| 00:01.02 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 00:11.02 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 00:17.01 | *** join/#asterisk smeet2002 (~smeet2002@dsl-69-172-120-166.acanac.net) |
| 00:18.26 | *** join/#asterisk Fabiano_heringer (~Fabiano@201.65.221.170) |
| 00:18.44 | greezmunkey | hehe I made it work! |
| 00:20.00 | greezmunkey | now I can clean up the dialplan, and set up for pstn access. |
| 00:20.47 | `mc | anyone install asterisk on ubuntu? |
| 00:21.24 | Fabiano_heringer | with asterisk 1.6 it´s possible to preserve the callerid when making blind/atx transfer? |
| 00:22.16 | WIMPy | blind only. For updates you need 1.8. |
| 00:22.44 | Fabiano_heringer | thks |
| 00:23.36 | Fabiano_heringer | im trying asterisk 1.8, i´m looking for a GUI for CEL, anybody know something? |
| 00:23.48 | greezmunkey | So an incoming call with cdi 12345, that is transfered from x100 to x200 will show x100 for caller id, not 12345?? |
| 00:24.16 | greezmunkey | s/cdi/cid |
| 00:26.15 | WIMPy | If it's blind, 12345, if it's attended x100 and if you're running 1.8 it will change to 12345 when you finish the transfer. |
| 00:26.44 | Fabiano_heringer | that´s exaclty I need... |
| 00:26.54 | greezmunkey | WIMPy: ok, I see. thanks. |
| 00:27.37 | Fabiano_heringer | what about CEL, that´s some graphical interface for the reports? |
| 00:28.33 | ChannelZ | smeet2002: Would there be a reason we'd want to? |
| 00:30.07 | drmessano^ | Why is www.iaxacat.com down? |
| 00:32.12 | *** join/#asterisk arielb27 (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 00:36.45 | smeet2002 | @ChannelZ The reason is fun and test |
| 00:40.07 | *** join/#asterisk cynicismic (~rclark@188-223-82-64.zone14.bethere.co.uk) |
| 00:41.05 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:41.32 | greezmunkey | Nice! In the mp-118 gateway you can tie a sip extension to a particular port. It's simple, and it works! |
| 00:42.11 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176003193.dsl.bell.ca) |
| 00:42.48 | dlynes | Is there a way to check from the cli to see what the deny and permit settings are for each sip peer? |
| 00:44.21 | p3nguin | greezmunkey: CallerID should always display the info of the person who will be on the other end when you pick up your phone: If a guy at 100 does an attended transfer to 200, 200's phone should show 100 since that is who will be on the other side. If 100 did a /blind/ transfer to 200, the display on 200 should show the original caller's CID since _that_ is who will be on the other side. |
| 00:44.40 | drmessano^ | smeet2002, I think the idea of centralized peering has long been abandoned in favor of URI dialing and centralized directories like ENUM |
| 00:44.52 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
| 00:45.29 | `mc | I downloaded AsteriskNOW and I am not sure if I should installed Asterisk+FreePBX or Asterisk+Asterisk-GUI..any suggestions? |
| 00:45.31 | `mc | I just wanna get started |
| 00:46.28 | drmessano | Forget Asterisk-GUI, IMO |
| 00:46.39 | drmessano | FreePBX is the way to go |
| 00:46.45 | `mc | thanks! |
| 00:46.53 | greezmunkey | p3nguin: ok, so if the guy at 100 does an attended transfer, then (prior to rel 1.8) the caller at 200 only ever sees "100", at least that was my take away. |
| 00:47.16 | smeet2002 | drmessano, thanks for answering.But the only thing I am new to this technology..so I have no idea what are you talking about. Can you give me some links? I just installed Asterisk 1.8, connected two IP phones, they are working. I would like to test it further. I can connect with DIgium technical support through their demo dialplan. But I would like to connect anybody else.. |
| 00:47.58 | drmessano | smeet2002, go to Flowroute.com, sign up for an account, and set up a SIP peer to them.. You have $.50 test credit to play with, I believe |
| 00:48.02 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 00:48.21 | greezmunkey | p3nguin: Is that about right? |
| 00:48.35 | drmessano | OOPS $.25 |
| 00:48.42 | drmessano | But that's plenty for a few test calls |
| 00:48.57 | dlynes | woah...asterisk 1.8 is out? |
| 00:49.02 | dlynes | Congratulations guys! |
| 00:49.04 | drmessano | Um cha? |
| 00:50.22 | smeet2002 | WOuld be great to connect with somebody's asterisk by IAX |
| 00:50.48 | p3nguin | greezmunkey: Even with 1.8, it should still operate the same way in that if 100 does an attended transfer to 200, 200 sees 100 since 100 is placing the call to 200... |
| 00:51.14 | p3nguin | greezmunkey: The difference is that in 1.8, the CID will update after the attended transfer has been completed. |
| 00:51.46 | p3nguin | The update allows the CID to show who is on the other side of the line. |
| 00:52.52 | greezmunkey | p3nguin: Got it. So then in the case of forwarding a call out to your cell phone, you should see the original callers cid. |
| 00:52.55 | p3nguin | If you always want to see the original caller's ID, modify the dialplan or don't do attended transfers. |
| 00:53.28 | p3nguin | greezmunkey: If you mean forward using the forward button on your phone, yes, you should see the caller's ID. |
| 00:53.31 | drmessano | smeet2002, if someone wants to allow a call, you can make a peer-peer call via IAX... |
| 00:54.02 | p3nguin | greezmunkey: But that can be broken if your dialplan always forces a caller ID number to be set on all outgoing calls. |
| 00:54.41 | smeet2002 | drmessano, I didn't get...what do you mean? |
| 00:54.45 | greezmunkey | p3nguin: the reason I get sticky about that, is especially in the case of forwarding, some systems I am familiar with either will not show the origcid, or you have to buy a license! Man I love this stuff! |
| 00:55.21 | greezmunkey | p3nguin: Good to know, I'll definately remember that. |
| 00:55.54 | smeet2002 | 1st Interval Sub Interval Rate |
