try lenny, thanks again. |
| 10:41.06 | scoof | tzafrir_laptop: sure, but I've plenty of asterisk to play around with :) |
| 10:41.13 | tzafrir_laptop | see sources.list on that system |
| 10:45.23 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 10:56.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 11:39.25 | cusco__ | hi |
| 11:40.19 | *** part/#asterisk superc (n=superc@ip-77-25-43-194.web.vodafone.de) |
| 11:40.49 | cusco__ | I have a ael file that says: Monitor(gsm,/var/log/asterisk/asterisk_rec/outbound-${UNIQUEID},mb) |
| 11:41.04 | cusco__ | how can I make them recorded files, have different permissions |
| 11:41.24 | cusco__ | so any other user in the box can read it |
| 11:42.18 | *** join/#asterisk QaDeS (n=mklaus@p4FC7284C.dip0.t-ipconnect.de) |
| 11:47.34 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:47.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:50.44 | TommyBotten | cusco__: That is more of a system option |
| 11:51.07 | TommyBotten | cusco__: You need to set the correct umask for the user spawning the process (probably user 'asterisk' or even 'root') |
| 11:51.15 | TommyBotten | or - you could set a "sticky bit" on the directory: |
| 11:51.33 | TommyBotten | cusco__: http://en.wikipedia.org/wiki/Sticky_bit |
| 11:52.50 | cusco__ | asterisk 28237 4.2 12.5 967852 256292 ? Ssl Jul27 59:24 /usr/sbin/asterisk -p -U asterisk |
| 11:52.56 | cusco__ | user asterisk is running it |
| 11:53.16 | cusco__ | so isn't there an option that sets the umask? |
| 11:53.35 | cusco__ | only asterisk user may access those files |
| 11:53.48 | cusco__ | sticky bit, let me look that up |
| 11:54.10 | TommyBotten | cusco__: I'm not really sure. I belive that asterisk adopts the system umask |
| 11:54.38 | cusco__ | "the Linux kernel ignores the sticky bit on files." |
| 11:54.55 | AlmightyOatmeal | got my polycom working, turns out the web UI for the phone sucks.. kept putting in |a as a username instead of what i put in the web UI |
| 11:55.21 | AlmightyOatmeal | time for work though |
| 11:55.27 | AlmightyOatmeal | lots of viruses to fix |
| 11:55.41 | TommyBotten | cusco__: True. But if you set it for a subdirectory, all files in that directory will adhere to the sticky bit. |
| 11:56.21 | cusco__ | ok |
| 11:56.22 | cusco__ | let me try |
| 12:01.41 | cusco__ | TommyBotten: did not work |
| 12:01.47 | cusco__ | You don't have permission to access /audio/recorded/1248782554.91092-in.gsm on this server |
| 12:01.59 | cusco__ | (that is apache) |
| 12:02.17 | cusco__ | 1761308 drwxr-xr-t 2 asterisk root 180224 2009-07-28 13:03 asterisk_rec |
| 12:02.26 | cusco__ | there is the "t" |
| 12:05.01 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:06.14 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 12:08.03 | TommyBotten | Are that newly created files? |
| 12:08.06 | TommyBotten | Or the old ones? |
| 12:08.14 | TommyBotten | A quick fix is merely a chmod 644 |
| 12:11.24 | cusco__ | TommyBotten: yea, but I don't want to be doing that every ten seconds... |
| 12:15.35 | leifmadsen | AlmightyOatmeal: the polycom webUI is probably the only thing that is really bad on those systems. You're much (much!) better off doing it from a configuration file (via FTP, HTTP, etc....) |
| 12:18.18 | kaldemar | cusco__: maybe the chmod could be put in the extension that does the recording |
| 12:24.21 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:25.17 | *** join/#asterisk Snoogan (n=asdf@115.69.179.5) |
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| 12:37.46 | *** join/#asterisk webman (n=adamg@200.179.233.220.static.exetel.com.au) |
| 12:38.17 | cusco__ | kaldemar: how ? |
| 12:39.39 | TommyBotten | hehe |
| 12:39.40 | kaldemar | cusco__: for example using app System |
| 12:39.42 | TommyBotten | you could do that |
| 12:39.49 | TommyBotten | with a simple system("chmod ....") |
| 12:39.52 | TommyBotten | not pretty though |
| 12:42.18 | webman | Can someone suggest how to resolve my one way audio problem? sip debug is at http://www.pastebin.ca/1509754 and I'm almost 100% sure it is not a firewall or nat issue.... |
| 12:43.30 | scoof | webman: wield tcpdump while you wait for somebody to look at it |
| 12:43.31 | afink | is it a common problem where * dials too fast for some analog lines? |
| 12:43.34 | scoof | (or wireshark) |
| 12:44.01 | afink | I was having a problem where the call didn't go through but adding a Wait(1) in front of the dial command seemed to help |
| 12:44.16 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:44.36 | webman | scoof: I used tcpdump on the asterisk client, and saw two way packet flow, also asterisk rtp debug showed two way audio.... |
| 12:44.48 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 12:45.13 | kaldemar | afink: nothing new there. |
| 12:45.23 | ariel_ | afink: you may need to add some www to your dial command |
| 12:45.24 | webman | afink: on a analog line, you need to wait for the other end to realise you have taken the line offhook, and find a dtmf processing channel for you... |
| 12:45.25 | scoof | webman: then do the same on each side of your nat and firewall gear. |
| 12:46.51 | webman | scoof: I can't do it on the firewall/etc, but I have done it on the asterisk server, and a voip soft phone client... and the packets are making it all the way round trip... |
| 12:47.16 | *** join/#asterisk ccesario_ (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 12:47.20 | webman | in fact, sip <-> sip works, only sip <-> PRI (TE410p) has the one way audio.... |
| 12:47.36 | webman | IAX2 <-> PRI works fine as well.... |
| 12:48.03 | afink | ariel_: thats what I was trying to remember. How does that work again? Do you put the w in before the dial? |
| 12:48.24 | ariel_ | afink: what do you use zap or dahdi? |
| 12:48.28 | webman | afink, no you can put the w before the number, or even in between the numbers |
| 12:49.57 | afink | zap on this box |
| 12:50.20 | afink | is each w a 1/2 second pause? |
| 12:50.23 | leifmadsen | yes |
| 12:50.27 | leifmadsen | 500ms |
| 12:50.36 | afink | awesome thanks |
| 12:50.43 | leifmadsen | Dial(DAHDI/g0/ww1726) |
| 12:51.04 | kaldemar | webman: SIP-SIP or IAX2-PRI working is not proof of it not being a firewall issue. |
| 12:51.28 | ariel_ | something like exten => s,8,Dial(zap/g4/www${EXTEN}) |
| 12:51.37 | ariel_ | is what i use for some e&m Wink setups |
| 12:52.14 | leifmadsen | ewwwww..... numbered priorities!!! |
| 12:52.14 | webman | kaldemar: I know that, but I know it's not firewall because I can see the packets arriving in asterisk (rtp debug) .... and on the client (tcpdump on the client machine) |
| 12:52.26 | TommyBotten | leifmadsen: You have a serious problem with them, don't you? |
| 12:52.36 | leifmadsen | I really really do |
| 12:52.48 | leifmadsen | they are so ridiculously outdated and useless, that no one should be using them |
| 12:52.54 | TommyBotten | Well.. understandable really. |
| 12:53.04 | leifmadsen | very much understandable |
| 12:53.13 | ariel_ | leifmadsen: there still used in allot of systems and until all the older 1.2 and 1.09 setups I have, I can't remove them all. |
| 12:53.15 | TommyBotten | True... but that goes for the dialplan language itself, If I've understood correctly |
| 12:53.16 | leifmadsen | numbered priorities are so Asterisk 1.0 |
| 12:53.27 | TommyBotten | dialplan vs. AEL |
| 12:53.32 | jaytee | yeah, it's so much fun to make a change, insert a new line and then renumber all the following priorities....just soooo much fun |
| 12:53.41 | leifmadsen | jaytee: I love doing that! |
| 12:53.48 | TommyBotten | hehehe |
| 12:53.52 | jaytee | :-) |
| 12:53.58 | leifmadsen | AEL *is* dialplan you realize right? |
| 12:53.59 | TommyBotten | using vim, thats actually quick |
| 12:54.19 | leifmadsen | all AEL does is convert from one syntax to dialplan |
| 12:54.30 | webman | don't forget to renumber the "jumps" +101 :) |
| 12:54.38 | TommyBotten | Ah... thats too bad |
| 12:54.48 | leifmadsen | yet another remnant that is unnecessarily restricting |
| 12:55.00 | TommyBotten | AEL? |
| 12:55.11 | TommyBotten | or the +101s? |
| 12:55.17 | leifmadsen | I could either match on just BUSY (+101), or I could use DIALSTATUS and match on many status |
| 12:55.19 | leifmadsen | +101 |
| 12:55.31 | leifmadsen | statii? |
| 12:55.36 | leifmadsen | statuses? :) |
| 12:55.41 | leifmadsen | (the latter) |
| 12:55.56 | TommyBotten | statuses I think :p |
| 12:55.59 | leifmadsen | hehe |
| 12:56.03 | leifmadsen | english is fun |
| 12:56.14 | leifmadsen | (and an unforgiving bitch) |
| 12:56.32 | TommyBotten | Nah... German is way worse |
| 12:56.38 | TommyBotten | Hope I'm not insulting anyone here ;) |
| 12:56.43 | leifmadsen | apparently I need to write an article on how using priority numbers and +101 kills kittens |
| 12:56.52 | TommyBotten | Do that |
| 12:56.54 | TommyBotten | Please |
| 12:57.00 | ariel_ | kills kittens.... wow |
| 12:57.01 | TommyBotten | Also include why 3cx is evil |
| 12:57.05 | leifmadsen | I just might.... if I can find the time |
| 12:57.08 | leifmadsen | ariel_: :) |
| 12:57.13 | webman | if the missing audio is the audio from the sip client to the PRI caller, and asterisk shows RTP being received from the SIP client, what reasons are there for the audio not getting sent on the PRI ? |
| 12:57.22 | leifmadsen | TommyBotten: I haven't used it yet... (and probably won't) |
| 12:57.44 | ariel_ | webman: incorrect codec or issues with transcoding? |
| 12:58.26 | ariel_ | I got to get back to doing some dynamic ivr menu's in a new 1.6 setup we are working on....bbl |
| 12:58.37 | webman | ariel: using g729, but there is plenty of license available.... and why can you transcode from pri to sip but not from sip to pri ? |
| 12:59.37 | leifmadsen | ariel_: have fun! |
| 12:59.43 | leifmadsen | (dynamic IVRs are awesome) |
| 13:00.02 | leifmadsen | wonders if ariel_ is using func_odbc, func_curl, or some other method for the dynamic bits |
| 13:00.07 | ariel_ | having issues with checking how many times to loop, But I will figure it out. |
| 13:00.20 | ariel_ | reading directly from the db |
| 13:00.24 | leifmadsen | ariel_: let me know if you need any help, as I've done dynamic IVRs a couple times now |
| 13:00.25 | ariel_ | via the mysql command |
| 13:00.40 | leifmadsen | ariel_: I suggest you go to func_odbc if you can -- much more stable than the MySQL stuff |
| 13:00.48 | ariel_ | leifmadsen: have any sample scripts? |
| 13:00.50 | leifmadsen | (plus many less lines -- don't need to setup the connection in the DB, etc...) |
| 13:01.14 | ariel_ | we have a read on dnis that gets all the variable up front |
| 13:01.16 | leifmadsen | I don't have full sample scripts as they were for other clients (and I don't have the dialplan access anymore), but I'd be happy to review or answer any questions you have |
| 13:01.35 | ariel_ | leifmadsen: great t/y |
| 13:02.01 | leifmadsen | I still suggest func_odbc over MySQL -- in 1.6 you can specify separate read and write DSN's and such too |
| 13:02.02 | ariel_ | trying to bridge an old 1.09 system and bring it to 1.6 type of setup |
| 13:02.08 | leifmadsen | so you can always read from one DB, and wrote to another |
| 13:02.19 | leifmadsen | s/wrote/write |
| 13:02.32 | leifmadsen | fish-bulb... I love it! :) |
| 13:02.45 | TommyBotten | leifmadsen: What are the basics of dynamic IVRs? |
| 13:02.47 | ariel_ | ok will see how I can work on it that way. |
| 13:03.04 | TommyBotten | (The one sentence version) ;) |
| 13:03.10 | leifmadsen | TommyBotten: basically, to create a structure where all the "what do I want to do" is filled in from some dynamic source, such as a database |
| 13:03.29 | ariel_ | In my case I have to make the menu for ether 17 ships, or 22 and in a few 5 to 10 ships only in the menu for dialing. |
| 13:03.38 | leifmadsen | basically a pattern match, the catches the digits, and then looks up, "what does option 5 do?" and then performs that logic |
| 13:03.50 | TommyBotten | leifmadsen: Ah. Neat. |
| 13:04.05 | leifmadsen | I should write a book or something... |
| 13:04.21 | TommyBotten | leifmadsen: You should. "The book" is a bit too basic |
| 13:04.33 | leifmadsen | well, it was always meant as an intro, not a bible |
| 13:04.50 | ariel_ | is the func_odbc faster then getting all the variables upfront via mysql? |
| 13:04.58 | leifmadsen | it's not any slower |
| 13:05.10 | ariel_ | but it goes and does multi reads |
| 13:05.10 | leifmadsen | ariel_: like, how are you doing it via MySQL? |
| 13:05.22 | leifmadsen | func_odbc in 1.6 has "multirow" |
| 13:05.37 | ariel_ | I driect calls to one inbound and do the connection and lookup based on the dnis |
| 13:05.37 | leifmadsen | so you can loop through all the values after they are read into memory |
| 13:05.51 | leifmadsen | ariel_: yep, you could do that with func_odbc |
| 13:06.02 | ariel_ | yes using the new 1.6 __name setups |
| 13:06.16 | leifmadsen | Set(HASH(ivr)=${ODBC_LOOKUP(${DNIS})}) |
| 13:06.18 | leifmadsen | something like that |
| 13:06.52 | leifmadsen | that is the single row lookup method |
| 13:07.09 | leifmadsen | or: Set(ARRAY(col1,col2,col3)=${ODBC_LOOKUP(${DNIS})}) |
| 13:07.12 | ariel_ | going to think about this, I am doing this with 3 different server that lookup to 2 main mysql that replicate themselfs |
| 13:07.22 | leifmadsen | yes, I do that as well |
| 13:07.24 | leifmadsen | ariel_: use pen |
| 13:07.29 | leifmadsen | then connect to 127.0.0.1 |
| 13:07.38 | leifmadsen | it'll control the failover between the replicated servers |
| 13:07.42 | leifmadsen | MySQL master-master setup right? |
| 13:07.55 | leifmadsen | or master-slave? |
| 13:07.59 | ariel_ | yes it's on a seperate systems |
| 13:08.08 | ariel_ | master-slave |
| 13:08.21 | leifmadsen | ok, so just setup func_odbc.conf to write and read from the appropriate servers |
| 13:08.29 | leifmadsen | you can do all your reads from the read-only servers if you want |
| 13:08.33 | leifmadsen | then write to just the master |
| 13:08.53 | leifmadsen | just setup the different DSN's |
| 13:09.23 | leifmadsen | asterisk can also failover between the slaves if one goes down |
| 13:10.13 | *** join/#asterisk flashnet (n=fdfdsf@200-127-214-8.cab.prima.net.ar) |
| 13:10.17 | webman | after re-explaining my one way audio problem, I suddenly had an "light bulb" moment.... |
| 13:10.26 | leifmadsen | fish bulb! |
| 13:10.34 | leifmadsen | what was it? |
| 13:10.41 | webman | disabling the jitter buffer in the chan_dahdi.conf and it is working perfectly now !! |
| 13:10.42 | ariel_ | is heading for his lab. need to setup 3 asterisk boxes and a seperate mysql system to start doing some testing... bbl |
| 13:11.28 | leifmadsen | ariel_: peas out |
| 13:12.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 13:16.24 | webman | in fact, I can enable the jb in chan_dahdi as long as jbimpl = fixed it works fine, but if jbimpl = adaptive then I get the one way audio problem |
| 13:17.42 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 13:18.33 | afink | could someone please help me get rid of this message on the cli? [Jul 28 08:17:38] WARNING[26321]: chan_dahdi.c:1774 dahdi_enable_ec: Unable to enable echo cancellation on channel 22 (No such device) |
| 13:19.07 | afink | I have hardware ec and don't think I should be getting this. Obviously I have something configured wrong somewhere |
| 13:19.13 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:21.54 | tzafrir_laptop | afink, what version of asterisk? dahdi/zaptel? |
| 13:21.56 | leifmadsen | webman: ya, the adaptive jb tends to be..... not working :) |
| 13:22.24 | afink | tzafrir_laptop: Asterisk 1.4.25.1, dahdi, PRI |
