| 00:10.13 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
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| 01:16.01 | Meaw | hi guys, getting registeration failed , no matching peer found |
| 01:16.19 | Penguin | Create a peer for the host. |
| 01:16.28 | Penguin | If you already did, pastebin it. |
| 01:16.59 | Meaw | I did not, should i create it through the asterisk cli ? or where exactly |
| 01:17.15 | Penguin | sip.conf |
| 01:18.01 | Penguin | What are you trying to register, anyway? |
| 01:18.28 | Meaw | sip |
| 01:18.45 | Penguin | If you did not create a peer entry in sip.conf for your device, where did you get a username and password for the device? |
| 01:19.09 | Meaw | im using an ip pbx phone system it has a gui |
| 01:19.41 | Meaw | i have a short sip.conf really let me paste it somewhere |
| 01:20.27 | Meaw | http://pastebin.ca/2442025 |
| 01:21.04 | Penguin | That's just the general section. |
| 01:21.22 | WIMPy | I don't see a [general] there. |
| 01:21.43 | WIMPy | So that's not a valid sip.conf. |
| 01:21.43 | Penguin | You have to create entries for any device that you want to peer with asterisk. Phones, another asterisk, an ITSP. Everythings needs a peer entry. |
| 01:22.24 | Meaw | so i should create peer for every sip client? |
| 01:22.30 | Penguin | That's what I just said. |
| 01:22.32 | Meaw | but people do login from other places |
| 01:22.49 | WIMPy | What does that mean? |
| 01:23.00 | Penguin | That's why we have the ability to set dynamic hosts. |
| 01:23.02 | Penguin | host=dynamic |
| 01:23.08 | Penguin | Then you don't have to hard-code an IP address. |
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| 01:43.06 | Meaw | Penguin, done that, and i did create new sip account, trying to login from eyebeam says wrong user/pass |
| 01:43.09 | Meaw | strange |
| 01:43.24 | Penguin | Did you "sip reload" after adding the entry to sip.conf? |
| 01:43.36 | Meaw | i did asterisk reload |
| 01:44.30 | Penguin | The username is between the square brackets, and the password is the secret. Be sure you have entered the data correctly. |
| 01:44.36 | Meaw | yes also did sip reload through asterisk cli |
| 01:49.40 | Meaw | it appears like this |
| 01:49.41 | Meaw | [Jan 1 01:49:20] NOTICE[183]: chan_sip.c:15580 handle_request_register: Registration from '"6009"<sip:6009@192.168.0.224>' failed for '0.0.0.0' - Wrong password |
| 01:50.07 | Penguin | Pastebin your new entry that you put into sip.conf. |
| 01:51.06 | Meaw | ok |
| 01:51.23 | WIMPy | Better the whole thing. |
| 01:51.36 | WIMPy | Do you see your peer with 'sip show peers'? |
| 01:52.05 | WIMPy | Hmm. I guess the log really says so. |
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| 02:34.54 | Penguin | It sure is taking a long-ass time to get that pastebinned. |
| 02:36.41 | j4jackj | Penguin: have you ever dialed between two computers without the use of a line simulator? |
| 02:37.04 | Penguin | Dialed? What is dialed? |
| 02:46.29 | j4jackj | ATD on one computer, ATA on the other, and then you see CONNECT 14400 ? |
| 02:47.05 | Penguin | I've never done that. I have Ethernet. |
| 02:47.48 | j4jackj | I do too. |
| 02:47.54 | j4jackj | I want to try it though :D |
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| 05:13.23 | volga629 | Hello Everyone, trying set-up remote extension on my android phone, but getting retransmissions problem, I am running stun server for this |
| 05:15.52 | j4jackj | Is the phone on the same network? |
| 05:16.37 | volga629 | no |
| 05:16.48 | volga629 | I my cell provider |
| 05:17.11 | volga629 | Canada telus 3g connection soft client |
| 05:17.42 | volga629 | I get registration, but call get dropped after 25 sec |
| 05:19.02 | volga629 | I see on stun that doing mapping, it look like on asterisk. I have pub ip and my local network also nat=yes |
| 05:19.32 | volga629 | in sip setting on asterisk |
| 05:19.53 | volga629 | I tried different clients and result the same |
| 05:20.57 | j4jackj | I mean on the same WiFi net |
| 05:21.11 | j4jackj | It needs nat=no |
| 05:24.10 | volga629 | remote extension is on mobile provider network |
| 05:24.47 | volga629 | If I set nat=no then no RTP at all |
| 05:25.27 | volga629 | REMOTE CELL --INTERNET---FIREWALL--ASTERISK |
| 05:26.51 | j4jackj | well no rtp anyway... |
| 05:27.01 | j4jackj | Maybe put the * where the firewall is. |
| 05:27.10 | j4jackj | Or put * in the DMZ. |
| 05:28.16 | volga629 | yes asterisk in private DMZ behind firewall |
| 05:28.23 | Penguin | Never put asterisk in the DMZ. |
| 05:28.27 | Penguin | Never. |
| 05:28.40 | j4jackj | It's a must for usefulness |
| 05:28.42 | Penguin | Learn to network or hire someone who knows how. |
| 05:28.55 | volga629 | private DMZ |
| 05:29.04 | Penguin | You'll have to have nat set to yes if your asterisk is behind a NAT. |
| 05:29.14 | volga629 | It done |
| 05:29.28 | Penguin | Try not using STUN. |
| 05:29.33 | volga629 | Also I am running my stun |
