| 00:01.07 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 00:02.07 | Nemus | I cannot get asterisk AMI in 1.8 to run the Action: Originate command http://pastebin.com/3yA0MTVz |
| 00:16.07 | epaphus | guys,.,. what is the best way to play the MOH .. simply invoking Background() will do? |
| 00:17.13 | carrar | it the hold button seems to work well |
| 00:18.29 | carrar | assuming you've setup moh and assigned a musicclass |
| 00:18.42 | carrar | or other options in sip.conf |
| 00:19.14 | epaphus | iam trying to use MOH after a Wait() |
| 00:19.37 | carrar | so this has nothing to with music on hold |
| 00:19.41 | carrar | You just want to play some audio |
| 00:19.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:19.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:20.07 | carrar | background or playback |
| 00:20.11 | carrar | check em out |
| 00:20.22 | epaphus | carrar, ohh. ok. then ill play the same .gsm from the MOH directory i guess |
| 00:21.42 | carrar | You can also force MusicOnHold |
| 00:21.51 | carrar | which may be what you want |
| 00:22.04 | carrar | but thats deprecated |
| 00:22.07 | epaphus | MusicOnHold()? |
| 00:22.12 | epaphus | ohh then.. :) |
| 00:23.10 | carrar | try it |
| 00:23.42 | carrar | core show application MusicOnHold |
| 00:23.47 | carrar | it's in 10 |
| 00:24.07 | carrar | once you set musicclass |
| 00:24.41 | carrar | maybe it's not deprecated |
| 00:28.05 | epaphus | oki |
| 00:29.33 | epaphus | it does work |
| 00:29.34 | epaphus | tnx |
| 00:33.24 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 00:38.39 | epaphus | the music goes ofrever.. it should be just 35 seconds.. hmm any idea iam doing something wrong i have MusicOnHold(35) |
| 00:43.08 | epaphus | ah, ok i just needed a "," in there |
| 00:43.23 | carrar | You didn't read "core show application MusicOnHold" |
| 00:44.27 | carrar | making your callers suffer through a intro? :) |
| 00:46.29 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 00:46.32 | epaphus | :) |
| 00:47.38 | epaphus | Guys.. so lets say that my SIP peers drop internet connection, or there PC softphone gets hard rebooted. .. why does there registration take sooo long to clear up in SIP SHOW REGISTRY ... anyway to avoid that |
| 00:48.45 | carrar | turn on qualify |
| 00:48.56 | carrar | or set it to a value |
| 00:49.23 | carrar | or ahve then quit the client first |
| 00:49.31 | epaphus | i had qualify set.. but no value |
| 00:49.35 | carrar | so it unregisters (zero expire time) |
| 00:50.47 | carrar | visit your example sip.conf |
| 00:50.55 | epaphus | thanks ill check that out carrar |
| 00:51.02 | epaphus | for quakify timeout |
| 00:52.18 | carrar | search for expire |
| 00:52.24 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 00:54.08 | glaz | I had an asterisk instance installed on FreeBSD, everything was working fine. I reinstalled my system with Ubuntu 12.04TLS i386, reinstalled asterisk, very basic config. I seem to have RTP issues now, I have no firewall installed, its not nat'd or anything... Any suggestion where to look at ? |
| 00:54.18 | glaz | tcpdump gives me traffic on RTP ports |
| 00:54.20 | glaz | from and to |
| 00:54.23 | glaz | but no audio |
| 00:55.06 | carrar | What command are you using? |
| 00:55.19 | glaz | command for what? |
| 00:56.00 | carrar | to capture |
| 00:56.14 | glaz | tcpdump -i eth0 |
| 00:56.33 | carrar | perhaps yoru audio is not going through asterisk? |
| 00:56.35 | glaz | eth0 is a dedicated interface for asterisk |
| 00:56.53 | glaz | what do you mean? |
| 00:56.59 | carrar | You can look in your SDP info |
| 00:57.04 | carrar | to find out where it's going |
| 00:57.10 | glaz | sdp show ? |
| 00:57.13 | carrar | no |
| 00:57.20 | carrar | from your tcpdump capture of the sip traffic |
| 00:57.28 | epaphus | carrar, you mean this? defaultexpiry= Number : Default duration (in seconds) of incoming/outgoing registration. Default 120 seconds. So in other words.. if i set this to 30 .. i force SIP devices to reregister at that period? |
| 00:57.46 | glaz | let me check |
| 00:57.55 | epaphus | if yes, i wonder then why anybody would use qualify and a timeout on that |
| 00:58.01 | glaz | but I think I would have realized if it was going to another IP |
| 00:58.48 | carrar | You would think |
| 00:59.38 | carrar | You're looking for a line in your INVITES that looks like "c=IN IP4" or in your 183 reply |
| 00:59.53 | carrar | or=and |
| 01:00.02 | glaz | sip set debug should show me that info, right? |
| 01:00.07 | carrar | should |
| 01:00.16 | carrar | I like to use ngrep |
| 01:01.10 | carrar | "ngrep -tqd any -W byline -l port 5060" is always a good line to start out with |
| 01:01.20 | glaz | ok ,it's showing me the right address |
| 01:01.38 | carrar | There are two sides you are looking for |
| 01:01.58 | glaz | yeah it shows the right IP on both sides |
| 01:02.07 | carrar | IP being your asterisk box? |
| 01:02.11 | glaz | yes |
| 01:02.20 | carrar | for both "c=IN IP4" |