| 13:22.35 | tzafrir_laptop | what version of dahdi? |
| 13:22.41 | tzafrir_laptop | also: lsdahdi | grep 22 |
| 13:23.21 | afink | ^^ it shows up on which ever channel being used |
| 13:23.31 | afink | dahdi linux 2.2.0 |
| 13:24.19 | afink | lsdahdi: http://pastebin.com/m51325493 |
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| 13:28.05 | afink | is it as easy just taking echocancel=yes out of chan_dahdi.conf? |
| 13:28.19 | tzafrir_laptop | afink, what happens if you add 'echocanceller=mg2,1-<whatever>' |
| 13:28.34 | tzafrir_laptop | to /etc/dahdi/system.conf |
| 13:28.40 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-533e40e87d5b65a1) |
| 13:28.46 | tzafrir_laptop | fix the range of channels accordingly |
| 13:29.12 | tzafrir_laptop | and then run: dahdi_cfg |
| 13:29.17 | tzafrir_laptop | and try the call again |
| 13:29.28 | afink | ill find out in a minute, waiting for asterisk to stop gracefully |
| 13:29.53 | tzafrir_laptop | btw: if you want to be sure that the software EC is not used (that the hardware EC is used instead) |
| 13:30.10 | afink | ^^ i do |
| 13:30.27 | *** join/#asterisk s14ck (n=s14ck@190-76-126-155.dyn.movilnet.com.ve) |
| 13:30.40 | tzafrir_laptop | use the echo canceller module "jpah" (instead of "mg2") |
| 13:31.00 | *** join/#asterisk maverick (n=maverick@83.240.229.38) |
| 13:31.06 | tzafrir_laptop | it's a "echo canceller" that messes up the audio badly |
| 13:31.20 | maverick | <PROTECTED> |
| 13:31.36 | maverick | and the call hang up |
| 13:31.41 | maverick | why ? |
| 13:34.27 | AlmightyOatmeal | leifmadsen: the point of me going to polycom is to avoid that.. i hated my cisco experience |
| 13:34.42 | leifmadsen | all phones should be configured from a central location though |
| 13:34.56 | leifmadsen | IMHO |
| 13:35.20 | afink | tzafrir_laptop: where do I put the jpah? in system.conf |
| 13:35.24 | AlmightyOatmeal | true, i can see the benefit to that |
| 13:36.37 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:37.34 | tzafrir_laptop | afink, right, instead of 'mg2' |
| 13:37.47 | afink | ok -> system.conf: echocanceller=jpah,1-23 |
| 13:37.52 | tzafrir_laptop | yes |
| 13:37.55 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 13:37.57 | afink | cool thank you |
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| 13:44.37 | afink | tzafrir_laptop: when I did that things got all out of whack. I wonder if my hardware ec isn't working? It sounded horrible. I changed it back to mg2 and it is ok now. but that seems really strange |
| 13:45.04 | tzafrir_laptop | so the software EC was actually in effect |
| 13:45.12 | afink | yes |
| 13:45.14 | tzafrir_laptop | time to check your settings |
| 13:45.28 | tzafrir_laptop | as for that, I'm not well familiar |
| 13:47.45 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:49.54 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
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| 13:50.59 | lowtek | Hey guys, anyone try compiling and have success running asterisk 1.4 + Zaptel + libpri on an ARM processor? |
| 13:53.17 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 13:54.14 | Kobaz | what's a good way to count the channels in use for a given iax/sip peer |
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| 13:57.24 | sysreq | Kobaz: you can add his channels to a given group with Set(GROUP()=mygroup) and then use the GROUP_COUNT() function to get the number of channels that are part of that group (ie. ${GROUP_COUNT(mygroup)}) |
| 13:57.36 | Psychobilly | lowtek yes it works quite good with some uclibc based systems i tried |
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| 13:59.36 | Kobaz | k |
| 13:59.55 | maverick | <PROTECTED> |
| 13:59.56 | maverick | and the call hang up |
| 13:59.58 | maverick | why ? |
| 14:00.19 | Kobaz | sysreq: can you do group= for sip/iax peers... i dont remember |
| 14:01.47 | tzafrir_laptop | Kobaz, no |
| 14:02.18 | Kobaz | k |
| 14:02.20 | tzafrir_laptop | lowtek, basically, yes |
| 14:03.11 | tzafrir_laptop | but see also https://issues.asterisk.org/view.php?id=15017 , which I didn't have time to follow-up |
| 14:03.37 | tzafrir_laptop | oh, and: go for asterisk 1.6.x , dahdi and libpri :-) |
| 14:04.24 | tzafrir_laptop | or basically just use the stuff from debs if you isntall debian nad let me know if you have any problem |
| 14:06.41 | Kobaz | [Jul 28 14:06:30] WARNING[4980]: chan_iax2.c:1141 iax2_sched_replace: Unable to cancel schedule ID 650411. This is probably a bug (chan_iax2.c: iax2_sched_replace, line 1141). |
| 14:06.45 | Kobaz | mmm |
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| 14:20.32 | Alfio | hi i want to know how i can play an ivr two times and then go to hangup |
| 14:21.13 | afink | http://www.voip-info.org/wiki/view/Asterisk+tips+IVR+menu |
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| 14:36.02 | imcdona | can someone explain what this means "exten => s,1+998" |
| 14:36.08 | leifmadsen | Alfio: create a loop.... |
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| 14:36.15 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 14:36.22 | leifmadsen | imcdona: means priority 999 |
| 14:36.41 | imcdona | why not just put priority 999? |
| 14:36.56 | leifmadsen | in that case, I have no idea |
| 14:37.02 | maverick | someone can help me ? |
| 14:37.06 | leifmadsen | why use that method at all, I have no idea |
| 14:37.11 | leifmadsen | that is a very old way of doing things |
| 14:37.21 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:37.40 | imcdona | thanks leif |
| 14:38.54 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:39.00 | Alfio | leifmadsen hi |
| 14:39.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:39.07 | leifmadsen | hello |
| 14:39.17 | Alfio | you wrote create a loop |
| 14:39.17 | Alfio | ? |
| 14:39.22 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:39.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.22 | leifmadsen | yes I did. |
| 14:39.31 | Alfio | how |
| 14:39.50 | leifmadsen | use While(), or Goto() plus a counter |
| 14:39.57 | leifmadsen | Set(COUNTER=$[${COUNTER} + 1]) |
| 14:40.16 | leifmadsen | it's pretty straight forward |
| 14:40.38 | Alfio | ok, whe the counter is greater than 2 goto hangup |
| 14:40.40 | Alfio | right |
| 14:40.41 | Alfio | ? |
| 14:40.44 | leifmadsen | yes |
| 14:40.46 | [TK]D-Fender | [10:35]<imcdona>can someone explain what this means "exten => s,1+998" <- means I hope you can dial apha-numberic |
| 14:41.07 | [TK]D-Fender | imcdona: Not legit |
| 14:41.20 | Alfio | leifmadsen thx |
| 14:41.24 | Alfio | i will try it |
| 14:43.14 | leifmadsen | [TK]D-Fender: huh? |
| 14:43.32 | [TK]D-Fender | leifmadsen: Last I checked dialplan priorities don't do math |
| 14:43.40 | leifmadsen | they do |
| 14:44.04 | [TK]D-Fender | leifmadsen: definitely news to me. This documented somewhere? |
| 14:44.13 | leifmadsen | possibly |
| 14:44.28 | [TK]D-Fender | leifmadsen: Because contextually speaking he asked this in #freepbx : |
| 14:44.30 | [TK]D-Fender | [10:39]<imcdona>anyone know why freepbx does this "exten => s,1+998(record),MixMonitor(${MIXMON_DIR}${CALLFILENAME}.${MIXMON_FORMAT},,${MIXMON_POST})" |
| 14:44.46 | [TK]D-Fender | leifmadsen: Being GUI created "bleh" I wouldn't trust much up front |
| 14:44.56 | *** join/#asterisk iewebguy (n=mark@65.19.81.253) |
| 14:44.59 | [TK]D-Fender | leifmadsen: And it has a label and that'd be the only reason I'd trust it might get called |
| 14:45.03 | leifmadsen | you can do: hello,6+101,Voicemail(100@default,b) |
| 14:45.04 | [TK]D-Fender | leifmadsen: Just be default |
| 14:45.19 | [TK]D-Fender | leifmadsen: wierd |
| 14:45.20 | leifmadsen | (if you were doing old style +101 jumping) |
| 14:45.32 | [TK]D-Fender | leifmadsen: ghastly. |
| 14:45.38 | leifmadsen | it is a totally outdated way of doing priority jumping |
| 14:45.45 | [TK]D-Fender | leifmadsen: the worst |
| 14:46.00 | leifmadsen | I'm writing an article today on why priority numbers and +101 jumping kill kittens |
| 14:46.13 | thehar | anyone mind giving me some FFA insight? |
| 14:46.29 | leifmadsen | thehar: that was not a useful question |
| 14:46.46 | thehar | fax for asterisk. hehe |
| 14:46.52 | leifmadsen | I know what FFA is |
| 14:46.52 | thehar | i have a pastebin.. let me get |
| 14:46.59 | *** join/#asterisk fernandojdk (n=fernando@189.88.64.188) |
| 14:47.03 | leifmadsen | thehar: my answer would be, to your question, "yes, I'm sure someone does." |
| 14:47.15 | fernandojdk | hi all |
| 14:47.25 | leifmadsen | well hello there |
| 14:47.29 | fernandojdk | leifmadsen: can you help with a issue? |
| 14:47.53 | thehar | http://pastebin.ca/1510024 |
| 14:48.26 | leifmadsen | fernandojdk: since I have absolutely no idea what your issue is, I have no idea. |
| 14:48.43 | fernandojdk | leifmadsen: is my scenario: I have a * with Realtime for peers and users. I would like to register some users with multiple domains. example: 1001@microsoft.com |
| 14:48.56 | leifmadsen | ok... |
| 14:49.01 | fernandojdk | leifmadsen: and another user registration with 1002@ibm.com |
| 14:49.29 | fernandojdk | but, i receive de message: Registration for '1001@...' failed: Not local domain |
| 14:49.45 | fernandojdk | leifmadsen: have u any idea? |
| 14:50.15 | leifmadsen | fernandojdk: you need to add domain= to sip.conf |
| 14:50.26 | leifmadsen | read the section regarding domains in the sip.conf.sample file |
| 14:50.42 | fernandojdk | yes, i did this |
| 14:50.51 | thehar | leifmadsen: mostly i keep getting res_fax.c:1792 acf_faxopt_write: channel 'Zap/32-1' set FAXOPT(filename) to '1248792582748099.tif' is unhandled! on setting a filename for inbound fax |
| 14:50.57 | fernandojdk | but, i have 100 domains or more |
| 14:51.42 | leifmadsen | fernandojdk: then you should create a script to generate the domain= lines I guess with #exec |
| 14:51.43 | fernandojdk | i like that * dont verify the domain, and only username, secret |
| 14:51.48 | leifmadsen | or some other method of loading the domains into asterisk |
| 14:52.01 | leifmadsen | turn off domain support then |
| 14:52.09 | fernandojdk | how? |
| 14:52.23 | leifmadsen | read the sip.conf.sample file |
| 14:52.27 | leifmadsen | heads off for a meeting |
| 14:52.40 | fernandojdk | ok, i'm see this now |
| 14:52.51 | fernandojdk | thanks for now |
| 14:52.57 | fernandojdk | i'm back |
| 14:52.59 | fernandojdk | rs |
| 14:53.48 | fernandojdk | i love you |
| 14:53.49 | fernandojdk | hahaha |
| 14:53.54 | fernandojdk | it's rigth |
| 14:54.05 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:54.05 | fernandojdk | leifmadsen: now is work fine |
| 14:54.20 | leifmadsen | amazing! documentation works! |
| 14:54.26 | thehar | heh |
| 14:54.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:55.35 | fernandojdk | yes, works |
| 14:55.38 | fernandojdk | thanks leifmadsen |
| 14:59.03 | Alfio | leifmadsen do i need to set the count in 2 if i want to times repeat? like Set(COUNT=2) and later Set(COUNT=$[${COUNT} + 1]) |
| 14:59.22 | Alfio | two* |
| 14:59.31 | leifmadsen | think about it.... |
| 14:59.41 | [TK]D-Fender | AlNo, you change what you compare the count to for dciding whether or not to loop |
| 14:59.42 | leifmadsen | you're taking the value of count, and incrementing it by one |
| 15:00.00 | leifmadsen | you want to increment until you hit the value you are checking for, and then you're going to do something when you hit that value |
| 15:00.09 | Alfio | yes when reach > 2 go to hang gup |
| 15:00.30 | leifmadsen | right |
| 15:00.36 | Alfio | ok |
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| 15:09.27 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 15:10.33 | DagMoller | there is a problem with video support in asterisk 1.6.1.1? it does not work, and the cleint aways show the message "No Compatible codecs" (on 1.6.0.9 works perfectly) |
| 15:11.12 | Chainsaw | DagMoller: It's not just that you compiled 1.6.1.1 with a different set of supported codecs? |
| 15:11.26 | DagMoller | Chainsaw, no, the same config |
| 15:11.50 | Chainsaw | DagMoller: Config, yes. I'm talking about how the daemon is compiled. |
| 15:12.20 | DagMoller | Chainsaw, as the same way of 1.6.0.9 |
| 15:14.56 | [TK]D-Fender | DagMoller: Go pastebin the failed call with SIP DEBUG enabled |
| 15:15.08 | [TK]D-Fender | DagMoller: Because nothing shoould have changed |
| 15:15.20 | DagMoller | ok |
| 15:15.26 | [TK]D-Fender | DagMoller: *'s codec support from those 2 are pretty much identical |
| 15:19.57 | *** join/#asterisk ArchGT (n=ArchGT@190.149.103.56) |
| 15:20.44 | [TK]D-Fender | ~pb |
| 15:20.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 15:25.09 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 15:25.54 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:26.09 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:27.14 | maverick | anyone with the knowledge to help me ? |
| 15:27.46 | leifmadsen | DagMoller: search issues.asterisk.org -- there is an open bug about video support in 1.6 |
| 15:29.12 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 15:30.21 | Alfio | leifmadsen www.pastebin.ca/1510067 |
| 15:30.28 | Alfio | thx that did the trick |
| 15:30.52 | leifmadsen | interesting |
| 15:30.55 | leifmadsen | how that works, I don't know |
| 15:31.03 | leifmadsen | your GotoIf() line is missing a closing brace |
| 15:31.11 | leifmadsen | and what an odd way of going about a simple problem |
| 15:31.32 | leifmadsen | you probably want WaitExten(), not Wait() |
| 15:32.39 | Alfio | leifmadsen ok i will check that |
| 15:33.45 | thehar | turns out that asterisk hated writing over nfs to our netapp for saving the tif file on fax |
| 15:36.35 | TrentCreek | I have a telemarketing pest who keep dialing in via SIP, but the call keeps getting transfer to ext. "s" What can I do to stop it? |
| 15:38.09 | [TK]D-Fender | TrentCreek: Who is it authing as? |
| 15:39.04 | TrentCreek | It's not...far as I know |
| 15:39.18 | leifmadsen | sounds like allowguest=no needs to be enabled |
| 15:39.21 | [TK]D-Fender | TrentCreek: Go look |
| 15:39.40 | TrentCreek | yeah, I thought I had that set |
| 15:40.09 | leifmadsen | 'sip show settings' |
| 15:40.38 | Alfio | leifmadsen www.pastebin.ca/1510076 look now |
| 15:41.25 | [TK]D-Fender | AlThat does not survive invalid dials |
| 15:41.31 | [TK]D-Fender | Alfio: That does not survive invalid dials |
| 15:41.36 | TrentCreek | Well this is not good: Allow unknown access: Yes |
| 15:41.45 | leifmadsen | TrentCreek: allowguest=no |
| 15:41.47 | leifmadsen | sip reload |
| 15:41.53 | [TK]D-Fender | And I wonder about Waitexten continuing priorities like that |
| 15:44.20 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:44.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:48.17 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 15:52.12 | elguero | Deeewayne: ping |
| 15:52.36 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 15:54.41 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 15:54.45 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 15:55.42 | Alfio | [TK]D-Fender www.pastebin.ca/1510083 look now |
| 15:56.54 | [TK]D-Fender | Alfio: Lots of mismated braces, and you aren't paying attention to where your Goto's lead |
| 15:57.02 | *** join/#asterisk fubada (n=notslak@ool-4b7faf03.static.optonline.net) |
| 15:57.05 | fubada | hello. |
| 15:57.14 | fubada | is it piss poor to use * stricly as sms gateway |
| 15:57.24 | fubada | sending our automated SMS messages |
| 15:58.15 | Alfio | <PROTECTED> |
| 15:59.57 | leifmadsen | fubada: asterisk can only send SMS in Europe (FYI) |
| 16:00.11 | fubada | oh |
| 16:00.14 | fubada | doh |
