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| 00:49.39 | Kalavera | hey guys I see an error on asterisk console [Jul 7 19:34:14] NOTICE[17810][C-000327ff] chan_sip.c: Failed to authenticate on INVITE to '"1557" <sip:1557@192.168.100.2> when I try calling by a second sip trunk |
| 00:49.56 | Kalavera | any idea what can I do to fix it |
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| 01:25.31 | [TK]D-Fender | Show the actual full call with "sip set debug on" |
| 01:25.38 | [TK]D-Fender | ~pb |
| 01:25.38 | 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. |
| 01:25.40 | [TK]D-Fender | ^^^ |
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| 01:32.17 | Kalavera | [TK]D-Fender: http://pastebin.com/0T8vsQCY |
| 01:33.23 | Kalavera | hope you can clarify it to me |
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| 01:41.44 | TECFALL | did they get rid of the dial command via cli in asterisk 13? |
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| 01:58.56 | [TK]D-Fender | no |
| 02:00.18 | [TK]D-Fender | Kalavera, "core set verbose 10" . New call |
| 02:01.50 | TECFALL | No such command 'dial 249@outbound' |
| 02:02.06 | TECFALL | core show help also doesn't show the dial command |
| 02:08.59 | [TK]D-Fender | Show the call |
| 02:09.02 | [TK]D-Fender | and the dialplan to match |
| 02:09.09 | [TK]D-Fender | wait.. |
| 02:09.14 | [TK]D-Fender | you can't just shove that on * CLI |
| 02:09.25 | [TK]D-Fender | What are you actually trying to accomplish? |
| 02:11.33 | TECFALL | I could always use dial to initiate calls from the cli for testing |
| 02:17.43 | Kalavera | [TK]D-Fender: I cant make another call right now cause they have turned the machines off and I am remotly , I will try again tomorrow morning and be back with that output |
| 02:17.55 | Kalavera | [TK]D-Fender: Thank you for your help |
| 02:19.18 | [TK]D-Fender | TECFALL, "help console dial" |
| 02:20.30 | TECFALL | ahhh, console dial |
| 02:20.39 | TECFALL | in 1.4 it was just dial |
| 02:21.10 | [TK]D-Fender | 1.4 was a lifetime ago |
| 02:22.29 | TECFALL | haha, right! I was given the great task of upgrading all of our asterisk servers |
| 02:22.56 | Kalavera | woww good luck |
| 02:22.59 | Kalavera | smiles |
| 02:23.23 | TECFALL | Haha, it isn't so bad. Looking at their current set up there was a lot of security holes. |
| 02:24.06 | TECFALL | the only hangup I am having at this point is getting two extensions to work between two remote locations. |
| 02:24.52 | TECFALL | But i think i'm going to call it a night. I'm sure i will be on here in the morning asking for some more help tomorrow. Appreciate you all! |
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| 02:32.08 | [TK]D-Fender | heads out for the night |
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| 04:21.42 | goatse-security | * * * * * Vote for weev! MAKE THE G.N.A.A. GREAT AGAIN!!! * * * * * |
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| 08:44.21 | pyroc | Hey all. Hoping for some help with custom contexts. I've had this working in the past, but I can't get it to work now. The trunk provider we use sends the SIP TO header differently, so make DID's work we need to change a few things by creating a custom context I believe |
| 08:44.29 | pyroc | incoming call: http://pastebin.com/62Vf38yG |
| 08:44.36 | pyroc | changes made to extensions_override_freepbx.conf: http://pastebin.com/T7u3ft5c |
| 08:44.43 | pyroc | trunk settings: http://pastebin.com/Wep7vN2A |
| 08:44.54 | pyroc | what we see in the CLI: chan_sip.c:25859 handle_request_invite: Call from '0884291510' (203.161.160.71:5060) to extension '0884291510' rejected because extension not found in context 'custom-from-engin' |
| 08:45.05 | pyroc | any help super appreciated |
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| 09:26.33 | Kunsi | pyroc: ask #freepbx |
| 09:27.22 | Kunsi | Also, i don't see extrnsion |
| 09:27.39 | Kunsi | Extension 0884291510 |
| 09:42.59 | pyroc | thats not an extension, thats the DID |
| 09:43.42 | pyroc | calls in to 0884291519 -> ext 2999 is what we're trying to achieve, but the SIP TO header has it in a weird format so we need to do somethng to get it to work |
| 09:44.01 | pyroc | this script worked previously, but this is a newer version of asterisk and it doesn't play the same |
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| 10:29.30 | WIMPy | Is QUEUE_VARIABLES(QUEUEHOLDTIME) the same as the expected wait time announced to the caller (if enabled)? Or is that something else again? |
