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11:14.35 | allizom | Hi all, I'm trying to configure asterisk to receive calls from my provider, but I can't still grasp how do I let calls from any number in. The error I receive is "No matching peer for '0039PHONENUMBERCALLINGME' from '213.205.21.8:5060'" and here are config and logs: https://paste.centos.org/view/raw/ac3072c8 |
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13:56.52 | Kobaz | allizom: you haven't configured an endpoint that matches either the username or the ip address that it's coming from |
13:58.02 | Kobaz | if the carrier is sending From: <PhoneNumber> in the sip headers, then you'll want to do an ip match |
13:58.27 | Kobaz | allizom: dpaste your pjsip set logger on, output |
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14:05.39 | Samot | Kobaz: How will the pjsip logger help in all this? |
14:07.06 | Kobaz | You'll see exactly how the call is coming in, so you can configure the endpoint to match |
14:07.29 | Kobaz | we can assume certain things, but it's good to know for sure |
14:07.51 | Kobaz | oh, that might be a chan_sip error |
14:07.56 | Kobaz | i didn't look at the logs yet actually |
14:08.03 | Samot | Riiight |
14:08.09 | Kobaz | sorry, haha, it's morning |
14:08.12 | Kobaz | he's got all the logs in there |
14:08.18 | Samot | Sure. |
14:08.21 | Kobaz | allizom: never mind, i didn't have coffee yet |
14:10.03 | Kobaz | allizom: insecure=invite .. your ITSP is not sending you any authentication and you want to match based on the ip. and you have a possible issue.. dns lookup ims.tiscali.net results in an empty response... are you using private dns? |
14:10.06 | allizom | Kobaz, Samot: thanks, I dug into my issue and with this configuration I've been able to receive calls: https://paste.centos.org/view/raw/1ba52f84 - I still can't place them probably because my To: header is not set to the requested one as you can see in the log: https://paste.centos.org/view/raw/ff222b6e - Yes, it's chan_sip due to my provider using tel uris |
14:10.47 | allizom | Kobaz: ims.tiscali.net is not resolvable via A record - but nevermind that |
14:12.03 | Samot | What does tel uris have to do with chan_sip? |
14:12.24 | allizom | Samot: AFAICT pjsip does not handle tel uris |
14:13.06 | allizom | at least in the snowflake version used by my provider |
14:13.14 | Samot | Uhm.. |
14:14.43 | Samot | Your provider doesnt care about chan_sip vs chan_pjsip |
14:14.44 | Kobaz | allizom: I see the cancel... so your carrier is stopping the call after proceeding |
14:15.03 | Kobaz | session progress and then cancel |
14:15.18 | allizom | Samot: they don't, it's just that if I use pjsip I can't handle tel uris they use/require |
14:15.51 | allizom | Kobaz: yes, they probably want my To: header to contain ims.tiscali.net |
14:15.52 | Samot | They use it in the from header |
14:16.19 | Samot | Not in the request header where asterisk gets the DID from |
14:16.36 | Kobaz | allizom: well... asterisk is sending the cancel |
14:16.49 | Kobaz | Reliably Transmitting (no NAT) to 213.205.21.8:5060: CANCEL sip:PHONENUMBERIMCALLING@213.205.21.8 SIP/2.0 |
14:17.53 | Samot | Cancel came from grandstreAm |
14:18.10 | Kobaz | i didn't get that far yet, but yeah it's passing it through |
14:18.11 | Samot | Then from asterisk to the provider |
14:18.24 | Samot | So the call was hung up |
14:18.26 | file | outgoing call to provider was placed, the dialed number was not found (and is likely being said in session progress), caller hears this and hangs up |
14:18.39 | allizom | file: correct |
14:18.46 | file | bows |
14:19.10 | Kobaz | do you have an audio recording? |
14:19.20 | Kobaz | is it saying 'call cannot be completed as dialed' or some such |
14:19.42 | Kobaz | and if so, contact your carrier and ask, umm, why? |