| 16:00.18 | leifmadsen | I don't know anyone who uses it just for an SMS gateway, but I don't see why not, assuming all the tools are available. |
| 16:00.22 | fubada | any idea what can send SMS in US |
| 16:00.35 | leifmadsen | no idea |
| 16:01.12 | fubada | http://www.kannel.org is O.S. sms gateway will give that a try first |
| 16:01.12 | fubada | thanks |
| 16:01.14 | *** part/#asterisk wwalker (n=wwalker@72.249.1.66) |
| 16:02.20 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 16:03.20 | *** join/#asterisk superbock (n=admin@mx-cln-1.netcanvas.com) |
| 16:03.27 | superbock | Hello all. |
| 16:03.36 | ehsjoar | Request for viewpoints regarding the voicemail application: Anybody feel that there should be a way to have the message deleted right away after it has been forwarded? Right now you'll have to push 7 to delete the message after it has been forwarded. I am thinking there could be something added to voicemail.conf to set the behavior here. Believe it or not, some people feel it is a pain to push the extra 7 :-) |
| 16:03.46 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 16:04.45 | superbock | I'm setting up a PRI in Italy, and incoming calls are working fine. But outgoing always fail with "Channel 0/1, span 1 got hangup, cause 44" right after the first ring. Any hint about this? |
| 16:04.46 | leifmadsen | ehsjoar: I'm not opposed to that idea |
| 16:04.56 | Slade- | why would asterisk limit itself to europe only for sms if other software can do it? |
| 16:05.22 | leifmadsen | Slade-: asterisk doesn't limit itself, the people who developed it were in europe, and thus that's what it supports |
| 16:05.32 | Slade- | ah |
| 16:05.32 | leifmadsen | it's not like someone said "fuck the US!" |
| 16:05.39 | ehsjoar | leifmadsen: Thanks for your feedback! |
| 16:05.43 | Slade- | oh i'm sure someones said that lots ;P |
| 16:05.49 | leifmadsen | (well, I'm sure someone did, but probably not in relation to SMS in Asterisk) |
| 16:06.21 | afink | tzafrir_laptop: found my problem with the hardware ec: VPMADT032 Failed! Unable to ping the DSP (2)! |
| 16:06.56 | jaytee | best business motto I've ever heard "We take the SH out of IT!" |
| 16:07.36 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 16:08.10 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 16:10.27 | TrentCreek | leifmadsen: thanaks for that help |
| 16:10.38 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:11.41 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 16:11.59 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 16:12.00 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:14.13 | Alfio | [TK]D-Fender www.pastebin.ca/1510094 i think i fixed |
| 16:15.38 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 16:16.07 | Alfio | [TK]D-Fender its playing the announce two times, when that happens it will hangup the line |
| 16:16.09 | jplank | anyone know if manxpower has found a job yet or not? |
| 16:18.06 | Qwell | jplank: ask him |
| 16:18.20 | jplank | I would if he was around |
| 16:18.33 | jplank | I have his email somewhere, and I'll dig it up in a few |
| 16:18.40 | Qwell | ~manxpower |
| 16:18.40 | infobot | well, manxpower is NOT an employee of Digium, but has been using Asterisk in production since late 2001. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org http://www.fnords.org/skillslist.html |
| 16:19.09 | jplank | aha, didn't know the bot knew that much info about him ;) |
| 16:19.25 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:20.03 | leifmadsen | ~qwell |
| 16:20.04 | infobot | qwell is, like, a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 16:20.09 | Qwell | mmhmm |
| 16:20.10 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:20.19 | leifmadsen | neato |
| 16:20.43 | *** part/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 16:20.43 | jplank | lol |
| 16:23.41 | cusco__ | hi |
| 16:23.43 | cusco__ | "require_mysql: Realtime table general@csr_queue: column 'time' cannot be type 'int(10) unsigned' (need char)" |
| 16:23.48 | cusco__ | what type must it be? |
| 16:24.02 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:25.11 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 16:25.44 | cusco__ | type char |
| 16:25.45 | cusco__ | ok |
| 16:25.46 | cusco__ | lol |
| 16:26.58 | grey-monkey | I'm doing a bulk provisioning of some polycom 330 phones. I reformatted then phones and now they are not connecting to the network. DHCP is on and everything should be back to factory settings. Has anyone else experienced this problem when reformatting? |
| 16:32.25 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
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| 16:33.32 | *** join/#asterisk errotan (n=errotan@5403E450.catv.pool.telekom.hu) |
| 16:33.55 | carrar | heh |
| 16:36.15 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:40.01 | *** join/#asterisk jareq (n=jarek@082139043136.radom.vectranet.pl) |
| 16:43.10 | *** join/#asterisk msetim (n=setim@189.32.44.21) |
| 16:43.17 | msetim | Hi :) What does I need to do to convert gsm to mp3 with lame? |
| 16:43.26 | leifmadsen | msetim: I'm wondering if you can do that with sox? |
| 16:43.33 | leifmadsen | as long as sox is compiled with mp3 support |
| 16:44.06 | Chainsaw | Otherwise you'd have to decode the GSM to plain PCM wave and then feed it to lame. |
| 16:44.21 | msetim | leifmadsen, I have it installed and I would use it without problem. I need to pass some kind of parameter to it work fine? |
| 16:44.21 | Chainsaw | Because it won't handle that directly. |
| 16:45.42 | *** join/#asterisk bbryant1 (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:45.59 | msetim | Chainsaw, oh... so it's my problem :) it doesn't work with gsm |
| 16:46.25 | Chainsaw | msetim: leifmadsen has some good ideas there. sox might be able to do it one go. |
| 16:46.36 | Chainsaw | +in |
| 16:47.00 | Chainsaw | A two-step process is always a bother, particularly when it comes to error handling & cleaning up temporary files. |
| 16:47.13 | msetim | I will try these ideas :) |
| 16:47.29 | msetim | leifmadsen, let me try and I go back to post results :-P |
| 16:48.52 | DagMoller | [TK]D-Fender, sip debug about video problems: http://asterisk.pastebin.com/m6cb9bf10 |
| 16:49.46 | [TK]D-Fender | Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x8000c (ulaw|alaw|h263) |
| 16:49.52 | [TK]D-Fender | DagMoller: P, non-P |
| 16:50.29 | DagMoller | [TK]D-Fender, ? |
| 16:51.12 | [TK]D-Fender | DagMoller: And before you mentioned SIP 488. ther is no 488 here |
| 16:52.11 | DagMoller | [TK]D-Fender, the same scenario works perfectly on asterisk 1.4 and 1.6.0.9 |
| 16:53.15 | [TK]D-Fender | DagMoller: I don't see a failure in that pastebin |
| 16:53.40 | DagMoller | [TK]D-Fender, but video does not work |
| 16:53.51 | [TK]D-Fender | DagMoller: I see you calling 2 playbacks, thats it. |
| 16:53.53 | DagMoller | the X-Lite show "No compatible codecs" |
| 16:54.15 | [TK]D-Fender | DagMoller: Actually.. not eveb. Just 1 |
| 16:54.49 | msetim | leifmadsen, By the way.. Do you know how convert gsm to pcm? Sox says that don't know 'pcm' format :( |
| 16:54.54 | [TK]D-Fender | DagMoller: Do a real test. This one does nothing except play 1 audio file and hangup |
| 16:55.05 | [TK]D-Fender | msetim: PCM? What for? |
| 16:55.22 | DagMoller | [TK]D-Fender, i doing an echo test for video test |
| 16:55.37 | [TK]D-Fender | DagMoller: There is no call to Echo () in there <---- |
| 16:55.42 | DagMoller | ok wait |
| 16:56.13 | msetim | [TK]D-Fender, I will convert GSM to PCM (using sox) and then feed it to lame to mp3 format... |
| 16:57.03 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:59.08 | DagMoller | [TK]D-Fender, http://asterisk.pastebin.com/m345ad66 |
| 16:59.48 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 17:01.14 | [TK]D-Fender | DagMoller: Do a PLAYBACK() between your answer() and Echo() |
| 17:03.28 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 17:03.29 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:03.34 | DagMoller | [TK]D-Fender, http://asterisk.pastebin.com/m65700a7e |
| 17:05.36 | [TK]D-Fender | DagMoller: Does audio echo? |
| 17:05.41 | DagMoller | yes |
| 17:06.03 | *** join/#asterisk Mango (n=Mango@S01060016b6853255.vc.shawcable.net) |
| 17:06.12 | [TK]D-Fender | DagMoller: Ok, not sure what to tell you on this one... |
| 17:06.48 | DagMoller | :/ |
| 17:07.31 | DagMoller | downgrade :( |
| 17:08.08 | Mango | Pardon me, but is this the correct way to do channel failover? Ie, if voipms is unavailable, use callcentric instead? |
| 17:08.09 | Mango | exten => NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms) |
| 17:08.09 | Mango | exten => NXXNXXXXXX,n,Dial(SIP/${EXTEN}@callcentric) |
| 17:08.09 | Mango | exten => NXXNXXXXXX,n,Hangup() |
| 17:09.35 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 17:10.49 | carrar | I'd probably check the status befor dialing again |
| 17:10.54 | Psychobilly | Mango maybe its better to use chanisavail() first to check the channel and then jump to the dial command of the available channel |
| 17:11.16 | carrar | ${DIALSTATUS} |
| 17:11.26 | Mango | Thanks. |
| 17:12.10 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:13.05 | Alfio | [TK]D-Fender is that the best way to create a two times playing ivr or there is a more easy way |
| 17:13.15 | Alfio | <PROTECTED> |
| 17:18.14 | *** join/#asterisk oglynn (n=me@173-9-102-81-NewEngland.hfc.comcastbusiness.net) |
| 17:18.44 | [TK]D-Fender | Alfio: You still broke the invalids.... |
| 17:19.12 | *** join/#asterisk many (n=many@dslb-088-067-166-204.pools.arcor-ip.net) |
| 17:19.34 | [TK]D-Fender | Psychobilly: chanisavail won't tell you the # is busy. |
| 17:19.45 | Alfio | how i can fix it |
| 17:19.46 | Alfio | ? |
| 17:20.11 | Psychobilly | [TK]D-Fender i think he was asking for sip peer availability |
| 17:20.18 | Alfio | because im sending the invalids to the first extension of the context |
| 17:20.22 | [TK]D-Fender | Alfio: Go pay attention to what you're doing. You clearly aren't looking at the flow step by step. |
| 17:21.41 | *** join/#asterisk propellerhead (n=yogurt2u@host62.200-117-118.telecom.net.ar) |
| 17:24.01 | oglynn | i am having a problem with context on incoming calls from a registered sip trunk config and error at http://pastebin.com/m54545f30 |
| 17:25.29 | [TK]D-Fender | oglynn: Go look at the actual SIP debug of the incoming call to what its matching. |
| 17:28.34 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:31.44 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:37.23 | oglynn | [TK]D-Fender: I added it to the pastebin http://pastebin.com/m7f9543eb |
| 17:37.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:42.42 | afink | is there any documentation on echocanceller=jpah? |
| 17:43.16 | afink | the guy at asterisk had never heard of it ;P |
| 17:43.29 | oglynn | [TK]-D-fender if i add include => partner1-incoming to [default] in extensions.conf it works. so i thought there was something wrong with my entry in sip.conf associating the context |
| 17:43.31 | afink | I meant digium |
| 17:43.37 | *** join/#asterisk fskrotzki (n=fskrotzk@mail.perspectivepartners.com) |
| 17:46.36 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:46.39 | *** join/#asterisk kc2tnk (n=fskrotzk@mail.perspectivepartners.com) |
| 17:50.06 | oglynn | anyone got any idea on why this inbound call is not ending up in the correct context? http://pastebin.com/m7f9543eb if I include the parner1-incoming in the default context it works |
| 17:51.24 | coppice | afink: what exactly is echocanceller=jpah supposed to be? |
| 17:52.01 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 17:52.14 | coppice | oh, I know. |
| 17:52.17 | [TK]D-Fender | oglynn: Only looking at the invite? crazy.. Look at the ENTIRE call attempt |
| 17:52.22 | afink | use only hardware ec |
| 17:52.34 | afink | and it does work |
| 17:52.52 | coppice | the hardware ECs have their own problems |
| 17:53.21 | afink | mine is working well now. I thought it had been for some time but setting that showed me that it wasn't working at all |
| 17:53.54 | Alfio | [TK]D-Fender http://pastebin.ca/1510189 can you please give it a look now |
| 17:54.06 | coppice | both Digium and Sangoma cards with the octasic EC chips have problems corrupting DTMF |
| 17:55.15 | afink | every time I see DTMF I think dial tone motha fu**** |
| 17:55.31 | jaytee | "pssst!, hey, I'll give ya fifty bucks if you drop 200hz for about 3 milliseconds" |
| 17:56.06 | [TK]D-Fender | Alfio: I'm wondering if you are loking at what you're doing at all or testing it. I'm prtty sure you aren't |
| 17:56.55 | errr | I am getting ready to plan a new dialplan for my system. I was wondering before I start if the #include directive supports globbing? where I can do #include peers/*.conf |
| 17:57.01 | oglynn | [TK]D-fender sorry i see an error now... No matching peer for '7815552000' from '150.150.150.150:5060' the 781number is a pstn callerid |
| 17:57.08 | Alfio | [TK]D-Fender i tested it and when i press an invalid extension it goes to loop label |
| 17:58.01 | Psychobilly | errr it does |
| 17:58.02 | [TK]D-Fender | Alfio: do a complete trace of timeouts & invalids. Then mix them |
| 17:58.07 | errr | Psychobilly: thank you |
| 18:02.31 | oglynn | [TK]D-Fender full call posted sorry if i am missing something silly |
| 18:02.32 | oglynn | http://pastebin.com/m4f07a817 |
| 18:03.29 | [TK]D-Fender | olgWell you don't have a matching peer. What more is there to say? |
| 18:03.52 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:04.27 | leifmadsen | anyone know of a simple script that will analyze existing CDR records (in the cdr-csv Master.csv file)? I just need it to pull back information for a particular phone to get the activity, calls, duration, etc... over a period of time |
| 18:04.47 | leifmadsen | I would probably create something myself if need be, but I'd rather not spend the time if I don't have to |
| 18:05.01 | oglynn | [TK]D-Fender i want to accept calls inbound to this number from any possible pstn source callerid. and want the calls to hit their own context (there will be a second number when I get the first one working) should i be configuring this differently? |
| 18:05.10 | ariel_ | leifmadsen: it's something that is really needed don't know of any good ones. |
| 18:05.36 | [TK]D-Fender | oglynn: You are masking things an I can't seew what looks like what. |
| 18:05.40 | leifmadsen | ya... |
| 18:05.47 | oglynn | ok |
| 18:06.03 | [TK]D-Fender | leifmadsen: Asterisk-stat? |
| 18:06.28 | leifmadsen | well, it seems to want to use a database |
| 18:06.36 | leifmadsen | I just want to analyze an existing .csv file |
| 18:07.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:07.37 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:08.15 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 18:09.26 | oglynn | [TK]D-Fender unchanged sip posted |
| 18:09.27 | [TK]D-Fender | leifmadsen: MS Excel :) |
| 18:10.39 | raden_work | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 18:10.49 | raden_work | it does exist |
| 18:11.05 | [TK]D-Fender | raden_work: then the running suer has no rights to it |
| 18:11.08 | [TK]D-Fender | user* |
| 18:11.14 | raden_work | im root |
| 18:11.20 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 18:11.34 | [TK]D-Fender | raden_work: Show us its running, etc |
| 18:11.45 | raden_work | what u mean ? |
| 18:11.51 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 18:12.19 | leifmadsen | ps aux | grep asterisk |
| 18:12.23 | guax | raden_work, are you shure asterisk is running? |
| 18:12.28 | lirakis | leifmadsen: what are you looking for specifically |