| 01:02.20 | glaz | and the other IP is the RTP server from the Metaswitch |
| 01:03.47 | glaz | tcpdump | ngrep -tqd any -W byline -l port 5060 ? |
| 01:03.54 | carrar | no |
| 01:03.57 | carrar | remove the tcpdump |
| 01:04.01 | epaphus | carrar, ? |
| 01:04.10 | carrar | ngrep is a capture program like tcpdump |
| 01:04.27 | carrar | works great for looking at SIP Headers |
| 01:04.51 | glaz | ok |
| 01:04.52 | carrar | epaphus, I would use qualify and then check the host to befor sending a call |
| 01:05.15 | carrar | set it something low |
| 01:05.31 | glaz | c=IN IP4 10.30.2.2 |
| 01:05.37 | glaz | which is fine |
| 01:05.45 | epaphus | when you say "check the host to" you are also referring to the qualify timeout right? |
| 01:05.51 | carrar | thats your * box? |
| 01:05.57 | glaz | yes carrar |
| 01:06.04 | carrar | epaphus, no |
| 01:06.15 | carrar | check to make sure the SIP device state |
| 01:06.19 | carrar | is there |
| 01:06.25 | carrar | before calling 2 parts |
| 01:06.28 | epaphus | carrar, how...? |
| 01:06.31 | carrar | qualify is the other part of that |
| 01:06.54 | carrar | look at ChanIsAvail |
| 01:07.01 | carrar | and it's coniditions |
| 01:07.05 | carrar | gotta go |
| 01:07.06 | glaz | I also see, c=IN IP4 66.11.***** |
| 01:07.16 | epaphus | tnx |
| 01:07.19 | glaz | which is the RTP server of the metaswitch |
| 01:09.27 | glaz | hummm, maybe the asterisk is telling my phone and the meta to re-invite |
| 01:09.39 | glaz | what do you think carrar ? |
| 01:10.34 | glaz | Also, I was using asterisk 1.4.X before, now 1.8.X |
| 01:14.43 | glaz | I'm lost. lol |
| 01:15.00 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 01:32.35 | mute | you've two interfaces? |
| 01:33.39 | mute | bsd probably uses the correct interface |
| 01:34.09 | mute | linux has a weak host model, whereas it may send packets out a different interface than they came in |
| 01:39.37 | mute | glaz: do you think that could be the issue? |
| 02:00.28 | epaphus | Is there any way to force my SIP peers to reregister every X seconds? |
| 02:10.51 | glaz | mute: YES! |
| 02:10.55 | glaz | that can be it |
| 02:11.18 | glaz | I have a route to the SIP server thought |
| 02:17.43 | ChannelZ | epaphus: That's usually up to the device |
| 02:18.57 | slav3_kitten | so if i'm not mistaken. say i want to create a server for people to sip into for like a party line. i need meetme right? |
| 02:19.09 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 02:19.18 | ChannelZ | or ConfBridge. Depends on your version and preference |
| 02:19.30 | ChannelZ | MeetMe is old, ConfBridge is new. |
| 02:21.58 | slav3_kitten | ah |
| 02:22.06 | slav3_kitten | *googles up confbridge* |
| 02:22.48 | slav3_kitten | i should upgrade from 1.8 to 1.whatever it is now |
| 02:24.33 | bchia | note that ConfBridge in 1.8 is not the same as the ConfBridge in 10+ |
| 02:25.07 | slav3_kitten | yea i noted, hence the upgrade bit |
| 02:25.21 | bchia | oh, read that wrong |
| 02:25.38 | bchia | current version is 11 |
| 02:25.44 | bchia | no more 1.anything |
| 02:25.58 | slav3_kitten | sighs |
| 02:27.14 | slav3_kitten | how old is 1.8 then |
| 02:27.31 | bchia | it's not that old, I think 2 years |
| 02:27.42 | bchia | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 02:28.31 | bchia | That was the problem with the 1.something name - it sounded like early software, but it was actually mature robust code, so the version after 1.8 was gonna be 1.10, but that was kinda silly so they dropped the 1. and just made 10 |
| 02:29.59 | slav3_kitten | so are there any big changes between 1.8 an 11 config wise that will screw me while i work through this book? |
| 02:30.34 | [TK]D-Fender | Go read the CHANGES file to see what changes. |
| 02:30.42 | [TK]D-Fender | And yes, clearly there are major additions |
| 02:30.44 | bchia | there's always updates to the configs, I can't think of anything that will really screw you hard. If you are learning Asterisk I'd go for 11 absolutely |
| 02:30.52 | slav3_kitten | because i run into things like that when configuring my legacy cisco stuff |
| 02:31.13 | slav3_kitten | [TK]D-Fender, i am reading the changes file as we speak. just wondering a yes/no on huge changes |
| 02:31.13 | bchia | yeah, and check out CHANGES or this https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 |
| 02:47.14 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 02:48.44 | [TK]D-Fender | slav3_kitten: We don't know what parts of * you'r using or how so how should we know what of your might break? Should we mention EVERY change jsut in clase? No.... read the file for yourself.... |
| 02:48.54 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 02:49.16 | slav3_kitten | [TK]D-Fender, the question was: will it break the things that the asterisk book which uses 1.8 teaches you. |
| 02:49.26 | [TK]D-Fender | MAYBE. |
| 02:49.44 | [TK]D-Fender | The book describes MANY modules. Many things changed. In ways that break? Maybe. |