| 05:29.47 | j4jackj | Try not using STUN. Just once. |
| 05:29.49 | Penguin | Set asterisk correctly for nat, set the device's peer entry correctly for nat. |
| 05:30.03 | Penguin | Turn off all nat traversal settings in the phone. |
| 05:30.04 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:30.18 | Penguin | Turn off STUN, ICE, TURN, or whatever else is enabled. |
| 05:30.22 | volga629 | yes peer set to nat=yes, reinvite=no |
| 05:30.37 | Penguin | s/reinvite/directmedia/ |
| 05:30.55 | Penguin | You just want to have the phone and asterisk without any other stuff involved. |
| 05:31.04 | j4jackj | directmedia=no |
| 05:31.08 | Penguin | Make sure SIP ALG is disabled on your router if it exists. |
| 05:31.42 | volga629 | yes, on fortigate I disabled everything related to sip signalling |
| 05:33.30 | volga629 | on let me try disable stun |
| 05:33.37 | volga629 | on soft client |
| 05:37.30 | volga629 | with disabled stun no rtp at all |
| 05:38.50 | Penguin | Enable sip debug, then make a call. If no audio, pastebin the entire sip debug of the call. |
| 05:41.48 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wghnghvfghrewkeu) |
| 05:42.49 | volga629 | ok on the way |
| 05:45.02 | Penguin | ~pb |
| 05:45.03 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 05:46.05 | volga629 | https://fpaste.networklab.ca/SgJ2/ |
| 05:50.04 | Penguin | I've seen the "retransmitting" lines before and I don't know what causes them. |
| 05:51.48 | volga629 | that exactly why I scratching my head, because I don't reason SDP look ok for me |
| 05:51.56 | volga629 | see |
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| 06:02.00 | volga629 | on rtp debug I see sent/received no problem until 15 sec pass then only sent rtp and never got reply back |
| 06:02.18 | kaldemar | it's not the SDP. asterisk expects an ACK in response to the OK. |
| 06:02.35 | apb1963_ | Capabilities: us - (ulaw|g729|g722|h264), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729) |
| 06:02.48 | kaldemar | INVITE sip:*97@10.18.130.50 looks weird by the way. |
| 06:03.13 | volga629 | it should use public ip right ? |
| 06:04.02 | volga629 | ok let me try set from-domain to pub ip |
| 06:06.34 | Penguin | Did you set your externaddr and localnet values in sip.conf? |
| 06:07.32 | kaldemar | that's an incoming message. it's set by the client or something else in between. |
| 06:08.47 | volga629 | yes I do. current settings nat=yes, external ip and my local subnet where asterisk is located |
| 06:09.09 | volga629 | From: <sip:*97@99.224.107.222>;tag=as452d7fb6 no difference |
| 06:11.35 | volga629 | Is any public live stun server that I can place the test ? |
| 06:16.30 | Penguin | Yeah. stunserver.org |
| 06:17.09 | volga629 | ok let me try |
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| 06:22.24 | volga629 | same think |
| 06:22.29 | volga629 | hmmm |
| 06:24.22 | *** join/#asterisk zordor (~zordor@202.94.145.1) |
| 06:24.39 | zordor | Hi everyone |
| 06:24.56 | zordor | First time around here :) |
| 06:28.26 | volga629 | this no make any sense it should work |
| 06:30.44 | volga629 | thank Everyone for suggestion today, I will continue trouble shoot tomorrow |
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| 11:19.33 | giucio | Hi, I'm writing an AMI script which uses Originate to place a call between Local/<exten> and a third party. I don't seem to be able to have the call go through the dialplan, it simply places the call and doesn't interact with the instructions I've put in extensions.conf. Is this normal? |
| 11:19.38 | giucio | Thanks! |
| 11:20.08 | Greenlight | If you use a Local channel then it'll get dumped into the dialplan. |
| 11:21.20 | Greenlight | When you say it "simply places the call", what exactly do you mean? |
| 11:21.35 | giucio | it actually establishes the call |
| 11:21.40 | Greenlight | To what |
| 11:21.43 | kaldemar | giucio: what does your originate action look like? |
| 11:21.59 | giucio | to the extension I specify in the AMI command |
| 11:22.08 | Greenlight | So, it *does* hit the dialplan |
| 11:22.31 | giucio | Greenlight: I've put Verbose() calls everywhere, but I cannot find where it enters it |
| 11:22.57 | giucio | Greenlight: I was actually worried it wasn't the right way to do it. If it is so, I'll investigate further |
| 11:22.58 | Greenlight | It'll enter it where you tell it, by specifying an extension. |
| 11:23.12 | Greenlight | If we can see your originate command to verify |
| 11:23.49 | Greenlight | Also, if you just turn up verbosity at the CLI, you should clearly see the call flow. |
| 11:23.51 | giucio | kaldemar, Greenlight: I'll debug the raw command from the library I'm using (pyst2) and see if the problem is there |
| 11:29.24 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 11:35.44 | giucio | kaldemar, Greenlight: this is a dump of the AMI traffic: http://dpaste.com/1363564/ |
| 11:36.29 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 11:36.32 | giucio | and this is the dialpan: http://dpaste.com/1363565/ |