14:19.43 | allizom | it said something like "the dialed number does not exist" |
14:20.02 | Kobaz | I've had this problem with verizon. They are VERY picky. You need to pass P-Asserted-Identity with your BTN in order to make calls |
14:20.18 | Kobaz | they will send you early media that the dialed number cannot go through, otherwise |
14:21.05 | Samot | Or perhaps they want calk |
14:21.18 | Samot | Callerid not in the from header |
14:21.47 | Kobaz | allizom: do you have a username credential for your carrier? |
14:22.06 | Kobaz | MYPHONENUMBER@ims.tiscali.net is that your username you're supposed to use? |
14:22.14 | allizom | Kobaz: it is identical to my full phone number |
14:22.21 | allizom | yes |
14:22.28 | Kobaz | is your username phonenumber... or phonenumber@...the whole thing |
14:23.33 | Kobaz | you'll want fromuser=MYPHONENUMBER@ims.tiscali.net assuming you need that for your username |
14:23.49 | Kobaz | and then probably, sendrpid=yes to pass callerid |
14:23.57 | allizom | let me try this |
14:25.37 | Samot | Or PAI. |
14:25.43 | Samot | That is a question for the ITSP. |
14:25.58 | Samot | What format do they want CallerID in? |
14:26.12 | Samot | Don't assume RPID, a failed RFC, is what is being used. |
14:26.42 | Kobaz | many, many systems support this, it's pretty common that it will work |
14:27.13 | Kobaz | in my experience PAI actually works less often, but yeah, either one is a consideration |
14:27.41 | Samot | Or perhaps they want "P-Called-Party-ID" which is what they use to send incoming callerid |
14:27.49 | Samot | In your experience? |
14:27.56 | Samot | PAI is an RFC standard |
14:28.04 | Samot | RPID is a failed RFC |
14:28.27 | Kobaz | really depends on the carrier |
14:28.33 | Samot | Exactly |
14:28.33 | Kobaz | or the device you're talking to |
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14:28.45 | Samot | Hence my statement of "This is a question for the ITSP" |
14:28.49 | Kobaz | sure |
14:28.53 | allizom | How do I set To: header to be <sip:PHONENUMBERIMCALLING@ims.tiscali.net> rather than <sip:PHONENUMBERIMCALLING@213.205.21.8> ? |
14:29.59 | Samot | You set the host as that |
14:30.17 | Kobaz | https://dpaste.com/8CGZR9A7P |
14:30.24 | Samot | It's using the host setting to set the 213.205.21.8 |
14:30.30 | Kobaz | that too |
14:31.05 | Kobaz | Samot: but his problem, is that ims.tiscali.net doesn't resolve |
14:31.11 | Samot | No. |
14:31.22 | Samot | ims.tiscali.net is probably a SRV record. |
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14:32.00 | Kobaz | not sure, it doesn't resolve any which way for me |
14:33.49 | Samot | allizom: What has the ITSP said? |
14:33.55 | allizom | "host -t SRV ims.tiscali.net" from my network gives "ims.tiscali.net has no SRV record" |
14:34.15 | Kobaz | why is your carrier giving you a dns with no records? |
14:34.41 | Samot | Most likely for routing/account purposes. |
14:34.56 | Samot | allizom: What has the ITSP said? |
14:35.09 | Samot | You've contacted them about not being able to make outbound calls? |
14:35.25 | Samot | They've looked at an attempt and told you "X, Y, Z is incorrect" |
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14:35.48 | allizom | they provided this one: core1.p.ims.tiscali.net which resolves to that IP address but is not identical to ims.tiscali.net |
14:35.58 | Samot | Sore? |
14:35.59 | Samot | So? |
14:36.04 | Samot | Use what they provide you |
14:36.04 | Kobaz | they provided that to do what with? |
14:36.14 | allizom | Samot: I can make calls without using asterisk, their service sorta works |
14:36.18 | Kobaz | Use as the proxy? the From: header? the what? |
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14:36.43 | allizom | let me read again the info they provided me |