| 18:12.44 | leifmadsen | lirakis: I just want to get data on a particular users calling habits |
| 18:12.55 | raden_work | guax, it is now |
| 18:13.00 | [TK]D-Fender | NOW? |
| 18:13.01 | lirakis | leifmadsen: okay .. so not some thing "simple" using sed / awk |
| 18:13.03 | raden_work | i dont understand why it didnt start at boot |
| 18:13.05 | [TK]D-Fender | What about THEN? |
| 18:13.28 | guax | because your init scripts/configuration are not ok? |
| 18:13.34 | leifmadsen | lirakis: ya, I'll probably just end up writing a script to import the Master.csv into a database, and then run asterisk-stat against it or something |
| 18:13.37 | [TK]D-Fender | raden_work: Maybe you could tell us why you believe it should have started at boot. |
| 18:13.47 | leifmadsen | [TK]D-Fender: turn down the "jerk" |
| 18:13.50 | lirakis | leifmadsen: okay .. i may mess around a little bit - im a bit bored right now |
| 18:14.10 | leifmadsen | lirakis: haha, coolio :) let me know if you get somewhere, and I can pick up where you left off. I'm most familiar with php if that helps |
| 18:14.14 | [TK]D-Fender | leifmadsen: that was a perfectly legit question. |
| 18:14.35 | [TK]D-Fender | leifmadsen: And you could also set up an ODBC connection against the CSV and save the import I suppose |
| 18:14.52 | leifmadsen | I'm not entirely sure what you mean by that |
| 18:14.56 | [TK]D-Fender | leifmadsen: My haed-office does that for one of our shipping systems. |
| 18:15.14 | [TK]D-Fender | leifmadsen: or does Asterisk-Stat only work against MySQL? |
| 18:16.14 | [TK]D-Fender | raden_work: I am quite serious. What did you do to have 8 start on boot? |
| 18:16.17 | [TK]D-Fender | * |
| 18:16.20 | [TK]D-Fender | ack |
| 18:16.24 | lirakis | [TK]D-Fender: it does pgsql as well |
| 18:16.42 | [TK]D-Fender | lirakis: hrm.... should be more agnostic.. |
| 18:17.02 | lirakis | [TK]D-Fender: yeah - i think its kinda ancient frankly |
| 18:17.42 | [TK]D-Fender | lirakis: Well * CDR formats haven't changed so... |
| 18:17.53 | xpot-mobile | Question: anyone have any suggestions of what would cause faxes to be intermittent on a 50/50 fiber connection with <20ms latency? Any preferences of using SIP vs. IAX2 for fax? |
| 18:18.18 | lirakis | [TK]D-Fender / leifmadsen: my friend and i are working on a modular cdr format reporting system for a qos based routing system we are working on. Im sure we will open source some form of it for the * community |
| 18:18.56 | lirakis | leifmadsen: doesnt solve your "immediate" problem .. but maybe in a few months there will be some thing more generic and modern that asterisk-stat |
| 18:19.07 | raden_work | [TK]D-Fender, i added it to one of our scripts i thought /etc/init.d/boot.local |
| 18:19.17 | ariel_ | leifmadsen: quick question about the dynamic ivr's, if I use ARRAY to get the variable do they service jumps to other contexts? |
| 18:19.33 | [TK]D-Fender | raden_work: Show us |
| 18:19.41 | raden_work | im looking for the script |
| 18:19.55 | raden_work | that was the least of my issues but im looking |
| 18:20.16 | [TK]D-Fender | raden_work: You've developed more? |
| 18:20.16 | oglynn | to look at my issue from the start rather than my non working situation. If I want to register with a sip provider and then recieve incoming calls from any source number what *should* i be doing |
| 18:20.27 | leifmadsen | ariel_: explain? |
| 18:20.45 | [TK]D-Fender | oglynn: It isn't the source number that matters. It isn't authing the call. |
| 18:20.45 | leifmadsen | ariel_: all ARRAY() does is take the values of the columns of a single row, and save those values to channel variables |
| 18:20.50 | ariel_ | when we are forking between context like added the __name |
| 18:21.09 | Psychobilly | oglynn the s extension or somehting like _X. |
| 18:21.26 | leifmadsen | ariel_: I still dont' follow... you meant transitive variables? |
| 18:21.48 | ariel_ | leifmadsen: like when we pull in info our varibles with the set command adding them with __Name |
| 18:22.04 | leifmadsen | you mean in Asterisk, or something external to asterisk? |
| 18:22.11 | ariel_ | in asterisk 1.6 |
| 18:22.13 | leifmadsen | I still don't follow what you mean |
| 18:22.16 | leifmadsen | example? |
| 18:22.25 | leifmadsen | Set(__VARIABLE=value) ? |
| 18:22.25 | ariel_ | ok just a sec |
| 18:22.51 | leifmadsen | all ARRAY() is doing is setting channel variables |
| 18:22.56 | leifmadsen | the same rules that apply to all channel variabels apply here |
| 18:22.59 | lucasb | Mornin.... is anyone available to help figure out why my Linksys 962 MWI light isn't lit when a new voicemail is present? |
| 18:24.31 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 18:25.00 | [TK]D-Fender | lucasb: What does your peer entry say? |
| 18:25.04 | lucasb | I'm running Asterisk 1.4.22 and am not using any GUI |
| 18:25.07 | raden_work | [TK]D-Fender, nm i dont have it on this system i thought i put it on after the os reinstall but guess not |
| 18:25.14 | lucasb | One sec [TK]D-Fender I'll grab that |
| 18:25.36 | [TK]D-Fender | leifmadsen: See? Perfectly legit, and confirmed cause :) |
| 18:25.56 | lucasb | [TK]D-Fender, Asterisk does show the subscription when I do a "sip show subscriptions" at the CLI... |
| 18:25.59 | [TK]D-Fender | leifmadsen: this revelation will save wondering the next time. |
| 18:26.13 | lucasb | [TK]D-Fender, You want what I have in the sip.conf? |
| 18:26.18 | ariel_ | leifmadsen: exten => X.,n,Set(__custname=${cname}) this is how I am setting up the variable |
| 18:26.22 | [TK]D-Fender | lucasb: Thats what I asked for. |
| 18:26.58 | leifmadsen | ariel_: transitive variables. Try: Set(ARRAY(__COL1,__COL2,__COL3)=${ODBC_LOOKUP(${DNIS})}) (for example) |
| 18:27.20 | leifmadsen | all ARRAY() does is let you set multiple variables on a single line, and assign the values from something external, like ODBC |
| 18:27.20 | Qwell | infobot: coin |
| 18:27.21 | infobot | somebody said coin was flipped... heads or tails? |
| 18:27.21 | lucasb | [TK]D-Fender, http://www.pastebin.ca/1510214 |
| 18:27.31 | Qwell | infobot: heads |
| 18:27.34 | leifmadsen | infobot: that's what qwell was asking you |
| 18:27.41 | Qwell | stupid bot |
| 18:27.43 | leifmadsen | ~coin |
| 18:27.44 | infobot | from memory, coin is flipped... heads or tails? |
| 18:27.46 | Qwell | glares at leifmadsen |
| 18:27.49 | leifmadsen | booo |
| 18:27.54 | leifmadsen | stares at Qwell |
| 18:28.31 | [TK]D-Fender | lucasb: PB "voicemail show users", and the SIP debug of your attempts |
| 18:28.53 | *** join/#asterisk Psychobilly (n=moi@adsl117-205.kln.forthnet.gr) |
| 18:29.00 | oglynn | [TK]D-Fender when you say it is not authing the call. I changed the insecure to very and it still does not work i dont understand why it works if i include my partner1-incoming context in the default context but is not picking the context otherwise? |
| 18:29.39 | [TK]D-Fender | oglynn: It doesn't match the peer so it isn't using the context specified by the peer. What don't you get? |
| 18:29.40 | raden_work | ok now i have the boot issue fixed |
| 18:29.49 | ariel_ | leifmadsen: great t/y |
| 18:29.51 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 18:29.54 | [TK]D-Fender | raden_work: One down... so whats your new preseeing problem? |
| 18:30.02 | [TK]D-Fender | pressing* |
| 18:30.06 | lucasb | [TK]D-Fender, http://pastebin.ca/1510218 for the CLI command |
| 18:30.15 | fubada | i need an SMS gateway, to send stuff to my member database. is asterisk a good choice |
| 18:30.18 | fubada | im in USA |
| 18:30.27 | leifmadsen | fubada: didn't I already answer this question? |
| 18:30.29 | raden_work | I can dial out have bi-directional communication dialing in the phone rings i pickup no one is there |
| 18:30.29 | fubada | s/an/a |
| 18:30.30 | lucasb | [TK]D-Fender, How do I see the sip debug of the attempts? |
| 18:30.36 | raden_work | and the origination phone keeps ringing |
| 18:30.47 | raden_work | lemee paste what ITSP gave me to work with |
| 18:30.47 | fubada | leif: you did but i want another opinion is that cool |
| 18:30.48 | [TK]D-Fender | fubada: No. |
| 18:31.00 | leifmadsen | my answer wasn't an opinion, it was a fact. |
| 18:31.11 | leifmadsen | Asterisk doesn't support SMS in North America. |
| 18:31.16 | fubada | I may implement voice down the road, thats why im inquiring about * |
| 18:31.35 | fubada | what is required to send SMS in North America |
| 18:31.45 | [TK]D-Fender | fubada: Down the road * may be capable of matter-replicated kittens. Ask us then. Right now the answer is "NO" |
| 18:31.54 | fubada | is it likely impossible with other open source tools |
| 18:31.56 | leifmadsen | http://www.voip-info.org/wiki/view/SMS+Service+in+the+USA |
| 18:31.57 | oglynn | [TK]D-Fender I need to register and my registration works. i want to accept calls from to my registered number what am i getting wrong or how would you do this? |
| 18:32.06 | carrar | I can do partial matter-replicated kittens |
| 18:32.12 | carrar | it's messy |
| 18:32.13 | [TK]D-Fender | oglynn: Fix your peer |
| 18:32.18 | [TK]D-Fender | carrar: Nasty |
| 18:32.37 | raden_work | 'http://pastebin.com/m69eae7ba |
| 18:32.57 | oglynn | [TK]D-Fender ok what should i try changing? |
| 18:33.30 | leifmadsen | btw: I got that link by typing "asterisk sms" into google |
| 18:33.41 | fubada | hey thanks you're awesome |
| 18:33.53 | leifmadsen | ~leifmadsen |
| 18:33.55 | *** join/#asterisk coppice (n=chatzill@193.194.17.210.dyn.pacific.net.hk) |
| 18:34.01 | leifmadsen | ~blitzrage |
| 18:34.02 | infobot | i heard blitzrage is a super cool fellow |
| 18:34.21 | leifmadsen | infobot: leifmadsen is blitzrage |
| 18:34.22 | infobot | leifmadsen: okay |
| 18:35.41 | raden_work | [TK]D-Fender, any idea |
| 18:36.17 | [TK]D-Fender | oglynn: I never got to see rela debug, with real configs. |
| 18:36.28 | [TK]D-Fender | oglynn: I'm not going to go guessing blind |
| 18:37.11 | [TK]D-Fender | raden_work: PB a failed call with SIP debug enabled |
| 18:38.54 | [TK]D-Fender | lucasb: not sure on yours... |
| 18:39.07 | raden_work | yes sir |
| 18:39.12 | lucasb | [TK]D-Fender, Just getting a PB of the sip debug... |
| 18:41.26 | raden_work | [TK]D-Fender, can i send output to a file :( ? ???? |
| 18:41.28 | raden_work | about 3 pages |
| 18:41.46 | [TK]D-Fender | raden_work: copy& paste, just like the rest of us. |
| 18:42.17 | lucasb | [TK]D-Fender, http://pastebin.ca/1510232 for the sip debug |
| 18:42.58 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:43.55 | [TK]D-Fender | lucasb: Use "sip relaod". Phones should not poll * for messages. * pushes NOTIFY's to phones for MWI |
| 18:43.59 | [TK]D-Fender | reload* |
| 18:44.45 | Alfio | [TK]D-Fender http://pastebin.ca/1510233 i finally get it |
| 18:44.53 | Alfio | check out please |
| 18:45.40 | [TK]D-Fender | Alfio: Wow, after all this time hacking your way you jsut go and rip the code right out of the apge you were given HOURS ago... |
| 18:46.22 | *** join/#asterisk bsteele (n=bbsteele@lb1.ihets.org) |
| 18:47.08 | Alfio | [TK]D-Fender the way i was using was a little confuse to me |
| 18:47.21 | bsteele | webmin.tgz from asterisk.org downloads.. Is this the server GUI or the Asterisk web interface? |
| 18:47.51 | Alfio | [TK]D-Fender i couldnt handle the timeouts by that way |
| 18:47.56 | *** join/#asterisk DigitalDaz21 (n=digitald@78-105-169-114.zone3.bethere.co.uk) |
| 18:48.26 | [TK]D-Fender | Alfio: Yes, because you weren't looking where it jumped to which reset the coutner every time <- |
| 18:48.34 | bsteele | I've installed PBX Manager from ThirdLane but there are license restrictions. Does anybody know of a better (free) Web GUI for asterisk? |
| 18:48.39 | [TK]D-Fender | bsteele: Neither |
| 18:48.43 | raden_work | http://pastebin.com/d1621b4b5 |
| 18:48.54 | [TK]D-Fender | bsteele: the Webmint hing is ancient garbage that was never really sueful for anything |
| 18:48.59 | bsteele | hehe |
| 18:49.00 | bsteele | ok |
| 18:49.24 | bsteele | The one I have installed won't let me auto deploy more than 4 phones.. and may limit me to do that which is a no-go |
| 18:49.54 | bsteele | Are you aware of a better Web interface? |
| 18:50.08 | DigitalDaz21 | for linux? |
| 18:50.29 | bsteele | yes |
| 18:50.38 | [TK]D-Fender | raden_work: Looks like you have an Aastra that isn't authing right as well |
| 18:50.38 | DigitalDaz21 | freepbx? |
| 18:50.39 | bsteele | centos 5.3 / asterisk 1.4 |
| 18:50.40 | Alfio | [TK]D-Fender thx for the advise and your patient |
| 18:50.48 | bsteele | digital: yes |
| 18:51.13 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:51.24 | raden_work | [TK]D-Fender, can you tell me exactly what the aastra isnt doing ???? |
| 18:51.32 | raden_work | cause i have a feeling it is something todo with the phone |
| 18:51.44 | bsteele | Kinda stuck trying to get two test phones to dial each other, right now with installing that App it overwrote my default settings and its not catching a context |
| 18:51.46 | raden_work | i can dial out but no inbound the phone does ring which i find odd |
| 18:51.52 | DigitalDaz21 | Don't you like freepbx? |
| 18:51.53 | [TK]D-Fender | raden_work: also : No user '17152815035' in SIP users list ---- No matching peer for '17152815035' from '204.11.192.37:5060 |
| 18:52.11 | [TK]D-Fender | raden_work: it isn't matching your peer so reinvites are NOT being disabled. great reason for 1-way audio. |
| 18:52.29 | bsteele | When I try to do: console dial 1000 I get this error |
| 18:52.31 | bsteele | chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 18:52.32 | bsteele | <PROTECTED> |
| 18:52.36 | [TK]D-Fender | bsteele: Then go fix your configs back the way they were |
| 18:52.41 | bsteele | yeah i have it all saved |
| 18:52.56 | [TK]D-Fender | bsteele: And chan OSS has no access to your DSp & fails |
| 18:53.12 | bsteele | I'm so clueless when it comes to setting up channels |
| 18:53.18 | [TK]D-Fender | bsteele: ... |
| 18:53.20 | [TK]D-Fender | ~book |
| 18:53.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:53.22 | [TK]D-Fender | ^^^^^^^ |
| 18:53.26 | bsteele | my boss tells me, " Just install the WebGUI" your making this too hard |
| 18:53.51 | bsteele | The book mainly focuses on systems with a hardware card |
| 18:53.57 | raden_work | LMAO |
| 18:54.03 | bsteele | i think |
| 18:54.07 | bsteele | if i'm wrong correct me |
| 18:54.08 | [TK]D-Fender | bsteele: Well are you going to install a UI like your boss asked? |
| 18:54.22 | bsteele | That was my first question coming in here ;) |
| 18:54.24 | lowtek | Any UI will limit your flexibility and options. |
| 18:54.30 | [TK]D-Fender | yup |
| 18:54.33 | WindowsUser | you dont need 50 pages to tell you how to check permissions on /dev/dsp :o |
| 18:54.43 | bsteele | I'm new to linux |
| 18:54.46 | bsteele | please be respectful |
| 18:54.47 | bsteele | intern |
| 18:54.58 | [TK]D-Fender | WindowsUser: True. And using it in the first place is usually a wasted test |
| 18:55.19 | WindowsUser | at least hes not testing asterisk with a winmodem? |
| 18:55.44 | [TK]D-Fender | bsteele: You're first time on the road in driving school shouldn't be a ferrari in a crowded street <- |
| 18:55.44 | bsteele | I have basic network admin classes that involve trivial tasks.. with a GUI (ubuntu) |
| 18:56.06 | [TK]D-Fender | bsteele: Ubuntu... network admin? Ew.. |
| 18:56.12 | raden_work | [TK]D-Fender, http://www.voltarclamps.com/sip2.png |
| 18:56.15 | bsteele | eh, more setting up a VM |
| 18:56.19 | bsteele | and playing with grep, and the like |
| 18:56.27 | bsteele | and adding users, etc via terminal |
| 18:56.36 | bsteele | basic admin |