| 00:55.54 | smeet2002 | REUNION ISLAND 262262 30 sec 6 sec $0.178 |
| 00:56.03 | smeet2002 | Who knows what it all means? |
| 00:56.12 | smeet2002 | Is it a price for 1 minute? |
| 00:56.19 | smeet2002 | What are intervals? |
| 00:56.26 | p3nguin | Which provider? |
| 00:56.51 | smeet2002 | http://flowroute.com/services/rates/ |
| 00:57.00 | p3nguin | Price is typically per minute and it will be billed every 6 seconds. |
| 00:57.03 | drmessano | .178 cents per minute.. Billed at 30 seconds minimum, then at each 6 second mark |
| 00:57.12 | dlynes | smeet2002, It's probably 17.8c/min, minimum 30 seconds, billed in 6 second increments |
| 00:57.17 | p3nguin | It looks like that particular number has a 30 seco.... what drmessano said. |
| 00:58.04 | smeet2002 | yes..it's definitely 17,89 cents for a minute.. |
| 00:58.06 | dlynes | drmessano, check your decimal spot again, and that dollar sign :) |
| 00:58.22 | p3nguin | hahaha Verizon math! |
| 00:58.28 | greezmunkey | heh |
| 00:58.36 | smeet2002 | so 6 secs is billing interval...thank you very much |
| 00:58.37 | drmessano | Yeah yeah |
| 00:59.10 | drmessano | I can't think and type |
| 00:59.18 | drmessano | So take what you get |
| 01:00.04 | drmessano | Trying to listen to the USC and Georgia games on the radio, laundry, and make excuses at the same time |
| 01:00.17 | drmessano | CANT DEW IT ALL |
| 01:03.12 | *** join/#asterisk alexshell (~chatzilla@unaffiliated/alexshell) |
| 01:08.47 | *** join/#asterisk jsjc (~jsjc@117.Red-79-146-228.dynamicIP.rima-tde.net) |
| 01:11.03 | *** join/#asterisk cabrel (~robin@c-68-62-212-11.hsd1.al.comcast.net) |
| 01:12.56 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 01:34.21 | *** join/#asterisk cynicismic (~rclark@188-223-82-64.zone14.bethere.co.uk) |
| 01:41.33 | `mc | anyone successfully install asteriskNOW on VirtualBox? |
| 01:43.02 | p3nguin | of course |
| 01:43.50 | `mc | have you? |
| 01:46.19 | p3nguin | Sure, and you can too if you'll download the iso image and just do it. |
| 01:52.15 | `mc | p3nguin, I have, its the networking that I was having issues with..accessing the configuration site |
| 01:52.30 | `mc | I actually just got it to work..I added a bridged network adapter |
| 01:52.51 | `mc | in VirtualBox and boot it back up..all is well now...BUT..now that I can acccess the FreePBX site...then what? |
| 01:55.19 | `mc | whats the command in AsteriskNOW to bring up the configruation/setup console...so I can add user and what not |
| 01:56.09 | p3nguin | All that is done in FreePBX. |
| 01:56.46 | `mc | I can't seem to log in as root |
| 01:57.18 | riscphree | user: maint |
| 01:57.37 | `mc | riscphree, didn't work |
| 01:58.30 | `mc | mind you, I just installed everything, it tells me in my login banner to access the FreePBX site but when I do, it doesn't accept my root user and password |
| 01:58.40 | p3nguin | I think you should read the AsteriskNOW startup guide. |
| 01:58.52 | p3nguin | You won't be using root. |
| 01:58.54 | `mc | looking for it |
| 01:58.58 | `mc | can't find it |
| 01:59.37 | `mc | they should make this more straight forward for ignorant people like myself |
| 01:59.49 | `mc | you guys are used to all this..so its easy |
| 01:59.51 | `mc | 'p |
| 02:03.43 | `mc | I can log into linux with my root login and pass |
| 02:03.48 | `mc | but when I go to the site, no dice |
| 02:05.22 | p3nguin | http://www.asterisk.org/AsteriskNOW-1.5-QuickStart |
| 02:05.40 | p3nguin | FreePBX (the default configuration interface for Asterisk) |
| 02:05.41 | p3nguin | SEE THIS |
| 02:06.01 | `mc | thanks! |
| 02:06.51 | `mc | haha |
| 02:07.02 | `mc | the default l/p is freepbx/fpbx |
| 02:07.06 | `mc | that doc is key..thanks |
| 02:08.40 | `mc | ok..how can I test Asterisk now that I can access FreePBX admin? |
| 02:08.59 | `mc | doc porn welcomed |
| 02:12.58 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:17.06 | `mc | anyone recommend a good document to get started on this whole voip+asterisk+freepbx? |
| 02:17.15 | `mc | other than just googling endlessly? |
| 02:20.19 | alexshell | what asterisknow version did you install? |
| 02:26.49 | ChannelZ | #asterisknow #freepbx ANYWHERE BUT HERE! |
| 02:32.40 | drmessano | ChannelZ, that's not necessary |
| 02:33.09 | *** join/#asterisk jazzy-J (~jazzy-J@92.249.107.65) |
| 02:36.57 | ChannelZ | You're right. FreePBX is totally not necessary |
| 02:38.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:40.16 | drmessano | You know there's been a lot of folks being banned lately for the sort of attitude you're showing right now, right? |
| 02:41.21 | ChannelZ | Telling people where they're more likely to get help? |
| 02:41.31 | jdoe | angrily telling anyone anything. |
| 02:42.23 | drmessano | ChannelZ, give me a break.. the whole "YOU CANT GET HELP HERE BECAUSE U HAVE FREEPBX AND IT SUCKS" is not "helpfully giving helpful help as to where to get help" |
| 02:42.29 | ChannelZ | If I'm angry I'll curse more |
| 02:42.49 | dive | I have just installed asterisk on a server and linphone on my laptop which connects to fine so far. Is there a way of having asterisk transmitting my linphone online status? |
| 02:43.13 | dive | I added myself as a contact but it says offline |
| 02:44.52 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:56.33 | *** join/#asterisk cynicismic (~rclark@188-223-82-64.zone14.bethere.co.uk) |
| 03:08.45 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 03:34.28 | greezmunkey | uh, why is a morse code module installed, from the latest book,"(speaking of analog telephony) That technology is over a hundred years old..." |
| 03:35.04 | greezmunkey | I know I can unload it, but still - really?!? |
| 03:36.15 | greezmunkey | Eh, ok I ccan think of a reason - for communicating with a ham readio interface, and then sending out morse code! |
| 03:36.36 | greezmunkey | needs to learn to type... |