| 02:50.09 | slav3_kitten | my dialplan is so simple nothing aside from a massive syntax change would break it as it stands. but i'm trying to work through the book, maybe is better then "absolutely" |
| 02:50.18 | [TK]D-Fender | Playing a guessing game is a waste of time. Just read it. |
| 02:50.28 | slav3_kitten | i am reading it |
| 02:50.47 | slav3_kitten | i've been reading it for a good bit of time |
| 02:51.00 | [TK]D-Fender | Odds are it'll only be a few little things. Serious problem depend of typically more spcial things through API's like AMI, CLI, etc |
| 02:51.45 | [TK]D-Fender | heads out for the evening. |
| 02:52.01 | slav3_kitten | have a great evening and thanks for the advice [TK]D-Fender |
| 02:52.11 | slav3_kitten | it's nice that there are so many helpful people in here |
| 02:53.10 | [TK]D-Fender | slav3_kitten: Sorry if I seem harsh, but this is one process you don't want to shortcut your way through. |
| 02:53.37 | [TK]D-Fender | slav3_kitten: These mistakes cost you serious time... |
| 02:53.45 | slav3_kitten | [TK]D-Fender, i'm learning it from compiling it up, i'm trying my absolute best not to shortcut anything |
| 02:54.21 | slav3_kitten | but since i'm learning if it's well known that v11 breaks many things in the book, i'd just learn the concepts an execution on 1.8 and then port things to 11 |
| 02:55.32 | slav3_kitten | if that makes any sense |
| 03:08.21 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
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| 03:41.46 | epaphus | HEY GUYS, should this match any 8 digit number after the 9? exten => _9XXXXXXXX,1,Dial(SIP/${EXTEN:1}@siptrunk) |
| 03:42.16 | epaphus | for some reason even though I send a 9 and an 8 digit number.. i still have to wait a lot for it to execute the Dial |
| 03:44.23 | *** join/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 03:45.06 | prologic | Hey all. I'm using the latest PBIAF (Asterisk 1.8). I'm having trouble receiving inbound calls on my DID. Here's a sip trace: http://codepad.org/3a5RjOvS ... Not sure why my pbx is sending back a 404. I've defined an Inbound route that matches |
| 03:46.35 | WIMPy | epaphus: Dial from where? What device is collecting you dialing digits? If it's Asterisk, then you have another longer extension that cuold match. |
| 03:48.28 | WIMPy | prologic: You should probably ask in #freepbx. We don't support configuration via 3rd party software here. |
| 03:49.04 | epaphus | WIMPy, i think in this case its my linksys PAP2... so i guess both my linksys PAP2 and asterisk should have the same dial plans? |
| 03:50.09 | WIMPy | epaphus: If you don;t want to wait, yes. Or you tell it to send everything immediately ("overlp") and let Asterisk do the work, if possible. |
| 03:52.05 | epaphus | i didnt know one could set a PAP2 to send everything immediately.. is that they keyword i should search google.. "overlap" ? |
| 03:52.45 | WIMPy | I have no idea if that device supports it.But yes, that option is called overlap dialing. |
| 03:53.07 | WIMPy | That's also often the only way to be able to dial #. |
| 04:06.38 | prologic | WIMPy, ok sorry :) |
| 04:06.59 | *** part/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 04:08.26 | epaphus | bah, it doesnt support it .. |
| 04:18.45 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 04:18.51 | slav3_kitten | god today sucked, i somehow lost my 16 gig sd card for my raspberry pi |
| 04:19.25 | epaphus | Anybody why when i do diaplan reload .... i see outputs such as: Registered extension context 'ael-default'; registrar: pbx_ael .. that kind of stuff i dont have in extensions.conf .. where does it come from? |
| 04:19.41 | epaphus | or would that always be there as default |
| 04:19.47 | WIMPy | extensions.ael |
| 04:20.09 | WIMPy | And that file is optional. |
| 04:21.36 | epaphus | tnx |
| 04:30.05 | epaphus | Is there any function out there that can play a DTMF tone? Like 123 .. its just an indication for a party. I saw PlayTones() but i dont think that its what i need |
| 04:31.08 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 04:31.21 | WIMPy | SendDTMF() |
| 04:31.41 | epaphus | tnx |
| 05:07.07 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
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| 05:58.38 | SeRi | waz up guys |
| 06:00.01 | SeRi | dijib: you in? |
| 06:02.01 | WIMPy | Hi SeRi |
| 06:03.01 | SeRi | yo WIMPy how are you |
| 06:04.25 | WIMPy | Not shut if I should be more awake or rather asleep. |
| 06:04.37 | SeRi | lol |
| 06:04.54 | WIMPy | s/shut/shure/ |
| 06:05.22 | SeRi | I am still recovering from this afternoon. |
| 06:05.26 | SeRi | long day with the fam |
| 06:06.00 | SeRi | my case/brik finally made it's way in this am to Houston. |
| 06:06.20 | *** join/#asterisk evilman_home (~evilman_h@94.28.229.193) |
| 06:07.22 | SeRi | I still have not recived an update from voip.ms on my port |
| 06:07.27 | SeRi | this is not good... :/ |
| 06:08.06 | WIMPy | Phone numbers should die. |
| 06:13.29 | SeRi | lol |
| 06:13.36 | SeRi | no a bad idea. |
| 06:14.59 | WIMPy | What do you find easier to remember? Someones phone number or their e-mail address? |