| 11:36.40 | Greenlight | Well it's not doing what you said. |
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| 11:36.43 | giucio | as you can see, there only one entry in the "webdialer" context |
| 11:36.45 | Greenlight | It's NOT dialling a Local channel. |
| 11:36.55 | Greenlight | It's dialling a SIP channel |
| 11:37.32 | Greenlight | So that'll not hit the dialplan, it'll call the trunk directly. Once the trunk answers, it'll get dumped to EXTEN |
| 11:39.25 | Greenlight | Which, depending on the behaviour you require, isn't a bad thing. There's no reason to use a Local channel unless you *need* to |
| 11:39.44 | Greenlight | Does SIP/giucio ring? |
| 11:40.45 | giucio | Greenlight: sorry I think I made confusion. Let me double check everything |
| 11:40.58 | giucio | Lots of new concepts here hehe :) |
| 11:41.26 | Greenlight | What are you trying to achieve? |
| 11:41.44 | Greenlight | Are you trying to call your SIP phone and have it dumped into the dialplan ? |
| 11:42.24 | giucio | yes |
| 11:42.39 | giucio | so that I can manipulate sip headers from there |
| 11:42.45 | Greenlight | Umm |
| 11:43.12 | Greenlight | At the point this hits the dialplan the SIP headers have already been sent. The calls already been answered by the SIP side. |
| 11:43.34 | Greenlight | Unless, you'lre planning to then Dial out to different SIP destination where you want to send custom headers. |
| 11:44.19 | giucio | I'm calling landline numbers |
| 11:44.30 | giucio | through a sip trunk of a local provider |
| 11:44.56 | Greenlight | Looks like you're trying to send the auto-answer header, to SIP/giucio ? |
| 11:45.00 | giucio | yes |
| 11:45.15 | giucio | we have agents with softphones |
| 11:45.24 | giucio | and these softphones should pick up the call automatically |
| 11:45.36 | giucio | so I'm trying to set the relevant sip headers |
| 11:45.46 | Greenlight | Which softphone? |
| 11:46.00 | giucio | I'm tring with Jitsi and CSipSimple |
| 11:46.13 | Greenlight | And you're 100% it supports auto-answer? |
| 11:46.19 | Greenlight | (Not many do) |
| 11:46.45 | giucio | yes they both do (at least, they have the option in their configuration dialogs) |
| 11:46.51 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:47.13 | giucio | You think there's a better way to achieve what I'm trying to do? |
| 11:47.28 | giucio | I don't want people to have to press any button to start the call |
| 11:48.18 | Greenlight | Yea, I get what you're trying to achieve. |
| 11:48.48 | Greenlight | And that's the header the softphone says it accepts ? |
| 11:48.51 | giucio | yes |
| 11:49.54 | Greenlight | Try without the 511 |
| 11:49.56 | Greenlight | *51 |
| 11:50.05 | Greenlight | And, include "'s |
| 11:50.10 | Greenlight | Eg |
| 11:50.20 | Greenlight | Variable: __SIPADDHEADER="Alert-Info: Auto Answer" |
| 11:51.26 | giucio | ok |
| 11:54.36 | Greenlight | Also, you shouldn't need the __ |
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| 12:07.54 | giucio | Greenlight: Ok I understand what I was doing wrong. Both the agent extension and the destination extension need to be declared in the webdialer context |
| 12:08.08 | giucio | that's why it wasn't hitting the agent extension in that context |
| 12:08.14 | giucio | it was hitting the default one |
| 12:08.50 | Greenlight | giucio: The "agents" extension does not. You're calling it directly. |
| 12:09.35 | giucio | I'll paste my current (apparently working) config |
| 12:10.51 | giucio | This is the dialplan: http://dpaste.com/1363601/ |
| 12:11.01 | giucio | This is the AMI command: http://dpaste.com/1363602/ |
| 12:11.23 | giucio | where channel="giucio", which is a Jitsi softphone with auto-answer set to true |
| 12:11.46 | giucio | it seems to work. Do you think it's incorrect in some aspect? |
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| 12:37.30 | giucio | Greenlight: sorry I didn't put your nick in my previous lines... I've pasted my small config in case you can have a look at it |
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| 12:38.53 | Greenlight | Your pastebin differs from your earlier AMI trace |
| 12:39.01 | Greenlight | You're now originating to a Local channel. |
| 12:39.40 | giucio | Greenlight: yes |
| 12:39.52 | giucio | Greenlight: I thought that was the correct way to do it |
| 12:39.55 | giucio | based on your feedback |
| 12:40.07 | giucio | so now it does hit the dialplan |
| 12:40.12 | giucio | in the webdialer context |
| 12:40.23 | giucio | and I'm able to manipulate sip headers from the diaplan |
| 12:40.41 | giucio | instead of doing it with the SIPADDHEADER var |
| 12:40.48 | Greenlight | Yea, you can do it that way |
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| 13:46.34 | hjf | i figured out why i was being unregistered on call |