14:36.45 | Kobaz | If their service soft of works with a soft phone, why would you configure asterisk without asking the carrier, "why does this kind of work?" |
14:37.31 | Samot | My guess, that's the OB Proxy |
14:38.17 | allizom | username: MYPHONENUMBER@ims.tiscali.net, outbound proxy ip address: 213.205.21.8 |
14:38.40 | allizom | domain/registrar: ims.tiscali.net |
14:38.42 | Samot | So that would mean fromuser=MYPHONENUMBER |
14:38.53 | Samot | fromdomain=ims.tiscali.net |
14:39.03 | Samot | outboundproxy=213.205.21.8 |
14:39.16 | Samot | So where does this core1.p.tiscali.net come into play? |
14:39.22 | Samot | What did they give that to you for? |
14:39.51 | allizom | it's just an alternate way they told me to find their outbound proxy |
14:39.57 | Samot | OK |
14:39.58 | allizom | they specify both |
14:40.00 | Kobaz | alternate way? |
14:40.10 | Kobaz | sounds like the only way? if the other domain has no dns records |
14:40.27 | allizom | core1.p.tiscali.net resolves to 213.205.21.8 |
14:40.36 | allizom | so just use the name or the ip address directly |
14:40.50 | Samot | So you lack the outboundproxy setting |
14:40.55 | Kobaz | Names are good, in case they decide to change this own the line |
14:41.16 | allizom | Samot: I'm adding it |
14:41.36 | allizom | with host=ims.tiscali.net |
14:44.41 | allizom | Samot: no luck. with host=ims.tiscali.net and outboundproxy=213.205.21.8 and fromuser=00390924507049 I'm back to not even being able to receive calls |
14:44.55 | Samot | Heh I figured. |
14:45.01 | allizom | sip show peers: Name/username Host Dyn Forcerport Comedia ACL Port Status Description |
14:45.01 | allizom | itsp (Unspecified) |
14:45.09 | Kobaz | right |
14:45.13 | Samot | You need to have two peers most likely |
14:45.20 | Samot | Are you in Germany? |
14:45.29 | allizom | nope, IT |
14:45.35 | Samot | Close enough. |
14:45.40 | Kobaz | heh |
14:45.45 | Samot | Those EU countries love themselves IMS |
14:46.04 | Samot | And using the old school incoming and outgoing users. |
14:46.19 | Samot | Basically incoming calls are going to match based on the host= setting |
14:46.35 | Samot | So whatever is in the host= setting needs to resolve to an IP if it is a FQDN |
14:46.55 | Samot | It's also the setting used for the domain part of Request/To headers. |
14:47.11 | Samot | This should be a Chan_PJSIP trunk to avoid all this BS. |
14:48.35 | allizom | Samot: I've tried and also read in the bug tracker, that I can't use that unfortunately due as said to those tel uris my provider uses |
14:48.45 | Kobaz | that doesn't make sense |
14:48.50 | Kobaz | why would chan_sip work any better |
14:49.02 | Kobaz | it has way less flexability with regards to that |
14:49.19 | Kobaz | can you link one of the bugs? |
14:49.25 | allizom | yes, one minute |
14:49.29 | file | chan_pjsip doesn't support Tel URIs as noone has put in the time to audit and add all the support for it |
14:49.41 | file | URIs in PJSIP are parsed into separate structures and you have to explicitly treat and use them differently |
14:49.47 | Kobaz | Oh, good to know. Never had to deal with tel: |
14:50.07 | file | treat it as a SIP URI and kaboom, you crash |
14:50.34 | Kobaz | Well okay then! hah |
14:50.38 | Kobaz | chan_sip it is |
14:51.33 | allizom | so my question stands, I need to use host=213.205.21.8 *and* to send To: headers with ims.tiscali.net, is this possible some way? |
14:52.03 | Samot | 9:45:13 AM <Samot> You need to have two peers most likely |
14:52.07 | Kobaz | https://dpaste.com/8CGZR9A7P |
14:52.21 | Kobaz | And to alter the To: or the From: header, you can additionally append the following to any of the above strings: |
14:52.32 | Kobaz | set up your peers with proper ips |
14:52.38 | Kobaz | and then modify it run-time |
14:53.30 | allizom | Kobaz: I'm currently calling it this way: Dial(SIP/${EXTEN}@itsp) |