| 04:02.01 | greezmunkey | *Serious* question: I built one of the examples from the book, where the system should read back numbers: The system makes no complaint about the logic, but throws this error: -- Executing [333@phones:3] SayNumber("SIP/1001-00000001", "1") in new stack -- <SIP/1001-00000001> Playing 'digits/1.ulaw' (language 'en') -- Auto fallthrough, channel 'SIP/1001-00000001' status is 'UNKNOWN' |
| 04:03.31 | greezmunkey | I initially ran the test, but found that I had not loaded the appropriate package. It threw a file not found error. I believe i fixed that, but get the above error now. I don't get it. |
| 04:03.57 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 04:04.08 | phix | :D |
| 04:05.49 | greezmunkey | Duh!, no goto statement...perhaps |
| 04:08.03 | greezmunkey | nope :/ |
| 04:12.13 | *** join/#asterisk thansen (~thansen@174.253.164.255) |
| 04:24.40 | *** join/#asterisk dobah (~dobah@cpe-98-150-177-51.hawaii.res.rr.com) |
| 04:30.45 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 04:45.35 | phix | heh |
| 04:49.11 | *** join/#asterisk eggie5 (~eggie5@adsl-71-135-164-39.dsl.pltn13.pacbell.net) |
| 04:49.55 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 04:53.29 | eggie5 | i installed asterisk (w/ homebrew on osx) but I can't find my conf files in /etc/asterisk - shouldn't that be there? |
| 05:02.58 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 05:10.56 | robl^laptop | eggie5: not sure about "home brew", but if you installed from source, the asterisk config files are npt created by default |
| 05:11.15 | robl^laptop | you have to "make samples" to get a copy of the default config files |
| 05:11.34 | eggie5 | robl^laptop: oh is that right? |
| 05:11.42 | eggie5 | robl^laptop: home brew is a package manager for osx |
| 05:12.26 | robl^laptop | eggie5: not sure about how homebrew packages it, but, from source, getting the sample files installed requires an extra step. |
| 05:15.27 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 05:15.55 | eggie5 | robl^laptop: where does the asterisk binary expect the config files to be located? |
| 05:16.07 | robl^laptop | those that are comfortable with asterisk tend to create their own config files from scratch anyway... the samples are just samples.. nothing that is "required". They tend to be a bit bloated for production use. |
| 05:16.08 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 05:16.32 | robl^laptop | by default /etc/asterisk, unless a different path is given at compile time |
| 05:18.38 | robl^laptop | shhh |
| 05:18.40 | robl^laptop | ahhh, even |
| 05:18.47 | robl^laptop | looked at homebrew's site |
| 05:19.05 | robl^laptop | it's building from source using a simple ruby script |
| 05:21.43 | ChannelZ | ruby ruby ruby ruuuubaaayyy |
| 05:22.10 | robl^laptop | eggie5: https://github.com/mxcl/homebrew/blob/master/Library/Formula/asterisk.rb <-- take a look , home brew is building and installing asterisks. "make" and "make install". there is no "make sample", so they were not installed anywhere on your system |
| 05:22.56 | eggie5 | robl^laptop: but so then is it looking in /etc/asterisk for the cofigs? |
| 05:23.05 | eggie5 | i just need to add them there? |
| 05:23.33 | robl^laptop | eggie5: yes, from what I can tell. it just needs to have config files in /etc/asterisk |
| 05:24.31 | eggie5 | robl^laptop: is running make samples pretty common thing? should I aadd it to that asterisk recipie? |
| 05:25.20 | robl^laptop | make samples is common for someone testing or learning asterisk.. but once you know what you are doing, most people don't use the samples.. they create custom config files from scratch |
| 05:25.52 | ChannelZ | I think the default is to overwrite so it might not be a good idea in the case where someone's upgrading |
| 05:26.12 | robl^laptop | again.. they are "samples", not necessarily what you want in a real working system. |
| 05:26.38 | robl^laptop | ChannelZ: that too. I forgot about the overwriting aspect. |
| 05:27.06 | eggie5 | robl^laptop: oh ok thanks |
| 05:27.09 | ChannelZ | personally I make a "dist" dir in my /etc/asterisk and copy all the sample configs out of the source dir into it so I have them for reference |
| 05:28.10 | robl^laptop | eggie5: also, note that the current formula is installing a rather outdated version of asterisk.. so there may be bugs in that version that have be fixed in newer versions. |
| 05:28.45 | eggie5 | robl^laptop: oh - i should update it then - what's a good current version? |
| 05:29.33 | robl^laptop | I have all of my config files in a private subversion repo..its a good way to be able to reinstall if something happens... |
| 05:30.12 | ChannelZ | I've actually thought about doing that |
| 05:30.44 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 05:30.56 | robl^laptop | eggie5: personally, I'd go with the latest release of 1.6.2.14. 1.8 is still a little too new for my tastes |
| 05:31.08 | eggie5 | robl^laptop: ok |
| 05:34.12 | robl^laptop | my main "concern" with 1.8 is that a lot of 3rd party add-ons haven't been updated to work with the changes in the new series |
| 06:06.11 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 06:06.11 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 06:10.01 | sshock | so I'm getting brute-force registration attacks all the time now; reminds me a lot of the constant SSH attacks I also see |
| 06:10.58 | sshock | of course, none of them are succeeding, but it still fills up my logs and it probably would help to at least try and slow them down |
| 06:12.23 | sshock | anyone else worked on ways to solve this, e.g., with some firewall rules? |
| 06:12.46 | sshock | I have some firewall rules to slow down SSH attacks; perhaps I could use something similar here... |
| 06:19.53 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:20.28 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:22.50 | sshock | any other ideas? |
