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| 00:17.25 | stercor | @search The keeper of the bees |
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| 02:53.03 | Slade | yea i'm using mobile phones for my office currently. i might need to give up on any kind of good call quality given how secondary voice is on a phone these days. every bluetooth headset just seems to make it worse |
| 03:02.02 | Slade | tho ringcentral seems to recommend some headsets i havent tried yet |
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| 13:40.48 | M1L0 | Hi averyone! somebody speak spanish? |
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| 14:47.36 | hlok | hey when I do the hello world tutorial using linphone it tells me user busy instead of the hello world |
| 14:47.47 | hlok | how do I fix it? |
| 14:49.03 | Samot | Well since we don't know what is wrong, we really can't say how to fix it. |
| 14:49.46 | hlok | okay, well I'm using the docker version of asterisk, it is version 15 point something |
| 14:50.03 | hlok | Samot: did the hello world thing work for your out of the box? |
| 14:52.32 | hlok | https://github.com/andrius/asterisk <-- this is the one I'm using |
| 14:53.47 | file | I would highly suggest not using a Docker Asterisk until you understand Asterisk and SIP, because it adds additional complexity |
| 14:54.49 | MLC | One of the SIP providers that I am considering (Call Centric) uses round robin DNS to load balance across their servers, and their registration expires in 60 seconds, so it is not unusual for the server that my Asterisk registers with to change throughout the day. This becomes a problem when there is a call in progress and the registration changes, there is no longer any SIP traffic through the firewall to keep the UDP port open |
| 14:54.49 | MLC | to the server on which that call originated and is connected to. When the remote party hangs up the provider sends a BYE command but the UDP port is no longer active in the firewall and so the call in Asterisk does not terminate. I'm looking for ideas on how to fix this. I can increase the UDP timeout in the firewall, but unless I make it 2+ hours long, there could always be a call that lasts longer than that. Is there somethin |
| 14:54.49 | MLC | g that would send an an occasional SIP packet of some kind on a call in progress? |
| 14:56.01 | hlok | file: hmmm, okay so I gotta do an adhoc install first? |
| 14:56.01 | igcewieling | like rtpkeepalive? |
| 14:56.35 | file | that would eliminate an aspect of things, SIP embeds IP addresses into things so sticking it in Docker can complicate that due to Docker networking |
| 14:57.03 | MLC | igcewieling I am unfamiliar with that, can you elaborate? |
| 14:57.28 | hlok | file: well the ip addresses seem to be working okay, cause I am registering just fine. |
| 14:58.09 | hlok | also I hear the busy tone, and ringing tone when I call it from my other SIP phone. |
| 14:58.32 | file | if you provide the console output then it may show what is up |
| 14:58.43 | igcewieling | MLC: no, but a grep can: grep keepalive asterisk-16.3.0/configs/samples/pjsip.conf.sample |
| 14:59.06 | MLC | I found it: rtp_keepalive that seems to send RTP packets. I need to send SIP packets |
| 14:59.21 | hlok | file: how do I provide a console output? |
| 14:59.22 | MLC | the RTP ports are staying open |
| 14:59.36 | igcewieling | MLC: try qualify and qualifyfreq then |
| 14:59.47 | file | I don't know how you are supposed to enter that container and retrieve logs or access the Asterisk console |
| 15:00.19 | MLC | igcewieling: I will try, but I'm guessing that those will go to the currently registered server, not the server that is active for a given call |
| 15:00.41 | igcewieling | MLC: Perhaps your problem doesn't have a solution. |
| 15:00.52 | MLC | igcewieling: it may not |
| 15:01.05 | MLC | other than the provider making some changes |
| 15:01.13 | igcewieling | Your best bet is get rid of the external firewall. |
| 15:01.40 | MLC | I've considered that |
| 15:02.03 | igcewieling | If nothing else, turn off SIP ALG |
| 15:02.59 | igcewieling | MLC: why not register by IP and use only one of their servers? |
| 15:03.22 | MLC | Thought of that too, but the provider made it clear that was undesirable. |
| 15:05.16 | hlok | file: I know how to access the container, so are the logs in /var/logs or how do I access the asterisk console if I'm in the machine? |