| 06:16.06 | mute | their name is the easiest :) |
| 06:21.10 | SeRi | email for sure |
| 06:22.11 | WIMPy | Same for me. But luckily, since I use Asterisk, you can call my e-mail address as well. |
| 06:22.35 | WIMPy | Only one easy to remember address for both. |
| 06:23.22 | SeRi | nice! |
| 06:23.50 | WIMPy | You don't have it? |
| 06:24.21 | SeRi | nope |
| 06:24.38 | WIMPy | Change that :-) |
| 06:26.02 | SeRi | Not a bad idea at all. I should do that. |
| 06:37.39 | SeRi | WIMPy: do you have your own mail server? |
| 06:37.55 | WIMPy | yes |
| 06:38.45 | SeRi | ah. ok that makes sense... |
| 06:39.05 | SeRi | my emails get redirected via a hosting service |
| 06:40.35 | WIMPy | Has anyone here ever tried a Samsung Anyweb phone and would share some opinions? google prefers to tell me about printers. |
| 06:42.09 | SeRi | WIMPy: I was given as a gift what apears to be an IBM pbx card. It has and RJ45 port and a FXO port. ANy ideas what are this cards? They dont have a model number |
| 06:43.11 | WIMPy | I didn't even know that IBM did anyhing in that area. What kind of card? |
| 06:43.28 | WIMPy | Not for PCs, I assume? |
| 06:44.32 | SeRi | It was on a 2U server and it was labeld "PBX" It had two of this cards |
| 06:44.40 | SeRi | They are pci |
| 06:45.06 | WIMPy | Then you could try to search for the PCI ID. |
| 06:45.15 | SeRi | cool thanks |
| 06:46.21 | WIMPy | Actually, googling for "ibm pbx pci" gives some results. |
| 06:48.23 | WIMPy | But noting interesting, it seems. |
| 06:48.45 | SeRi | Is similar to this |
| 06:48.48 | SeRi | http://www.tmcnet.com/channels/pbx-pci-adapter/ |
| 06:50.38 | WIMPy | Oh, an appliance in PCI format? |
| 06:50.52 | WIMPy | I never understood that concept. |
| 06:53.51 | SeRi | WIMPy: Yes I think it is. |
| 06:53.54 | SeRi | not sure though |
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| 10:15.02 | loggiew | hey guys |
| 10:15.10 | loggiew | first time setting up asterisk |
| 10:15.21 | loggiew | got to a point where i can make a call between two devices |
| 10:15.37 | loggiew | no audio and the CLI shows no log data upon initiating the call |
| 10:15.48 | loggiew | both devices connect, but no sound and the CLI doesn't report it |
| 10:15.59 | loggiew | any thoughts on where I might start reading? |
| 10:17.14 | x2s | did you set core debug higher than 0 and core verbose, too? |
| 10:17.33 | tonylyliu | you need to check firewall if audio streams are blocked assume you are using sip/iax2 phones |
| 10:17.52 | loggiew | no firewall, all internal network. Just toying to learn |
| 10:18.01 | loggiew | x2s, looking now |
| 10:18.04 | *** join/#asterisk dgeary2 (~david@49.176.100.68) |
| 10:18.10 | tonylyliu | the future of telephony |
| 10:18.18 | tonylyliu | you can google it |
| 10:18.26 | tonylyliu | the online book |
| 10:21.24 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 10:22.32 | tonylyliu | any guys has tested confbridge function on asterisks 11.0.1 with large volums calls? |
| 10:22.50 | tonylyliu | or put it into production use? |
| 10:27.12 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 10:29.08 | loggiew | ok, so now im seeing the devices, that resolved my CLI question. But Im still not certain what's wrong with the sound |
| 10:34.24 | ChannelZ | What are the devices? |
| 10:34.52 | loggiew | a soft phone and my cell |
| 10:35.11 | ChannelZ | the cell running a SIP client on the LAN I take it? |
| 10:35.22 | loggiew | when i type core show channels, the phone lists 'None' under location, that seems odd |
| 10:35.27 | loggiew | yes |
| 10:36.00 | loggiew | everything is internal with no firewalls turned on (intentionally.. lord knows what ive forgotten) |
| 10:36.32 | ChannelZ | well there are many places it can go wrong |
| 10:37.11 | loggiew | ya :/ |
| 10:37.12 | ChannelZ | start with 'core set verbose 3' and then pastebin a test call |
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| 10:54.46 | loggiew | thanks for the help guys, maybe I just need to stop banging my head on the wall for the night and work on it tomorrow |
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| 11:26.44 | and7ey | Can somebody please help me - I am trying to forward all incoming calls to another number, here is my config - http://pastebin.com/ZudNAKbX, but I see the following in incoming call - handle_request_invite: Call from '' (10.25.0.50:5060) to extension '7499952XXXX' rejected because extension not found in context 'default'. What am I missing in my setup? |
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| 11:30.14 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
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| 11:42.42 | ChannelZ | and7ey: as it says, that extension is not found in the context "default". I see no context named that. |
| 11:43.48 | ChannelZ | and I hope your [beeline] was actually in sip.conf and not extensions.conf as your paste notes |
