00:00.07 | [TK]D-Fender | tvirus: it is recommended that if you value your faxes you will keep them as far away from * as possible. |
00:00.59 | tvirus | The fax line carries the DSL, though, and the boss wants to use it. Why doesn't * play well with faxes? |
00:03.55 | Elijah` | [TK]D-Fender, could you take a look at this SIP debug and see what you think? I thought I had the NAT problem resolved, but still not working... here's the pastebin: http://elijah.pastebin.com/m3ffe7d8e |
00:05.21 | [TK]D-Fender | Elijah`: from the START of the call, and include your sip.conf masking only passwrds |
00:05.21 | *** join/#asterisk brodiem (n=brodiem@e2.72.1243.static.theplanet.com) |
00:06.09 | Elijah` | That was from the start... but here I'll add in the sip.conf... |
00:06.25 | axisys | [TK]D-Fender: there are soo many emails.. i wish there is one article where it covers the setup.. i know wishful thinking |
00:06.38 | [TK]D-Fender | Elijah`: No, that wasn't the start |
00:07.15 | [TK]D-Fender | axisys: JFGI |
00:07.39 | axisys | the only thing is not working is when i receive a call from outside the pstn port redirects to lne port.. everyelse is working |
00:07.50 | axisys | whats jfgi ? |
00:09.21 | tvirus | Just Fucking Google It |
00:11.00 | Elijah` | lol |
00:11.17 | Elijah` | ok here it is, I double checked and did another call, this is where it starts http://elijah.pastebin.com/d6b8fddf7 |
00:11.27 | axisys | tvirus: heh |
00:11.46 | axisys | man people are sooo angry.. wow! |
00:11.54 | axisys | sorry for asking questions |
00:12.47 | axisys | if anyone here can help w/o getting pissed of please help |
00:12.55 | axisys | thnx |
00:13.09 | tvirus | I'd help but I have no idea what I'm doing yet :) |
00:13.22 | axisys | tvirus: appreciate it really |
00:13.45 | Elijah` | same... newbie here too LOL |
00:14.06 | axisys | Elijah`: heh |
00:14.32 | axisys | i have been googling for about 2 weeks.. nothign came out to be same issue as mine |
00:14.58 | axisys | but heh.. as long as I dont ask [TK]D-Fender i guess i am safe from another blasphemy.. heh |
00:15.52 | jtodd | axisys: you've seen these, right? |
00:15.53 | jtodd | http://voxilla.com/tools/device-configuration-wizard/ |
00:15.54 | [TK]D-Fender | Elijah`: Where is your Cisco located relative to *? |
00:16.12 | Elijah` | it's behind a separate NAT |
00:16.27 | axisys | jtodd: yes.. i followed and it helped a lot |
00:16.40 | axisys | jtodd: i can call to my cell now from my line port |
00:16.42 | [TK]D-Fender | axisys: http://www.google.ca/search?hl=en&q=SPA-3102+setup+guide+asterisk&btnG=Search&meta= |
00:17.03 | [TK]D-Fender | Elijah`: What do you have forwarded to *? |
00:17.11 | axisys | jtodd: i can recv call to.. but did not go thru sip proxy |
00:17.17 | Elijah` | port 5060, and the RTP ports 10000-10010 |
00:17.23 | Elijah` | UDP |
00:17.31 | clintc | tvirus: faxes won't work well if asterisk is trying to echo cancel |
00:17.40 | [TK]D-Fender | Elijah`: You should widen that a healthy bit. I've seen things go bad due to over tightening |
00:17.54 | [TK]D-Fender | Elijah`: But as we're not seeing an answer it may be an issue on the Cisco end |
00:17.55 | Elijah` | so widen the RTP to say, 50 or 100? |
00:18.16 | Elijah` | aah, ok makes sense... now thing is... |
00:18.22 | Elijah` | if the cisco places the call, things work fine |
00:18.32 | tvirus | So the fax should just be left alone with the analog line. |
00:18.45 | [TK]D-Fender | Elijah`:have you forwarded on their side? |
00:18.47 | axisys | [TK]D-Fender: thanks.. i will go throw the search results again.. thanks again |
00:19.08 | Elijah` | No, as I understood you shouldn't have to.. so no nothing is fwded |
00:19.22 | clintc | tvirus: that's generally the easiest thing to do |
00:19.28 | jtodd | axisys: I don't have any specific solutions; it's been a long time since I messed with a Sipura. If I were you, I'd set up wireshark and look at what is happening on the ethernet between your * server and the Sipura. |
00:19.55 | tvirus | Sounds good. So for a single analog line, would an ATA work just fine getting it setup in *? |