| 13:47.55 | hjf | i was trying G.722 and i noticed the audio clock rate was 16KHz so i tried to increase it to 44.1khz... it worked, but wasn't a major difference. so i changed to 32KHz and forgot about it. it was crashing when it tried to select 32KHz sampling rate. i put it back to 16K and it works |
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| 13:48.30 | Blashyrkh | i have a problem with disa not recognizing my dtmf tones |
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| 14:01.15 | Blashyrkh | what exactly does relaxdtmf=yes do? |
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| 14:03.59 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 14:05.06 | Blashyrkh | so now i try to pinpoint where the dtmf tones get lost |
| 14:05.25 | Blashyrkh | in this post someone just turned up his logger level |
| 14:05.27 | Blashyrkh | https://issues.asterisk.org/jira/browse/ASTERISK-15525 |
| 14:05.44 | Blashyrkh | where do i do that? |
| 14:07.24 | hjf | Blashyrkh: are you using in-band dtmf or out of band signaling? |
| 14:07.34 | Blashyrkh | out of bound rfc |
| 14:07.44 | Blashyrkh | configured at the sip provider and on the peer side |
| 14:10.29 | Greenlight | We had to change the source code to make DTMF work with our ITSP |
| 14:10.49 | Greenlight | Seems Cisco and Asterisk disagree about what the RFC dictates |
| 14:12.04 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:12.26 | Greenlight | Can't remember which way around it is, but one of them wants to use 101 and the other 96 |
| 14:12.43 | Greenlight | The "payload type" i think it's called. |
| 14:12.46 | hjf | hm... i'm unable to "force" g.729 with an spa2102, to some internet voip provider. their website says they support it but on the status page i only see g711u |
| 14:13.30 | Greenlight | hjf: Is Asterisk inbetween these ? |
| 14:13.37 | hjf | no idea |
| 14:13.48 | Greenlight | Ok, well, good luck with that |
| 14:13.53 | hjf | yeah |
| 14:14.06 | hjf | it's a borrowed device eevn |
| 14:14.07 | hjf | even |
| 14:14.12 | hjf | i don't even have the account password |
| 14:14.25 | hjf | so if i screw up i dont know what to do lol |
| 14:14.52 | Greenlight | You're in #asterisk. I you don't even know if Asterisk is involved in your setup, then I think you're going to struggle... |
| 14:15.28 | hjf | won't make much of a difference anyway, as i'm not in control of the server |
| 14:16.29 | Greenlight | Yea, it's not like "disallow=all allow=g729" would force it or anything crazy. |
| 14:16.29 | Blashyrkh | i see the dtmf coming in as 101 and it works in a similar setup |
| 14:17.06 | hjf | Greenlight: that's what i did in my home asterisk server. but the client is a linksys device |
| 14:17.06 | Greenlight | Blashyrkh: I can't recall which way around it was, but we had to change the source from what it's set to by default to the other one. |
| 14:17.45 | Blashyrkh | the source is my sip provider |
| 14:17.59 | Blashyrkh | if you mean source by that? |
| 14:18.09 | Greenlight | No, I mean source code |
| 14:18.19 | Greenlight | rtp_engine.c iirc |
| 14:18.40 | Greenlight | Asterisk is *hard coded* to use either 96 or 101. Some ITSP's use the other one. |
| 14:19.10 | Greenlight | Seems silly to me, but that's the way it is. |
| 14:19.56 | *** join/#asterisk felipealmeida (~user@177.205.235.186.dynamic.adsl.gvt.net.br) |
| 14:20.43 | Greenlight | You may have better luck with in-band or sipinfo |
| 14:30.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:33.24 | hjf | does anyone know the SPA2102 or, i guess, the linksys/cisco adapter things? this has two fxs ports, and i'd like to use the dial plan to dial either my home extensions, or an external SIP provider, from an analog phone. but i don't know if a "line" is hardwired to a "physical phone port" in the device? |
| 14:38.53 | hjf | heh well now it decided to call with g.729 |
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| 15:09.02 | *** join/#asterisk [sr] (~kvirc@213.228.181.48) |
| 15:09.04 | [sr] | hi WIMPy |
| 15:10.28 | WIMPy | jo [sr] |
| 15:11.15 | [sr] | wazza |
| 15:11.37 | WIMPy | Got up too early. |
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| 15:29.55 | Blashyrkh | i have disa set up but with some calls i have the problem that the disa call ends with |
| 15:30.11 | Blashyrkh | http://pastebin.com/BFuTP1xt |
| 15:30.29 | Blashyrkh | is there an easy way to find ot what non-zero this is? |
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| 15:33.13 | WIMPy | Someone hung up instead of dialling an extension. |
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| 16:39.07 | carbinemonoxide | Hello. I'm having issues compiling asterisk |
| 16:39.28 | Greenlight | carbinemonoxide: What sort of issues ? |
| 16:39.32 | carbinemonoxide | /usr/bin/ld: logger.o: invalid string offset 521680684 >= 12626 for section `.strtab' |
| 16:39.36 | carbinemonoxide | Lots of those |
| 16:41.31 | Greenlight | You did ./configure ... did it report any issues ? |
| 16:41.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:41.50 | carbinemonoxide | None that I saw |