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| 11:49.19 | and7ey | ChannelZ: should I create section [default] in extensions.conf? What should be there? Sorry for such questions, I am new to Asterisk |
| 11:50.19 | ChannelZ | the fact that it's going to default implies that the incoming call is not being matched to any peer in sip.conf |
| 11:50.58 | ChannelZ | So you should figure out why that is probably. The error you pasted said the call was coming from 10.25.0.50 which is a LAN address |
| 11:57.29 | [sr] | guys |
| 11:57.40 | [sr] | what headers are needed for ogg compilation? |
| 11:58.12 | ChannelZ | libvorbis probably |
| 11:58.14 | [sr] | i already have libogg-dev but asterisk doesn't let me enable it yet |
| 11:58.28 | ChannelZ | ogg is the container, vorbis is the audio codec |
| 11:58.44 | [sr] | ahhhhhh |
| 11:59.07 | [sr] | makes sense, dumb me |
| 11:59.13 | [sr] | thanks channel X |
| 11:59.13 | [sr] | ;) |
| 11:59.34 | ChannelZ | Sure |
| 12:00.56 | [sr] | and for misdn? |
| 12:01.29 | [sr] | capi headers right? |
| 12:01.39 | [sr] | libcapi i mean |
| 12:02.54 | ChannelZ | dunno on that one besides what menuconfig says |
| 12:03.07 | ChannelZ | isdnnet, misdn, suppserv.. whatever those are |
| 12:07.36 | [sr] | it could be handy for me |
| 12:07.52 | *** join/#asterisk bytemaster (~qweq@host81-150-217-167.in-addr.btopenworld.com) |
| 12:07.58 | [sr] | i dont see any "supp" headers |
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| 12:18.10 | WIMPy | capi is for chan_capi. |
| 12:18.25 | [sr] | hi WIMPy |
| 12:18.39 | WIMPy | And capi doesn't go with misdn1, but it can be used with misdn2. |
| 12:18.54 | weinerk | Hi. Please help. How can I make the agi-script wait for pickup if I am doing this: |
| 12:18.54 | weinerk | asterisk -rx "originate local/18885551212 extension 1234@mycontext" |
| 12:19.25 | WIMPy | And there are two different chan_capi (off course). Bot seem a little dead. |
| 12:21.17 | [sr] | WIMPy: idea is to make use of two HFC cards, the one's i have, just for internal use |
| 12:21.49 | WIMPy | Well, you can go all the ways with them. |
| 12:22.20 | WIMPy | It's just up to you to decide wich drivers to use. |
| 12:22.54 | [sr] | WIMPy: not now but in a few months ago, i testes loading misdn and dahdi drivers, and they didn't liked to live toghether |
| 12:23.44 | WIMPy | No, you need to load only one of the drivers, off course. |
| 12:24.18 | [sr] | but i have cards that needs dahdi |
| 12:24.30 | WIMPy | The can live alongside on the same machine, but with two of the same cards, you have to use the same driver for both. |
| 12:24.31 | [sr] | my error is to think i can have then both on the same system i think |
| 12:24.51 | WIMPy | No, that should be fine. |
| 12:28.04 | [sr] | damn i was trying 11.x on my tests, and have a siemens ip phone doesn't work with it dont know why |
| 12:28.32 | WIMPy | With SIP firmware? |
| 12:28.42 | [sr] | yes, a c610IP |
| 12:28.56 | [sr] | i can see in sip show peers (phone is here at home and server at work) |
| 12:29.06 | WIMPy | I haven't seen a Siemens IP phone for real, yet. |
| 12:29.21 | [sr] | that the IP it sent to the server was it's the internal IP instead of the public |
| 12:29.29 | WIMPy | But I would like to see hardware to connect the regulart Up0 phones to Asterisk. |
| 12:29.32 | [sr] | well, what do you mean by sip firmware then? :) |
| 12:30.08 | WIMPy | Usually they talk HFA (AKA CorNetIP) instead of SIP. |
| 12:30.17 | [sr] | ah no |
| 12:30.44 | [sr] | sip show peers show's me now: 100/100 192.168.2.23 D A 5060 UNREACHABLE |
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| 12:31.05 | [sr] | it should be the public ip there, instead of 192.168.2.23 |
| 12:31.15 | WIMPy | You know the nat setting in sip.conf, don't you? |
| 12:31.27 | [sr] | with 10.x works ok, and no conf was changed |
| 12:32.52 | [sr] | WIMPy: yes, but i have a droid phone with 3cx, and it works ok, anyway extensions are with nat = never |
| 12:33.08 | WIMPy | has only one user behind NAT, but that looks ok. |
| 12:34.09 | [sr] | ah |
| 12:34.12 | [sr] | now works! |
| 12:34.20 | WIMPy | If you don't use Asterisk's nat support, yuo need it on the phone, like enabling a stun server. |
| 12:34.48 | [sr] | for somehow i did this on the past and didn't worked, maybe some router problem |
| 12:35.13 | WIMPy | Absolutely possible. |
| 12:35.29 | [sr] | for sure |
| 12:35.38 | [sr] | or ISP problems |
| 12:35.44 | [sr] | never mind |
| 12:35.48 | [sr] | works! |
| 12:36.17 | [sr] | funny is the droid 3cx client works with nat = never |
| 12:39.52 | [sr] | well going to lunch... WIMPy thanks for your time |
| 12:40.08 | [sr] | WIMPy: ah, tomorow we are going to have a visit from your "boss" :p |
| 12:40.23 | [sr] | your(a) |
| 12:40.46 | WIMPy | doesn't really have a boss. |
| 12:41.18 | [sr] | WIMPy: hehe.. just playing with you |
| 12:41.39 | WIMPy | Bad luck then. |
| 12:42.08 | WIMPy | I guess my real boss would be muself. And I wouldn't mind a little holliday :-) |
| 12:42.22 | [sr] | WIMPy: how i understand you... |