| 18:56.50 | lowtek | bsteele: Are you deploying a production Asterisk system? |
| 18:56.57 | bsteele | Eventually, yes. |
| 18:56.57 | lowtek | bsteele: Rather, are you tasked with doing this? |
| 18:57.01 | WindowsUser | so you'd know how to add a user that expires in a week? |
| 18:57.15 | bsteele | This system must be in production by September |
| 18:57.49 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:57.52 | bsteele | So please, don't dog me as I am trying to do my job |
| 18:58.11 | *** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:58.30 | [TK]D-Fender | [14:54]<[TK]D-Fender>bsteele: Well are you going to install a UI like your boss asked? |
| 18:58.38 | WindowsUser | I'm an equal opportunity sasser |
| 18:59.00 | bsteele | Do you have any opinions on Web GUI's for configuring asterisk? |
| 18:59.03 | iewebguy | bsteele good luck with your work. asterisk has a fun but steep learning curve in places |
| 18:59.05 | bmoraca | bsteele: read the book. then decide whether a gui is what you want. if it is, make sure you understand the drawbacks, etc. |
| 18:59.14 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:59.33 | bsteele | My boss apparently doesn't know anything about actually coding this and tells me install GUI. |
| 18:59.38 | [TK]D-Fender | bsteele: As you've been told. Several work, but you give up control to the concepts it implements |
| 18:59.42 | bsteele | I am trying to understand what is happening |
| 19:00.09 | bsteele | So, technically I need both. Mainly because my boss wnats the web interface for maintenance / admin for 'the next guy.' |
| 19:01.08 | [TK]D-Fender | bsteele: Then trash what you have now and just install AsteriskNOW and be done with it |
| 19:01.50 | bsteele | Are there any drawbacks to NOW? |
| 19:02.12 | bmoraca | yes |
| 19:02.48 | [TK]D-Fender | bsteele: Have you not been listening? |
| 19:02.59 | [TK]D-Fender | bsteele: Its been repeated several times now. |
| 19:03.01 | bsteele | Ok, thanks for your previous help fender |
| 19:03.27 | [TK]D-Fender | bsteele: go with a GUI and you are stuck in their cookie-cutter way of doing things. Odds are if you don't like it, you're SOL |
| 19:03.34 | bsteele | You are obviously just being a dick by this point |
| 19:03.49 | bsteele | All I asked was are there any drawbacks to something COMPLETELY different than I was talking about |
| 19:03.53 | bsteele | and accusing me of not listening |
| 19:04.00 | [TK]D-Fender | bsteele: No, I'm being extremelyclear (not that I wasn't before,m but you didn't seem to catch on) |
| 19:04.17 | bsteele | We were speaking Web GUI, not Asterisk NOW (which does have a GUI) |
| 19:04.23 | [TK]D-Fender | bsteele: *NOT is just a distr with the GUI installed for you. |
| 19:04.39 | WindowsUser | any gui has limitations that you may or may not run into :-/ |
| 19:04.57 | WindowsUser | unless its like a horribly complex gui :) |
| 19:04.57 | [TK]D-Fender | bsteele: So rather than watch you hack away at installing FreePBX and related bits by hand having little experince in Linux, going with AsteriskNOW is doing yourself a favour |
| 19:05.00 | *** join/#asterisk |Rain| (i=rain@ev.il.net) |
| 19:05.28 | |Rain| | is it possible to set non-local variables in a gosub in 1.6? |
| 19:05.30 | bsteele | My linux experience is not hampering me too much. It is the fact that I have no telephony experience |
| 19:05.44 | [TK]D-Fender | |Rain|: What is a "non-local variable"? |
| 19:06.18 | [TK]D-Fender | bsteele: Well feel free to visit freepbx.org and try to install it yourself along with everything else it requires and prepare to rinstall * to match it. |
| 19:06.22 | [TK]D-Fender | reinstall* |
| 19:06.54 | [TK]D-Fender | |Rain|: * vars don't have scope within a call. |
| 19:07.09 | bsteele | I already have freepbx installed |
| 19:07.12 | grey-monkey | Should asterisk be setting the time on my ip phones? Or do I need to run a separate time server? (The time/date on my server is correct, but the phones are an hour behind) |
| 19:07.18 | [TK]D-Fender | |Rain|: Globals are global to all calls but are not persistent across * restarts |
| 19:07.25 | [TK]D-Fender | bsteele: then use it |
| 19:07.25 | bsteele | my phones are seeing the server, tftp is sending the configs out correctly |
| 19:07.50 | [TK]D-Fender | grey-monkey: Completely separate of * |
| 19:08.36 | bsteele | i just need to figure out the dialing / config of dialplan and channels. Since most guides i've read talk about using Zap cards and a VOIP service for handing calls (trunks?), etc. I am a little confused as to where to go next |
| 19:08.42 | grey-monkey | [TK]D-Fender: Thanks! |
| 19:09.08 | bsteele | but I'll figure it out |
| 19:09.15 | bsteele | Thanks. |
| 19:09.17 | *** part/#asterisk bsteele (n=bbsteele@lb1.ihets.org) |
| 19:09.18 | |Rain| | [TK]D-Fender: 1.6 uses LOCAL(foo) to scope variables within the current gosub frame |
| 19:09.29 | [TK]D-Fender | bstYou don't configure the dialplan with FreePBX. It does it for you |
| 19:09.40 | WindowsUser | aww he left |
| 19:09.44 | |Rain| | or, at least, it does so when you're using a macro within AEL |
| 19:10.39 | |Rain| | but I have some macros that depend on the unscoped behavior (e.g. to set a bunch of variables based on data from REALTIME()) |
| 19:11.20 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d88b:2100:5e39:6178) |
| 19:11.29 | [TK]D-Fender | |Rain|: Sorry, can't help with AEL-specific bits that it parses in its own way |
| 19:12.03 | *** part/#asterisk jareq (n=jarek@082139043136.radom.vectranet.pl) |
| 19:13.01 | |Rain| | hrm. it kinda looks like it's not actually wrapping the rest of my sets in LOCAL(), so maybe I've got something else going wrong |
| 19:13.53 | |Rain| | woo, don't mind me... |
| 19:15.12 | *** join/#asterisk jkroon (n=jkroon@dsl-240-153-59.telkomadsl.co.za) |
| 19:17.03 | iewebguy | Grey-monkey: what brand of phones? |
| 19:17.16 | grey-monkey | polycom |
| 19:17.22 | grey-monkey | 330 and 320 |
| 19:18.09 | iewebguy | I don't know anything about polycom. the cisco phones I have can use a seprate time server |
| 19:18.30 | [TK]D-Fender | grey-monkey: ntp settings are in your sip.cfg |
| 19:19.50 | grey-monkey | Thanks. I just updated that in my sip.cfg. I'll reset my phones and see if the time is right. |
| 19:19.51 | superbock | I'm setting up a PRI in Italy, and incoming calls are working fine. But outgoing always fail with "Channel 0/1, span 1 got hangup, cause 44" right after the first ring. Any hint about this? |
| 19:20.32 | bmoraca | superbock: http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php <--- very helpful |
| 19:22.15 | *** join/#asterisk vicscandl (n=vicscand@adsl-99-130-13-113.dsl.irvnca.sbcglobal.net) |
| 19:23.38 | ariel_ | leifmadsen: I have another question, dial plan wise, when using Read, there is a note for 1.4 on: Warning: |
| 19:23.40 | ariel_ | In Asterisk 1.4: the "filename" field does not accept concatenation (file1&file2&file...). Does 1.6 accept concatenation? |
| 19:23.54 | leifmadsen | yes it does |
| 19:24.09 | ariel_ | t/y |
| 19:26.09 | *** part/#asterisk |Rain| (i=rain@ev.il.net) |
| 19:26.41 | *** join/#asterisk af_ (n=getsmart@88-149-240-229.dynamic.ngi.it) |
| 19:27.26 | *** join/#asterisk NateHB (n=NateHB@72.34.90.74) |
| 19:29.58 | *** join/#asterisk propellerhead (n=yogurt2u@host62.200-117-118.telecom.net.ar) |
| 19:31.05 | *** join/#asterisk datacompboy (n=datacomp@213.187.251.250) |
| 19:36.54 | jaytee | filename field doesn't accept concatenation in 1.4? since when? |
| 19:37.26 | ariel_ | jaytee: read: see bottom of this: http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 19:39.29 | jaytee | ok, but that would just be for Read. I know Playback supports concatenated files. |
| 19:39.40 | ariel_ | yes |
| 19:39.51 | ariel_ | but trying to get what they pressed for a dynamic menu |
| 19:43.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:43.52 | *** join/#asterisk cyyaw (n=cyyaw@201.240.163.9) |
| 19:49.33 | [TK]D-Fender | ariel_: Dynamic menu? |
| 19:49.46 | ariel_ | Dynamic menu/ivr yes |
| 19:50.09 | bmoraca | that sounds like a pain in the balls |
| 19:50.17 | [TK]D-Fender | ariel_: then use Background() and run a normal IVR and not Read() |
| 19:50.50 | ariel_ | yes that is what I have setup, was trying to just figure out which would have been better when we get to 22 items |
| 19:51.06 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 19:53.05 | *** part/#asterisk mphill (n=mphill@174.37.19.92) |
| 19:53.38 | buttons840 | I'm writing a python script which will control asterisk though the AMI. I'm hoping to learn more about a behavior of the AMI I've noticed; when you perform an originate over a SIP channel, you get one unique NewChannel event, however, when I plug in to the PSTN and issue a originate I receive multiple NewChannel events? Can someone give some insight to this behavior? |
| 19:54.42 | datacompboy | buttons840: i have tried to post bug about that, but got reply to set up random unique variable on originate, and catch call later on it |
| 19:56.31 | raden_work | [TK]D-Fender, u have any idea why i cant get a inbound phone call ? |
| 19:56.48 | buttons840 | when you say set up a unique variable you mean a literal variable, as in I'd issue a Variable: uniquevar=random and watch for this uniquevar in all NewChannel events? |
| 19:57.26 | buttons840 | i wasn't aware that a newchannel event would list variables, but i havn't tested it either, but it's a good suggestion to try |
| 19:58.46 | datacompboy | buttons840: moment plz |
| 19:58.58 | [TK]D-Fender | raden_work: I already gave you the reasons i suspect. I havent seen you do anything about it yet. |
| 19:59.40 | datacompboy | buttons840: https://issues.asterisk.org/view.php?id=8173 |
| 20:00.19 | [TK]D-Fender | buttons840: "plug into PSTN". Maybe you could clarify this.. a lot |
| 20:01.15 | buttons840 | i connect to the pstn through a sangoma card... bah, i forget what they're called, but it the card you can plug a regular phone line into (the card that receives signal) |
| 20:01.34 | *** join/#asterisk ingenius (n=alektro@host246.190-138-201.telecom.net.ar) |
| 20:01.37 | buttons840 | FXO/FXS I forget the difference |
| 20:02.07 | buttons840 | working primarily with SIP, but sometimes tests things on the PSTN |
| 20:02.31 | [TK]D-Fender | buttons840: pastebin some complete samples we can compare |
| 20:03.06 | buttons840 | [TK]D-Fender, maybe later, i want to look into datacoms suggestion first |
| 20:03.20 | *** join/#asterisk s14ck (n=s14ck@190-76-126-155.dyn.movilnet.com.ve) |
| 20:04.08 | [TK]D-Fender | buttons840: That was for 1.2, was resolved as "no change required" and has been a dead issue for over a year |
| 20:04.16 | [TK]D-Fender | buttons840: But you're welcome to sit on it :) |
| 20:04.34 | datacompboy | [TK]D-Fender: i just want to show him last message in thread :)) |
| 20:06.02 | buttons840 | datacompboy, well, thanks for specifying that ;) |
| 20:06.21 | datacompboy | buttons840: agh! also, one thing: sometimes it send "NewChannel" instead of "NewState". just ignore -- that bug, and hope it get fixed. but i'm still on ooold 1.4 |
| 20:06.59 | buttons840 | datacompboy, are you suggesting this behavior is improved in 1.6? |
| 20:07.39 | datacompboy | buttons840: I hope so, but can't test, since not tried to upgrade to 1.6. |
| 20:08.02 | buttons840 | this killer for me is that a single originate can produce multiple events with multiple uniqueids, and so it's hard to manage these events with all the uniqueids of unknown origin |
| 20:08.19 | datacompboy | buttons840: just forget about uniqueid. |
| 20:08.47 | datacompboy | buttons840: its not for calls, but for command -- so rely on variable that _you_ set only :) |
| 20:08.54 | lucasb | [TK]D-Fender, Sorry I stepped away from the 'puter -- The last thing you mentioned for me to do to get MWI working was to issue a "sip reload" command at the * CLI... I did that but nothing has changed |
| 20:10.10 | *** join/#asterisk autojack (n=owen@nerdnetworks.org) |
| 20:10.24 | *** join/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 20:10.37 | buttons840 | datacompboy, will this solution work even if the channel is not answered? |
| 20:10.39 | autojack | if I'm upgrading between minor versions of Asterisk (1.4.x to 1.4.x) do I need to do anything beyond configure/make/make install? |
| 20:11.01 | *** part/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 20:11.09 | datacompboy | buttons840: http://pastebin.ca/1510341 not about <sid> appendix in session var. yes, it works even for not answered calls, that I have catch via deadagi |
| 20:11.36 | buttons840 | agi :( |
| 20:11.43 | datacompboy | autojack: yes, check your patches and additional modules you have installed |
| 20:11.53 | buttons840 | i was hoping to avoid using the agi by simply watching the newexten events in my AMI script |
| 20:12.05 | buttons840 | it's a real pain to have to use two interfaces |
| 20:12.11 | datacompboy | autojack: if everything are stock -- should be ok from that |
| 20:12.14 | autojack | datacompboy: none, in this case. I have a stock install. |
| 20:12.16 | autojack | OK |
| 20:12.45 | datacompboy | buttons840: nope, i'm use fastagi+ami. fastagi to control channel, and ami only for originate and redirect :) that's works very well |
| 20:12.52 | [TK]D-Fender | lucas You should see the PUSH NOTIFY's going out for the VM |
| 20:13.00 | autojack | I'm trying to figure out what changes are in 1.6 and if I should consider upgrading to that. |
| 20:13.26 | buttons840 | datacompboy, i hope i can avoid that or my program is going to double in complexity |
| 20:14.08 | [TK]D-Fender | autojack: Go read the docs in the tarball |
| 20:14.10 | buttons840 | i figure the dialplan can manage my IVR, and I can monitor IVR responses using newexten events through the AMI |
| 20:14.34 | leifmadsen | autojack: lots :) |
| 20:14.39 | autojack | heh |
| 20:14.52 | autojack | I just figured there would be some page that explains what new stuff is in it. |
| 20:14.52 | leifmadsen | you'll want to read UPGRADE-1.6.txt and CHANGES |
| 20:14.56 | autojack | nod |
| 20:14.58 | leifmadsen | there is :) |
| 20:14.59 | datacompboy | buttons840: that vary from what you do. i'm completely replace extensions.{conf,ael} with my own session interpreter, and use only app_conference to do bridging. |
| 20:15.01 | [TK]D-Fender | autojack: And 1.4.x series minor changes rarely affect your configs expcept for the changes regarding ZAPTEL/DAHDI |
| 20:15.09 | autojack | [TK]D-Fender: OK |
| 20:15.31 | buttons840 | datacompboy, i'm not familiar with most of those features, nor do i need them |
| 20:15.34 | autojack | I probably shouldn't bother, but I'm trying to solve a nagging one-way audio problem. |
| 20:15.49 | buttons840 | datacompboy, may i ask what your program does, and what is language is it in? |
| 20:17.05 | datacompboy | buttons840: www.expertcall.com, f.e. working on it. its are incoming calls to rounds, direct calls, scheduled orders etc. core are on erlang, backend on php :) |
| 20:17.08 | [TK]D-Fender | autojack: Most commonly that is a NAT / routing issue |
| 20:17.17 | raden_work | [TK]D-Fender, im baffled on howto correct that and google is not being any help |
| 20:17.21 | buttons840 | datacompboy, for myself, i'm writing a simple script that make automated outbound calls, when answer the the dialplan will be used for a simple IVR, and (as said) I will monitor ivr responses through newexten events in the AMI |
| 20:17.42 | raden_work | how does the inbound phone number have to match the peer ? |