00:20.19 | axisys | jtodd: i already installed wireshark.. i will followup on your suggestion.. thnx |
00:21.00 | clintc | tvirus: we never had good luck getting a fax to work with an ata... we send them out via an fxs configed port and that works well |
00:21.09 | jtodd | axisys: change one thing at a time, note what you did, test further. Too many options in the Sipura - easy to get confused. |
00:21.23 | tvirus | clintc: I meant for a separate analog line, not connected to the fax. |
00:21.31 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-bf902bcc8805d395) |
00:21.37 | clintc | tvirus: ah, sure |
00:21.46 | Elijah` | I would bet it's on the cisco end, because there's other phones registered elsewhere (behind other NATs) that work ok |
00:21.52 | axisys | jtodd: i know :-) |
00:22.01 | jtodd | axisys: I think part of the "magic" for the configuration was "insecure=very" on the Asterisk side in sip.conf for that peer. |
20:33.26 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
20:33.27 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
20:33.43 | Nivex | jbot: welcome back |
20:33.44 | jbot | Good to see you, back! |
20:33.58 | Nivex | heh |
20:34.51 | ShaunWing | and then what do I do after that? |
20:35.14 | [TK]D-Fender | eric_hill: Please dump your dialplan from CLI |
20:35.19 | *** part/#asterisk iotashan (n=shan@adsl-71-150-254-145.dsl.mdsnwi.sbcglobal.net) |
20:35.38 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
20:36.28 | eric_hill | [TK]D-Fender: It's huge. I tried to boil it down to the smallest working demonstration of the problem rather than spamming this chat with 1000 lines of irrelevant crap. |
20:37.23 | eric_hill | [TK]D-Fender: And I can work around the problem by setting a variable before the first Gosub. It just smells like a bug. ARG1 gets dropped if returned from an invalid context extension. |
20:37.37 | eric_hill | [TK]D-Fender: But I guess it could be as-designed. |
20:37.38 | [TK]D-Fender | eric_hill: -- Executing [1002@foo-a:2] Return("Zap/16-1", "") <-- I stragnly see it starting at priority *2* and now I'm starting to complete not trust what you're showing me. |
20:37.44 | [TK]D-Fender | Either way heading home, BBIAB |
20:38.09 | *** join/#asterisk MindTheGap (n=MindTheG@201.17.149.252) |
20:39.33 | jeev | Fender. i did it |
20:39.34 | jeev | crap |
20:41.21 | drmessano | HA |
20:41.25 | drmessano | Sounds like it |
20:41.45 | *** join/#asterisk barakuda (n=barakuda@91.78.92.63) |
20:44.14 | jeev | hmm, why aren't you ignored |
20:53.43 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
20:53.58 | LemensTS | this is repeating over and over in the cli: -- parse_srv: SRV mapped to host sphone.vopr.vonage.net, port 5061 |
20:54.07 | bkw_ | its doing SRV lookups |
20:54.08 | bkw_ | thats normal |
20:54.19 | kfife | Qwell: func_odbc depends upon somethign called res_odbc. Do you have any idea what CentOS/RHEL5 calls it in their repository? I can't find a match. |
20:54.34 | LemensTS | bkw_ : ok thanks, never had it do that with other itsp's |
20:54.35 | Qwell | res_odbc is part of asterisk |
20:54.51 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
20:54.58 | bkw_ | LemensTS: if you have ITSP's without SRV records they are clearly doing it wrong |
20:55.07 | bkw_ | SRV records are a must to do thing correctly |
20:55.30 | kfife | I see. I just need to re-run configure and make menuselect, and the dependencies will show as satisfied? |
20:55.37 | ShaunWing | Say: [root@messaging etc]# service asterisk start |
20:55.38 | ShaunWing | Starting asterisk: [ OK ] |
20:55.40 | hsv-al | i still cant get rid of this crap |
20:55.40 | ShaunWing | [root@messaging etc]# ps aux | grep asterisk |
20:55.41 | ShaunWing | root 8437 0.0 0.0 65924 596 pts/0 S 22:52 0:00 /bin/sh /home/shaunw/asterisk-bin/sbin/safe_asterisk -U shaunw -G shaunw -C /home/shaunw/asterisk-bin/asterisk |
20:55.43 | ShaunWing | shaunw 8446 0.1 0.1 447312 9468 pts/0 Sl 22:52 0:00 /home/shaunw/asterisk-bin/sbin/asterisk -f -U shaunw -G shaunw -C /home/shaunw/asterisk-bin/asterisk -U shaunw -vvvg -c |