| 12:42.52 | [sr] | sometimes its better to be the employ than the employer... |
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| 12:44.10 | WIMPy | Not sure. |
| 12:45.47 | [sr] | of course i preffer to be my own boss also |
| 12:46.27 | WIMPy | This evening I will have an external :-) boss. |
| 12:49.31 | [sr] | :) |
| 12:49.37 | [sr] | going to lunch |
| 12:49.40 | [sr] | brb |
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| 13:46.41 | [sr] | back |
| 13:46.47 | [sr] | with 11.x |
| 13:47.29 | [sr] | the notify information keeps appearing on the console, is there any parameter to supress this information? like the one for the manager to supress the login/logout info |
| 13:49.38 | wonderworld | i wanto to set the IP of a SIP peer as channel variable. how could i do that? |
| 13:49.43 | [sr] | i mean, not just when it changes |
| 13:50.40 | WIMPy | wonderworld: What do you want to do? |
| 13:51.18 | wonderworld | i want the ip a peer with a specific sip channel is using as channel var |
| 13:51.31 | wonderworld | like a call starts, peer gets sip channel. i want the ip of the peer |
| 13:52.21 | WIMPy | Like CHANNEL(peerip)? |
| 13:52.39 | wonderworld | ohh, does that exist? yeah. exactly that |
| 13:53.08 | wonderworld | thanks |
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| 15:46.07 | teknoprep | hi all |
| 15:46.31 | teknoprep | i have a simple setup for asterisk ? ITSP < -> Asterisk < - > Another Sip Server |
| 15:46.59 | teknoprep | what i have currently is a single line that anything comming into the box gets forwarded to an extension that is registered |
| 15:47.16 | teknoprep | my questino is how to i tell it to go to a secondary extension if the first extension is not registered |
| 15:47.28 | teknoprep | [inbound-ezuce] |
| 15:47.29 | teknoprep | exten => _X.,1,Dial(SIP/${EXTEN}@ast1-eZuce) |
| 15:47.37 | teknoprep | sorry i was going to paste that into patebin |
| 15:48.12 | tonikasch | imho it's better if you paste that into pastebin or similar |
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| 16:07.24 | [TK]D-Fender | teknoprep: that doesn't look like you're dialing a "registered extension". It's calling some other server. If they don't answer then just dial the next thing right after. |
| 16:09.15 | mute | dial at the same time |
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| 16:28.28 | pzn | I need help about asterisk/grandstream gxw-4108 fxo adapter. PSTN -> Voip calls OK. Voip->PSTN calls ok. problem: all voip->PSTN calls should go out from an FXO port choosen by grandstream device... need to go out using specific FXO port. how? thanks in advance |
| 16:29.38 | pzn | sorry for my bad english. currently grandstream is choosing the outgoing FXO port. I'd like to be able to choose the port from asterisk. |
| 16:35.48 | tonikasch | bye |
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| 16:58.32 | enzo | <PROTECTED> |
| 16:58.38 | enzo | (hello) |
| 17:00.23 | enzo | I think, it's hwec, you confirm ? |
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| 17:11.50 | EmleyMoor | enzo: As far as I know it just works - as long as, of course, it's detected |
| 17:12.02 | EmleyMoor | I have an AEX410P with one |
| 17:14.06 | enzo | you know how I can check that EmleyMoor ? |
| 17:15.08 | enzo | the fact is I've lanched dahdi_genconf, it has generated /etc/asterisk/dahdi-channels.conf, and I can read in it: line="1 WCTDM/0/0 FXSKS (In use) (EC: VPMOCT032 - INACTIVE)" (same thing for all lines) |
| 17:19.32 | EmleyMoor | It's not mentioned in mine |
| 17:21.43 | enzo | and with some asterisk command, it's surely possible to check the echo canceller used, someone knows the command ? |
| 17:23.03 | EmleyMoor | In chan_dahdi.conf I have echocancel=yes and it says that for hardware EC that's all you need |
| 17:24.52 | enzo | ok, I've done that, so it works I guess :) |
| 17:26.20 | EmleyMoor | I don't know how to check for sure... but see if that helps. |
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| 17:42.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:45.46 | enzo | in which place (on last ubuntu) it's advised to put sounds and moh ? /usr/share/asterisk or /var/lib/asterisk directory ? |
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| 17:57.11 | mute | make install put mine in /var/lib/asterisk/sounds/en |
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| 18:08.00 | wonderworld | which codec should be choosen for sip peers to have low translation cost for confbridge use? |
| 18:10.56 | dlynes | wonderworld, if you can afford the bandwidth, ulaw or alaw |
| 18:11.15 | wonderworld | it's mostly about CPU cost |
| 18:11.25 | dlynes | wonderworld, then again, ulaw or alaw |
| 18:11.34 | dlynes | wonderworld, they're the two simplest codes |
| 18:12.00 | wonderworld | i was thinking about what codec confbridge uses internaly |
| 18:12.00 | pzn | wonderworld, g711 ulaw or alaw as dlynes said... the cpu cost is about 0% |
| 18:12.43 | dlynes | wonderworld, confbridge just bridges two calls...it doesn't use a codec |