| 20:17.48 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 20:17.49 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:17.58 | datacompboy | buttons840: may be simpler to put .call file in /var/spool/asterisk/outgoing ? and just monitor newexten ? |
| 20:18.47 | autojack | [TK]D-Fender: yeah except I am running on a colo'd server without NAT, and using commercial VOIP termination :/ |
| 20:19.08 | autojack | it's an intermittent problem. at this point I'm guessing it's an issue with one or some of the routes my provider uses. |
| 20:19.30 | buttons840 | datacompboy, i considered it, but i need to make thousands of calls as fast a possible and bumping thousands of call files around the hard drive seemed a little unruly |
| 20:19.45 | autojack | what I really need to do is capture the behavior with full debugging on. |
| 20:19.50 | [TK]D-Fender | autojack: Re-invites will kill it jsut the same |
| 20:20.14 | autojack | [TK]D-Fender: well, I also tried it without reinvite. |
| 20:20.28 | autojack | eventually it happened again. |
| 20:20.44 | datacompboy | buttons840: well... then, you do originate with some unique variable set, and on newchannel message do getvar to bind channel<->call |
| 20:20.46 | autojack | that did seem like a promising solution. |
| 20:21.12 | buttons840 | datacompboy, yes |
| 20:21.27 | [TK]D-Fender | autojack: Well we've got nothign else to comment on at this point.. |
| 20:21.42 | datacompboy | buttons840: there will be no problem to have several NewChannel on one channel -- it will just same bind :) |
| 20:22.29 | *** join/#asterisk RonDutt (n=aiontsu@69.7.43.20) |
| 20:23.03 | RonDutt | Not directly related to * but how can I check to see what FXO lines are detected on a A400 Sangoma card? |
| 20:23.13 | autojack | [TK]D-Fender: yeah I know, thanks anyway :) I'll try to capture a full snapshot of the issue and then see if anyone can help me out. |
| 20:23.54 | autojack | I'm going to be bummed if it's my provider because they've been really good otherwise. |
| 20:24.43 | *** part/#asterisk autojack (n=owen@nerdnetworks.org) |
| 20:25.36 | raden_work | [TK]D-Fender, how do i correct the no matching user in sip users ? |
| 20:25.37 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:27.01 | [TK]D-Fender | raden_work: Make it match. |
| 20:27.06 | [TK]D-Fender | checkout time, BBIAB |
| 20:28.00 | raden_work | how do i make all inbound calls match my peer ? |
| 20:28.03 | *** join/#asterisk cryptanthus (n=newview@wsip-72-214-233-12.om.om.cox.net) |
| 20:32.19 | cryptanthus | I've been working on getting an asterisk system setup with a sangoma A200 card with 4 FXO ports. I was able to get the system to answer a call. Can someone help me get the system to allow me to call out. I have an Aastra 6757i phone I'm working with. |
| 20:33.27 | datacompboy | raden_work: _X.,1,... ? |
| 20:33.31 | cryptanthus | In my extensions.conf file I have... exten => 1000,n,Dial(Zap/1,10) |
| 20:33.53 | buttons840 | datacompboy, so, earland, you say you can dynamically upgrade the code, what does that mean? |
| 20:34.25 | raden_work | datacompboy, ??? |
| 20:35.02 | *** join/#asterisk d4rkstar (n=bruno@ip-233-233.sn2.eutelia.it) |
| 20:36.06 | datacompboy | buttons840: http://www.erlang.org/course/advanced.html#load |
| 20:36.28 | datacompboy | raden_work: in extensions, _X. will match all dialed numbers. |
| 20:36.30 | leifmadsen | _X. will match any 2+ digit extension, starting with 0-9. If you need to match on peer names (like in SIP URI dialing), then you can use something like _[0-9A-Za-z]. which is any 2+ alphanumberic extension |
| 20:36.36 | d4rkstar | good evening |
| 20:36.51 | d4rkstar | hi |
| 20:36.52 | d4rkstar | :D |
| 20:37.16 | d4rkstar | sorry, just to say if someone is interested in a Open Source CTI project |
| 20:37.16 | datacompboy | raden_work: if you place inbound peer into separate context, and put into that context wildcard rule -- you will get what you want |
| 20:37.31 | raden_work | lemee paste my extensions |
| 20:37.41 | datacompboy | d4rkstar: what you mean exactly? |
| 20:37.59 | buttons840 | datacompboy, so it appears you can modify the program without taking it offline? I presume this is a needed ability in your program? |
| 20:38.04 | raden_work | http://pastebin.com/d510d366 |
| 20:38.07 | raden_work | brb 3 min |
| 20:38.22 | d4rkstar | http://astcti2.googlecode.com |
| 20:38.46 | d4rkstar | datacompboy, i mean that |
| 20:38.49 | datacompboy | buttons840: yes. i have now zero-downtime code upgrade :) |
| 20:38.54 | buttons840 | datacompboy, fortunately, i have the luxury of being able to take my system offline for updates ;) |
| 20:39.29 | buttons840 | datacompboy, sounds like quite the software |
| 20:39.46 | datacompboy | raden_work: well, and problem is ...? |
| 20:40.08 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 20:41.16 | datacompboy | d4rkstar: why you use asterisk 1.4 ? |
| 20:41.34 | d4rkstar | the project is already migrated to asterisk 1.6 |
| 20:41.40 | d4rkstar | silently :) |
| 20:41.49 | d4rkstar | some documentation needs upgrade |
| 20:41.54 | d4rkstar | but the codebase is for 1.6 |
| 20:42.18 | d4rkstar | i need help |
| 20:42.21 | datacompboy | d4rkstar: and why you need one-more-asterisk-gui? |
| 20:42.40 | d4rkstar | datacompboy, this project is quite different from that |
| 20:43.55 | d4rkstar | check the docs to learn why |
| 20:44.20 | lucasb | d4rkstar, are you looking to develop a middleware-type app? |
| 20:44.34 | datacompboy | d4rkstar: no, thanks. you needed in help, but you can't describe you you need it. so, that will be still your problem. |
| 20:44.49 | d4rkstar | wow datacompboy :) |
| 20:45.19 | datacompboy | d4rkstar: if you have problem -- describe it. but send to docs ;) |
| 20:45.37 | d4rkstar | datacompboy, i don't have any problem |
| 20:45.58 | d4rkstar | i would like to get help from some skilled asterisk programmer |
| 20:46.09 | d4rkstar | to bring the community a software that asterisk doesn't have :D |
| 20:46.20 | d4rkstar | i don't earn money or similar things from that |
| 20:46.24 | datacompboy | d4rkstar: 03:42d4rkstari need help == so, you need help, but you have no problem? :) |
| 20:46.34 | d4rkstar | help in develop |
| 20:47.38 | datacompboy | d4rkstar: have you project some community, or you still alone? have you some usage scenario of app, or only imaginations? |
| 20:47.41 | d4rkstar | lucasb, it's a client-server application that is connected from one side to AMI and from another to CTI Clients |
| 20:48.20 | d4rkstar | datacompboy, check here: http://code.google.com/p/astcti2/source/browse/#svn/trunk |
| 20:48.45 | d4rkstar | there's almost a codebase and a full project with full analysis |
| 20:49.11 | d4rkstar | and no imaginations |
| 20:49.21 | datacompboy | d4rkstar: and where is any one real applications? |
| 20:50.44 | d4rkstar | datacompboy, i don't understand where is your a frontal attack or not :) |
| 20:51.16 | d4rkstar | i just said before that this is a project on which i need help: this means that the project needs programmers |
| 20:51.20 | d4rkstar | to be completed |
| 20:51.59 | datacompboy | d4rkstar: if project have no deadline, one programmer is enough. |
| 20:52.27 | d4rkstar | datacompboy, you're right! I would like to share a deadline with someone |
| 20:53.35 | d4rkstar | i don't want to insist in publicize this project |
| 20:54.04 | d4rkstar | if some of you is interested, my address is public |
| 20:54.19 | d4rkstar | and the project is open to anyone |
| 20:56.16 | raden_work | datacompboy, the problem is when i call in rings phone but when answer it doesnt pick up just keeps ringing on dial side |
| 20:57.21 | datacompboy | raden_work: well, what is your call flow? |
| 20:58.06 | datacompboy | raden_work: i mean, from where call in, and to what context it come. |
| 20:59.18 | *** part/#asterisk cryptanthus (n=newview@wsip-72-214-233-12.om.om.cox.net) |
| 20:59.24 | raden_work | http://pastebin.com/m69eae7ba |
| 21:00.52 | raden_work | [Jul 28 15:59:42] NOTICE[4293]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '101' rejected because extension not found. |
| 21:01.00 | _ShrikE | I am attempting to write an LCR dialplan that is giving me fits. It builds a variable that is ultimately run through the SORT function. The problem is that the sort function does not return the proper/complete order when the keys and values are passed as a variable. Anyone have any ideas? http://pastebin.ca/1510382 |
| 21:01.01 | raden_work | [from-callcentric] |
| 21:01.01 | raden_work | exten => s,1,Answer() |
| 21:01.01 | raden_work | exten => _X,1,Dial(SIP/101) |
| 21:01.56 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:02.08 | datacompboy | => _X.,1,Dial(SIP/101) |
| 21:02.28 | raden_work | whats the . do ? |
| 21:03.05 | d4rkstar | raden_work, it let your dial plan to match all the numbers |
| 21:03.39 | d4rkstar | raden_work, _X. = regexp every sequence of numbers of any length |
| 21:03.56 | raden_work | thanks :) |
| 21:04.29 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.133.61) |
| 21:04.51 | raden_work | still same BS i dial in phone rings i pickup nothing there and the phone im dialing from keeps ringing |
| 21:04.55 | datacompboy | raden_work: here is short-cut from original extensions.conf sample http://pastebin.ca/1510393 |
| 21:05.08 | *** join/#asterisk yziquel (i=55da623a@gateway/web/freenode/x-d24bea064f41fd0d) |
| 21:05.12 | datacompboy | raden_work: show "sip show peers" output |
| 21:05.32 | raden_work | Name/username Host Dyn Nat ACL Port Status |
| 21:05.32 | raden_work | 101/101 192.168.1.101 D 5060 OK (60 ms) |
| 21:05.32 | raden_work | callcentric/17772445766 204.11.192.31 5080 OK (70 ms) |
| 21:05.42 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:06.08 | datacompboy | tryo to comment-out "s,1,Answer()" line |
| 21:08.06 | NateHB | hey guys, I have a wird issue i think it might be related to my having two nics on my asterisk box, i have a softphone on my handheld, it registers with asterisk, but sip show peers shows UNREACHABLE, but I can ping the ip from the asterisk box just fine, you guys have any insight on whats going on? |
| 21:08.09 | d4rkstar | raden_work, it's obvious to say but do also a dialplan reload |
| 21:08.24 | raden_work | sorry this my first time with asterisk |
| 21:08.41 | raden_work | steep learning curve been reading and googling and trying to work with the ITSP as much as possible |
| 21:08.43 | datacompboy | NateHB: qualify=yes in [peerconf] |
| 21:09.04 | datacompboy | NateHB: and what with registration -- have it bee success? |
| 21:09.24 | raden_work | how do i comment out ? |
| 21:09.34 | datacompboy | raden_work: add ";" at line start |
| 21:09.39 | raden_work | thank u |
| 21:09.52 | NateHB | datacompboy, it registers, but only without a password, i get an error if I try it with a password |
| 21:10.21 | raden_work | just keeps ringing |
| 21:10.21 | datacompboy | NateHB: pasebin your sip.conf's section and console output (hide password) |
| 21:10.55 | raden_work | [Jul 28 16:09:46] WARNING[4293]: chan_sip.c:2803 retrans_pkt: Maximum retries exceeded on transmission 33910395-3457804198-852233@msw1.telengy.net for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 21:11.07 | datacompboy | raden_work: doesn't answers?! "core set verbose 5" "core set debug 5" and do one more call and post whole output in pastebin |
| 21:11.38 | d4rkstar | datacompboy, you're really fast |
| 21:12.14 | d4rkstar | datacompboy, I appreciate a lot your skill and fastness |
| 21:12.31 | *** join/#asterisk bobJR (n=lulz@adsl-153-124-150.tys.bellsouth.net) |
| 21:12.38 | [TK]D-Fender | raden_work: sounds like a networking/NAT issue |
| 21:12.59 | _ShrikE | Actually.. this pastebin illustrates the problem a little better. http://pastebin.ca/1510400 |
| 21:13.41 | datacompboy | d4rkstar: only on easy problems. trouble like _ShrikE have i can't help |
| 21:14.30 | d4rkstar | datacompboy, i missed the first part of _ShrikE trouble |
| 21:14.38 | d4rkstar | if i can be of some help |
| 21:14.48 | datacompboy | d4rkstar: 04:01_ShrikEI am attempting to write an LCR dialplan that is giving me fits. It builds a variable that is ultimately run through the SORT function. The problem is that the sort function does not return the proper/complete order when the keys and values are passed as a variable. Anyone have any ideas? http://pastebin.ca/1510382 |
| 21:15.23 | d4rkstar | datacompboy, ok i'm giving a look at |
| 21:16.27 | _ShrikE | d4rkstar: When I pass my variable to the SORT function the output is incorrect. If I hard enter the exact same content of the variable then it works fine. |
| 21:16.32 | raden_work | here is about 10 seconds of debugging http://pastebin.com/ |
| 21:16.35 | raden_work | crap |
| 21:17.00 | raden_work | http://pastebin.com/d6e2c2196 |
| 21:17.10 | *** join/#asterisk bobJR (n=lulz@adsl-153-124-150.tys.bellsouth.net) |
| 21:17.29 | d4rkstar | _ShrikE, let me study one minute your macro and check it on my system |
| 21:18.42 | d4rkstar | _ShrikE, the first thing that came in my mind is that the input to the SORT function is not-well generated |
| 21:19.01 | d4rkstar | but this is my first approach to the problem |
| 21:19.21 | lucasb | [TK]D-Fender, wb! I have some more info that might be useful to you.... I am using Asterisk as an app server for voicemail/IVR only... I do not register my phones with it... could that be part of the problem? |
| 21:20.03 | d4rkstar | _ShrikE, otherwise the SORT should give an incorrect return also if you hard-pass the value |
| 21:20.19 | *** join/#asterisk pdani_ (n=pdani@5403A2E5.catv.pool.telekom.hu) |
| 21:20.47 | datacompboy | raden_work: aggghh! :) add 'regexten=101' into sip.conf/[101] |
| 21:20.53 | raden_work | core and verbose at 5 http://pastebin.com/d9b6f939 |
| 21:21.08 | _ShrikE | I thought the same. But if I NoOp the variable and then cut and paste the value it works. http://pastebin.ca/1510400 See priorities 50 + |
| 21:21.15 | lucasb | [TK]D-Fender, Could that be why asterisk is ignoring the SUBSCRIBE requests? I have a feeling that those are needed to have * issue the NOTIFY. The thing I'm confused about is "sip show subscriptions" does show the phone subscriptions. |
| 21:21.24 | d4rkstar | _ShrikE, let me see |
| 21:21.58 | raden_work | datacompboy, under [101] ? |
| 21:22.13 | datacompboy | raden_work: ys |
| 21:23.40 | datacompboy | raden_work: and do sip reload after |
| 21:24.41 | raden_work | <PROTECTED> |
| 21:24.42 | raden_work | <PROTECTED> |
| 21:24.42 | raden_work | <PROTECTED> |
| 21:24.42 | raden_work | <PROTECTED> |
| 21:24.42 | raden_work | <PROTECTED> |
| 21:24.48 | raden_work | keeps ringing thats what i get |
| 21:25.26 | datacompboy | O_O who keeps ringing? 101 or you hears beeps in calling phone? |
| 21:26.22 | raden_work | the dial phone keeps ringing the VOIP receiving phone ring i pickup no one there Dialing phone still ringing and that was message i got @ console |
| 21:27.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:27.36 | d4rkstar | _ShrikE, it seems that something happens on prio 52 |
| 21:27.39 | datacompboy | raden_work: put "canreinvite=no" to [general] of sip.conf |
| 21:27.44 | d4rkstar | _ShrikE, where the variable is expanded |
| 21:27.53 | raden_work | dont have a general |
| 21:28.11 | raden_work | oh yeah i do now nevermind |
| 21:28.16 | raden_work | added that last night sorry |
| 21:28.40 | datacompboy | raden_work: you have writes all configs from scratch? wow.. i prefer to cut-down default:) |
| 21:28.59 | _ShrikE | The variable that is NoOp'd on priority 50 is the same thing that is passed to SORT in priority 52. I am not sure why the NoOp would handle it correctly but SORT wouldn't. |