| 15:05.21 | MLC | what are "session timers"? controlled by parms timers and timers_min_se. I see the doc but didn't find anything that really explains what it is. |
| 15:05.46 | igcewieling | They are another keepalive type thing, but again, it only operates on the current call. |
| 15:06.29 | file | asterisk -r opens the Asterisk remote console |
| 15:06.31 | igcewieling | When they are set, but should not be, you'll get disconnected calls at 10 or 20 mins, I don't recall which. (assuming the default session timer value) |
| 15:07.28 | MLC | default is 1800 = 30 minutes |
| 15:07.37 | MLC | does that only serve to disconnect calls? |
| 15:08.27 | hlok | file: okay I'm in it now what do i do? |
| 15:08.50 | file | you have to do a call, and then provide a link to the output so someone hear can look at it, interpret what is going on, and give you an answer |
| 15:08.52 | *** join/#asterisk hfb (~hfb@47.139.20.75) |
| 15:10.08 | hlok | it is not producing any output when I do a call, can I check from the console if I registered properly? |
| 15:11.25 | MLC | hlok: pjsip show registrations |
| 15:11.40 | MLC | holk: also asterisk -rvvv will give more console output |
| 15:12.33 | hlok | hmmm it says no objects found |
| 15:13.10 | MLC | does your pjsip.conf have a type=registration section? |
| 15:14.16 | hlok | MLC: nope |
| 15:14.43 | hlok | i just copied the hello world doc https://wiki.asterisk.org/wiki/display/AST/Hello+World |
| 15:15.18 | MLC | did you change sip.conf or pjsip.conf ? |
| 15:15.30 | hlok | I made pjsip.conf |
| 15:15.59 | MLC | some providers need a registration section, some do not. Check with your provider and see if they have a sample pjsip.conf file |
| 15:16.31 | hlok | k |
| 15:18.19 | hlok | I don't have a provider to my knowledge. |
| 15:18.46 | MLC | are you trying a call between 2 internal phones? |
| 15:22.32 | hlok | linphone says registration failed: not implemented |
| 15:22.59 | hlok | MLC: |
| 15:23.08 | hlok | MLC: yeah that's what the tutorial says |
| 15:23.39 | MLC | try pjsip show contacts and pjsip show endpoints |
| 15:29.20 | hlok | pjsip show endpoints found 1 object |
| 15:30.33 | hlok | hmmm linphone says "Could not start udp transport on port 5060, maybe this port is already used." |
| 15:30.47 | hlok | does it have to be on a different machine from the asterisk server? |
| 15:31.04 | MLC | you can't run asterisk and a soft phone on the same computer without changing the ports |
| 15:31.18 | MLC | would be easiest to put the phone on a different computer |
| 15:32.09 | hlok | MLC: ah thanks |
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| 15:43.48 | MLC | So back to my earlier question, are session timers only for disconnecting sessions, or could they also be used to keep the SIP port open with a keep alive? |
| 15:44.58 | hlok | hmmm I'm having trouble getting the ipv6 transports to show up in pjsip show transports |
| 15:45.13 | hlok | do I have to load a special module for ipv6 or anything? |
| 15:45.59 | file | no |
| 15:46.12 | igcewieling | none of my phones support ipv6 so I've never bothered working with it. |
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| 15:50.18 | hlok | shocked face |
| 15:51.25 | hlok | okay, I guess I'll put it up on one of my VPSs and try it over ipv4 |
| 15:58.09 | Samot | Why do you need another VPS? Can't you just run IPv4 on the current system? |
| 15:58.25 | Samot | Crazy but Asterisk can support _both_ IPv4 and IPv6. |
| 15:58.53 | igcewieling | hlok: none of my users use softphones. |
| 16:03.00 | hlok | weird it's not showing the udp ports to be open |
| 16:04.12 | hlok | firewall says they are open, hmmm I'll check the docker config |
| 16:04.20 | hlok | igcewieling: what do they use? |
| 16:05.34 | hlok | yeah it was the docker config |
| 16:06.31 | igcewieling | hlok: Polycom and VTech SIP hardphones. |
| 16:07.16 | igcewieling | I think we have a few people setup with softphones recently, but those are on mobile phones and so work terribly. |
| 16:08.48 | Samot | What softphone? |
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| 16:23.47 | hlok | android has builtin softphone which depending on android version doesn't work great, linphone is also a viable softphone on android |
| 16:27.01 | Samot | I was actually asking igcewieling |