| 06:28.32 | pabelanger | sshock: change your UDP port |
| 06:29.05 | sshock | hmm, not a bad idea |
| 06:31.05 | sshock | as long as my voip provider lets me specify a port for inbound calls (from DID) |
| 06:31.40 | pabelanger | sshock: If you are registering with them, it should not be an issue |
| 06:32.42 | sshock | ok |
| 06:33.19 | sshock | well, anyway, I just looked at my provider (flowroute) and it supports both a host[:port] or a SIP URI for inbound routes |
| 06:33.54 | sshock | the SIP URI doesn't support a port, |
| 06:34.10 | sshock | but the host one does, also it says "(Not required if using SIP registration for inbound. DIDs route via SIP Registration by default.)" |
| 06:34.58 | dive | does anyone know how to show and online/offline status for a user? |
| 06:36.51 | *** join/#asterisk tyman (~tyler@c-67-160-233-60.hsd1.ca.comcast.net) |
| 06:43.50 | dive | or extension, I probably mean |
| 06:44.32 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 06:45.11 | sshock | dive: you mean like whether the line is on-hook or off-hook? |
| 06:45.36 | dive | yeah |
| 06:46.15 | dive | what I'm doing is using linphone and added myself as a contact but it always shows me offline |
| 06:46.31 | dive | for myname@mydomain.com |
| 06:47.03 | sshock | sip show channels |
| 06:47.12 | sshock | might be what you're looking for |
| 06:47.45 | dive | hmm shows 0 active dialogs |
| 06:47.47 | sshock | but I don't think a channel exists just when your phone is off the hook; it has to actually be connected to another phone I think |
| 06:48.13 | dive | but ideally I would like it to show if I am running linphone as online, and if not then offline |
| 06:48.34 | dive | to other users |
| 06:49.19 | dive | not just local users, but from the internet |
| 06:49.22 | sshock | well, I know if you turn up the debug level (e.g., core set verbose 10) |
| 06:49.36 | sshock | then you will see a message when your phone registers or unregisters with * |
| 06:49.47 | dive | yeah but I don't want to read it in the console |
| 06:49.59 | dive | I want it to show in linphone as online |
| 06:50.08 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 06:51.10 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 06:51.29 | sshock | yeah, I'm not sure what you can do |
| 06:54.50 | sshock | hmm, "sip show peer" might help |
| 06:55.50 | sshock | these fields are different depending on whether the device registered or not: "Expire", "Addr->IP", "Useragent", "Reg. Contact" |
| 06:56.44 | sshock | are you a linphone developer? |
| 06:57.19 | dive | no |
| 06:57.55 | sshock | and where exactly do you want this online/offline to show up? |
| 06:58.00 | dive | hmm Status: unmonitored looks interesting |
| 06:58.13 | dive | in other users linphones |
| 06:58.19 | sshock | mine shows Status: unmonitored all the time |
| 06:59.10 | sshock | so are you planning to to create a special version of linphone by modifying the source? |
| 07:00.03 | dive | no, I just want it to show when an extension is online or not |
| 07:00.22 | sshock | but how are you going to change what linphone shows without modifying linphone? |
| 07:00.26 | dive | currently it shows I'm offline, when I'm online |
| 07:00.56 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 07:00.58 | dive | I just want asterisk to relay the extension status |
| 07:01.31 | sshock | but you are talking about status of other extension, not your own extension, right? |
| 07:01.37 | dive | yeah |
| 07:02.32 | sshock | but linphone doesn't have a place where it shows status about other extensions, does it? |
| 07:02.44 | dive | yes it has a contact list with status |
| 07:02.56 | sshock | oh, ok; I hadn't seen that before |
| 07:03.19 | sshock | ok, the presence status... |
| 07:03.30 | dive | yes |
| 07:04.00 | sshock | ok, yeah I'm not sure what is required to get that working properly |
| 07:04.33 | sshock | maybe it is a bug in * |
| 07:04.42 | sshock | what version of * you have? |
| 07:06.06 | sshock | try doing some google searches for Asterisk presence or something |
| 07:06.09 | dive | asterisk-1.6.1.6 |
| 07:06.20 | sshock | here you go; read this: http://www.voip-info.org/wiki/view/Asterisk+presence |
| 07:06.53 | sshock | looks like you may need to add some extra stuff to extensions.conf and sip.conf to get presence working |
| 07:07.04 | sshock | anyway, gotta run; good luck with it |
| 07:07.08 | *** part/#asterisk sshock (~sshock@sshock.net) |
| 07:09.07 | eggie5 | hi - so i set up my asterisk to work w/ x-lite phone |
| 07:09.19 | eggie5 | what is a quick test I can do? |
| 07:09.32 | eggie5 | like add something to my extensions.conf file |
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| 08:08.07 | Weazel | heys guys, i'm searching for a solution to send ppl to an IVR before they reach VoiceMail, to give an option to either leave a message to the person or go to another extensions, but i'm pretty noobish... can anyone help ? |
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| 08:23.46 | ChannelZ | Well you basically just do 'Background' on a sound ("your party did not answer, if you wish to try a different extension dial it now") maybe followed by a WaitExten(3) or something |
| 08:24.22 | ChannelZ | then send them to voicemail for the 'current' extension. |
| 08:27.28 | Weazel | ChannelZ: thats definitely what i wanna do, prob is, i dunno how to write that dialplan for the 'current' extension part |
| 08:27.47 | ChannelZ | it's just sequential |
| 08:28.04 | ChannelZ | exten => 111,1,Dial(SIP/Whoever,20) |
| 08:28.20 | ChannelZ | exten => 111,n,Background(nobody-answered-or-is-busy) |
| 08:28.25 | Weazel | but thats writing a line for every single exten, and future extens |
| 08:28.36 | Weazel | can't i like make a default theme ? |