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| 11:06.41 | jkroon | hi all, i've got a client that wants inbound SIP calls based on ext@his.ip, so basicaly he wants someone to send an invite to extension@his.ip and IF that extension is valid on the local machine, dial that specific extension. |
| 11:06.53 | jkroon | i'm guessing that's the whole thing with sip and guest? |
| 11:07.21 | WIMPy | Just allowguests and you're done. |
| 11:08.06 | WIMPy | Just make sure you don't play some announcement when someone calls an invalid extension. |
| 11:08.11 | jkroon | into which context do they go - the default global? |
| 11:08.37 | WIMPy | What you configured in the global section, yes. |
| 11:09.23 | jkroon | that just feels damn risky. |
| 11:09.59 | WIMPy | Why? |
| 11:10.06 | jkroon | dunno, it just does. |
| 11:10.24 | jkroon | what happens if someone sends an invite to partynumber@his.ip and somehow that thing routes out. |
| 11:10.44 | jkroon | ok, so context=guests and in guests I just include the contexts to which access is required? |
| 11:10.48 | Kunsi | Your default context should never route out |
| 11:11.06 | WIMPy | You better don't allow that in the context you use. |
| 11:11.39 | jkroon | hahahaha, just busy double checking that now :) |
| 11:21.34 | jkroon | thanks WIMPy |
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| 13:18.54 | TECFALL | When dialing between local extensions I have to either use insecure=port or insecure=port,invite why can't i use insecure=no? http://pastebin.com/gXqRJKfv |
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| 13:29.33 | [TK]D-Fender | You don't |
| 13:29.38 | [TK]D-Fender | tthat is a bad idea |
| 13:29.53 | [TK]D-Fender | You should NOTT be using "insecure" for phone devices. |
| 13:29.55 | [TK]D-Fender | Those send auth |
| 13:30.07 | [TK]D-Fender | providers often send un-authed calls, but a phone device you have should not |
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| 13:35.07 | TECFALL | so insecure=port,invite is correct? |
| 13:37.18 | [TK]D-Fender | No, you should NOT be using incesure at all for a phone device |
| 13:38.12 | TECFALL | so what is the default insecure if you don't define it? |
| 13:38.55 | [TK]D-Fender | "no" |
| 13:39.01 | [TK]D-Fender | which is what it should be |
| 13:39.33 | TECFALL | right: but that is what is causing be to receive the errors I described in the paste. |
| 13:40.39 | [TK]D-Fender | You aren't showing the call |
| 13:40.44 | [TK]D-Fender | Call from '' (192.168.113.103:49156) to extension '248' rejected because extension not found in context |
| 13:40.44 | [TK]D-Fender | <PROTECTED> |
| 13:40.48 | [TK]D-Fender | and that is a DIALPLAN error |
| 13:41.10 | [TK]D-Fender | Except that the context looks off. |
| 13:41.20 | [TK]D-Fender | Which leads to believe that you are allowing un-authed guest calls |
| 13:41.26 | [TK]D-Fender | but lets see the actual call to back it up |
| 13:41.46 | [TK]D-Fender | username=249 <- wrong, should be authuser, not username |
| 13:41.56 | [TK]D-Fender | insecure=invite,port ; WORKS <- NO |
| 13:42.36 | [TK]D-Fender | Next, are these 2 devices behind a NAT away from your server? |
| 13:42.43 | [TK]D-Fender | behind the SAME NAT that is |
| 13:42.54 | TECFALL | no, they are both on the same server in the same location. |
| 13:43.09 | [TK]D-Fender | clarify that |
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| 13:43.13 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:43.40 | [TK]D-Fender | how are they ON a server? Are these multiple soft-phone RUNNING on the actual server? |
| 13:44.25 | TECFALL | sorry, no they are two desk phones in the same location on the same pbx server |
| 13:45.51 | [TK]D-Fender | Show the actual call. |
| 13:45.57 | [TK]D-Fender | and "sip show peers" |
| 13:47.11 | TECFALL | http://pastebin.com/cqqWKxWp |
| 13:48.26 | TECFALL | and i did change username to authuser |
| 13:49.56 | [TK]D-Fender | remove the port= |
| 13:50.10 | [TK]D-Fender | directmedia=no |
| 13:50.23 | [TK]D-Fender | and change type=friend |
| 13:51.34 | TECFALL | changing type to friend works |
| 13:51.42 | TECFALL | but i thought that was frowned upon |
| 13:52.40 | [TK]D-Fender | Depends on circumstance. If you're keeping them local then use permit/deny to lock them into your local subnet |
| 13:53.08 | TECFALL | Well, these will be calling to different remote office locations as well |