14:53.41 | allizom | so, let me see.. |
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14:54.28 | allizom | it should be the second case: SIP/username@domain (SIP uri) |
14:55.25 | allizom | can I specify todomain without touser? I guess I should write touser anyway, with a variable |
14:58.57 | allizom | I'm going to try with Dial(SIP/${EXTEN}@itsp!${EXTEN}@ims.tiscali.net) |
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15:05.47 | allizom | Samot: while it did in fact send the To: header I intended, I still can't place calls. How should I configure those two peers? |
15:06.05 | Kobaz | show your new debugs? |
15:07.50 | allizom | of course, let me save them |
15:13.57 | allizom | here it is: https://paste.centos.org/view/raw/08e686bb |
15:16.36 | Samot | OK |
15:16.40 | Samot | I'm going to ask this again |
15:16.52 | Samot | Have you asked the ITSP why they are rejecting your calls? |
15:17.28 | Kobaz | Yeah, now's definitely the time to do that |
15:17.56 | Kobaz | because the headers look fine now (according to what I believe you should be sending (based on your limited information from your carrier)) |
15:18.32 | allizom | nope, I can try and hope for the best, but I really think I'll be unable to talk with someone knowledgeable enough. I'll do that though |
15:19.27 | Samot | GD. |
15:19.34 | Samot | Find a better provider then. |
15:19.58 | allizom | Thank you. This is what I can get right now unfortunately |
15:20.05 | Kobaz | Yeah, if your carrier can't answer a basic question on 'why did this fail?', then move on |
15:20.12 | allizom | But I will talk to them |
15:20.18 | Samot | Sorry but I'm just sick of that excuse. |
15:20.20 | Samot | Or "reason" |
15:20.24 | allizom | If all else fails, I'll take a log from my ATA |
15:20.29 | allizom | and compare the two |
15:20.35 | Samot | Everyone seems to always have a provider that is worse at this then them. |
15:20.49 | Kobaz | Samot: haha true |
15:21.05 | Kobaz | allizom: the ata is working? |
15:21.24 | allizom | yes, if I connect directly with my provider credentials |
15:21.29 | Kobaz | sdkfjhasdfhjaskhfasdf |
15:21.34 | Kobaz | and you waited until now to share this? |
15:22.04 | allizom | well, yes? |
15:22.05 | Kobaz | You realize there is 100% less guessing involved if you have a working configuration to go against |
15:22.33 | allizom | that's why I said I want to compare the two |
15:22.38 | Kobaz | The last half hour was a complete waste then |
15:22.44 | allizom | oh, sorry |
15:22.48 | Kobaz | It's all just guessing at what's supposed to work |
15:23.08 | allizom | I see |
15:23.10 | Kobaz | Knowing for sure, is the way to go |
15:23.23 | Kobaz | "This works" Lets replicate this to another system |
15:23.27 | Kobaz | versus starting from scratch |
15:24.03 | Kobaz | So yeah. show a full call via the ATA, and then that'll be the roadmap on what to have asterisk send |
15:24.51 | allizom | it's unfortunately going to take a while, but I'm going to do exactly that |
15:24.59 | Kobaz | Yeah, that's the best way |
15:25.06 | Kobaz | Then you know what it's supposed to look like |
15:25.17 | Kobaz | That's literally my motto at the office. "We need to know 'for sure'" |
15:25.30 | Kobaz | No guessing, or, as little as possible |
15:26.25 | Samot | ð¿ |
15:27.05 | Samot | This is awesome. |
15:27.27 | allizom | oh, actually I told you: 15:36:14 - allizom: Samot: I can make calls without using asterisk, their service sorta works |
15:27.36 | Samot | Kobaz: You literally joined this conversation with assumptions and guesses before even looking at the provided data. |
15:27.40 | Kobaz | allizom: you didn't provide many details |
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15:27.56 | seanbright | Samot: STOP ARGUING WITH KOBAZ |
15:28.00 | Kobaz | haha |