| 08:28.44 | Weazel | so even if i add a new exten it'll go there by default |
| 08:28.50 | Weazel | and choose the 'corrent' ext ? |
| 08:28.51 | ChannelZ | So make it a macro or Goto a common handler |
| 08:29.37 | Weazel | ok but how does the line for 'current' ext will look like in the macro ? |
| 08:30.55 | ChannelZ | Well go google "asterisk macros" and read up, it's a little more complicated than explaining here |
| 08:31.58 | ChannelZ | but basically you create an extension in a special 'macro context' of sorts, and then call it like a dialplan application with arguments |
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| 08:35.29 | ChannelZ | Also I guess you can use the 'd' flag in VoiceMail() to let people type in a different extension while someone's greeting is playing. But you're sort of asking two questions anyway |
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| 08:40.35 | Weazel | ChannelZ: only relevant to greeting though thanks a lot for ur help |
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| 10:13.51 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 10:52.04 | *** join/#asterisk greezmunkey (~greezmunk@173.241.161.62) |
| 10:54.54 | greezmunkey | Heres kind of a wierd one. Asterisk 1.6.2.14 built by root @ localhost.localdomain on a i686 running Linux (CentOS) Data at: http://pastebin.com/PLSdYFd0 - the issue is that dialing 333 should playback digits. i checked /var/lib/asterisk/sounds/en/digits - it's there and has all of the files needed. |
| 10:55.35 | ChannelZ | Answer() first probably |
| 10:55.58 | greezmunkey | ChannelZ: It can't be that simple...I'll add it. |
| 10:56.32 | ChannelZ | It might be. Some apps answer the channel if it's not automatically, but some do not |
| 10:56.41 | ChannelZ | SayDigits might be one of the |
| 10:57.41 | ChannelZ | s/the/them/ |
| 10:57.47 | greezmunkey | ChannelZ: you rock, it needed it, as well as a Goto statement - this was not in the example in the book! |
| 10:59.13 | ChannelZ | adds it to the list |
| 11:00.30 | greezmunkey | ChannelZ: Yeah, theres even more to add, I'll make it work and paste what I come up with. |
| 11:00.53 | greezmunkey | ChannelZ: not that I am any kind of expert (yet) |
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| 11:01.24 | ChannelZ | MusicOnHold is one, I think MeetMe acts funny IIRC, etc. |
| 11:01.44 | ChannelZ | In general if you're going to play something you should always Answer the channel yourself first |
| 11:03.01 | greezmunkey | ChannelZ: you mean with Answer(), or something I've done wrong in here? |
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| 11:05.17 | ChannelZ | yes with Answer() |
| 11:07.00 | ChannelZ | Like I said a few apps will automatically Answer the channel if it hasn't been already (like Playback) but others do not.. so in general if you are doing something in your dialplan which is going to either make noise, it's a good idea to explicitly Answer() the channel yourself first |
| 11:07.38 | ChannelZ | s/make noise/make noise or listen to any/ |
| 11:07.53 | ChannelZ | yadda yadda |
| 11:08.00 | ChannelZ | I should be in bed |
| 11:08.03 | greezmunkey | ChannelZ: I see. |
| 11:08.36 | greezmunkey | ChannelZ: You've been here since *early* yesterday. |
| 11:09.23 | greezmunkey | ChannelZ: I see how this works. it's a lot like programming in basic. |
| 11:12.25 | ChannelZ | heh right down to the line numbers |
| 11:13.29 | ChannelZ | There's always AEL which is a little more structured |
| 11:13.33 | greezmunkey | ChannelZ: The new book is nice. i can't wait until i can get a hardcopy of it. |
| 11:13.49 | ChannelZ | which? |
| 11:14.12 | greezmunkey | ChannelZ: Future of Telephony 3 |
| 11:14.49 | greezmunkey | ChannelZ: Here's the pre print version: http://ofps.oreilly.com/titles/9780596517342/ |
| 11:15.40 | ChannelZ | oh didn't know that was posted. Knew they were working on it |
| 11:16.52 | greezmunkey | ChannelZ: I was in a a few days ago asking questions, Russel Bryant was in here, and posted the link. |
| 11:27.21 | greezmunkey | ChannelZ: Is there a document that details all of the applications and functions? |
| 11:28.02 | ChannelZ | help |
| 11:28.19 | ChannelZ | I mean "core show applications" |
| 11:28.37 | ChannelZ | then find something that sounds interesting and "core show application xxxxx" |
| 11:29.54 | ChannelZ | and conversely "core show functions" and "core show function xxxxx" |
| 11:31.18 | greezmunkey | ChannelZ: I see. I ran dialplan show, and it output what it made out of my mess. Nice and clean. I haven't tried any of this stuff. Good call. |
| 11:32.49 | greezmunkey | mmm Almond Roca, I'd give you one but...I can't encode it into g711 yet. |
| 11:35.18 | ChannelZ | heh |
| 11:35.43 | ChannelZ | Where is my GD teleporter or flying car I was promised? |
| 11:36.10 | greezmunkey | ChannelZ: My jetpack is out of gas, can't get a refill. |
| 11:37.11 | greezmunkey | ChannelZ: can you imagine 50 million senior citizens in flying cars... |
| 11:39.35 | ChannelZ | No worse than on the roads I suppose. |
| 11:39.47 | ChannelZ | If we had flying cards they'd mostly fly themselves |
| 11:39.54 | Weazel | whenever i get to voicemail i hear vm-intro.gsm is there a way to add the file vm-reachoper.gsm after or before the vm-intro is played ? |
| 11:41.18 | ChannelZ | Besides hacking the source, not that I'm aware of. You could make a new sound that has both. |
| 11:41.34 | Weazel | and replace the current vm-intro.gsm ? |
| 11:41.47 | ChannelZ | yeah |
| 11:42.04 | ChannelZ | actually.. |
| 11:42.08 | Weazel | how do i edit gsm sound files ? any good known softwares i can use ? |
| 11:42.37 | ChannelZ | just get the wav versions, edit with whatever you like, and then recompress to gsm if that's what you want |
| 11:42.42 | ChannelZ | or leave it as wav and let asterisk do it |
| 11:42.54 | Weazel | how do i recompress to gsm ? |