| 16:28.30 | igcewieling | Samot: Zulu for FreePBX customers (I think those are desktop only) and gloCOM a softphone from our Hosted vendor., |
| 16:28.59 | Samot | Oh, so vendor specific |
| 16:29.45 | igcewieling | *nod* That way our normal techs can support it and use the vendor support for help. |
| 16:29.53 | Samot | Not sure about gloCom but honestly, Zulu gets a lot of reported problems. |
| 16:31.55 | igcewieling | *nod* I don't know if the customer is still using Zulu. |
| 16:33.45 | igcewieling | Thankfully, gloCOM is licensed with a per seat cost which makes for far fewer "installed it and they only used it once" situations. |
| 16:34.25 | Samot | Yeah, that's why I like the Bria Stretto platfrom. |
| 16:34.29 | Samot | Yeah, that's why I like the Bria Stretto platform. |
| 16:34.42 | Samot | Support, per user seats and re-usable licenses. |
| 16:43.43 | hlok | weird now it's giving me the password prompt but is not accepting the password in the config file |
| 16:52.14 | hlok | i am getting results in endpoints and contacts but it has their local ip |
| 17:28.44 | igcewieling | Do they provide end user support? |
| 17:33.01 | Samot | No, you provide end user support. |
| 17:35.58 | *** join/#asterisk gohardtech (~VOIPGuy@S01069401c297e127.ed.shawcable.net) |
| 17:36.49 | gohardtech | I know this is sort of unrelated to asterisk, but can I ask a question about an SMS gateway? |
| 17:44.03 | Samot | Ask it. |
| 17:45.34 | gohardtech | Do my DID providers need to have SMS capabilities, or can I redirect them to a gateway that strips SMS first, and forwards SIP Voice traffic? Is that even a thing? Appreciate any feedback you may have. |
| 17:47.18 | Samot | Yes, they need it. |
| 17:48.23 | gohardtech | too bad ;) thanks for the answer. Was hoping that wasn't the case. |
| 17:51.33 | hlok | okay for w/e reason it works via tcp |
| 17:52.04 | hlok | but for some reason I'm still not getting any audio output on my linphone desktop app |
| 17:52.15 | hlok | what's the default codec? |
| 17:57.35 | MLC | you specify that in your pjsip.conf with disallow and allow |
| 18:02.58 | hlok | oh so can i do allow=opus? |
| 18:03.46 | hlok | anyway I can see what the setting is from the console? |
| 18:03.53 | MLC | you can try that |
| 18:04.01 | MLC | pjsip show endpoint xxxxxx |
| 18:04.10 | MLC | will show the codecs defined |
| 18:04.20 | file | if codec was a problem the call should fail, you wouldn't get no audio |
| 18:06.59 | hlok | file: hmmm |
| 18:07.10 | file | are you still using Docker? |
| 18:09.10 | hlok | file: yep |
| 18:09.53 | hlok | it lets me place the call to extension 100 but I don't hear anything, though i think it maybe the linphone app, cause i tried with my voip.ms account can't hear anything there either |
| 18:16.43 | hlok | on the asterisk end it says it plays the hello world thing |
| 18:17.02 | file | "rtp set debug on" will show you RTP traffic that is sent/received in Asterisk |
| 18:17.03 | hlok | also it seems i can't login via my other softphone that is on my phone where I know the audio works *sighs* |
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| 21:02.08 | igcewieling | Example #384912 of how stupid people are about phones: Customer tries "testing the fax" by sending a fax to a fax number on the same PRI, causing a hairpin call. |
| 21:11.14 | wyoung | igcewieling: I wouldn't call them stupid though. |
| 21:12.54 | wyoung | Who still uses a fax these days? Are emails considered legal documents now? |
| 21:13.31 | Samot | Nope. |
| 21:14.33 | Samot | Specially with things like HIPAA. |
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| 21:20.36 | allizom | Hi, feel free to point me to a more appropriate resource as this is not strictly related to asterisk. I'm trying to configure an ATA to use the SIP account my ISP gave me. I'm not able to register, even as the settings appear to be correct and I'm using the same ones on another device. What I'd like to understand is what receiving a 'SIP/2.0 408 Request Timeout' error, along with 'User-Agent: ZTE-SBC' and 'X-ZTE-Cause: "SBC-434 |
| 21:22.48 | snuff-work | allizom: trying to use same SIP details on two different devices behind a NAT is unlikely to work well |
| 21:23.22 | allizom | snuff-work: ok, is that true even if I use different local ports? |
| 21:23.46 | snuff-work | partially depends on your router and what 'sip' fixes it tries to do |