| 16:42.14 | carbinemonoxide | Configuring again |
| 16:42.14 | Greenlight | What distro are you on ? |
| 16:42.17 | carbinemonoxide | Debian |
| 16:42.19 | carbinemonoxide | 7 |
| 16:43.17 | Greenlight | The linker's not happy about something, but I'm not sure what. |
| 16:43.24 | Greenlight | Hopefully someone else will have an idea |
| 16:44.29 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:44.45 | carbinemonoxide | I saw no errors running ./configure |
| 16:45.02 | carbinemonoxide | ls |
| 16:45.06 | carbinemonoxide | Whoops. :P |
| 16:47.05 | carbinemonoxide | From config.log: gcc version 4.7.2 (Debian 4.7.2-5) |
| 16:47.05 | carbinemonoxide | configure:3683: $? = 0 |
| 16:47.05 | carbinemonoxide | configure:3672: gcc -V >&5 |
| 16:47.05 | carbinemonoxide | gcc: error: unrecognized command line option '-V' |
| 16:47.05 | carbinemonoxide | gcc: fatal error: no input files |
| 16:47.07 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:47.08 | carbinemonoxide | compilation terminated. |
| 16:47.27 | carbinemonoxide | conftest.c:9:28: fatal error: ac_nonexistent.h: No such file or directory |
| 16:47.27 | Greenlight | Do you have some old compiler or something |
| 16:47.31 | Greenlight | Try updating |
| 16:47.59 | Greenlight | Also |
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| 17:01.59 | John1001 | hello everyone |
| 17:03.11 | navaismo | wasaaaaaaa |
| 17:03.36 | John1001 | i successfully installed the new asterisk 11.5 with the asterisk-gui and wow it looks nice, internal calls are working great, but there is a weird problem with the context, it looks for the 's' extension and i have to include it manually in the config files itself .. |
| 17:03.48 | John1001 | Hey there Navaismo |
| 17:04.25 | John1001 | <PROTECTED> |
| 17:04.38 | [TK]D-Fender | asterisk-gui <--- clarify this please... |
| 17:04.55 | [TK]D-Fender | The actual product named "Asterisk GUI" has been dead for some time now... |
| 17:05.09 | John1001 | oh |
| 17:05.18 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:05.43 | [TK]D-Fender | So if you're referring the the blue/orange motif coloured one... I would avoid if I were you |
| 17:05.52 | [TK]D-Fender | It has not really been developed for years |
| 17:06.13 | John1001 | im refering ot the 'new' 2.0 branch (its what they told me) |
| 17:06.17 | John1001 | i will get the link for oyu |
| 17:06.41 | navaismo | if you want to use a GUI with your asterisk better use FreePBX |
| 17:07.04 | [TK]D-Fender | John1001: Avoid that one |
| 17:07.09 | [TK]D-Fender | John1001: It is dead |
| 17:07.24 | John1001 | Copyright (C) 2006-2010 Digium, Inc. |
| 17:07.27 | John1001 | omg you're right :( |
| 17:07.35 | John1001 | i loved that gui but it clearly had some bugs |
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| 17:16.21 | carbinemonoxide | And... Another compile attempt. |
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| 17:44.59 | P3r3gr1n3 | I have been attempting to get my voicemail to send to email for quite a while. I have dont this on other servers with no problem, but this one is giving me greif. I am on asterisk 11.4.0. I have successfully sent email using sendmail. I cannot get my voicemail to send to my email. |
| 17:46.52 | P3r3gr1n3 | what is the correct mailcmd for sendmail? mailcmd=/usr/sbin/sendmail |
| 17:46.59 | P3r3gr1n3 | Should I have to specify? |
| 17:49.29 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 17:52.39 | [TK]D-Fender | Asterisk's sample configs call the common sendmail script in the usual way. |
| 17:52.46 | [TK]D-Fender | What MTA you actually have that using is another matter |
| 17:52.53 | [TK]D-Fender | as is the responsibility of configuring it |
| 17:53.28 | carbinemonoxide | What distro? |
| 17:53.54 | [TK]D-Fender | In common barebones installs this will send directly from your server which won't have any auth info, probably no domain info, etc and could get rejected for a huge number of reasons including reverse DNS lookup failures, etc |
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| 18:02.55 | carbinemonoxide | D-Fender, you seem pretty good with Asterisk... |
| 18:03.49 | carbinemonoxide | [TK]D-Fender: Happen to know what might cause this: /usr/bin/ld: logger.o: invalid string offset 521680684 >= 12626 for section `.strtab' |
| 18:04.06 | [TK]D-Fender | no |
| 18:04.10 | carbinemonoxide | Bleh. |
| 18:04.51 | carbinemonoxide | At least I learned that 521680684 is greater than 12626... |
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| 18:31.41 | bacobart | when i make a call on my softphone to another softphone via my asterisk server it shows up as extension@ip-address-of-asterisk-server. is it possible to just have only the extension show up, not have the @ip-address bit included? |
| 18:32.46 | [TK]D-Fender | bacobart: that's you softphone that chooses to display it like that |
| 18:32.53 | [TK]D-Fender | bacobart: Asterisk is not responsible for that |