| 21:29.21 | raden_work | datacompboy, everything i had wasnt working ITSP what they sent me was a joke people been helping me for 3 days :( |
| 21:30.05 | datacompboy | raden_work: well, is canreinvite=no fix problem? sip reload and retest |
| 21:30.27 | raden_work | datacompboy, ok now the dial phone keeps ringing i pickup receiving phone its silent for about 5 seconds and i get a dial tone |
| 21:30.46 | datacompboy | raden_work: dialtone?? what in console? |
| 21:31.11 | raden_work | <PROTECTED> |
| 21:31.11 | raden_work | <PROTECTED> |
| 21:31.11 | raden_work | <PROTECTED> |
| 21:31.11 | raden_work | <PROTECTED> |
| 21:31.11 | raden_work | <PROTECTED> |
| 21:31.13 | raden_work | [Jul 28 16:29:39] WARNING[4293]: chan_sip.c:2803 retrans_pkt: Maximum retries exceeded on transmission 33915600-3457805383-839631@msw1.telengy.net for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 21:31.16 | raden_work | <PROTECTED> |
| 21:31.18 | raden_work | sorry that was a lil long |
| 21:33.24 | datacompboy | raden_work: well... what are your network setup? its seems to be NAT problem |
| 21:33.26 | raden_work | I just dont get why dialing in is such a issue :( |
| 21:34.05 | raden_work | there is a westell DSL modem bridged to netgeat prosafe VPN Router in NAT mode then server connected to lan as well as phone |
| 21:34.20 | datacompboy | ha! nat mode. |
| 21:34.30 | datacompboy | have you mapped all UDP ports to asterisk? |
| 21:35.51 | raden_work | like in the router ? |
| 21:36.22 | datacompboy | raden_work: yes. |
| 21:36.31 | raden_work | why dont i just put asterisk box in DMZ ? |
| 21:36.42 | raden_work | no i havent done any routing or mapping |
| 21:37.02 | *** join/#asterisk bobJR (n=lulz@adsl-153-124-150.tys.bellsouth.net) |
| 21:37.27 | d4rkstar | _ShrikE, on my system it seems to sort them right |
| 21:37.37 | raden_work | datacompboy, what exactly do i have todo ? |
| 21:37.49 | datacompboy | raden_work: you should configure asterisk for nat. http://pastebin.ca/1510425 == here is nat part from default config |
| 21:38.15 | raden_work | do i need todo anything in the router ? |
| 21:38.55 | datacompboy | raden_work: forward all udp ports to asterisk |
| 21:39.06 | d4rkstar | _ShrikE, http://pastebin.ca/1510427 |
| 21:39.36 | datacompboy | raden_work: or, at least, 5060 + ports from rtp.conf: 10000-20000 by default |
| 21:40.09 | _ShrikE | d4rkstar: hrmm.. I am running 1.4 |
| 21:40.09 | d4rkstar | datacompboy, asterisk 1.6 should also support stun |
| 21:40.25 | d4rkstar | _ShrikE, uh |
| 21:40.35 | d4rkstar | what release of 1.4? |
| 21:40.50 | d4rkstar | _ShrikE, maybe i can do a test on a 1.4 system i have |
| 21:41.09 | raden_work | datacompboy, the NAT stuff goes under general ???? |
| 21:41.17 | datacompboy | raden_work: yes, in general |
| 21:42.03 | datacompboy | oggghhhh... 4:40... here sunrise... i'm really needs to get sleep:) raden_work: i'm be there a 4-6 hours later. but hope, you will get system working before:) good morning everyone! |
| 21:42.20 | d4rkstar | bye DarkLogik |
| 21:42.23 | d4rkstar | bye datacompboy |
| 21:42.24 | d4rkstar | :) |
| 21:42.31 | raden_work | datacompboy, thanks for all your help |
| 21:42.37 | raden_work | its greatly appreciated |
| 21:43.13 | *** part/#asterisk datacompboy (n=datacomp@213.187.251.250) |
| 21:43.20 | raden_work | externip = should be our static ip from ISP correct ? |
| 21:43.35 | d4rkstar | raden_work, yes |
| 21:43.45 | d4rkstar | i can suggest you a nat configuration example |
| 21:43.51 | d4rkstar | if you wait 4 minutes |
| 21:43.56 | d4rkstar | i pastebin that to you |
| 21:47.29 | d4rkstar | raden_work, take a look here: http://pastebin.ca/1510434 |
| 21:48.33 | raden_work | working on router at moment one sec |
| 21:50.44 | *** part/#asterisk [TK]D-Fender (n=zsirc@161.216.133.61) |
| 21:51.02 | d4rkstar | _ShrikE, found a way? |
| 21:52.06 | raden_work | I need port forwarding right ?????? |
| 21:52.17 | raden_work | to fordward all UDP to asterisk box ? |
| 21:52.37 | *** join/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 21:52.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:53.06 | d4rkstar | raden_work, if you use stun server |
| 21:53.11 | d4rkstar | you don't need to forward ports |
| 21:53.20 | _ShrikE | d4rkstar: still working on it |
| 21:53.36 | d4rkstar | if you don't use stun, you need to forward ports |
| 21:53.51 | raden_work | huh ? |
| 21:53.55 | d4rkstar | :D |
| 21:53.57 | raden_work | whats stun ? |
| 21:55.44 | raden_work | i can open all udp 5060 - 20000 thats it |
| 21:56.08 | d4rkstar | STUN stands for Simple Traversal of UDP Through NATs |
| 21:56.41 | d4rkstar | so if you use STUN asterisk will cooperate with an external server that allow it to avoid nat problems |
| 21:57.02 | d4rkstar | if you don't have stun you can do this simple thing |
| 21:57.25 | d4rkstar | in sip.conf, under [general], put externalip = your external ip |
| 21:57.40 | d4rkstar | in rtp.conf modify your ports range to 10000-20000 |
| 21:57.50 | raden_work | already have that |
| 21:58.00 | d4rkstar | on your firewall nat UDP/5060 to your asterisk |
| 21:58.11 | d4rkstar | and nat range UDP/10000 to UDP/20000 to your box |
| 21:58.20 | d4rkstar | i suggest you to nat also UDP/5061 |
| 21:58.26 | lucasb | Is anyone available to help me with getting Asterisk MWI to work? I am using v1.4.22 and I am using * as an app server only (Voicemail & IVR). When I set my phones for MWI I see them subscribed in the * CLI ("sip show subscriptions) however there are no NOTIFY messages being sent to the phones |
| 21:58.29 | d4rkstar | that's all |
| 21:58.36 | raden_work | i was just going todo 5000 - 20000 at asterisk box |
| 21:58.45 | d4rkstar | raden_work, it's excessive :D |
| 21:59.39 | drmessano | 5060 UDP and whatever is in rtp.conf, UDP |
| 21:59.39 | raden_work | ok |
| 21:59.49 | drmessano | 5061 = useless |
| 21:59.50 | raden_work | 10k to 20k |
| 21:59.55 | d4rkstar | raden_work, in sip.conf i suggest you to set also localnet |
| 22:00.01 | d4rkstar | under general |
| 22:00.05 | raden_work | did that |
| 22:00.27 | d4rkstar | ok |
| 22:00.27 | drmessano | externhost/externip? |
| 22:00.41 | d4rkstar | use only externip |
| 22:01.01 | drmessano | or externhost |
| 22:01.08 | d4rkstar | you can set externhost to your external FQDN if have any |
| 22:01.46 | raden_work | FQDN ? |
| 22:01.57 | d4rkstar | raden_work, from the cli do a "sip show settings" and it will list all settings |
| 22:01.57 | raden_work | i cannot find forwardi9ng anywhere in this POS router |
| 22:02.14 | d4rkstar | raden_work, :D |
| 22:04.02 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:05.02 | raden_work | wow it doesnt have port forwarding :( |
| 22:05.07 | raden_work | port triggering |
| 22:05.10 | raden_work | thats iot |
| 22:05.22 | d4rkstar | what kind of router you've ?? |
| 22:05.33 | *** join/#asterisk lwizardl (n=chatzill@c-67-177-142-128.hsd1.mi.comcast.net) |
| 22:05.36 | lwizardl | hi |
| 22:05.52 | raden_work | d4rkstar, Netgear FV538 |
| 22:06.22 | d4rkstar | raden_work, i have a FVS338 and it have a powerful portforwarding section |
| 22:06.33 | d4rkstar | i don't know if FV358 lacks of that |
| 22:07.49 | raden_work | i looked it up in the manual it reffered to port triggering section |
| 22:08.22 | d4rkstar | raden_work, is this your router: http://www.netgear.com/Products/VPNandSSL/WiredVPNFirewallRouters/FVX538.aspx?detail=Specifications |
| 22:08.45 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:09.03 | raden_work | omfg this dns issue is pissing me off |
| 22:09.15 | raden_work | d4rkstar, yes sir it is |
| 22:09.24 | drmessano | haha |
| 22:09.32 | drmessano | X-Lite 3.0 (software for phones) <--- or "softphone" |
| 22:10.32 | d4rkstar | raden_work, let me check your manual :) |
| 22:10.40 | raden_work | please soo |
| 22:10.45 | d4rkstar | i'm curious |
| 22:10.48 | raden_work | it has port range forwarding |
| 22:11.04 | raden_work | d4rkstar, http://documentation.netgear.com/fvx538/enu/202-10062-05/FVX538-06-03.html#wp53766 |
| 22:11.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:11.29 | d4rkstar | it's opening |
| 22:11.30 | d4rkstar | wait |
| 22:11.36 | raden_work | should have bought a cisco |
| 22:12.34 | raden_work | my cisco switch has more routing options oh yeah thats in for warranty :( |
| 22:13.26 | lwizardl | anyone here using a Linksys NSLU2 for an asterisk pbx ? |
| 22:14.53 | d4rkstar | raden_work, i can download the manual |
| 22:14.55 | d4rkstar | :( |
| 22:15.00 | d4rkstar | the download goes in timeout |
| 22:15.22 | d4rkstar | i've a good skill on netgear routers |
| 22:16.06 | raden_work | well ive worked with many routers this is the first one ive seen without port forwarding |
| 22:16.45 | d4rkstar | raden_work, got it |
| 22:17.23 | d4rkstar | raden_work, go in Security |
| 22:17.32 | d4rkstar | then go in Services |
| 22:18.06 | raden_work | yeah im there |
| 22:18.07 | *** join/#asterisk DigitalProcessor (i=ifone@ns2.mpolitan.net) |
| 22:18.24 | raden_work | 60 Asterisk UDP 5060 20000 Maximize-Reliability |
| 22:18.27 | d4rkstar | ok add a new service and call it RTP Traffic |
| 22:18.31 | d4rkstar | ok |
| 22:18.35 | d4rkstar | you just defined it |
| 22:18.36 | d4rkstar | :) |
| 22:18.37 | raden_work | that work ? |
| 22:18.39 | d4rkstar | let's go on |
| 22:18.50 | raden_work | yeah still cant get a connect on dial in |
| 22:19.03 | d4rkstar | Security -> Firweall |
| 22:19.10 | d4rkstar | Security -> Firewall* |
| 22:19.13 | raden_work | yteap |
| 22:19.51 | *** join/#asterisk TrentCreek (n=chatzill@129.113.113.149) |
| 22:19.58 | raden_work | RTSP:UDP Allow Always ANY ANY Maximize-Reliability Always |
| 22:20.06 | d4rkstar | you should create an Inbound service (a new inbound rule) using that custom Service created under Security->Services |
| 22:20.12 | DigitalProcessor | Hey guys, If I had to choose between hosted VoIP and setting up my own. What would be the biggest advantace of setting up my own vs getting it hosted by someone else? |
| 22:20.12 | TrentCreek | Anyone using IpKall and Asterisk? |
| 22:20.18 | DigitalProcessor | I don't know which way to go. |
| 22:20.31 | d4rkstar | raden_work, on the bottom of the page you should have an add.. button |
| 22:20.33 | DigitalProcessor | I am very familair with linux |
| 22:20.37 | d4rkstar | under inbound services |
| 22:20.43 | raden_work | yeah |
| 22:20.51 | yziquel | http://paste.lisp.org/display/84352 hi. i've been following the o'reilly asterisk book. why is dialplan reload not available? |
| 22:20.52 | raden_work | oh i can configure the fordward to server in threr ? |
| 22:20.58 | d4rkstar | yes |
| 22:21.02 | DigitalProcessor | My company is giving me 100 grand to play with |
| 22:21.10 | d4rkstar | raden_work, whe you add an inbound service |
| 22:21.15 | raden_work | RTSP:UDP Allow Always 192.168.1.100 ANY WAN1 Always |
| 22:21.29 | d4rkstar | you can check to send all the traffic to a specific server on your lan |
| 22:21.56 | raden_work | 192.168.1.100 is asterisk |
| 22:21.59 | d4rkstar | check the page 80 of the manual |
| 22:22.49 | d4rkstar | you should select in the service combo the service defined before "60 Asterisk UDP 5060 20000 Maximize-Reliability" |
| 22:22.57 | d4rkstar | i don't know hou you called it |
| 22:26.44 | raden_work | u lost me |
| 22:26.45 | d4rkstar | raden_work, is that more clear than before? |
| 22:27.08 | raden_work | there is no service defined before asterisk |
| 22:27.36 | d4rkstar | raden_work, let's start from begin |
| 22:27.41 | raden_work | ok |
| 22:28.01 | d4rkstar | i'll try to explain before the logic of netgear unit |
| 22:28.18 | raden_work | ok |
| 22:28.25 | raden_work | lol |
| 22:28.39 | d4rkstar | the router want you define a thing called "Service" that is the port or port range and the protocol |
| 22:29.06 | *** part/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 22:29.07 | d4rkstar | the router have a lot of preset, but in our case, the range 10000-20000 udp for example, doesn't exists in the presets |
| 22:29.34 | d4rkstar | once you defined a "Service", you should tell to the router how to use |
| 22:29.50 | d4rkstar | so you can ask the router to match that traffic and discard |
| 22:30.01 | d4rkstar | or you can ask the router to match that traffic and nat it to a specified ip |
| 22:30.04 | d4rkstar | :) |
| 22:30.25 | raden_work | ok |
| 22:30.30 | *** join/#asterisk iewebguy (n=mark@c-75-68-250-121.hsd1.vt.comcast.net) |
| 22:31.25 | raden_work | i got it so in firewall rules i have to set it to asterisk :P |
| 22:31.35 | d4rkstar | to define a "Service" you should go in "Security"->"Services" and add two presets |
| 22:31.56 | d4rkstar | one for udp/5060 that you can call SIP/UDP |
| 22:32.16 | d4rkstar | and one for udp/10000-20000 that you can call RTP/UDP |
| 22:32.59 | d4rkstar | after defined these two Services, you can go in "Security"->"Firewall rules" and add two inbound rules that selects these patterns |
| 22:33.02 | raden_work | is it ok just for testing to leave it 5060-20000 for now ? |
| 22:33.09 | raden_work | i get what u are saying |
| 22:33.16 | d4rkstar | i think that it's good for testing |
| 22:33.33 | d4rkstar | but in production i suggest to have two patterns |
| 22:33.38 | lwizardl | anyone here using a Linksys NSLU2 for an asterisk pbx ? |
| 22:33.42 | d4rkstar | so you can use the proper QoS for them |
| 22:33.51 | raden_work | once i get things working i will be going through everything again documenting etc.... |
| 22:34.20 | raden_work | ok lemee pastebin my conf files |
| 22:35.15 | d4rkstar | ok |
| 22:35.55 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 22:36.03 | raden_work | http://pastebin.com/d1addde6b |
| 22:36.09 | raden_work | did everything in fire3wall |
| 22:36.14 | raden_work | i can call out hear both ways |
| 22:36.27 | d4rkstar | is that working? |
| 22:36.44 | raden_work | when calling in the phone will ring but pickup its silence then 3 - 5 seconds later i get a dial tone |
| 22:37.01 | raden_work | it was just silence before we commented out ; exten => s,1,Answer() |
| 22:37.20 | raden_work | so i can get the phone to ring inbound thats about all |
| 22:37.44 | d4rkstar | can you pastebin also a full cli with verbosity 10 |
| 22:37.47 | d4rkstar | core set verbose 10 |
| 22:37.54 | d4rkstar | then try to make an inbound call |
| 22:37.59 | d4rkstar | the output pastebin |
| 22:38.01 | raden_work | i can do that one moment |
| 22:38.42 | raden_work | what the verbose anyways ? |
| 22:38.56 | d4rkstar | put it at 10 |
| 22:40.46 | raden_work | i did |
| 22:40.58 | raden_work | i some people tell me 10 some say 5 just wondering what it is ? |
| 22:40.59 | raden_work | :( |
| 22:41.12 | raden_work | http://pastebin.com/d6e34ef63 |
| 22:41.13 | d4rkstar | it the verbosity level :) |
| 22:41.14 | Qwell | raden_work: more verbose is never bad |
| 22:41.15 | d4rkstar | i prefer 10 |
| 22:41.25 | raden_work | what does it control ? |
| 22:41.29 | raden_work | what does it control ? |
| 22:41.30 | raden_work | http://pastebin.com/d6e34ef63 |
| 22:44.07 | d4rkstar | raden_work, it seems that your machine have ip 192.168.1.101 and not .100 |
| 22:44.24 | raden_work | server *.100 phone *.101 |
| 22:44.47 | d4rkstar | the phone is in your lan? |
| 22:44.57 | raden_work | yes |
| 22:45.24 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 22:45.25 | raden_work | router 192.168.1.1 server 192.168.1.100 phone [101] 192.168.1.101 |
| 22:45.31 | d4rkstar | ok |
| 22:45.45 | d4rkstar | if you're sure to have setup right the nat |
| 22:45.58 | raden_work | what u mean ? |
| 22:46.22 | d4rkstar | if you're sure that your router is properly configured to nat all udp traffic to your asterisk box |
