| 00:01.11 | klow | Testing an "echo" test call from a remote endpoint behind NAT , and seeing that Asterisk is sending RTP to the private IP of the endpoint. Using pjsip, and I do have rewrite_contact=yes on the endpoint , any other reason this would be happening? |
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| 02:42.38 | lunaphyte | [TK]D-Fender: i was looking for a device which can provide a means for an ip phone to use an analog phone line |
| 02:42.54 | lunaphyte | sort of the opposite of a traditional ata, so to speak |
| 02:44.01 | MaliutaLap | lunaphyte: an ATA would do the job |
| 02:45.43 | lunaphyte | it would? |
| 02:45.56 | lunaphyte | atas have fxs ports, right? |
| 02:53.49 | MaliutaLap | fxo |
| 02:54.08 | MaliutaLap | some even have both |
| 02:54.14 | MaliutaLap | depends on the unit |
| 03:07.57 | lunaphyte | right, yeah. that what i meant by traditional ata. |
| 03:08.17 | lunaphyte | i should have said what people typically think of when they say ata. e.g. fxs ports |
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| 03:11.08 | hiyo | Hello, is there a way to change the default recording extensions for confbridge to something else? |
| 03:11.48 | hiyo | I want to be able to record to ogg, as the wav files get large very quickly. |
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| 03:51.20 | lunaphyte | for a device like that which allows an ip phone to make calls out an analog phone line, could the ip phone possibly just communicate directly with the device, without the need for other components? or should i expect that some sort of server will need to be involved to handle communication between the two in some capacity? |
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| 05:38.03 | [TK]D-Fender | SPA has it's own call handling as to the auth it'll require to place calls if youwa nt to try to set a phone to use it directly |
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| 07:12.38 | Lope | any ideas for a free or cheap SIP providers for making calls to USA phones? I just need it to test my Asterisk system. I've been using "Freetone" the android app. It works okay except doesn't seem to support RFC2833 DTMF and has no DTMF options. |
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| 07:42.05 | Lope | I'm experiencing a strange phenomenon. When I call my asterisk server from a new SIP account, I don't hear any of the greetings, there's no audio coming through at all. but the moment I press a DTMF tone etc, then the sound comes through? |
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| 07:44.16 | Lope | I noticed the microphone icon was not being shown on my PC. I switched ALSA/pulse-audio and now the microphone icon is displayed and audio comes through immediately. |
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| 08:52.52 | Lope | I've got a context for incoming calls in extensions.conf [from-foo] starting with "exten => 1234,1,NoOp()" which is working properly. Then I've got [from-foo] in voicemail.conf with "1234 => 0000,Test,foo@bar.com". When I try run the voicemail command in my dialplan I get an error message: leave_voicemail: No entry in voicemail config file for '1234'. Any ideas? |
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| 08:54.32 | Lope | I've got the core set to verbose 10 debug 10. It's not telling me more than what I mentioned. |
| 08:59.28 | Lope | Oh looks like I need to add @context when I run the voicemail app. |
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| 09:05.06 | Lope | okay it plays my custom mailbox greeting but then it still plays the default mailbox greeting (vm-intro) afterwards and then it plays auth-thankyou after that. I want to silence all of those. |
| 09:17.40 | Lope | I've got directoryintro=/path/to/my/sounds under [from-foo] inside my voicemail conf. But it's still playing the standard vm-intro and so on. |
| 09:18.06 | Lope | I've added my own vm-intro and auth-thankyou wav files in /path/to/my/sounds |
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| 09:22.48 | Lope | oh maybe it's supposed to be a file... |
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| 09:29.14 | Lope | I also tried directoryintro=/path/to/silence (where silence is a wav file) |