| 18:13.03 | dlynes | wonderworld, it only performs translation if the two legs don't match for codecs |
| 18:13.30 | wonderworld | ok, i see, so the codec doesn't matter to confbridge, in case all peers use the same? |
| 18:13.38 | dlynes | wonderworld, correct |
| 18:13.46 | dlynes | wonderworld, however |
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| 18:14.11 | dlynes | wonderworld, when two or more people are talking at a time, it will need to do merging of the legs |
| 18:14.48 | wonderworld | ..and the cost for this mergin would be the lowest with alaw/ulaw ? |
| 18:14.54 | dlynes | wonderworld, correct |
| 18:15.00 | dlynes | wonderworld, and the simplest |
| 18:15.06 | wonderworld | cool, thanks |
| 18:15.36 | dlynes | wonderworld, ulaw/alaw do not use any compression; it's the raw almost purely unadulterated audio stream |
| 18:16.17 | wonderworld | thats 64kbit/sec ? |
| 18:16.50 | dlynes | wonderworld, correct |
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| 18:26.24 | mtbf | I am having an issue with missing recording files on asterisk 1.8.13, it happens from time to time and I don't know how to reproduce it, logfile contents look normal to me, at least I don't see anything indicating an error, http://pastie.org/pastes/5361623 but I can't figure it out. |
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| 18:35.40 | mtbf | Now I found the description would fit to 2012-04-27 19:26 +0000 [r364277] Matthew Jordan <mjordan@digium.com> (http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8-current) this issue, however 1.8.13 was released at 2012-06... |
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| 19:23.02 | loggiew | Im having a delay in all audio of about 1 second between two sip devices |
| 19:23.17 | loggiew | suggestions on where to begin reading? |
| 19:25.41 | dlynes | loggiew, are they both on the same LAN? |
| 19:25.47 | loggiew | yes |
| 19:25.59 | loggiew | my entire setup is just a test all within my LAN |
| 19:26.03 | dlynes | loggiew, do they need to go outside the LAN to connect to each other? |
| 19:26.08 | loggiew | no |
| 19:28.11 | *** part/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 19:28.57 | loggiew | i should be fairly solid on processing power etc because I have an i7 with 24 gig of RAM, way over kill for this |
| 19:29.17 | loggiew | the two devices are going across internal wireless but I didnt expect the delay to be quite this bad |
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| 19:45.08 | dlynes | loggiew, have you checked ifconfig to make sure you don't have any framing or other errors? |
| 19:51.01 | *** join/#asterisk vlad_sta20:43.29 Opened logfile log/2012/1111. |
| 20:44.39 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:44.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.10.0 (2012/11/06), 1.8.18.0 (2012/11/06), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:45.43 | ketas | oh fun, x crapped just after enter :P |
| 20:46.34 | dlynes | ketas, /usr/sbin/asterisk -rx 'sip show peers' |
| 20:47.13 | dlynes | ketas, btw...why'd you name yourself after a fish? |
| 20:49.44 | ketas | no, i'm "disk" |
| 20:50.34 | *** join/#asterisk ruied (~AndChat66@po-217-129-155-146.netvisao.pt) |
| 20:50.43 | ketas | i don't have any access to run things like this |
| 20:54.28 | ketas | i only know that there is way to get list of all sessions from client itself |
| 20:54.59 | ketas | still confusing? |
| 20:57.27 | mute | like presense information...? |
| 21:00.42 | ketas | almost, and about client itself |
| 21:01.25 | ketas | "what are my other registered sessions" |
| 21:08.40 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 21:20.28 | n8ideas | anyone here happen to have some experience with ODBC scalability? |
| 21:20.32 | n8ideas | and res_odbc optimization? |
| 21:20.48 | n8ideas | specifically interested in the "share_connections" option and concurrency limit |
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| 21:32.09 | loggiew | stabs windows firewall |
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| 21:48.19 | pabelanger | Anybody tried zoiper on bell's 4g network? Seem to be having to RTP issues |
| 21:48.24 | pabelanger | and not sure if bell is just blocking them |
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| 22:02.41 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 22:35.48 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 22:43.25 | *** join/#asterisk CajaCaliente (~bloat@CPE-72-135-216-250.wi.res.rr.com) |
| 22:44.57 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:01.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:08.06 | *** join/#asterisk jeffspeff (~Jeff@67.231.40.120) |
| 23:08.18 | jeffspeff | how can i tell if my ISP is blocking port 5060? |
| 23:08.32 | jeffspeff | my box was working fine, and now it's not. my IP hasn't changed |
| 23:09.12 | jeffspeff | when i do a port scan on my external ip to see if it's listening for 5060, it says it's not; but that could mean that the ISP is blocking it |