| 22:46.24 | d4rkstar | we can check better sip.conf parameters |
| 22:46.40 | raden_work | ill double check routert |
| 22:46.42 | d4rkstar | it seems that codecs are well configured |
| 22:46.53 | raden_work | services |
| 22:46.53 | d4rkstar | so it can be a misconfiguration in sip.conf |
| 22:46.55 | raden_work | Asterisk UDP 5060 20000 Maximize-Reliability |
| 22:47.05 | raden_work | sip.conf is on pastebin |
| 22:47.11 | *** join/#asterisk bobJR (n=bob@adsl-153-124-150.tys.bellsouth.net) |
| 22:47.23 | raden_work | http://pastebin.com/d1addde6b |
| 22:47.39 | d4rkstar | i saw it |
| 22:47.39 | d4rkstar | :) |
| 22:47.44 | d4rkstar | wait i check some things |
| 22:47.49 | raden_work | outbound |
| 22:47.51 | raden_work | Asterisk Allow Always ANY ANY Maximize-Reliability Always |
| 22:48.01 | raden_work | inbound |
| 22:48.02 | raden_work | Asterisk Allow Always 192.168.1.100 ANY WAN1 Always |
| 22:51.44 | *** join/#asterisk Alfio (n=Amunoz@adsl-55-172.tricom.net) |
| 22:51.59 | raden_work | off subject but our bandwidth here is 800 kbps / 5 mbps when i call outbound the conversation is slightly choppy |
| 22:52.16 | *** join/#asterisk [netman] (n=netman@88.27.57.136) |
| 22:52.40 | d4rkstar | raden_work, try to change sip.conf as http://pastebin.com/d5193e236 |
| 22:53.09 | *** part/#asterisk lwizardl (n=chatzill@c-67-177-142-128.hsd1.mi.comcast.net) |
| 22:54.08 | raden_work | [from-callcentric] |
| 22:54.08 | raden_work | ; exten => s,1,Answer() |
| 22:54.08 | raden_work | exten => _X.,1,Dial(SIP/101) |
| 22:54.08 | raden_work | [to-callcentric] |
| 22:54.08 | raden_work | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@callcentric) |
| 22:54.11 | raden_work | <PROTECTED> |
| 22:54.35 | d4rkstar | i don't looked at that for now |
| 22:54.53 | d4rkstar | i changed some settings in general and in the peer from-callcentric |
| 22:55.30 | d4rkstar | for that you can change it to exten => _0X.,1,Dial(SIP/${EXTEN:1}@callcentric) |
| 22:55.35 | raden_work | i can dial in will ring the phone pickup phone the dial phone keeps ringing receiving phone is silence |
| 22:55.36 | d4rkstar | so you can dial 0 + number to call |
| 22:56.37 | raden_work | only thing that changed is before it would ring i would pickup and about 3-5 seconds have a dial tone now it stays silent |
| 22:57.16 | d4rkstar | can you pastebin your cli output, but without sip debug (sip set debug off) ? |
| 22:57.17 | raden_work | <PROTECTED> |
| 22:57.18 | raden_work | <PROTECTED> |
| 22:57.18 | raden_work | <PROTECTED> |
| 22:57.18 | raden_work | <PROTECTED> |
| 22:57.18 | raden_work | <PROTECTED> |
| 22:57.20 | raden_work | <PROTECTED> |
| 22:57.46 | raden_work | sorry pastebin taking like 20 seconds to dns driving me nutz |
| 22:57.53 | d4rkstar | ahhahah |
| 22:57.54 | d4rkstar | ok |
| 22:58.16 | d4rkstar | the phones are mute in both directions? |
| 22:58.28 | raden_work | lol |
| 22:58.37 | raden_work | ok im dialing with my cell here is what is happeneing |
| 22:59.13 | raden_work | Dial with CELL > VOIP PHONE RINGS > PICKUP VOIP PHONE > CELL PHONE STILL RINGING > VOIP PHONE IS SILENCE |
| 22:59.19 | *** join/#asterisk jplank_ (i=6218a9ec@gateway/web/freenode/x-4d0d1b0a518eb90e) |
| 22:59.25 | raden_work | im not getting a connection basically |
| 22:59.34 | d4rkstar | raden_work, but if you speak in one phone, you can ear in the other? |
| 22:59.45 | d4rkstar | or there's mute in both ways |
| 22:59.45 | d4rkstar | ? |
| 23:00.27 | raden_work | nope |
| 23:00.31 | raden_work | mute both ways |
| 23:00.39 | raden_work | my cell phone rings the entire time like no one picked up |
| 23:00.58 | d4rkstar | <PROTECTED> |
| 23:01.42 | d4rkstar | this let us know that asterisk have routed the call and asked the phone in your lan to "talk" directly with the remote peer |
| 23:02.03 | raden_work | so is the problem in the phone config ? |
| 23:02.06 | d4rkstar | try to change nat=yes to your [101] exten |
| 23:02.10 | raden_work | AASTRA 9133i |
| 23:02.21 | d4rkstar | in sip.conf |
| 23:02.21 | raden_work | www.voltarclamps.com/sip1.png |
| 23:02.27 | d4rkstar | then do a sip reload |
| 23:02.30 | d4rkstar | and try again |
| 23:02.48 | raden_work | crap |
| 23:03.00 | raden_work | http://www.voltarclamps.com/sip2.png |
| 23:05.03 | drmessano | get rid of the outbound proxy server |
| 23:05.06 | *** join/#asterisk _abc_ (n=no@67-208-18-57.ip.tor.radiant.net) |
| 23:05.22 | _abc_ | what kind of phone number would be starting with 188888xxxxx ?! |
| 23:05.26 | d4rkstar | raden_work, sorry, away |
| 23:05.46 | _abc_ | i can't find it anywhere and i have a call from something like that |
| 23:05.57 | _abc_ | does anyone have a pointer ? |
| 23:06.03 | _abc_ | it's not telemarketing |
| 23:06.36 | drmessano | Its tollfree of some sort |
| 23:06.42 | raden_work | it might be a VOIP company i know mine is 177777 |
| 23:06.44 | d4rkstar | raden_work, configuration of your aastra seems to be fine |
| 23:06.58 | *** join/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 23:07.00 | d4rkstar | try to change nat=yes to your [101] exten |
| 23:07.05 | raden_work | yeah working on it |
| 23:07.47 | raden_work | d4rkstar, the same |
| 23:08.04 | d4rkstar | i think that there's some trouble with nat |
| 23:08.05 | d4rkstar | :( |
| 23:08.15 | d4rkstar | i would like to continue to help you |
| 23:08.37 | raden_work | still a nat issue ehhh |
| 23:08.46 | d4rkstar | try to pastebin a sip show settings |
| 23:08.56 | _abc_ | 1-888-trixbox is a valid number fyi |
| 23:09.04 | _abc_ | and self-explaining |
| 23:09.29 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 23:09.46 | raden_work | http://pastebin.com/d73f6dcf7 |
| 23:10.55 | d4rkstar | uhm Internal IP: 127.0.0.2:5060 |
| 23:11.07 | raden_work | wtf ????? |
| 23:12.03 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:12.03 | raden_work | why is local net showing diffrent then what i have set as well ? |
| 23:12.49 | drmessano | What do you have set? |
| 23:13.02 | d4rkstar | drmessano, http://pastebin.com/d5193e236 |
| 23:13.06 | raden_work | localnet=192.168.1.0/255.255.0.0 |
| 23:13.19 | raden_work | how do i set internal ip ? |
| 23:13.30 | d4rkstar | raden_work, try to bind asterisk to 192.168.1.100 |
| 23:13.49 | *** part/#asterisk sjobeck (n=Adium@208-151-246-17.dq1sn.easystreet.com) |
| 23:13.51 | d4rkstar | bindaddr=192.168.1.100 in sip.conf |
| 23:13.51 | raden_work | bindaddr=192.168.1.100 ? |
| 23:13.51 | drmessano | Wait, multiple NICS? |
| 23:14.01 | raden_work | one nic @ moment |
| 23:14.28 | raden_work | how do i set internal ip ?> |
| 23:14.35 | _abc_ | raden_work: maybe you configured with net down and then brought it up and the scripts guessed wrong ? |
| 23:14.45 | drmessano | Does your router have some SIP aware bullshit that you can turn off |
| 23:14.46 | d4rkstar | try to do also a restart |
| 23:14.49 | d4rkstar | full restart of asterisk |
| 23:14.55 | drmessano | I saw some "SIP" crap listed in its flyer |
| 23:15.07 | _abc_ | the default route and outgoing network must be up when you run the auto config scripts |
| 23:15.48 | raden_work | _abc_, net was up but our connection has been glitchy last week so possible |
| 23:16.03 | *** join/#asterisk pwden (n=domin8@pool-72-80-205-195.nycmny.east.verizon.net) |
| 23:16.13 | raden_work | drmessano, ill check the manual |
| 23:16.20 | raden_work | core restart now ? |
| 23:16.21 | _abc_ | so 1-888-88xxxxx would be someone from the trixbox or asterisk domain ? or both ? who is the genius who allocates itu numbers and then loses the allocations, but tracks IPs ? oh, wait, the riaa is fighting the isps not the itu |
| 23:16.29 | d4rkstar | raden_work, yes |
| 23:18.38 | drmessano | _abc_: what reasoning do you have that an 888-88 number has ANYTHING to do with asterisk or trixbox??? |
| 23:18.51 | raden_work | <PROTECTED> |
| 23:18.53 | raden_work | still the same |
| 23:19.02 | _abc_ | drmessano: ouija board |
| 23:19.03 | drmessano | Fix your SIP aware router |
| 23:19.09 | _abc_ | aka google |
| 23:19.33 | d4rkstar | raden_work, it's strange and weird that |
| 23:19.41 | drmessano | _abc_: Which search? |
| 23:20.09 | drmessano | http://barrymoltz.com/business/just-dial-1-888-88-barry |
| 23:20.19 | drmessano | Like that one for Barry J Moltz? |
| 23:20.52 | drmessano | http://188888solar.com/contact_us |
| 23:20.54 | drmessano | or that one |
| 23:21.15 | drmessano | http://www.thepinkygroup.com/1_888_88_pinky_com <-- or that |
| 23:21.19 | _abc_ | drmessano: i found it it seems to be telemarketing or debt collection or something. 1-888-882-3811 if you don't use the dashes the idiotic genius does not find it |
| 23:21.37 | _abc_ | so much for 411 and so on |
| 23:21.52 | drmessano | Should have tried google |
| 23:22.10 | raden_work | drmessano, you think thats what it is ? |
| 23:22.48 | drmessano | raden_work: Routers that have anything to do with SIP are pure crap as middlemen |
| 23:23.01 | _abc_ | http://800notes.com/Phone.aspx/1-888-882-3811/3 |
| 23:23.22 | _abc_ | the twist is the call seems to imply possible identity theft |
| 23:23.38 | _abc_ | that's why i am digging it, they called the right number but asked for the wrong person |
| 23:23.42 | DigitalProcessor | Guys if I have lets say 300 stores with 2 phones each. With enough bandwidth on every end. Will one asterisk suffice or should I have 2 or 3 for redundacy? |
| 23:23.42 | raden_work | drmessano, reading manual bro |
| 23:23.44 | _abc_ | and it's a bank |
| 23:23.54 | _abc_ | that tends to be worrying |
| 23:23.57 | raden_work | brb i need to change DNS and thatll probally drop me |
| 23:24.14 | DigitalProcessor | raden_work, nah |
| 23:24.21 | DigitalProcessor | it shouldn't |
| 23:24.24 | d4rkstar | raden_work, i think yes |
| 23:24.35 | d4rkstar | raden_work, check your /etc/hosts file |
| 23:24.48 | d4rkstar | and see if your hostname is binded to 127.0.0.2 |
| 23:24.55 | d4rkstar | or something like that |
| 23:25.02 | *** join/#asterisk JonCup (n=JonCup@72.34.90.74) |
| 23:26.01 | d4rkstar | my /etc/hosts looks like this: |
| 23:26.02 | d4rkstar | -bash-3.2# cat /etc/hosts |
| 23:26.02 | d4rkstar | 127.0.0.1 localhost.localdomain localhost |
| 23:26.18 | raden_work | wow im still here |
| 23:26.28 | DigitalProcessor | Told you |
| 23:27.02 | raden_work | 127.0.0.1 localhost |
| 23:27.22 | raden_work | DigitalDaz21, i know it shouldnt but when suse resets config sometime it restarts eth* |
| 23:27.30 | raden_work | DigitalProcessor, dont ask me |
| 23:27.39 | raden_work | so do i need to change that ? |
| 23:27.45 | raden_work | lemee paste entire hosts |
| 23:27.55 | d4rkstar | ok |
| 23:28.30 | raden_work | http://pastebin.com/d55f6100e |
| 23:28.41 | raden_work | wow im lighting fast now that i setup own dns ISP sucks |
| 23:29.06 | d4rkstar | check line 24 |
| 23:29.32 | d4rkstar | asterisk sees ttssusa.site or ttssusa as first on line 24 |
| 23:29.44 | d4rkstar | so your internal ip is 127.0.0.2 |
| 23:29.54 | raden_work | so what should i do ? |
| 23:29.54 | d4rkstar | try to comment that line |
| 23:29.59 | d4rkstar | and restart asterisk |
| 23:30.11 | d4rkstar | and check if something change :) |
| 23:31.24 | raden_work | just a sec |
| 23:31.52 | DigitalDaz21 | 127.0.0.2 is NOT a valid IP address |
| 23:32.19 | DigitalDaz21 | the only valid address in the 127 range is 127.0.0.1 |
| 23:32.26 | d4rkstar | DigitalDaz21, some distro use to put that entry in /etc/hosts |
| 23:33.47 | raden_work | grr yast keeps rewriting it |
| 23:34.06 | d4rkstar | a suggestion |
| 23:34.15 | d4rkstar | change the file and then set it immutabile :) |
| 23:34.34 | d4rkstar | chattr +i /etc/hosts |
| 23:34.35 | d4rkstar | :) |
| 23:34.56 | d4rkstar | so yast can't change it anymore |
| 23:37.02 | raden_work | thanks :) |
| 23:37.25 | raden_work | <PROTECTED> |
| 23:37.33 | d4rkstar | ok that's great |
| 23:37.41 | d4rkstar | try now |
| 23:37.48 | raden_work | same thing |
| 23:38.09 | raden_work | [Jul 28 18:37:22] WARNING[26713]: chan_sip.c:2803 retrans_pkt: Maximum retries exceeded on transmission 33938846-3457813054-239051@msw1.telengy.net for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 23:38.30 | d4rkstar | nat problems dear my friend :D |
| 23:38.49 | d4rkstar | post your rtp.conf file |
| 23:39.34 | raden_work | http://pastebin.com/de9c57e9 |
| 23:40.06 | d4rkstar | mumble mumble |
| 23:40.39 | raden_work | nothing showing in console anymore when i call in or out ? |
| 23:41.01 | d4rkstar | raden_work, let move us to stun |
| 23:41.09 | d4rkstar | callcentring seems to support them |
| 23:41.14 | d4rkstar | do this: |
| 23:41.31 | raden_work | u think it in the router ? |
| 23:41.33 | d4rkstar | remove any sip configuration from your netgear firewall |
| 23:41.59 | d4rkstar | oh wait |
| 23:42.19 | d4rkstar | Callcentric has developed its NAT traversal technology without using STUN. |
| 23:42.39 | raden_work | :((((((( |
| 23:42.42 | raden_work | thats bad isnt it |
| 23:43.00 | raden_work | should i get my crappy linksys router up here and see if that works ? |
| 23:43.02 | raden_work | lol |
| 23:43.31 | d4rkstar | you followed this guide: http://www.callcentric.com/support/device/asterisk |
| 23:43.33 | d4rkstar | ? |
| 23:44.33 | raden_work | yeah that caused more problems than i have now |
| 23:44.46 | raden_work | callcentric support is a joke i been working on getting asterisk up like 10 days |
| 23:45.09 | d4rkstar | raden_work, i've to go bed now |
| 23:45.31 | d4rkstar | but if you want tomorrow i'll back online and try to help again |
| 23:45.42 | d4rkstar | i think you should check better the nat |
| 23:45.46 | raden_work | d4rkstar, you have a email addy or some way to contact you im on service tommorow |
| 23:45.53 | raden_work | definatly in the nat ? |
| 23:46.14 | d4rkstar | when you've no voice, the problem is that the rtp traffic is dropped somewhere |
| 23:46.28 | raden_work | could it be our ISP / |
| 23:46.39 | d4rkstar | i can suggest you to use a tool to inspect traffic inbound on your linux box |
| 23:46.46 | d4rkstar | like iptraf or tcpdump |
| 23:46.48 | raden_work | should i bridge modem to asterisk box and hook phone up on other nic to test ? |
| 23:47.06 | d4rkstar | try a piece at time |
| 23:47.18 | raden_work | the router one piece :) |
| 23:47.27 | d4rkstar | but the real thing you should do is see if udp traffic arrive to your box |
| 23:47.32 | d4rkstar | check iptraf |
| 23:47.35 | d4rkstar | or tcpdump |
| 23:48.31 | d4rkstar | check this: http://www.erg.abdn.ac.uk/users/alastair/tcpdump.html#rtp#rtp |
| 23:48.32 | raden_work | tcpdump |
| 23:48.38 | raden_work | im getting hundreds of lines |
| 23:48.52 | d4rkstar | raden_work, that's normal, you should filter traffic |
| 23:48.58 | raden_work | howso ? |
| 23:49.06 | d4rkstar | try with: tcpdump -T rtp -vvv src |
| 23:49.39 | d4rkstar | and place a call to your provider |
| 23:50.13 | raden_work | i can call out just fine |
| 23:50.15 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 23:50.16 | raden_work | inbound the problem |
| 23:50.34 | d4rkstar | and place a call from your provider :) this was the sense |
| 23:50.35 | d4rkstar | :D |
| 23:50.42 | raden_work | tcpdump -T rtp -vvv src |
| 23:50.42 | raden_work | <PROTECTED> |
| 23:50.56 | d4rkstar | after src try to put the ip of your provider |
| 23:52.20 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:53.18 | raden_work | ok i called out OMG like 1000 lines that normal |
| 23:53.24 | raden_work | for like 5 seconds |
| 23:54.50 | raden_work | when i dial inbound nothing happens |
| 23:55.46 | raden_work | whats SIP AGL SUPPORT ? |
| 23:59.03 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