20:55.44 | Qwell | kfife: if you've got the deps installed, yeah :) |
20:55.44 | ShaunWing | root 8482 0.0 0.0 61160 712 pts/0 S+ 22:53 0:00 grep asterisk |
20:55.45 | hsv-al | [Aug 18 09:22:41] WARNING[9071]: res_smdi.c:1264 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. |
20:55.52 | kfife | You've just saved me a lot of time! Thanks a million! |
20:55.53 | ShaunWing | Is this now running as non root? |
20:55.53 | Qwell | kfife: make sure you have unixODBC-devel, or whatever your distro calls it |
20:55.55 | hsv-al | how do i prevent that module from loading up on start |
20:56.17 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
20:57.22 | ManxPower | hsv-al: chances are the same way as you stop any other module from loading in Asterisk |
20:57.51 | hsv-al | but the thing is, its not set to load up |
20:57.53 | ManxPower | You had better hope no modules require res_smdi.so. |
20:58.14 | ManxPower | hsv-al: pastebin your /etc/asterisk/modules.conf |
20:58.30 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
20:59.05 | *** part/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
20:59.11 | ManxPower | hsv-al: you can always mv /usr/lib/asterisk/modules/res_smdi.so |
20:59.17 | ManxPower | drat |
20:59.33 | ManxPower | hsv-al: you can always mv /usr/lib/asterisk/modules/res_smdi.so /usr/lib/asterisk/modules/disabled-res_smdi.so |
20:59.38 | *** join/#asterisk gerphimum (n=trekkie@cpe-67-10-147-45.satx.res.rr.com) |
20:59.48 | ManxPower | But as I said, you had better hope no other modules require res_smdi.so |
21:01.00 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
21:01.21 | ShaunWing | say when I run as non root: asterisk -r I get back... |
21:01.23 | ShaunWing | -bash: asterisk: command not found |
21:01.41 | Qwell | ShaunWing: if you aren't root, you need to use the full path to asterisk |
21:01.44 | ManxPower | ShaunWing: then the directory asterisk is in is not i your path |
21:02.03 | ShaunWing | ok, but how can I create a short cut for myself? |
21:02.07 | ShaunWing | please |
21:02.21 | ManxPower | Oh, ShaunWing is one of *those* people. |
21:02.39 | ManxPower | (the people that expect us to tutor them in basic Linux stuff) |
21:02.47 | drmessano | Right Click on the desktop |
21:02.52 | drmessano | Go to New --> Shortcut |
21:02.56 | seanbright | Alt-F4 |
21:02.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
21:02.57 | drmessano | Go to the C; Drive |
21:03.05 | drmessano | find the exe file... |
21:03.09 | drmessano | Oh, wait |
21:03.22 | ManxPower | drmessano: no, no. He said "short cut", not "shortcut. For a short cut, do not cut as much |
21:03.24 | hsv-al | http://pastebin.ca/1178006 |
21:03.26 | seanbright | ShaunWing: first, in your irc client type /server create-asterisk-shortcut |
21:03.28 | drmessano | LOL |
21:03.30 | hsv-al | no reference to it |
21:03.32 | hsv-al | weird |
21:03.57 | drmessano | ShaunWing: freenode has a built in IRC Linux documentation server |
21:03.59 | ManxPower | hsv-al: now you know what to do. |
21:04.01 | ShaunWing | TX 4 teh tip |
21:04.02 | drmessano | You need to add it to your server list |
21:04.08 | drmessano | do /server linux-docs |
21:04.30 | hsv-al | what is that module typically used for |
21:04.31 | seanbright | i get the medal if it works |
21:04.39 | hsv-al | but since its not in my system,, ill issue the command you pasted |
21:04.58 | ManxPower | hsv-al: just add the noload line like I originally suggested |
21:05.24 | ShaunWing | Unknown host âlinux-docsâ |
21:05.27 | ManxPower | I suspect you'll lose voicemail if you don't load res_smdi.so. |
21:05.28 | seanbright | drat |
21:05.43 | eric_hill | Can anyone else reproduce this problem? http://pastebin.com/d63f52f43 |
21:06.05 | seanbright | eric_hill: yes -> http://pastebin.com/m7fd1bce |
21:06.24 | drmessano | ShaunWing: Did you do it in this window? |
21:06.31 | seanbright | an exact reproduction |
21:06.33 | ShaunWing | yes |
21:06.39 | seanbright | stops being a smartass and goes home |
21:06.48 | eric_hill | needs more beer on a Monday. |
21:07.04 | drmessano | try /server linuxdocs |