| 09:29.39 | Lope | The console did not mention any warnings or anything at all about directoryintro. It seemed to completely ignore it. |
| 09:31.27 | Lope | I've checked that there are no permissions issues. asterisk can play those sounds for other things. |
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| 11:17.25 | angryuser | Good day people, quick question i've got some 183 with SDP from carrier, asterisk converts it to 180 when sending to the customer and stripts the SDP, how can i get as* to just relay the 183 ? |
| 11:33.21 | julius | hi |
| 11:34.56 | WIMPy | lo |
| 11:37.31 | julius | any isdn persons around? i was looking into building my own Y-cable to connect a fritzbox to DSL and ISDN. theres a picture here: http://service.avm.de/support/media/de/33.png |
| 11:37.36 | julius | let me draw a picture, its easier |
| 11:38.20 | WIMPy | What's wrong with that picture? |
| 11:39.47 | julius | nothing, just that they turn the rj45 jacks "head to head" but do not turn the numbering |
| 11:52.29 | julius | assuming that in the picture the left rj45 goes to the DSL splitter, the right one into the fritzbox (the one going to the ISDN box isdn shown) i would connect 4 to 4 and 5 to 5 |
| 11:56.36 | WIMPy | That's DSL. PSTN is the upper one. |
| 12:06.25 | angryuser | i've found how to solve that, however if my provider send me 183 and than 404 asterisk takes a LONG time to forward the message, and forwards the 503 instead of 404 |
| 12:07.24 | angryuser | What a pain, is there a reason why as* do not forward the 404 from the provider immediately ? I have a ery simple dialplan, so its not dialplan related |
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| 12:30.00 | julius | ah crap, just found out that the 5140 does not support ip phones, but the manual of the 7170 says you can add "ip phones" - thats what i want to redirect the call to asterisk, right? |
| 12:58.53 | WIMPy | yes |
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| 13:18.48 | julius | oh, i didnt post the picture to my second question: https://www.wti.com/userimages/straight-cable-300.png |
| 13:19.53 | julius | one goes to the fritzbox, the other to the dsl splitter (isdn not shown). do i connect wire 4 to 4 and 5 to 5 OR 4->5 , 5->4 ? |
| 13:20.13 | WIMPy | Doesn't matter. |
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| 14:08.20 | alsaiduq | hello guys |
| 14:08.28 | alsaiduq | how do i download asterisk header files? |
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| 14:20.09 | [TK]D-Fender | You compile them |
| 14:20.26 | [TK]D-Fender | for the source tarball that's right there on the website |
| 14:21.08 | [TK]D-Fender | Under the well-hidden "Downloads" page |
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| 14:25.13 | mutilator | hey all, if i wanted to do a 'voicemail reload' or 'dialplan reload' from a windows server via something scripted (powershell) what would be my best route? |
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| 14:25.49 | [TK]D-Fender | Something that talks tto your * server |
| 14:26.03 | [TK]D-Fender | SSH, AMI, etc |
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| 14:30.24 | mutilator | hmmm ami is like a telnet directly to asterisk console |
| 14:30.27 | mutilator | ? |
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| 14:31.21 | [TK]D-Fender | ? book |
| 14:31.27 | [TK]D-Fender | ~book |
| 14:31.27 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:31.28 | [TK]D-Fender | ^^^ |
| 14:31.51 | [TK]D-Fender | just one of the things you can do is issue CLI commands |
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| 16:05.44 | angryuser | Hello, my operator sends me an 183 with SDP which is OK, then sometimes i got 404 not found, however asterisk forwards me 503 instead of that 404, does anyone know how to relay 404 instead ? |
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| 16:06.34 | [TK]D-Fender | That' just how that stack works. |
| 16:06.45 | [TK]D-Fender | dThis should call for a code change |
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| 16:07.14 | [TK]D-Fender | As you shouldn't gt progress for something that tries to say "not found" afterwards. |
| 16:07.21 | [TK]D-Fender | Doesn't sound like "up to spec" |