| 11:43.07 | ChannelZ | sox or something |
| 11:43.41 | ChannelZ | or 'file convert' on the console |
| 11:44.05 | coppice | if you just want to trim the GSM audio in time, don't decompress and recompress. Just trim frames. A recompression looses a *lot* of quality |
| 11:44.56 | Weazel | can i "file convert vm-intro.gsm vm-intro.wav" ? |
| 11:45.10 | Weazel | coppice: i don't want to trim, i want to add another file into it |
| 11:45.12 | ChannelZ | That's why I say get the wav prompts, edit those, and then compress back to gsm |
| 11:45.37 | Weazel | ok i'll try tnx |
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| 11:45.40 | ChannelZ | You might be able to just 'cat' the files into each other... do gsm files have headers? |
| 11:45.49 | ChannelZ | isn't sure |
| 11:45.58 | coppice | well, any decompression and recompression of low bit rate codecs has a horrible effect on quality |
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| 11:46.50 | Sheeplet | lo all |
| 11:47.14 | ChannelZ | the gsm files might just be raw data. cat vm-reachoper.gsm >> vm-intro.gsm |
| 11:47.27 | ChannelZ | (make a copy first.) see what happens |
| 11:47.35 | Sheeplet | hi all |
| 11:47.40 | ChannelZ | ahoyhoy |
| 11:51.51 | greezmunkey | g'night all. |
| 11:51.58 | ChannelZ | night |
| 11:53.57 | Weazel | ChannelZ: worked like a charm, though there is not delay to give a user a chance to hit the 0 before the beep |
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| 11:54.48 | ChannelZ | cat a second or so of silence onto the end.. from the silence directory |
| 11:55.31 | ChannelZ | (but I think they can hit 0 while the message is recording and it will pop out, won't it?) |
| 11:56.20 | Weazel | yea but it'll make a big confusion |
| 11:56.24 | Weazel | but silence worked |
| 11:56.26 | Weazel | thanks a million |
| 11:56.32 | v1s | if all my calls are coming from a provider that supports sip and iax and i have to asterisk boxes connected together and all calls are terminating to gsm. I have just a few sip clients should I try to use iax as much as i can or should I use sip? |
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| 11:57.18 | ChannelZ | sip is fine |
| 11:58.19 | ChannelZ | IAX is good for connecting *'s together or trunking a lot of calls over one connection than SIP because it's more bandwidth effecient |
| 11:58.47 | v1s | so i should use iax if i am trying to save some bandwidth |
| 11:59.22 | ChannelZ | between you and your provider sure, particularly if they can do IAX trunking |
| 11:59.54 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 11:59.56 | [sr] | howdy |
| 12:00.09 | v1s | ChannelZ: thanks for the feedback |
| 12:01.01 | ChannelZ | sure. Bed time for me. |
| 12:01.56 | Weazel | can i choose different extensions for operator extensions ? like lets say exten 500 has 200 as the operator and 501 has 201 ? (when hit 0 on vm) |
| 12:02.55 | [sr] | gud night ChannelZ |
| 12:02.56 | ChannelZ | I believe it jumps to the 'o' extension so the only way to make them different would be by context |
| 12:03.59 | Weazel | where does the context appear ? which file i mean |
| 12:05.43 | ChannelZ | it's whatever context you're in when you went to voicemail |
| 12:06.00 | ChannelZ | although actually I think there is a channel var you can set to make it "exit" to a different context |
| 12:06.52 | ChannelZ | or no that's actually a config option, not a variable. |
| 12:07.10 | Weazel | well i know it goes to default context in voicemail, question is, where do i see default's dialplan itself |
| 12:07.18 | ChannelZ | no not voicemail context, but the dialplan context. |
| 12:07.40 | Weazel | confused |
| 12:08.07 | ChannelZ | If you were in the 'incoming' context when you called Voicemail(xxx) then it'd jump to the 'o' extension of the same 'incoming' context if they hit 0 |
| 12:08.24 | ChannelZ | Voicemail contexts are different than dialplan contexts |
| 12:10.10 | Weazel | well since i'm using freepbx, i guess it'll be as u say at o@macro-vm, which means freepbx will overwrite my changes :S |
| 12:10.14 | Weazel | stupid freepbx |
| 12:13.19 | ChannelZ | well with freepbx I have no idea |
| 12:14.17 | Weazel | oh great found a solution, if i enable vmx locator and change the dial 0 there, it affects the ext |
| 12:14.18 | Weazel | :D |
| 12:14.23 | Weazel | hurray! |
| 12:14.37 | Weazel | ChannelZ: thanks a lot for ur help :D |
| 12:15.05 | ChannelZ | FWIW. I didn't know we were talking freepbx but glad you figured out how to make it do it |
| 12:16.49 | Weazel | we weren't talking freepbx, i did all the changes in console and shell til the last part |
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| 13:11.17 | z4nD4R | somebody with experiens how to make tls peering between asterisk and SipXec ? |
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| 13:59.37 | cvnet | hi all |
| 13:59.52 | z4nD4R | hi |
| 14:00.55 | cvnet | i want a registered user to check his voicemail wihtout entering password, but cant do it, need help http://pastebin.com/2EG3Lbac |
| 14:01.21 | cvnet | voicemial mailbox =6100 pass=1111 |
| 14:05.32 | cvnet | anyone? |
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| 14:06.42 | WIMPy | core show application voicemailmain |
| 14:09.57 | cvnet | read taht and still teh same |
| 14:10.01 | cvnet | i did exten => _X.,n,VoiceMailMain(6100,6100, 1111) |
| 14:10.26 | cvnet | wont work , it asks for password, and the funny thing is it doesnt read the password, but if i can outside line dtmf is working fine |
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| 14:11.11 | z4nD4R | <PROTECTED> |
| 14:11.35 | WIMPy | cvnet: That does not fit in to the syntax I get displayed. |
| 14:11.54 | cvnet | oo i have to do exten => _X.,n,VoiceMailMain(6100,6100, s) |
| 14:11.57 | cvnet | correct? |
| 14:12.49 | cvnet | woow |