| 18:32.58 | bacobart | oh, okay |
| 18:33.12 | bacobart | that's too bad then :P thanks |
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| 18:39.15 | apb1963_ | carbinemonoxide: If I had to guess (and I'm guessing), I'd say your logger.o file got corrupted somehow or just didn't link right for some reason. Recompile it, see if that helps any. It's also possible somehow you're using the wrong options, or mixed different versions together. All guesses. |
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| 18:43.06 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 18:49.30 | volga629 | Hello Everyone, having trouble with RTP stream for softphone on my cell with 3G connection, getting retransmissions. Here call debug https://fpaste.networklab.ca/eZJm/ |
| 18:51.45 | volga629 | test case: CELL 3G soft client --- INTERNET---FIREWALL ----(private dmz) ASTERISK |
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| 19:21.04 | carbinemonoxide | editline/libedit.a: could not read symbols: Bad value |
| 19:21.04 | carbinemonoxide | collect2: error: ld returned 1 exit status |
| 19:21.04 | carbinemonoxide | make[1]: *** [asterisk] Error 1 |
| 19:21.04 | carbinemonoxide | make: *** [main] Error 2 |
| 19:21.05 | carbinemonoxide | root@hestia:/usr/src/asterisk-11.5.1# |
| 19:25.02 | navaismo | ~pastebin |
| 19:25.02 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:25.47 | carbinemonoxide | Didn't think 5 lines would warrant a pastebin but thanks. |
| 19:26.10 | carbinemonoxide | http://pastie.org/8292275 |
| 19:26.12 | carbinemonoxide | There. |
| 19:30.06 | navaismo | which OS, what version of asterisk? |
| 19:30.26 | carbinemonoxide | Debian, Asterisk 11 |
| 19:30.46 | carbinemonoxide | Debian 7 more specifically |
| 19:37.06 | *** join/#asterisk Hammertalk (~Hammertal@static-98-113-96-227.nycmny.fios.verizon.net) |
| 19:37.50 | Hammertalk | hey I am having an RTP issue with a remote phone can someone help me out |
| 19:37.54 | Hammertalk | please |
| 19:41.58 | ChannelZ | If you would define the problem, maybe. |
| 19:44.41 | navaismo | carbinemonoxide, sounds more like an OS issue than asterisk |
| 19:44.47 | navaismo | did you installed correctly the system? |
| 19:45.55 | carbinemonoxide | It's a fresh install of Debian. Asterisk is the first thing I've tried to install on it besides DAHDI and PRI |
| 19:46.11 | Hammertalk | <PROTECTED> |
| 19:47.49 | Hammertalk | and yes I can make calls if the phone is on the same network as the PBX |
| 19:47.50 | ChannelZ | Hammertalk: Minimally you need to properly config localnet and externaddr in sip.conf, and then also forward some ports for incoming RTP to your asterisk box, as defined in rtp.conf |
| 19:48.41 | ChannelZ | (the default is ridiculous like 10000-20000 but you can narrow that down to a few depending on how many calls you plan on processing at any given time. Need some buffer room, I arbitrarily pick 25 or 50 for a few calls) |
| 19:48.54 | Hammertalk | well I have already forwarded all the ports I needed to and as far as eddeding the config file I cant the PBX is a switchvox |
| 19:49.42 | navaismo | carbinemonoxide, have you tried with an older version like 11.4.0 or 11.5.0 |
| 19:49.55 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:49.57 | ChannelZ | what direction of audio doesn't work? |
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| 19:50.17 | Hammertalk | Both |
| 19:50.35 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 19:50.49 | ChannelZ | well you seem to have firewall issues |
| 19:50.59 | carbinemonoxide | I'm running 11.5.1 |
| 19:51.32 | Hammertalk | in the packet capture however I can see the phone sending RTP and the called party sending RTP however this is off my router |
| 19:51.45 | ChannelZ | Asterisk needs to be able to send out to UDP on almost any port (you can't control what the remote end requests) and needs to be able to accept UDP on whatever port range is defined in rtp.conf |
| 19:51.47 | Hammertalk | on my BPX I see no RTP at all |
| 19:52.27 | carbinemonoxide | Gonna try the 11.2-cert2 |
| 19:52.27 | Hammertalk | ChannelZ Port 10,000 to 30,000 is wide open |
| 19:52.34 | ChannelZ | Assuming your asterisk is giving the remote end the right (external) IP address to send do, then you'll have to investigate why those packets aren't making it in through your firewall/router |
| 19:54.36 | Hammertalk | So thats were I am needing help. does anyone have experince using a Switchvox with a cisco router and can point me in the right direaction |
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| 19:56.48 | ChannelZ | it's more of a router thing than a switchvox thing. But we'd really need to see a sip debug to prove it's not an asterisk config issue |
| 19:57.10 | [TK]D-Fender | Hammertalk: Switchvox is a closed platform and only directly supported by Digium, not here |
| 19:57.51 | volga629 | I have the same issue with RTP, registration work fine |
| 20:01.46 | Hammertalk | volga629> what router are you using? |