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| 16:27.31 | znf | Hello. I'm having some issues with elastix/asterisk, and I can't seem to figure out what's going on. I've configured the system, I have extensions, I can make internal calls, I can make outbound calls, but I can't seem to be able to receive calls from external sources, all I get is a busy signal. I've turned on sip debugging, but I'm at a loss with the outputs, other than a "SIP/2.0 401 Unauthorized". Could anyone give me some pointers on where to look |
| 16:27.31 | znf | next? I'm using SIP trunks |
| 16:29.58 | angryuser | [TK]D-Fender: well i theory its in the spec, as 183 tells that you are in progresss not ringing |
| 16:30.41 | angryuser | its not like i get 180 ringing and that not found |
| 16:30.47 | angryuser | then* |
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| 16:32.31 | angryuser | what i want to say is that after 183 you are not yet got the called device for sure |
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| 16:33.10 | angryuser | anywyas its a shame ;( |
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| 16:41.29 | [TK]D-Fender | znf: does SIP debug report back "found peer X for ..." for that call? |
| 16:42.16 | znf | [TK]D-Fender: yes |
| 16:42.22 | znf | Damn it, left the office |
| 16:47.27 | [TK]D-Fender | then your system is challenging the inbound call and it is failing to auth |
| 16:47.36 | [TK]D-Fender | what is "external sources" exactly? |
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| 17:13.57 | julius | WIMPy, doesnt matter? |
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| 18:05.28 | XATRIX | Hi guys, i'm back again with the same problem. I have WAN on my router, where i've installed asterisk 11. And i set INCOMING policy to allow of my firewall. The remote SIP client can connect to. But no sound is coming from and to |
| 18:07.35 | *** join/#asterisk theron (~theron@2620:10d:c091:200::5:4066) |
| 18:12.26 | [TK]D-Fender | And we should be looking at the actual call... |
| 18:15.19 | XATRIX | yea, give me a sec. i'm collecting info |
| 18:17.41 | XATRIX | [TK]D-Fender, http://www.fpaste.org/333155/02904914/ this is what i have on |
| 18:18.09 | XATRIX | maybe it's a situation when the client is behind NAT ? |
| 18:19.45 | XATRIX | nat=force_rport,comedia ? |
| 18:19.54 | [TK]D-Fender | What about your server? |
| 18:22.39 | XATRIX | what do you mean ? |
| 18:22.49 | XATRIX | It has a real IP. Not nated |
| 18:22.58 | XATRIX | All incoming connection to are allowed |
| 18:23.13 | hypknight|Away | is now away - Reason : Auto-Away (Away from Keyboard for 60 minutes) |
| 18:24.23 | [TK]D-Fender | Show us |
| 18:26.09 | XATRIX | sorry, looks like we solve it |
| 18:26.13 | XATRIX | Still testing |
| 18:26.17 | XATRIX | give a sec. |
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| 19:04.53 | voip | woot |
| 19:26.19 | drmessano | lol |
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| 19:27.15 | jpastore | hi, is it possible to silence the sip messages scrolling by in the CLI so I can see the output of the commands I'm trying to run? |
| 19:27.59 | [TK]D-Fender | disable your SIP debug |
| 19:28.46 | znf | [TK]D-Fender, sorry, I'm back now, had to leave the office and drive to a meeting with my boss. |
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| 19:29.23 | znf | [TK]D-Fender, by "external sources" I mean just getting calls on my DID |
| 19:30.09 | jpastore | [TK]D-Fender, sip set debug off; I saw it say that it was shutting off, but I still have a ton of messages scrolling by |
| 19:30.26 | [TK]D-Fender | Very few providers actually sends their users AUTHED calls and you should not be challenging them. For those you should have "insecure=port,invite" in your peer |
| 19:30.42 | [TK]D-Fender | jpastore: What kind of messages? |
| 19:31.33 | jpastore | [TK]D-Fender, "AMD: Channel [SIP...." |
| 19:31.47 | [TK]D-Fender | pastebin it |
| 19:32.09 | jpastore | others about executing an agi script, user log on notifications from cronjobs. |
| 19:32.14 | jpastore | [TK]D-Fender, ok 1 sec |
| 19:35.26 | jpastore | [TK]D-Fender, http://pastebin.com/i87MnM2s |
| 19:37.46 | [TK]D-Fender | That isn't SIP debug |
| 19:37.59 | [TK]D-Fender | that is all sort of other basic dialplan processing stuff |
| 19:38.01 | jpastore | [TK]D-Fender, how can I shut that off? |