21:07.09 | *** join/#asterisk jks (n=jks@193.189.93.254) |
21:07.27 | ShaunWing | [ERROR]Unknown host âlinuxdocsâ connecting to irc://linuxdocs/,isserver (irc://linuxdocs/,isserver). [Help] Reconnecting in 15 seconds. [Cancel] |
21:07.52 | drmessano | Sounds like your IRC client isn't RFC2188 compliant |
21:07.55 | drmessano | Try google |
21:08.15 | drmessano | Oh |
21:08.21 | drmessano | You have ChatZilla |
21:08.22 | ManxPower | eric_hill: I suspect you do not understand how gosub works. |
21:08.45 | eric_hill | ManxPower, I'm just trying to figure out if this is a bug or a feature. |
21:08.58 | ShaunWing | yes |
21:09.02 | drmessano | With ChatZilla, ShaunWing, you can join channel 0 on here for linux help |
21:09.02 | ShaunWing | ChatZilla |
21:09.04 | sn9 | drmessano: doesn't work in xchat, either, and that *is* compliant |
21:09.10 | drmessano | Just do a /join 0 |
21:09.17 | eric_hill | ManxPower, I wouldn't have expected the Macro argument to disappear, but saving it to a variable first works just fine. |
21:09.23 | ManxPower | eric_hill: It's working as intended (subroutines start with exten => s,1,Whatever |
21:09.25 | [TK]D-Fender | eric_hill: -- Executing [1002@foo-a:2] Return("Zap/16-1", "") in new stack <- why do we not see a PRIORITY 1 execute for this? |
21:09.46 | ShaunWing | *ChanServ*[#0] Welcome to #0 - the channel that shouldn't exist... |
21:09.48 | ManxPower | [TK]D-Fender: "s" is converted to real priorities at runtime |
21:10.05 | [TK]D-Fender | ManxPower: No, read the PB |
21:10.06 | jeev | Fender, i've succeeded in passing the variable with _XXX then $CHANNEL 1:4. now i've got to add _XXX to the dial plan also so the database get doesn't fail. |
21:10.12 | sn9 | ShaunWing: i edited the /topic there |
21:10.13 | ManxPower | Â Â -- Sent into invalid extension 's' in context 'foo-b' on Zap/16-1 |
21:10.14 | drmessano | ShaunWing: Indeed not compliant |
21:10.27 | drmessano | ShaunWing: Try Ctrl-Alt-Shift-F4 |
21:10.33 | [TK]D-Fender | ManxPower: -- Executing [s@macro-example:2] Gosub("Zap/16-1", "foo-a|1002|1") in new stack followed IMMEDIATELY by : -- Executing [1002@foo-a:2] Return("Zap/16-1", "") in new stack |
21:10.53 | sn9 | drmessano: what is that supposed to do? |
21:11.05 | eric_hill | [TK]D-Fender - That was a verbose = 5, do you want a verbose 9? |
21:11.14 | [TK]D-Fender | ManxPower: the first gosus is completely untrustoworthy |
21:11.17 | ManxPower | [TK]D-Fender: We are working on different problems. The one I'm working on is the invalid gosub |
21:11.17 | [TK]D-Fender | gosub |
21:11.39 | [TK]D-Fender | ManxPower: I'm working on #1 which tells me that what we are looking at is DOCTORED |
21:11.41 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
21:11.49 | ShaunWing | its ##linux |
21:12.11 | [TK]D-Fender | ManxPower: because #2 also would seem to indicate that the ${ARG1 } got wiped |
21:12.11 | ManxPower | If he changes exten => i,1, to exten => s,1, I bet stuff will start working |
21:12.30 | [TK]D-Fender | ManxPower: 2 problems, my impossible priority, and a MISSING var |
21:12.35 | eric_hill | [TK]D-Fender: You keep thinking I'm faking the result... http://pastebin.com/d40cac214 |
21:12.56 | hsv-al | tisk tisk @ "dr" messano for using |
21:12.58 | hsv-al | the alt-f4 trick |
21:12.58 | ManxPower | eric_hill: your pastebin of the dialplan executing does NOT match the CLI output. |
21:13.18 | eric_hill | Would you like to remote control my machine and see what I'm seeing?! |
21:13.29 | eric_hill | I'd be happy to set up a webex. I'm not faking this. |
21:13.50 | ManxPower | Remember when you do a macro inside a macro the arguement values are not saved. I assume it is the same for gosub |
21:14.27 | ManxPower | eric_hill: That's why I think it is important to fix the obvious wrongness (as I told you just moments ago) than to try to find the hidden problems. |
21:14.29 | [TK]D-Fender | ManxPower: That makes a logto of sense if they tried to make a cheap-out implementation of Gosub |
21:14.37 | [TK]D-Fender | lot* |
21:14.46 | ManxPower | [TK]D-Fender: I got bitten by the issue years ago |