| 14:12.51 | cvnet | thanks it works now |
| 14:13.19 | cvnet | but it does not read my dtmf (anything i press is ignored) but if i call outsideline dtmf is working fine... what could that be? |
| 14:13.31 | WIMPy | With three parameters? Interesting. i only see two documented. |
| 14:13.57 | WIMPy | What kind of phones? What kind of connection outside? |
| 14:14.46 | cvnet | lets say if i call outside line to order pizza, it says press 1 to place your order now, that is read by other system |
| 14:15.17 | cvnet | but when im directly connected to asterisk box via sip, and i want to check my voicemail, it does not read 1 when i try to hear my messages |
| 14:16.40 | WIMPy | Theck the dtmf settings of you phone and Asterisk. bet choice is usually RFC2833. |
| 14:17.18 | cvnet | got it working |
| 14:17.20 | cvnet | thanks alot for your help |
| 14:17.24 | cvnet | really appreciate it |
| 14:18.04 | WIMPy | should learn not to leave out too many letters. |
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| 15:19.02 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:19.02 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 15:22.27 | cvnet | can we changed the voicemail format to mp3? |
| 15:28.15 | smeet2002 | cvnet.sorry it's an asnwer to your question, I am a newbie...the only thing I want to metnion that it's said in the book mp3 puts a heavy load on you CPU, WAV doesn't. SO if you have a big enough hard drive its' better to have all voice files in WAV |
| 15:28.24 | smeet2002 | it's NOT an answer |
| 15:28.34 | smeet2002 | to mention |
| 15:31.49 | WIMPy | No. mp3 is playback only. |
| 15:32.29 | cvnet | so can i have it emailed to me mp3 format? |
| 15:32.34 | cvnet | so i can check it from my iphone? |
| 15:32.56 | WIMPy | Only if you convert it yourself. |
| 15:33.10 | WIMPy | playback only = no recording |
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| 15:33.31 | smeet2002 | cvnet..I beleive it's possible to write a script, that could check folders for voice file on regular basis and do conversion |
| 15:33.40 | cvnet | ok |
| 15:33.41 | cvnet | thanks |
| 15:35.11 | russellb | there is an option to have a script executed every time voicemail state changes (new message added, message deleted, message listened to, ..) |
| 15:35.15 | russellb | you could use that as your hook for conversion |
| 15:35.26 | russellb | look for externnotify in voicemail.conf.sample |
| 15:38.40 | cvnet | thanks |
| 15:40.49 | smeet2002 | may be somebody here still wants to test my home Asterisk installation? |
| 15:41.08 | smeet2002 | I don't want to buy DID right now.. |
| 15:41.38 | smeet2002 | or may be somebody knows about free SIP on the Internet? |
| 15:41.45 | russellb | are you in the US? |
| 15:41.50 | smeet2002 | in Canada |
| 15:42.16 | smeet2002 | but I don't have PSTN phone |
| 15:42.30 | russellb | ah, was going to recommend google voice, but that won't really work for you. |
| 15:57.39 | jsjc | is there any posibility in voicemail to avoid the voicemail greeting? or configure where to get the greeeting from? |
| 16:03.12 | cvnet | jsjc you can create your own greeting, or just make it like 1 sec blank |
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| 16:44.40 | doolittlework | hi there a looking a way to protect my dialplan code, can one include your extensions.conf and store it on a remote server |
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| 16:51.05 | merlin8282 | Hi all. Do you know if Junghanns Hardware is working with asterisk 1.6 or 1.8 ? We're currently running 1.4.24.1-BRIstuffed-0.4.0-RC3e. |
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| 19:25.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 19:54.37 | leifmadsen | hey all |
| 19:55.02 | leifmadsen | anyone know if there is a way to tell from the hangup routine ('h' extension) who performed the hangup? |
| 19:56.42 | tzanger | leifmadsen: DIALSTATUS might help that |
| 19:57.18 | tzanger | leifmadsen: also it's my understanding that 'h' only occurs when the far side hangs up. My asterisk knowledge is dated though, so take it with a grain of salt |
| 19:57.34 | tzanger | unlike some people, I don't have a book contract to motivate me to know this stuff. :-p |
| 19:58.16 | WIMPy | That's what I remember as well, but I've never used it. |
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| 19:59.45 | WIMPy | The first call wuld continue in the dialplan if you want it to. |
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| 20:10.46 | dive | is there a way of *forcing* an extention to be online/available? |
| 20:11.03 | dive | as seen externally that is |
| 20:11.41 | dive | I've tried using hints but can't it to work with linphone, currently my status is always shown as offline |
| 20:11.50 | p3nguin | Extensions are never "offline" as long as pbx_config has loaded them. |
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| 20:12.19 | dive | well, maybe wrong choise of word |
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| 20:12.59 | dive | but when I look at my extention from outside my server it shows offline in linphone for some reason |
| 20:13.17 | dive | <PROTECTED> |
| 20:13.24 | dive | in cli if that is a clue |
| 20:13.40 | p3nguin | That's showing a device state, not an extension. |
| 20:14.52 | dive | hmm |
| 20:15.41 | dive | what can I do to show my linphone as online from external/internet? |
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| 20:20.47 | riddlebox | man I hate how many people are always scanning my ip my iptables and fail2ban are working overtime |
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| 20:22.25 | leifmadsen | tzanger: DIALSTATUS presumes I'm using Dial() |
| 20:22.31 | [sr] | eia WIMPy |
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| 20:32.10 | tzanger | leifmadsen: yes, it does. :-) |