| 19:38.10 | [TK]D-Fender | that likee 10 different things |
| 19:38.21 | jpastore | is it possible to shut off all at once? |
| 19:38.24 | [TK]D-Fender | "that off" emans nmothing |
| 19:38.28 | [TK]D-Fender | If yuuo want to see nothig ... |
| 19:38.31 | [TK]D-Fender | The close CLI |
| 19:38.44 | jpastore | the equivalent of sip debug off for everything |
| 19:38.55 | [TK]D-Fender | You might have CORE debug enabled |
| 19:39.03 | [TK]D-Fender | core set debug 0 |
| 19:39.11 | jpastore | let me try that. thank you |
| 19:39.28 | [TK]D-Fender | which.... I'm not suspecting is in play just yet |
| 19:39.40 | [TK]D-Fender | This looks like basic verbose |
| 19:42.51 | [TK]D-Fender | indeed it is |
| 19:43.00 | [TK]D-Fender | This isn't even core debug from the look of it |
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| 20:11.03 | mub | Hello friends! |
| 20:11.46 | klow | anyone know why asterisk (pjsip) would send it's internal IP in an SDP , despite the fact that I have set the external_signaling_address to the public IP? |
| 20:12.52 | mub | My boss decided to virtual the database of our vicidial cluster last night... Without telling me. Vicidial's documentation says NOT to do this, but it doesn't say why. |
| 20:13.18 | mub | Can someone provide a reason why it shouldn't be virtualized? |
| 20:13.34 | [TK]D-Fender | what does "virtual the database" mean? |
| 20:14.20 | mub | [TK]D-Fender: Yesterday it was running on a HP Proliant g5. Now it's a vm in our ovirt cluster |
| 20:15.43 | mub | And so far there are no problems, but I want there to be |
| 20:16.47 | klow | I have no idea what vicidial is until I just looked, but it looks like, as with many other things, certain aspecs of the system may rely on extremely accurate clock timing which is historically a problem in virtual environments, sometimes |
| 20:17.13 | klow | i have issues with conference calls on my vmware asterisk systems, which I suspect is because of clocking problems . |
| 20:19.16 | mub | klow: Yeah, I told him that, but he shot me down saying "only the asterisk servers will stay bare metal" |
| 20:19.47 | mub | "The database doesn't need accurate timing" |
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| 20:24.16 | [TK]D-Fender | Depends what THEY mean. |
| 20:24.25 | [TK]D-Fender | If the DB is SEPARATE from * or nott |
| 20:24.38 | [TK]D-Fender | because that is different than just VMing a complette system |
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| 21:21.32 | znf | [TK]D-Fender, mind having a look on this? http://pastebin.com/xWjwE1GV I've added insecure=port,invite yet I still can't get it to wor k |
| 21:22.08 | znf | [TK]D-Fender, I think the "important" part is at line ~221 |
| 21:23.49 | [TK]D-Fender | Show me the peer |
| 21:26.37 | znf | http://pastebin.com/QhG98vMG <-- it's MYNUM2 |
| 21:26.50 | znf | (neither work, btw, I just tested different things with both) |
| 21:27.48 | [TK]D-Fender | you can't have multiple peers with the same host |
| 21:27.52 | [TK]D-Fender | that is a big screwup |
| 21:27.58 | [TK]D-Fender | it will matcht he FIRST one all the time |
| 21:28.06 | [TK]D-Fender | And you'd see that if you looked at the unmasked debug |
| 21:28.14 | [TK]D-Fender | the others DON'T have what I told you to add |
| 21:28.49 | znf | wait, what unmasked debug? |
| 21:28.52 | [TK]D-Fender | And you configured each of these different in ways that tell me you're doing it wrong for most if not all of them |
| 21:29.16 | [TK]D-Fender | trunk-4021569MYUSER |
| 21:29.20 | [TK]D-Fender | the NUMBERS you're hiding |
| 21:29.29 | [TK]D-Fender | thae part that PROVES what peer I'ms supposed to be looking at |
| 21:29.34 | znf | Ah. |
| 21:29.36 | [TK]D-Fender | And you have made sure we can't verify |
| 21:29.45 | znf | it's MYNUM2 |
| 21:29.54 | [TK]D-Fender | It is not matching the one that has the setting I told you to put in |
| 21:30.16 | znf | sorry for giving out confusing output, just thought I'd mask the more sensitive parts |
| 21:30.19 | [TK]D-Fender | type=friend <- should be PEER |
| 21:30.36 | [TK]D-Fender | your inconsistetntt settings are a mess theree |
| 21:31.11 | znf | I started with that, it's just that I kept trying different things to maybe make this work somehow |