21:15.18 | [TK]D-Fender | eric_hill: It does look like the var is getting eaten by the gosub so far. |
21:15.22 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
21:15.26 | ManxPower | the easiest thing is to NEVER EVER use ${ARG2}, ${ARG2} etc when calling a macro or gosub |
21:15.53 | [TK]D-Fender | ManxPower: Actually its more like never call "gosub" from within a macro |
21:15.54 | eric_hill | Changing to exten => s,1 will cause the Gosub to never return on an invalid extension, hence the second Gosub will not be executed. |
21:16.18 | ManxPower | s/e ${ARG2}/e ${ARG1}/ |
21:16.23 | eric_hill | Ok, I'll write that down in the list of stuff not to do inside a dialplan. |
21:16.27 | LemensTS | I got vonage setup in asterisk using the soft phone account they offer, i was hoping to use it as DID like most other ITSP's offer...i called there tech support they dont know what DID is...do any of you know if they do this service or not? |
21:16.58 | [TK]D-Fender | LemensTS: DID = a phone number they receive that they route to you |
21:17.00 | Qwell | LemensTS: don't all Vonage accounts have phone numbers? |
21:17.11 | [TK]D-Fender | LemensTS: and of course their tech support drones don't know shit about * |
21:17.12 | ManxPower | [TK]D-Fender: recursive macros are the heart of my macros! 9-) |
21:18.04 | ManxPower | eric_hill: the 2nd invocation of the macro/gosub will overwrite the ARG1, ARG2, etc and not restore them when returning to the original calling macro/gosub |
21:18.18 | ManxPower | this is by design as I understand it. |
21:18.25 | [TK]D-Fender | ManxPower: And the look of revusion on my face upon seeing your spaghetti&meatsauce&ice-cream&brusselssprout&caviar code LEGENDARY |
21:18.38 | LemensTS | qwell: tkd: i mean so i can use one phone number with multiple outgoing lines...thats did isnt it... |
21:18.46 | [TK]D-Fender | LemensTS: NO |
21:18.48 | ManxPower | LemensTS: no. |
21:18.52 | [TK]D-Fender | LemensTS: a DID is a PHONE NUMEBR |
21:18.59 | eric_hill | ManxPower, tx. Fender, classic, as always. |
21:19.09 | [TK]D-Fender | LemensTS: this has no impact over how it may be delivered to you. |
21:19.09 | Qwell | voip doesn't really have the concept of "lines".. presumably, they allow multiple incoming/outgoing calls at once |
21:19.31 | *** part/#asterisk chackal_sjc (n=pet@200-168-20-85.dsl.telesp.net.br) |
21:19.35 | *** join/#asterisk pascal_alm (n=pascal_a@66.11.93.129) |
21:20.21 | ManxPower | Qwell: They are called "trunks" in the VoIP world. 8-| |
21:20.56 | LemensTS | manx : trunks yes that is the work i mean. I want a did with multiple trunks. |
21:20.57 | [TK]D-Fender | :p |
21:21.04 | LemensTS | *word |
21:21.10 | ManxPower | chortles evily |
21:21.21 | [TK]D-Fender | LemensTS: You just got "Punk'd" by ManxPower |
21:21.25 | jeev | Fender! http://al.pastebin.com/d22899ae9 uh, i dunno how to now make the number going out check the _XXX, extension then dial out after. do i need more variables? |
21:21.32 | [TK]D-Fender | ManxPower: You are a very bad person.... |
21:21.39 | LemensTS | tkd: its fine i have a bad memory |
21:22.53 | [TK]D-Fender | jeev: No, its PART of the exten that was dialed. |
21:26.25 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
21:26.50 | LemensTS | http://www.voip-info.org/wiki/view/Vonage+Business+Plus here is sip trunking with vonage i was looking for, helped when i used the work trunk. |
21:27.22 | eric_hill | Is there a switch somewhere to turn on outbound Caller ID sending over an ISDN line? Inbound name/number is coming through, and outbound number are working fine. |
21:27.36 | ManxPower | eric_hill: yes |
21:27.50 | roe_ | has anyone done any performance testing with the atom processor? I didn't see anything on the dimensioning page |
21:28.09 | eric_hill | offers ManxPower a beer for the switch name. |
21:28.16 | ManxPower | eric_hill: If the outbound number works, then the telco has already set that option |
21:28.49 | eric_hill | ManxPower: It's for an internal switch-to-switch call, not going over the PSTN. |