| 20:02.26 | volga629 | I have Fortigate 110 in front and I disabled all SIP features |
| 20:07.04 | Hammertalk | do you get a rining before the call is connected |
| 20:13.27 | ChannelZ | volga seems to have 100 different problems. Always. |
| 20:17.05 | Hammertalk | I know its not the PBXs issue I am hoping someone as run in to this before |
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| 20:26.23 | volga629 | My internet went down heavy rain in Ottawa |
| 20:26.46 | volga629 | So back to story |
| 20:27.16 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 20:27.56 | Hammertalk | <PROTECTED> |
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| 20:28.04 | volga629 | I am getting retransmissions when trying dial from remote extension |
| 20:28.14 | volga629 | What firewall you have ? |
| 20:28.54 | Hammertalk | I am using a router I havnt setup a firewall yet |
| 20:29.53 | Hammertalk | have you turned off all the ALG SIP opations |
| 20:30.42 | volga629 | I have port forward. On fotigate language is vip which point to asterisk on private dmz and the same with RTP range. To help with NAT issue I am running stun server |
| 20:31.03 | volga629 | what kind of router ? |
| 20:31.16 | Hammertalk | Cisco SRP%$!W |
| 20:31.22 | Hammertalk | 541 klol |
| 20:32.23 | Hammertalk | what kind of appliance are you using? |
| 20:32.30 | Hammertalk | I mean PBX |
| 20:32.35 | volga629 | on cisco router enabled sip inspection did you disabled ? |
| 20:32.45 | volga629 | asterisk |
| 20:34.03 | Hammertalk | ill look in to that but I believe I have |
| 20:34.07 | volga629 | for right now until I will get all infrastructure working correctly then I might be decide get some pbx based on asterisk. |
| 20:34.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.22) |
| 20:34.44 | Hammertalk | ya I disabled ALG SIP |
| 20:34.54 | volga629 | Do you access to cli or you usinf ASDM ? |
| 20:34.57 | volga629 | using |
| 20:35.14 | Hammertalk | CLI |
| 20:36.03 | Hammertalk | well more like its GUI |
| 20:36.56 | Hammertalk | So why go thur the bothor of progarming something you arent really going to use |
| 20:37.10 | volga629 | Are you using nat overload or one to one ? |
| 20:38.06 | volga629 | to learn all trouble in test environment instead invest 20 k in system which will be half functional |
| 20:39.07 | Hammertalk | One to many I think |
| 20:39.55 | Hammertalk | no sorry one to one |
| 20:40.10 | volga629 | yes one to one is good |
| 20:40.37 | volga629 | so do port forward from outside to you asterisk box |
| 20:41.11 | Hammertalk | I already setup the port forwarding |
| 20:43.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 20:43.10 | volga629 | on remote extension can register to asterisk box ? |
| 20:43.18 | Hammertalk | yes |
| 20:43.33 | Hammertalk | It can even make and recive calls |
| 20:43.42 | Hammertalk | just cant talk |
| 20:44.01 | volga629 | RTP range is forwarded ? |
| 20:44.12 | Hammertalk | yes |
| 20:44.43 | volga629 | asterisk box sip settings should be set nat=yes and public plus you local subnet where sit asterisk box |
| 20:45.22 | volga629 | public ip |
| 20:45.56 | Hammertalk | already done |
| 20:46.03 | volga629 | is remote end is ata or softphone ? |
| 20:46.14 | Hammertalk | Polycom |
| 20:46.22 | volga629 | so what you see on debug ? |
| 20:47.47 | Hammertalk | well when I run a packet capture on the router where the PBX sits I see RTP flowing to the IP address of the PBX over when I run it off the PBX I see no RTP at all |
| 20:48.06 | volga629 | polycom is know as trouble maker for NAT environment, is any chance you can try test with different softphone ? |
| 20:48.34 | Hammertalk | good Idea ill try it know one second |
| 20:49.09 | volga629 | try rtp debug on asterisk box see where is loosing stream |
| 20:51.16 | mitchrodrigues | so anyone else experience sigill when starting asterisk 11.5.1 ubuntu 12.10 (=> 0x00000000005bd768 <+152>:vmovq %rdi,%xmm1) |
| 20:52.17 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:53.36 | Hammertalk | I am using Switchvox which is nercomplietly locked down no sure Ill be able to runa rtp debug |
| 20:53.52 | Hammertalk | best I could do is run a TCP Dump |
| 20:55.48 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-3-122.home.otenet.gr) |
| 20:56.12 | volga629 | try see on tcpdump one sent one received and which ips |
| 20:57.42 | Hammertalk | So If I run it of my router I see the router getting the RTP but it doesnt forward it on and if i do a tcp dump of the PBX I see it never recived the RTP \ |
| 20:57.45 | Hammertalk | at all |
| 20:57.49 | Hammertalk | From both sides |
| 20:57.53 | Hammertalk | of the call |
| 20:58.35 | volga629 | better is cap file with tcpdump and review with wireshark there you can check |
| 20:59.31 | Hammertalk | that is how I am reviewing them |
| 21:00.49 | volga629 | cap file can be open with wireshark |
| 21:01.00 | volga629 | and decode rtp stream |