15:28.03 | seanbright | this is your SECOND WARNING |
15:28.14 | Kobaz | Samot: My brain skipped over the url, that's not my normal modus |
15:28.40 | seanbright | is just kidding, hopefully that is obvious |
15:28.42 | Kobaz | Samot: as little guessing as possible... and... based on what information I had... well.. that was the only start |
15:29.59 | Kobaz | allizom: and i asked in response: .... [ask] the carrier, "why does this kind of work?" |
15:30.09 | Samot | Kinda work? |
15:30.12 | Samot | OK. |
15:30.16 | Samot | ATAs, softphones... |
15:30.21 | Kobaz | 'sorta works' 'kind of works' |
15:30.24 | allizom | was not allocating blame :P |
15:30.24 | Samot | They kinda have a set purpose.. |
15:30.27 | Kobaz | that's when you ask the carrier |
15:30.38 | Samot | They have nice fields in their GUI for you to put details in. |
15:30.48 | Samot | Asterisk is a telephony kit. |
15:31.01 | Kobaz | What doesn't work about it? |
15:31.10 | Samot | Kinda got to do all those configs that the ATAs and softphones already have in place. |
15:31.18 | Samot | So when it works on an ATA vs Asterisk... |
15:31.27 | Kobaz | allizom: What's the 'sorta' part? |
15:31.34 | Samot | It either means Asterisk is configured wrong or they are blocking PBXes on the service. |
15:31.52 | Kobaz | You can always set the user agent |
15:31.53 | allizom | that it uses tel uris in a particular, nonstandard way |
15:32.21 | Kobaz | Hi, I'm a grandstream |
15:32.26 | Kobaz | Hi, I'm a mac |
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15:42.01 | allizom1 | This is the log of a successful call I placed: https://paste.centos.org/view/raw/a124191a |
15:44.06 | Kobaz | that looks like wireshark |
15:44.16 | allizom | yes, sorry it's rather unwieldy |
15:44.21 | Kobaz | can you upload the pcap instead |
15:44.34 | Kobaz | It's a little easier to see when it's unformatted |
15:44.57 | allizom | I know, I just don't have a place to upload it to |
15:44.57 | Kobaz | because then you see the actual headers as they are, literally |
15:45.21 | allizom | maybe there's a way to just extract the sip part |
15:45.25 | Samot | That was made straight from the Grandstream? |
15:45.28 | Kobaz | https://gofile.io/ or such |
15:45.36 | Samot | Where was this capture done at? |
15:45.56 | allizom | it's the capture from the ATA taken from my home router |
15:46.22 | Samot | So again |
15:46.28 | Samot | Either Asterisk is configured wrong |
15:46.32 | allizom | I'm trying to upload it somewhere |
15:46.36 | Samot | Or they are blocking the use of Asterisk |
15:46.50 | Samot | Call. The. ITSP. |
15:46.53 | Samot | JFC. |
15:47.12 | allizom | ok, I will, as I said before |
15:47.19 | Samot | I know, i know. |
15:47.23 | Kobaz | The main things I get out of this: P-Preferred-Identity and <number>@ims.tiscali.net |
15:47.26 | Samot | They know less than you. |
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15:47.57 | allizom | shrugs |
15:48.21 | Samot | Again, either Asterisk is configured wrong and they need to tell you what to correct. |
15:48.36 | Samot | Or they are blocking the use of it. |
15:48.49 | Kobaz | well you can figure out what needs to be corrected based on this dump vs one that's not working |
15:48.56 | Kobaz | including the user agent |
15:49.07 | Kobaz | if that's causing a block or whatever |
15:49.49 | allizom | Kobaz: I guess if they were actively blocking that I would not be able to even receive calls, but. Still a guess |
15:50.22 | allizom | call them and figure out someway. That's fine |
15:50.28 | allizom | thank you all |
15:52.36 | Kobaz | Now you have some tools to figure this out, and... how to get better help in the future |
15:52.40 | Kobaz | You're welcome |
15:52.49 | allizom | Kobaz: point taken |
15:57.32 | Kobaz | here's what you can do |
15:57.45 | Kobaz | allizom: take the pcap from the ata |