21:28.51 | ManxPower | eric_hill: the option is set by the telco, not the user. Also most terminating telcos overwrite the name sent with whatever name the phone company has on file for that number. |
21:29.05 | ManxPower | eric_hill: then I have no idea. |
21:30.08 | *** join/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
21:30.12 | nny_2 | hmmp |
21:30.13 | nny_2 | h |
21:30.33 | ManxPower | eric_hill: you mean internal asterisk-asterisk call right? |
21:31.06 | nny_2 | ha nm |
21:31.08 | nny_2 | hi all |
21:33.04 | eric_hill | ManxPower: Asterisk -> Nortel PBX |
21:33.38 | eric_hill | ManxPower: Otherwise I'd just use SIP->SIP or IAX->IAX. |
21:33.43 | ManxPower | eric_hill: It must suck to be you. |
21:34.02 | eric_hill | I can't wait to get off this crappy PBX. |
21:34.19 | ManxPower | eric_hill: if you do a PRI debug I suspect you'll see Asterisk send the Caller*ID name. If so then it is entirely up to the Nortel. |
21:35.03 | *** join/#asterisk fury_ (n=flolry@ool-4b7fd6a2.static.optonline.net) |
21:35.26 | eric_hill | I've been debugging and see a *bunch* of stuff, including "Received caller name'8670 Eric H'", but I never see a "Sending caller name". |
21:35.28 | ManxPower | eric_hill: I believe Caller*ID name can be sent in one of two ways on PRI. You might want to search the mailing list archives to see if Asterisk can be told to send it the other way. |
21:35.42 | eric_hill | Good to know. Tx. |
21:36.07 | eric_hill | I also found a post that said I should try switching from Q.SIG to DMS100. May try that tonight after-hours. |
21:36.33 | fury_ | is there any addon or library or something that will help me with getting my phone system to be able to pronounce human names? |
21:36.57 | [TK]D-Fender | fury_: "core show application record" |
21:37.14 | eric_hill | fury_: ran across http://www.cepstral.com/ today... |
21:37.59 | fury_ | [TK]D-Fender, I don't think that's what I want |
21:38.05 | fury_ | eric_hill: thanks |
21:38.37 | [TK]D-Fender | fury_: Probable. And I want $1,000,000. Guess nobody is happy today. |
21:38.52 | fury_ | no its no problem :) |
21:39.05 | fury_ | just saying that's not what I asked. actually it will be useful anyway though. |
21:39.10 | ManxPower | fury_: most text-to-speach programs allow you to spell names phonetically |
21:39.18 | ManxPower | words, not names |
21:41.30 | fury_ | eric_hill: it has a demo, might work good enough :) |
21:41.35 | fury_ | my name is kind of hard to pronounce |
21:41.48 | eric_hill | fury seems pretty easy to me... |
21:42.14 | sn9 | Abby Normal? |
21:43.58 | *** join/#asterisk ta^3 (n=tacvbo@189.146.172.3) |
21:44.43 | ManxPower | eric_hill: what if its pronounced "furry"? |
21:46.04 | *** part/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
21:46.21 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
21:47.18 | kfife | Qwell: Func_odbc Installed! Thank you! |
21:50.09 | kfife | Does anyone know: soft hangup SIP/.... : How do I get the unique ID of the channel? It's the string that comes across the CLI when a call comes in. It's not the string that comes in when you type: core show channels or even sip show channels. What command will show the string? |
21:50.55 | kfife | ...you can't always draw a relationship between the channel you see that is hung, and the string that came across the CLI when the call originated. |
21:51.14 | exothermc | ManxPower: I figured out a solution to my 503 ambiguity the other day. |
21:51.45 | kfife | exothermic: 503 as in Grandstream HT503? |
21:52.17 | [TK]D-Fender | kfife: "core show channels concise" |
21:52.22 | exothermc | No sip response code 503 (congestion) I belive. |
21:52.58 | kfife | Thanks for the distinction. I've got a question for anyone using the GS HT503. |
21:53.24 | eric_hill | MansPower: I think my CID problem may be solved with prilocaldialplan=local in the zapata.conf. One hour and counting to a maintenance window... :) |
21:53.31 | exothermc | I was having the issue of not knowing how best to handle getting a 503 from a carrier (who wanted me to treat the 503 as a network wide response not just the server, which is how the RFCs say to use it for) |