| 21:02.28 | volga629 | check on router Voice--->Sip---->Rtp parameter if enabled then make sure asterisk port range not overlapping |
| 21:03.39 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-31-243.home.otenet.gr) |
| 21:03.58 | volga629 | http://www.digium.com/en/products/switchvox/support/switchvox-on-your-network |
| 21:04.44 | Hammertalk | OMG that worked thank you soooo much |
| 21:05.04 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 21:05.39 | Hammertalk | you are a genius |
| 21:06.09 | Hammertalk | how did you know that would be an issue? |
| 21:07.40 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 21:08.02 | volga629 | :-) many broken routers |
| 21:08.51 | Hammertalk | lol man your the best |
| 21:09.12 | Hammertalk | can you chat me a email address of yours? |
| 21:10.22 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:17.31 | *** part/#asterisk Hammertalk (~Hammertal@static-98-113-96-227.nycmny.fios.verizon.net) |
| 21:17.38 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-33-143.home.otenet.gr) |
| 21:18.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:18.47 | *** join/#asterisk SGjunior (~sgjunior@out-pq-198.wireless.telus.com) |
| 21:30.14 | volga629 | in my case look like fortigate on the way, but I see proper traffic come in and going out, but some place is get lost and can't confirm if it is asterisk side or border |
| 21:32.24 | volga629 | Ottawa power outage plan to be restored at 8:30 pm :-( |
| 21:36.51 | *** part/#asterisk coreyf1513 (~cfarrell@24-177-3-55.dhcp.nwtn.ct.charter.com) |
| 21:43.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.19) |
| 21:44.21 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 21:47.49 | *** join/#asterisk Blashyrkh (~chatzilla@213-33-14-146.adsl.highway.telekom.at) |
| 21:48.41 | Blashyrkh | iam trying to debug an asterisk call that gets disconnected suddenly and at different locations in the dialplan |
| 21:51.59 | Blashyrkh | http://pastebin.com/JDfQk1r1 |
| 21:52.40 | Blashyrkh | how can i find out more about what disconnected here |
| 21:53.51 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:12.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:48.36 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 22:54.33 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 22:57.12 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 22:59.57 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:202b:9ef7:f647:5dd0) |
| 23:06.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.26) |
| 23:16.29 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:17.22 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 23:33.40 | carbinemonoxide | Would anyone here be able to help me in debugging my Asterisk server? |
| 23:42.15 | carbinemonoxide | I've got my server set up now. Just trying to figure out what I'm doing wrong. I don't know if my Asterisk server has connected to Google Voice and I cannot connect a SIP phone to it. |
| 23:48.37 | *** join/#asterisk russum (~russum@ool-4a599d14.dyn.optonline.net) |
| 23:50.49 | carbinemonoxide | Ugh |
| 23:51.03 | carbinemonoxide | Can't connect my SIP phone to my Asterisk server on the same network. |
| 23:51.12 | igcewieling | ~ask |
| 23:51.12 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:51.55 | carbinemonoxide | I don't know how to debug this asterisk server. All I know is to check /var/log/asterisk/messages and that doesn't show me anything. |
| 23:52.18 | igcewieling | ~book |
| 23:52.18 | 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 |
| 23:52.23 | igcewieling | read that then come back |
| 23:52.31 | igcewieling | at least the first few chapters. |
| 23:52.52 | j4jackj | Does anyone know how to get a google voice account when yours has been crippled? |
| 23:55.42 | navaismo | carbinemonoxide, did you allow the sip port on your pbx firewall? |
| 23:56.32 | igcewieling | navaismo: as far as I can tell he doesn't even know how to get into the Asterisk console. |
| 23:57.08 | carbinemonoxide | asterisk -rrrv |
| 23:57.12 | carbinemonoxide | I know that much |
| 23:57.32 | igcewieling | then sip debug would be useful. on a pastebin. |
| 23:57.33 | igcewieling | ~pb |
| 23:57.34 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:57.51 | carbinemonoxide | At this point my brain is fried because I've spent the past 2 days dealing with issues while compiling, so sorry if I'm not super descriptive |
| 23:57.53 | igcewieling | carbinemonoxide: I hope that was a typoe. |
| 23:58.10 | carbinemonoxide | I'm getting all of my information from bad tutorials |
| 23:58.27 | *** join/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 23:58.29 | igcewieling | carbinemonoxide: try getting it from the asterisk book. |
| 23:58.36 | carbinemonoxide | I am right now |
| 23:58.39 | igcewieling | I hope "asterisk -rrrv" was a typoe |
| 23:59.34 | ChannelZ | reconnect! reconnect! reconnect god damnit! |
| 23:59.50 | *** part/#asterisk russum (~russum@ool-4a599d14.dyn.optonline.net) |