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| 13:53.27 | TECFALL | which is the next problem i am having |
| 13:53.51 | [TK]D-Fender | friend will be fine. |
| 13:54.12 | [TK]D-Fender | make sure to be properly monitoring hack attempts and firewall yourself accordingly |
| 13:54.49 | TECFALL | which i am. |
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| 14:00.56 | TECFALL | Here is the problem I am having with calls between remote locations: http://pastebin.com/MYEvcNZw |
| 14:05.37 | [TK]D-Fender | more regular peer mistakes |
| 14:06.55 | TECFALL | ?? |
| 14:07.12 | [TK]D-Fender | http://pastebin.com/dXNjj2KQ |
| 14:07.20 | [TK]D-Fender | use that as the basis for your inter-PBX peers |
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| 14:31.35 | TECFALL | Does the fromuser and defaultuser have to match on both servers? |
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| 14:34.42 | TECFALL | or would i swap them for each server |
| 14:35.17 | [TK]D-Fender | they ahve to match obviously |
| 14:35.39 | [TK]D-Fender | You can't have one side say "I'm Fred" and the other side have a peer saying the user is "John" |
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| 14:37.23 | TECFALL | what is the difference between fromuser and defaultuser? |
| 14:38.05 | [TK]D-Fender | One changes the From: |
| 14:38.46 | [TK]D-Fender | because when you don't set it... it'll take the current CallerID, typically carried inbound From: |
| 14:42.32 | TECFALL | It just seems weird to have both fromusers be the same |
| 14:42.47 | TECFALL | or is it just the defautluser that needs to match for auth? |
| 14:43.52 | [TK]D-Fender | the names ahve to match |
| 14:43.57 | [TK]D-Fender | you use the from calling BOTH WAYS |
| 14:44.10 | [TK]D-Fender | You can have one suide saying they are JOHN and the other expecting MIKE |
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| 14:52.42 | TECFALL | Ok, so what is with this: http://pastebin.com/by8qKKwQ |
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| 14:54.54 | [TK]D-Fender | Add up username as well then |
| 15:01.26 | TECFALL | added username=trusted to both peers and still get the same error |
| 15:01.54 | TECFALL | or did you mean to add that to the extension peers? |
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| 15:06.24 | [TK]D-Fender | <PROTECTED> |
| 15:06.28 | [TK]D-Fender | Shouldn't have that name |
| 15:06.32 | [TK]D-Fender | show the call |
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| 15:10.17 | TECFALL | http://pastebin.com/kY561M6y |
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| 15:25.21 | TECFALL | Ok, so if I change username=omaha2, fromuser=omaha2 on both peers I can dial from omaha to texas but not the other way. If I change username=texas, fromuser=texas on both peers i can dial from texas to omaha but not the other way. |
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| 15:50.21 | TECFALL | is there a way to show the caller id of the called extension on the phone that is initiating the dial |
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| 15:58.28 | rmudgett | TECFALL: You mean connected line? |
| 16:02.30 | TECFALL | rmudgett: yes |
| 16:02.52 | TECFALL | like once you are connected to the call, your phone would show the caller id that you are connected to. |
| 16:03.15 | rmudgett | Yes. That is the connected line feature. |
| 16:03.24 | TECFALL | where is that at? |
| 16:04.14 | TECFALL | ah, it is part of the dialplan. |
| 16:29.19 | Kalavera | [TK]D-Fender: is there a way to onli run core set verbose for a specific extention ? |
| 16:29.30 | [TK]D-Fender | no |
| 16:32.39 | TECFALL | http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE I can't get this working on my cisco 7942. What exactly do they mean on note for the cisco phones by using progress() before dial? |
| 16:33.38 | [TK]D-Fender | they mean exactly what they say |
| 16:33.44 | [TK]D-Fender | Want X? Do Y. |
| 16:34.28 | TECFALL | interesting. i can see the caller id getting set in the console, but it isn't displaying on the phone |
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| 16:36.01 | [TK]D-Fender | You still need tho RPID options on those peers the same as you have for the inter-PBX ones I gave you |
| 16:36.05 | [TK]D-Fender | sendrpid & trustrpid |
| 16:54.53 | TECFALL | That was it. thanks |