15:57.58 | Kobaz | use tcpdump to filter out only the ITSP traffic.. use host xxxx, etc |
15:58.22 | Kobaz | tcpdump -r <pcapfile> host 213.205.21.8 -A -s0 > /tmp/working |
15:58.29 | Kobaz | take a pcap on the asterisk box |
15:58.37 | Kobaz | tcpdump -r <pcapfile> host 213.205.21.8 -A -s0 > /tmp/notworking |
15:59.00 | Kobaz | and throw them in a diff tool. Line everything up in terms of the INVITEs and see what needs to be changed |
16:06.06 | Samot | Or call support |
16:06.59 | Kobaz | *and call support |
16:07.01 | Kobaz | and be like wtf |
16:07.40 | allizom | I'll do both. This is one: https://paste.centos.org/view/raw/2e54f539 and tomorrow (today it's holiday) I'll call them |
16:08.07 | seanbright | rport |
16:08.42 | seanbright | that's the first difference that jumps out at me |
16:09.06 | seanbright | actually the request URI is different too |
16:09.19 | Kobaz | and Route: is missing on the 'not working' |
16:09.22 | Samot | The easy fix |
16:10.00 | Samot | Add the domain and IP to /etc/hosts |
16:10.27 | Samot | So it will resolve locally to the right IP |
16:10.30 | Kobaz | working as P-Preferred-Identity, notworking, does not |
16:10.45 | Samot | The ATA isnt trying to resolve it |
16:11.08 | Samot | Its resolving the ob procy |
16:11.13 | Samot | Proxy |
16:11.15 | Kobaz | allizom: sort the fields alphabetically in each INVITE. and go line by line, and get the appropriate setting in asterisk |
16:11.51 | Samot | Again |
16:12.10 | Kobaz | And if that doesn't work, definitely something going on with the carrier |
16:12.15 | Samot | The ATA is not trying to resolve the host domain |
16:12.20 | Samot | Asterisk is |
16:12.26 | Samot | No |
16:12.31 | Samot | Jfc |
16:12.41 | seanbright | heh |
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16:14.09 | seanbright | Kobaz: Samot: we're going to play roshambo to see who helps allizom: please PM your choice of one of the following: rock, paper, scissors |
16:14.23 | Samot | Naw |
16:14.29 | Samot | Its all Kobaz |
16:14.39 | seanbright | don't give up |
16:15.07 | allizom | easy people, I can try on my own :) |
16:15.08 | Samot | Meh. Im in negative fucks |
16:15.09 | Kobaz | It definitely can work. Samot's all trying to throw in the towel so soon |
16:15.22 | Samot | Yeah that is it |
16:15.25 | seanbright | Kobaz sent me 'Apricots' via PM so instructions must have been unclear |
16:15.36 | Kobaz | haha |
16:16.08 | Kobaz | I've missed you guys |
16:16.23 | Kobaz | I literally took like a 4 year hiatus from active participation in #asterisk |
16:16.36 | Kobaz | Been so friggin busy |
16:17.56 | allizom | Samot: if I add "213.205.21.8 ims.tiscali.net" to /etc/hosts do I need outboundproxy= and fromdomain= anymore ? |
16:18.10 | Samot | Yes you do |
16:18.14 | allizom | ok |
16:18.26 | Samot | That just solves dns resolution |
16:18.49 | Samot | Because Asterisk is doing a lookup on the domain |
16:18.57 | Samot | That doesnt have a record |
16:19.13 | Samot | The grandstream is not |
16:19.24 | Samot | Its using the ob proxy |
16:20.29 | allizom | Samot: that was it! I've succesfully placed a call right now |
16:20.45 | allizom | oh goodness |
16:20.45 | Kobaz | yay |
16:20.51 | seanbright | nice. i knew Kobaz would get it resolved... |
16:21.05 | Kobaz | And... you guys... I did point this out 25 minutes ago, that the domain had no dns |
16:21.12 | Samot | Its like ATAs were configured a certain way |
16:21.20 | Samot | I know |
16:21.51 | Samot | Im explaining why the ATA worked and Asterisk did not |
16:22.05 | Samot | Asterisk wasnt configured right |
16:22.07 | Kobaz | seanbright: the propos has to be to allizom for sticking around so long |
16:22.52 | allizom | I can be stubborn if I want to |
16:22.57 | allizom | which is not always a pro |
16:23.28 | Samot | I dont allow pure IP requests either |