21:54.06 | exothermc | So basically they wanted me to route advance their network on a 503, not just keep hammering all their different end points for the call. |
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21:57.14 | *** part/#asterisk barnseenio (n=bb@90.204.160.230) |
21:59.06 | mltlnx | Does anyone know why when I restart my Sangoma 104 card my channel banks rings on all channels? Its really strange and we are deploying them for use in a hospital. Of course those lines will be in the patients rooms! |
21:59.45 | denon | hah |
21:59.57 | denon | I'd talk to Sangoma and see if they've got some tips |
22:00.25 | denon | they've probably seen it a hundred times |
22:00.45 | mltlnx | Called them this morning and they did not know. |
22:03.09 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
22:03.43 | kfife | . |
22:05.34 | mchou | ok, is this possible with asterisk? On an outbound call, dial plan extension pattern is matched (after the phone hsa sent the dial string). I want to play a recording (more or less saying which extenion rule matched) before asterisk actually performs the Dial cmd. What would be a good way to do this? |
22:05.57 | mchou | has* |
22:06.42 | [TK]D-Fender | mchou: "core show application playback" |
22:14.18 | *** part/#asterisk Netgeeks (n=chris@204-16-157-172-static.ipnetworksinc.net) |
22:15.53 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
22:16.08 | jeev | Fender, did it again.. with the dial plan, used exten 4:11 and it picked it up, perfect! |
22:16.12 | jeev | god i'm the best |
22:19.52 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
22:21.57 | *** join/#asterisk hadronzoo (n=user@m765f36d0.tmodns.net) |
22:22.35 | hadronzoo | Hello, how can I add some time before asterisk answers a call on either an inbound route or on an inbound trunk? |
22:22.49 | riddlebox | can someone help me, when I make a call out, I can talk for a little bit, then it hangs up? -- Executing [16186544596@DLPN_LocalLong:1] Macro("SIP/524-0822a840", "trunkdial-failover-0.3|Zap/g1/16186544596||trunk_1|") in new stack |
22:22.49 | riddlebox | <PROTECTED> |
22:22.49 | riddlebox | <PROTECTED> |
22:22.49 | riddlebox | <PROTECTED> |
22:22.52 | riddlebox | <PROTECTED> |
22:22.54 | riddlebox | <PROTECTED> |
22:22.56 | riddlebox | <PROTECTED> |
22:22.59 | riddlebox | <PROTECTED> |
22:23.03 | riddlebox | <PROTECTED> |
22:23.05 | riddlebox | <PROTECTED> |
22:23.07 | riddlebox | <PROTECTED> |
22:23.09 | riddlebox | <PROTECTED> |
22:23.12 | riddlebox | <PROTECTED> |
22:23.12 | jeev | pastebin.com |
22:23.14 | riddlebox | shoot sorryt |
22:23.16 | riddlebox | kick me |
22:23.23 | jeev | give me your adddress! |
22:23.24 | riddlebox | I did, but I guess it didnt copy the link |
22:23.24 | wwalker | kicks riddlebox |
22:23.37 | kd8ikt | leaps out and begins bludgeoning riddlebox |
22:23.54 | riddlebox | how about that http://pastebin.ca/1178100 |
22:24.21 | wwalker | can yo7u bludgeon with a sharp object? or only blunt ones? |
22:24.33 | kd8ikt | howbout i castrate you for not having atleast a unique nick |
22:25.23 | riddlebox | thank goodness [TK]D-Fender hasnt seen that mistake yet ;) |
22:26.18 | [TK]D-Fender | riddlebox: I have |
22:26.21 | heedly | hadronzoo: Wait(time) before Answer or Dial. |
22:27.02 | [TK]D-Fender | riddlebox: What are you dialing out of? |
22:27.10 | hadronzoo | heedly: in what conf file? |
22:27.21 | [TK]D-Fender | hadronzoo: extensions.conf, where else? |
22:27.28 | [TK]D-Fender | hadronzoo: this is all DIALPLAN |
22:27.34 | hadronzoo | [TK]D-Fender: cool, thanks |
22:28.08 | riddlebox | [TK]D-Fender, a pots line |
22:28.24 | Qwell | ~whatnow > Deeewayne |
22:28.31 | Qwell | silly bot |
22:28.37 | [TK]D-Fender | riddlebox: First guess "callprogress=yes", #2 = other side hung up. |
22:29.10 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
22:30.49 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
22:31.39 | hardwire | anybody handle motorola bsr2000 cmts's here? |
22:31.46 | hardwire | it's.. slightly.. related.. :) |
22:31.58 | heedly | hadronzoo: extensions.conf |