| 16:57.52 | TECFALL | on to the next problem. if I call a phone, and hang up after a few rings the call doesn't terminate on the phone that was called. it shows the line ringing and you can't end the call. http://pastebin.com/Jafymmi1 |
| 17:04.47 | Samot | You're going to need to turn on AGI debugging. |
| 17:05.10 | Samot | Pretty much most of anything that happened during that call is happening during your AGI calls. |
| 17:05.21 | Samot | None of that is written into the log. |
| 17:05.30 | Samot | Or seen by -vvvvvvvvvvv |
| 17:06.17 | Samot | But this is probably something to look at |
| 17:06.22 | Samot | [Jul 8 11:51:00] WARNING[20534]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission a8b1d4fb-76150008-09a9f8b9-48d04b0e@192.168.113.103 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 17:06.22 | Samot | Packet timed out after 6399ms with no response |
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| 18:47.55 | TECFALL | I still can't figure out why my phones are recognizing hangups. This only hapens if the call isn't answered and you hang up. The call on the phone that was called never hangs up. http://pastie.org/private/samrlfxiw8ajcxuegcf1g |
| 18:49.52 | [TK]D-Fender | not sure if I'm recalling correctly, give "nat=never" a try on those phones |
| 18:49.58 | [TK]D-Fender | Not sure if this was related |
| 18:54.51 | TECFALL | still the same thing |
| 18:56.41 | TECFALL | call from phone1 to phone2. Let it ring a few times and without ever picking up phone2 I end the call on phone1 and the call on phone2 just keeps showing that there is a call. However, phone2 doesn't keep ringing, it just still shows the line ringing |
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| 20:21.24 | talntid | i have a MOH class, and inside the folder I have a single .gsm file. when I set the musicclass and do moh, it appears to pull from the right dir, but there is no actual sound |
| 20:21.33 | talntid | no errors in the console |
| 20:21.34 | talntid | ideas? |
| 20:21.48 | [TK]D-Fender | Maybe you just have an audio issue |
| 20:21.55 | talntid | if i do a playuback() on the same file, it works |
| 20:21.59 | [TK]D-Fender | and you you should probably be showing the complete attempt |
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| 20:32.21 | talntid | [TK]D-Fender, http://pastebin.com/vv3N9SMh |
| 20:36.10 | [TK]D-Fender | Answer() first and ensure the audio path is up |
| 20:36.19 | [TK]D-Fender | (playback of silence, etc |
| 20:36.23 | [TK]D-Fender | heads home |
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| 21:24.48 | talntid | [TK]D-Fender, answer() and playback silence before worked. |
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| 22:38.26 | apb1963 | so I guess I must be the last person still using google voice/xmpp |
| 22:38.41 | apb1963 | :/ |
| 22:38.54 | apb1963 | or rather... not using it. |
| 22:39.04 | apb1963 | since it's been down for a few days at leaste |
| 22:39.12 | apb1963 | s/e// |
| 22:39.26 | apb1963 | s/sinc/since |
| 22:39.34 | apb1963 | hmm |
| 22:39.43 | apb1963 | !! |
| 22:52.12 | Kalavera | [TK]D-Fender: I have this issue at the moment , when I call to cellphones it is threatening the call this way |
| 22:52.21 | Kalavera | [TK]D-Fender: Executing [993374853@nivel-15:8] Dial("SIP/1557-0004fa2e", "SIP/optical2/9993374853") in new stac |
| 22:52.40 | Kalavera | in extensions.con I have this line for the dial : |
| 22:53.17 | Kalavera | [TK]D-Fender: exten => _9xxxxxxxxx,n,Dial(SIP/optical2/${EXTEN}) |
| 22:53.48 | Kalavera | [TK]D-Fender: and the message I receive from the softphone is no route to host |
| 22:59.39 | talntid | another issue - in my queues.conf, in my queue, I have a context=billing |
| 23:00.10 | talntid | then in dialplan, I have a [billing] context, with a single digit extension - asterisk doesn't react at all when I press a single button though - no errors. |
| 23:00.19 | talntid | if I do * to hang up, it hangs up though, when configured to do so |
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| 23:35.55 | [TK]D-Fender | kavShow the actual call |
| 23:36.24 | [TK]D-Fender | Kalavera, ^ |
| 23:36.34 | [TK]D-Fender | ~pb |
| 23:36.34 | 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. |
| 23:36.35 | [TK]D-Fender | ^^^ |
| 23:47.20 | Kalavera | [TK]D-Fender: Its calling now for landlines , cellphones and provinces , had to fit the _xxxxxx in the extentions.conf |
| 23:47.27 | Kalavera | [TK]D-Fender: Thank you again |
| 23:47.51 | Kalavera | [TK]D-Fender: Sorry for the delay was in a call |