| 23:15.11 | *** join/#asterisk mathi (Matthew@ip-62-235-244-174.dsl.scarlet.be) |
| 23:15.12 | mathi | hi |
| 23:16.23 | mathi | I would like to log the messages in a file, but I don't understand how to specify what level of verbosity I want. Just like I would do -rvvvv for the console for exmample |
| 23:16.31 | p3nguin | jeffspeff: Your scan was most likely a TCP scan, so your result was not going to be an accurate reflection of your actual situation. |
| 23:18.14 | jeffspeff | p3nguin, i used the t1 shoppers scanner, can you recommend one that has udp? |
| 23:19.20 | p3nguin | I would use netcat to create a listening server and netcat on a remote system to send some data to that listening server. |
| 23:19.22 | mathi | anyone? |
| 23:19.56 | p3nguin | A pair of netcats is really good for testing if a port is firewalled or not. |
| 23:22.07 | ChannelZ | mathi: see logger.conf for creating the logfile, and cli.conf for setting the level |
| 23:23.44 | jeffspeff | p3nguin, if i change the value of "port" within the general context and within the peer context of sip.conf it should change the udp listening port of sip shouldn't it? |
| 23:24.20 | p3nguin | To change the listening port of the SIP channel driver, you have to change the bind port. |
| 23:24.58 | jeffspeff | so, there should be a bindport= as well as a port=? |
| 23:25.17 | p3nguin | udpbindaddr=0.0.0.0:5080 for example |
| 23:26.41 | mathi | ChannelZ, doesn't work for me... in logger.conf I have: messages => error,dtmf |
| 23:27.01 | mathi | ChannelZ, in cli.conf I have: core set verbose 30 = yes |
| 23:27.03 | jeffspeff | p3nguin, thanks, got that resolved |
| 23:27.10 | mathi | ChannelZ, I have all the logs in the console |
| 23:27.12 | jeffspeff | going to change it on other side of sip trunk and test |
| 23:27.15 | mathi | but not in the "messages" file |
| 23:27.57 | mathi | (when I restart asterisk i see some logs in messages, but I want the logs of Verbose(1, ...) in my Dialplan |
| 23:29.33 | p3nguin | mathi: You probably want to add the "verbose" level to your log mode. |
| 23:29.52 | p3nguin | messages => error,dtmf,verbose |
| 23:30.07 | mathi | p3nguin, I have "verbose" level in my logfile called "full", but it doesn't has any logs when my dialplan is executed |
| 23:30.15 | mathi | but I cans ee all the logs in the console |
| 23:30.40 | p3nguin | Check the full log file instead of messages log file. |
| 23:30.58 | p3nguin | If you have the full mode enabled, a new file for full will be created and used for those levels of logging. |
| 23:31.05 | p3nguin | Each mode has its own file name. |
| 23:31.07 | mathi | I know... |
| 23:31.15 | mathi | I have the "full" logfile in front of me |
| 23:31.25 | mathi | it has no logs |
| 23:31.27 | mathi | it does sometimes |
| 23:31.36 | p3nguin | If you still don't get the results you want, consider the Log() application. |
| 23:31.46 | mathi | :-( |
| 23:32.00 | mathi | Verbose() is only for console? |
| 23:32.23 | p3nguin | Unless you are logging the verbose level, I believe so. |
| 23:32.58 | mathi | I have verbose level |
| 23:33.20 | jeffspeff | well, i changed asterisk to listen on 6050 instead of 5060, and all works fine now... i guess this conclusively means that Ritter is blocking my 5060 traffic |
| 23:34.34 | mathi | when i restart asterisk I have a bunch of messages in "full" logfile, but when my dialplan is executed, the Verbose() messages aren't logged at all |
| 23:34.38 | p3nguin | I wouldn't say conclusively, but it is certainly an indication of the possibility. |
| 23:37.15 | jeffspeff | p3nguin, well, it was working fine on 5060, then it stopped working (without me changing a thing, i haven't had time to touch this box) then I change the port from 5060 to 6050 and now it magically works again. i am the network admin, and i know for a fact that it wasn't blocked on my work side, so that just leaves my home side, which i'm not in as much controll over the ISP provisions, etc. |
| 23:37.58 | p3nguin | Call 'em up and ask WTF they did. |
| 23:39.21 | jeffspeff | i am, as we speak |
| 23:39.22 | jeffspeff | :) |
| 23:39.44 | p3nguin | "WHAT'D YOU DO?!" |
| 23:48.26 | jeffspeff | p3nguin, funniest thing ever, i finally got through to their tech support... The guy introduced himself as Dennis, I asked how he was doing this evening, to which he responed that he's doing well, I then inquired if they practice port blocking on residential internet account, he then hung up on me. |
| 23:49.59 | p3nguin | That's interesting. Are you calling again? |
| 23:50.08 | jeffspeff | no, i'm giving up on them |
| 23:50.38 | jeffspeff | think i'll find/make a friend that's very near an ATT fiber loop |
| 23:52.04 | jeffspeff | Hi, my name's jeffspeff. Yes, this is a gun, and here are some boxes of equipment i'll be installing in your closet today. Oh, almost forgot, you won't have to pay for internet anymore. :) |
| 23:53.22 | jeffspeff | that was a joke btw |
| 23:53.27 | jeffspeff | nobody was laughing. |
| 23:53.28 | jeffspeff | :S |