| 21:24.48 | allizom | what do you mean by sip fixes? there's an option "SIP alg" and I disabled that |
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| 21:26.14 | snuff-work | that would be it.. someone else may be able to provide more 'experience' on how running many SIP clients behind a NAT.. i had some a long time ago but my memory is hazy |
| 21:29.00 | Samot | A 408 Request Timeout is pretty self-explanatory. |
| 21:29.12 | Samot | The request to the host timed out. |
| 21:29.25 | Samot | You're not getting a reply back from the ISP. |
| 21:29.37 | Samot | So either there is something wrong with the account at the ISP or you have a NAT issue. |
| 21:29.42 | [TK]D-Fender | Or the request never made it to them in the first place |
| 21:29.48 | Samot | Either way, it sounds like you need to call the ISP. |
| 21:29.52 | Samot | Correct. |
| 21:30.11 | seanbright | i tried calling the ISP but the request timed out |
| 21:30.15 | seanbright | zing |
| 21:30.29 | Samot | But if this ISP is providing you both Internet and Voice, then it's 100% their issue. |
| 21:30.29 | snuff-work | first check.. ensure the device has a default gateway set.. and a DNS (if u connect via dns) |
| 21:30.41 | Samot | What device? |
| 21:30.55 | allizom | well, I'm definitively receiving replies from my ISP because I can see their User agent string |
| 21:31.00 | snuff-work | runs away from most NAT issues =) |
| 21:31.04 | allizom | Samot: I'm using a Grandstream ATA |
| 21:31.25 | Samot | And where are you seeing their User Agent string? |
| 21:31.32 | snuff-work | i'd almost build tunnels than deal with NAT in a substantial fashion |
| 21:31.49 | Samot | Tunnels to where? |
| 21:31.56 | allizom | Samot: in a log on my ATA |
| 21:31.58 | Samot | The other side would have to have a destination to tunnel to... |
| 21:32.11 | seanbright | china. a tunnel to china. |
| 21:32.27 | Samot | OK, the ATA sent the REGISTER request. It got a 408 Timeout. |
| 21:32.36 | allizom | exactly |
| 21:32.45 | Samot | That means the ATA never got a provisional reply. |
| 21:33.03 | Samot | So they aren't responding to your request for some reason. |
| 21:34.00 | Samot | You should send the REGISTER, they should reply with a 401 or a 407 challenge. You'll send another REGISTER and they should reply with a 200 OK if everything is good. |
| 21:35.23 | allizom | it appears so. I'd like to make sure it's not an issue on my end (you talked about NAT) |
| 21:37.19 | Samot | Is the ISP providing you the Internet connection as well? |
| 21:37.26 | allizom | yes |
| 21:37.40 | Samot | Did they provide you with a modem/router combo or are you using your own router? |
| 21:37.48 | allizom | the latter |
| 21:38.07 | Samot | So do you have DMZ setup for your router in ISP modem router? |
| 21:38.27 | allizom | I'm not using their box at all |
| 21:38.45 | Samot | So you're doing Layer 3 in your own router? |
| 21:39.15 | Samot | And what kind of router do you have? |
| 21:39.44 | allizom | I'm using a modem router which was provided by my previous ISP, and can be configured for use with other ISPs, I reconfigured it. It's a Technicolor, one minute for the model |
| 21:40.10 | allizom | TG789VAC |
| 21:42.29 | allizom | It has fxs ports which I'm currently using with the SIP account from my current ISP |
| 21:44.56 | Samot | Well I would suggest visiting their forums or support site. |
| 21:45.11 | Samot | Because not much help is going to be here in regards to dealing with it's settings and programming. |
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| 21:46.04 | allizom | Samot: I don't want to deal with this router, I just wanted to use the ATA but so far I have not been able to |
| 21:46.20 | allizom | sure, I will try to contact them anyway |
| 21:50.39 | allizom | Samot: do you think this router could be involved with my issue? And if so why? Because of its NAT function, or because of its telephony function? |
| 21:51.22 | Samot | Could be both, I'm not sure. |
| 21:51.32 | Samot | But this could be a NAT issue. |
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| 23:13.42 | allizom | Just for the records, my issue was caused by a confusing naming of the required fields, that is username/authenticate id is *not* the same thing as uri/user id. It was not related to NAT |
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