| 21:31.41 | [TK]D-Fender | znf: Show unmasked info |
| 21:31.48 | [TK]D-Fender | So I can prove what's happening |
| 21:31.55 | znf | ok, give me 1 minute |
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| 21:34.15 | znf | [TK]D-Fender, sip debug http://pastebin.com/b7zCfJBy |
| 21:34.51 | [TK]D-Fender | and the configs |
| 21:34.59 | [TK]D-Fender | masking the SECRET is fine |
| 21:35.18 | znf | http://pastebin.com/YYBERLtZ |
| 21:36.17 | znf | Oh, crap. I removed that and I don't have a default route. |
| 21:38.25 | [TK]D-Fender | packs up and heads home.... |
| 21:38.56 | znf | I hope it wasn't something I said |
| 21:41.58 | znf | Oh, that made it work! |
| 21:42.05 | znf | wish I could thank him now |
| 21:42.47 | WIMPy | It won't be long until he's bsck. |
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| 21:45.48 | znf | Cool |
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| 22:09.04 | WIMPy | znf: See. it wasn't long at all. |
| 22:09.12 | znf | :-) |
| 22:09.24 | znf | [TK]D-Fender, thanks. that worked, I get inbound calls now! |
| 22:09.53 | znf | Now, what was that about multiple peers with the same host? I've added a 2nd one and it seems to work the same as the first one. Where will it go wrong? |
| 22:10.05 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:11.20 | [TK]D-Fender | It will ONLY hit the first one |
| 22:11.38 | znf | by "HIT" what do you mean? each peer has its own DID |
| 22:11.42 | [TK]D-Fender | no |
| 22:11.55 | [TK]D-Fender | when a call comes in * tries to find a MATCH for the source of the call |
| 22:12.02 | [TK]D-Fender | peer matches by IP. |
| 22:12.24 | [TK]D-Fender | It will take the FIRST peer you defined with a matching host and all of your calls will get matched against that peer |
| 22:13.21 | znf | So what will I see happening? I'm not sure I follow |
| 22:13.49 | [TK]D-Fender | it will use the FIRST peer you defined. |
| 22:13.57 | [TK]D-Fender | it will not look to your foloowing ones. |
| 22:16.29 | znf | so how do I add multiple accounts from the same provider? |
| 22:18.06 | [TK]D-Fender | Step 1 : stop calling it "adding accounts". |
| 22:18.11 | [TK]D-Fender | Realize how calls are matched |
| 22:18.22 | [TK]D-Fender | Because they will NOT match the next one on INBOUND |
| 22:19.00 | [TK]D-Fender | You can use whatever peer you want for OUTBOUND, but inbound matches on the host and the first one that does wins |
| 22:21.27 | znf | so what you're saying, if I have peer1 and peer2, and my inbound routes are like peer1 -> ext1, and peer2 -> ext2; but peer1 and peer2 are on the same host, the matching won't work like I want it to work? |
| 22:21.51 | [TK]D-Fender | ? |
| 22:21.59 | [TK]D-Fender | You using FreePBX? |
| 22:22.01 | znf | Yes |
| 22:22.06 | [TK]D-Fender | ALWAYS mention that first |
| 22:22.07 | znf | Elastix, actually |
| 22:22.12 | znf | I did, actually. |
| 22:22.17 | [TK]D-Fender | And no, Inbound routes is not even a part of this conversation |
| 22:22.31 | [TK]D-Fender | that is call processing AFTER a call is accepted in the first place |
| 22:22.38 | [TK]D-Fender | We are talking about PEER MATCHING |
| 22:22.52 | [TK]D-Fender | HowWhich TRUNK it will consider the call to be COMING FROM in the first place |
| 22:23.18 | znf | Not to sound like an idiot (which I probably do), but in my scenario, I'm not sure how that will affect me |
| 22:24.13 | [TK]D-Fender | Be aware of the impact on CDR, and for those contexts you are specifying |
| 22:24.22 | [TK]D-Fender | Because it won't be hitting the peer you think it will |
| 22:26.27 | znf | I'm not being picky there, we're just using it for our internal phone system, hopefully I won't be needing to deal with that, ever. |
| 22:29.25 | [TK]D-Fender | Just be aware of how * matches |
| 22:30.54 | znf | Would it be appropiate to ask a OpenCNAM question here? |
| 22:31.48 | znf | If yes, is there a way to change the number to international format (+40... for Romania) before querying OpenCNAM? |
| 22:35.27 | [TK]D-Fender | That's a GUI question. it looks up what you tell it to look up, and your GUI is what defines what it asks for |
| 22:35.35 | [TK]D-Fender | You need to ask in their channel |
| 22:36.07 | znf | I'm aware of how it works. Thanks :0 |
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