16:23.48 | Kobaz | right, avoids a lot of crap calls coming in |
16:24.04 | Samot | And that would have been the answer if you called my support |
16:24.15 | Samot | And you were my user |
16:24.28 | Kobaz | but you're off on holiday today |
16:24.36 | Samot | No im not |
16:24.42 | Kobaz | i'm just saying, if you were the carrier |
16:24.49 | Samot | Im on call 24/7/365 |
16:24.58 | Kobaz | you completely missed it |
16:25.01 | Samot | Im a carrier |
16:25.08 | Samot | I have holiday support |
16:25.16 | Kobaz | [2021-01-06 11:07:40] <allizom> ..... and tomorrow (today it's holiday) I'll call them |
16:25.24 | Samot | I know |
16:25.36 | Kobaz | I'm giving you a hard time, you know |
16:25.38 | seanbright | so friday is my birthday |
16:25.42 | Samot | So the carrier lacks holiday support? |
16:25.46 | Kobaz | Happy astribirthday |
16:26.00 | Kobaz | Yeah that's crazy, what carrier is actually closed... ever? |
16:26.07 | seanbright | i hope that you've sent gifts via UPS/FedEx as USPS has been pretty slow |
16:26.34 | Kobaz | BTC can make it over the same day |
16:26.45 | seanbright | sure, i'll take 1 |
16:27.04 | Kobaz | i just about trippled my money lately |
16:27.38 | Kobaz | Need to set a good sell trigger when the impending crash occurs |
16:30.48 | Kobaz | allizom: how's your tel: uris working |
16:31.29 | allizom | Kobaz: they send it on inbound calls |
16:31.37 | allizom | and you still need to handle that |
16:31.46 | Kobaz | I need to handle that? |
16:32.05 | allizom | yes, 'you' |
16:32.08 | Kobaz | haha |
16:32.12 | Kobaz | :P |
16:32.14 | allizom | for some values of 'you' |
16:34.34 | grummund | Hi, what's the right way to replace the default voicemail greeting? |
16:35.02 | Kobaz | you can A) replace the wav/etc files in the sound path, or B) write your own voicemail |
16:35.23 | seanbright | grummund: "comedian mail, mailbox?" you mean? |
16:35.57 | grummund | replacing the file /usr/share/asterisk/sounds/en_US_f_Allison/vm-intro.wav works but seems a bit "hacky", so just checking on the right way. |
16:36.19 | Kobaz | seanbright: I just have mine say 'Mailbox?' |
16:37.26 | seanbright | https://github.com/asterisk/asterisk/blob/master/CHANGES#L1280-L1283 |
16:37.30 | seanbright | grummund: ^^ |
16:37.47 | grummund | it is presumably comedian mail, whatever is the stock VoiceMail() app. |
16:40.27 | grummund | "Please leave your message after the tone, when done hangup or press the pound key." |
16:40.39 | grummund | sorry, i mean that message ^ |
16:42.16 | seanbright | just scanning the sample voicemail.conf i don't see it |
16:43.17 | grummund | the stock file is vm-intro.wav and just replacing works, but i guess might be clobbered on a upgrade. |
16:44.37 | seanbright | yeah, looks like you may have to do that |
16:45.16 | grummund | that's fine. |
16:59.36 | grummund | just to be sure... there are no free British accent voice prompts for asterisk, right? |
17:19.54 | seanbright | i could record some for you |
17:20.05 | seanbright | g'day govna |
17:20.21 | seanbright | pop pop cherrio |
17:20.27 | seanbright | cheerio* |
17:20.40 | grummund | yeah i found some good ones like that ;) |
17:21.06 | file | seanbright: you doing a British accent... that would be... something |
17:21.41 | seanbright | put another shrimp on the barbieeeeee |
17:22.38 | grummund | https://ymstat.com/dyn/community/13012_orig.mp3 |
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19:14.21 | cloud9 | hey everyone. I have asterisk realtime setup with mysql reading the extensions table. very slick. I don't see a way to insert "include => context" into the extensions table. I don't want to have to edit the extensions.conf file each time I want to add a new context. Can you shed any light? |
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