22:32.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
22:32.25 | *** join/#asterisk jksM (n=jks@193.189.93.254) |
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23:03.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
23:07.44 | lesouvage | When I change the value of the variable "emailbody" in voicemail.conf it doesn't seem to effect the content of the e-mail send when a voicemail is received. I run asterisk 1.4.21.1. Isn't that where the variable "emailbody" is ment for? |
23:10.27 | [TK]D-Fender | lesouvage: Did you restart *? |
23:10.36 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
23:11.12 | lesouvage | [TK]D-Fender: no I did a reload, restart is needed? |
23:11.22 | [TK]D-Fender | lesouvage: You need to reload the app at a minimum |
23:15.06 | lesouvage | [TK]D-Fender: I did a restart of Asterisk but still the default content of the e-mail ."New 0:24 long msg in box 505 from 505, on Tuesday, August 19, 2008 at 01:00:54 AM" |
23:15.15 | *** join/#asterisk brodiem_mobile (n=brodiem_@32.171.51.154) |
23:15.35 | [TK]D-Fender | lesouvage: pb your voicemail.conf |
23:15.56 | lesouvage | Point is that the "from" is not correct. |
23:16.25 | lesouvage | [TK]D-Fender: thanks |
23:16.33 | [TK]D-Fender | lesouvage: And that has WHAT to do with the "email body" exactly? |
23:17.25 | [TK]D-Fender | lesouvage: Don't show me one thing and then pull a bait & switch like that |
23:17.48 | lesouvage | [TK]D-Fender: If the message that I can't change would have been correct there would have been no need for change. |
23:18.34 | [TK]D-Fender | lesouvage: First you complain about the body now you complain about the "from" Holy shit make up your mind, and SHOW ME. |
23:24.31 | jeev | god i'm so pissed, my friend called and bout 2 minutes into the conversation.. got disconnected. |
23:24.36 | jeev | i dunno if it's the ITSP or what.. |
23:25.16 | lesouvage | [TK]D-Fender: I paste it on http://www.pastebin.be/13238 |
23:27.02 | unpaidbill | last t38 |
23:27.18 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
23:28.11 | [TK]D-Fender | lesouvage: And please PB a sample e-mail |
23:29.19 | unpaidbill | ~opal |
23:32.39 | *** part/#asterisk jkirby (n=jkirby@dsl-240-111-11.telkomadsl.co.za) |
23:35.43 | kfife | when I type CLI> show channels, the last few characters of the channel name are always truncated. SIP/ptassessor-b7d004a0 shows up as SIP/ptassessor-b7d00. This means I have to complete the name before using a command such as soft hangup. Granted it will autocomplete with a TAB press, but only if the first part of the name happens to be unique. Is there a way to show the WHOLE channel name? |
23:36.12 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
23:38.03 | [TK]D-Fender | kfife: "core show channels concise" |
23:40.45 | *** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
23:40.56 | kfife | Thanks! You always seem to know! You are the one that turned me on to local channels a while back which brought a great deal of efficiency to my dialplans. |
23:40.58 | [TK]D-Fender | ok, heading to the gym. Back later |
23:41.19 | kfife | [TK]D-Fender: that thanks was for you! |
23:41.34 | [TK]D-Fender | kfife: np |
23:42.09 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
23:42.55 | anonymouz666 | [TK]D-Fender going to the gym? |
23:42.58 | anonymouz666 | WTF |
23:43.13 | lesouvage | [TK]D-Fender: New 0:24 long msg in box 505 from 505, on Tuesday, August 19, 2008 at 01:00:54 AM |
23:44.42 | anonymouz666 | [TK]D-Fender: you are gold medal of asterisk olympic |
23:45.37 | lesouvage | [TK]D-Fender: This is the whole e-mail. It seems to be the pager message. |
23:46.12 | anonymouz666 | imagine [TK]D-Fender asking for SIP DEBUG, extensions.conf and dmesg in the gym :P |
23:50.14 | jeev | hahaah |
23:50.19 | jeev | he's the bear grylls of asterisk |
23:57.59 | *** join/#asterisk spartan7 (n=spartan7@wsip-70-169-241-66.oc.oc.cox.net) |
23:59.35 | lesouvage | [[tK]D-Fender: sorry, it was a stupid typo. I have the voicemail to e-mail working as it should with msmtp instead of sendmail and a gtalk account. msmtp is very very easy to configure. |