| 20:32.41 | tzanger | leifmadsen: when trying to figure out stuff like this I generally find it useful to try and dump all available variables and see if I can find something I can differentiate with |
| 20:32.56 | leifmadsen | agreed |
| 20:33.02 | leifmadsen | DumpChan() ftw |
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| 21:40.58 | alex5771 | hi is there any plac to get a DB of the area-code NPA NXX to switch or rate center,or address of the switch or company who owns that block,i remember a few years ago i was able to get TXT files either from FCC or NANPA with all that data for free? |
| 21:51.09 | ChannelZ | nanpa.com has an areacode list |
| 21:51.37 | alex5771 | ChannelZ: i need NXX also |
| 21:56.02 | ChannelZ | NXX is an area code but you also want prefixes which that I don't know |
| 21:56.43 | alex5771 | right i think they closed it down for "security" reasons a few years ago i could get nice files from NANPA |
| 21:58.49 | ChannelZ | Probably |
| 22:02.33 | greezmunkey | ChannelZ: did alex5771 leave, he could have gone to http://www.localcallingguide.com/lca_prefix.php - and doodled around a bit to get what he wanted. |
| 22:03.50 | greezmunkey | brb |
| 22:05.49 | ChannelZ | hmm interesting |
| 22:06.09 | ChannelZ | Guess he could query each area code and crawl all the results.. |
| 22:08.27 | greezmunkey | ChannelZ: yeah, I read the data into databases, and than can run queries against them to pull out what I need. |
| 22:10.31 | greezmunkey | ChannelZ: that method is very useful for programming least cost routing where local areacodes and ld areacodes are the same. |
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| 22:45.50 | starkovv | Hello everybody |
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| 22:47.18 | starkovv | Does anybody know how to monitor progress of sending fax through Asterisk with res_fax_digit module installed? |
| 22:47.31 | Nwallins | Hi, is there a way to use asterisk to make free calls to POTS numbers using VOIP? I see "VOIP boxes" with no recurring fees going for like $250, but I'm wondering if there is linux software that can do the same thing |
| 22:48.12 | starkovv | Misprint "res_fax_digium" |
| 22:48.26 | Nwallins | I have used SIP with Android / Ekiga / Diamondcard, but there are per-minute charges for calls within the US |
| 22:49.35 | starkovv | I use AMI to monitor fax events, but there only events for beginning of fax transmission end for finish |
| 22:50.43 | starkovv | But I need to see the middle of that, to calculate percentage of transmission progress. |
| 22:52.31 | starkovv | In CLI when i make "fax show session N" i see only "frames sent" parameter changing real-time |
| 22:54.16 | starkovv | If this is only one way to get fax real-time progress status, then how to calculate how many lines was sent of whole document? |
| 22:59.20 | Nwallins | how do these one-time-purchase VOIP boxes make free calls? |
| 23:01.17 | ChannelZ | Those devices aren't on my radar, can you name some? |
| 23:03.28 | Nwallins | checking.. i am thinking maybe $25 goes to the box, and $225 goes to the service provider, who figures the one-time income now is a good bet |
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| 23:08.42 | ChannelZ | Chances are yes; or up-sell other services, charge for support, etc. |
| 23:11.59 | ChannelZ | something like this: http://news.cnet.com/8301-13845_3-20010241-58.html |
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| 23:24.30 | Nwallins | ChannelZ: yep, exactly |
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| 23:34.01 | DEMNVT | Morning all |
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| 23:40.05 | DEMNVT | I'm having some trouble with getting one of our IAX trunks to register. Hoping you can help me |
| 23:40.33 | DEMNVT | It's a link between the pabx in our satellite office and our main office |
| 23:40.56 | DEMNVT | The only thing that's changed is that the satellite PABX was moved onto a new internet connection |
| 23:41.25 | DEMNVT | As far as I can tell there is no firewalling etc stopping it... but when the trunk tries to register it times out |
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| 23:44.05 | DEMNVT | What commands can I use to troubleshoot in the CLI |
| 23:44.08 | DEMNVT | ? |
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| 23:49.23 | DEMNVT | Actually... I think I just spotted the problem... It looks like the satellite PABX is sending through it's internal IP in the IAX packets rather than the WAN IP |
| 23:50.30 | greezmunkey | DEMNVT: I'm no expert here but it sounds like a natted connection to me. |
| 23:53.28 | drmessano | ~sipnat |
| 23:53.28 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:54.03 | drmessano | DEMNVT, Asterisk does that by design.. you need to use the parameters mentioned above to force asterisk to use the external IP |
| 23:54.13 | drmessano | and under which conditions to use it |
| 23:54.16 | drmessano | Read up |
| 23:54.49 | drmessano | Oh, IAX.. nevermind |
| 23:54.53 | DEMNVT | It's not SIP I'm worried about though. It's an IAX trunk |
| 23:55.09 | DEMNVT | we have another one with an external provider that is registered fine |
| 23:55.17 | DEMNVT | so they have outbound calls |
| 23:55.34 | DEMNVT | It's only the IAX link between satellite and head office that's not working |
| 23:55.55 | DEMNVT | Rx-Frame Retry[Yes] -- OSeqno: 003 ISeqno: 000 Type: IAX Subclass: LAGRQ |
| 23:55.55 | DEMNVT | <PROTECTED> |
| 23:55.55 | DEMNVT | Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 005 Type: IAX Subclass: ACK |
| 23:55.55 | DEMNVT | <PROTECTED> |
| 23:56.40 | DEMNVT | sorry to flood... but that''s what I'm seeing when I do an iax2 debug |
| 23:58.51 | DEMNVT | The satellite PABX doesn't seem to be getting any IAX frames back from head office... I'm assuming because head office is sending packets back to 192.168.101.10 |