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| 09:03.47 | Hobbes` | I see a bunch of these warnings before an incoming call via chan motif is connected 'Exceptionally long queue length queuing to Motif/blahfoo' |
| 09:04.48 | Hobbes` | the warnings stop even before my extension is called |
| 09:05.09 | Hobbes` | but once I answer I can only hear my own voice back |
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| 09:07.06 | Hobbes` | and I see a lot more udp packets coming in than those going out |
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| 11:11.22 | kemmler | I have a local asterisk machine I'm trying to open up to the internet. I can register to it on lan but can't touch the asterisk service using my WAN ip. "netstat -nap |grep 5060" shows "udp 0 0 0.0.0.0:5060 0.0.0.0:* 30645/asterisk". I know the port is open in my router because I've nmapped my wan ip and a friend has as well |
| 11:12.02 | kemmler | I also flushed my iptables and have run "iptables -A INPUT -p tcp -m tcp --dport 5060 -j ACCEPT" to no avail. |
| 11:12.31 | kemmler | I see nothing in aserisk -r with sip debug on and verbosity at 5 when trying to connect |
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| 11:40.50 | kemmler | nvm got it, router typo |
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| 14:45.45 | Penguin | How could changing the transport for a phone from udp to tcp make asterisk no longer match the peer on calls? The phone registers, but calls to asterisk from the phone end up with "no matching peer." |
| 14:47.15 | Penguin | The peer entry in sip.conf is set to transport=tcp,udp and asterisk is listening on tcp. |
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| 14:47.51 | Penguin | With the phone's transport set to udp, the phone registers and calls inbound to asterisk match the peer correctly. |
| 14:48.21 | Penguin | Change the phone's transport to tcp, the phone registers and calls inbound to asterisk no longer match the peer. |
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| 14:57.51 | Penguin | Additionally, calls TO the phone work just fine while on tcp. It is only not matching the peer on calls from the phone. |
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| 15:26.04 | starks | hi, what's the quickest way to ring-all or sequentially dial a range of DIDs? a call queue wouldn't be a DID dialing, right? |
| 15:43.20 | Penguin | Do you know what DID is? |
| 15:50.40 | starks | penguin: i'm pretty sure i have a good idea. maybe if i rephrase my question will make more sense: i have 100 phones but we aren't sure if all of the dids are still registered. what would be fastest/easiest way to test the range without dialing every number manually? |
| 15:51.19 | Penguin | Check with your provider. They maintain your DIDs. |
| 15:51.27 | WIMPy | It it was DID, there wouldn't be any point in testing each single number. |
| 15:52.32 | starks | well, my line of work involves reselling DIDs for disaster recovery |
| 15:52.49 | starks | so there's always a question of who is assigned which ranges |
| 15:53.33 | Penguin | Those aren't records I keep. Perhaps your accounting would be a better place to look for such information. |
| 15:54.00 | WIMPy | The question is extremely vague. |
| 15:54.29 | starks | well, if i wanted to dial 100 phone numbers and to make sure each did is assigned to a phone properly, how would i do it? i know the internal extensions work |
| 15:54.59 | starks | our call rules are very tight |
| 15:55.20 | Penguin | You simply want to dial a bunch of numbers automatically? |
| 15:55.24 | starks | right |
| 15:55.31 | WIMPy | If you're sure you're distributing them the way you want, i@m unsure what else you want to test. |
| 15:55.50 | Penguin | I use a list of numbers in plain text and a shell script to loop and dial each number in the list. |
| 15:56.28 | starks | that sounds easy enough, i thought i'd have to do something more sinister or twilio-ish |
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| 16:24.44 | Penguin | [tk]d-fender: Have you ever encountered a situation where a particular peer will not match when switched to tcp transport? Using udp transport, it's 100%. Change to tcp and calls from the phone inbound to asterisk fail with no matching peer found; calls from asterisk to the device are still good. |
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| 16:27.36 | [TK]D-Fender | Penguin: Never used TCP for it before |
| 16:28.24 | Penguin | I was going to change over to TCP to try to save my battery a little. |
| 16:30.22 | Penguin | It's kind of stupid that it is behaving this way. Go in sip.conf, enable tcp, set transport=tcp,udp for the phone, save, sip reload. The phone registers normally and can receive calls. Calls from the phone end with no matching peer. |
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| 17:28.31 | kaawee | Hello! How can I make asterisk call using "CLIP - no screening" and determine for incoming calls which of the numbers are the "CLIP - no screening" numbers and which are the "network provided" numbers? |
| 17:29.18 | WIMPy | That question doesn't really make sense. Which way round? |
| 17:29.48 | kaawee | WIMPy: basically both |
| 17:30.21 | kaawee | WIMPy: so in case I get a call with two caller IDs, I want to know (in the SIP protocol) where I can find these 2 caller IDs (and which are which) |
| 17:30.22 | WIMPy | Ok, as two questions that works. So 1. you don't you tell your telco. |
| 17:31.01 | WIMPy | 2. I think it's in CALLERID(ton), even if that doesn't match the documentation. |
| 17:31.30 | WIMPy | With SIP, ask your ITSP. Most don't seem to support that. |
| 17:32.03 | WIMPy | The best I've seen is using from and PAI, but no real indication of the SI. |
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| 17:34.18 | kaawee | WIMPy: of the "SI"? |
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| 17:34.32 | WIMPy | screening indicator |
| 17:34.46 | kaawee | ah |
| 17:35.44 | kaawee | so I may get a "P-Asserted-Identity" or a "Remote-Party-ID" or a "From" header, but I do not know whether these headers are "user-provided, not screened", "user-provided, verified and passed", "user-provied, verified and failed", "network-provided"? |
| 17:36.03 | WIMPy | I've never seen that. |
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| 17:36.30 | WIMPy | The best is that if from and PAI are different, from would be user provided and PAI network provided. |
| 17:36.40 | WIMPy | But no guarantees anyway these days. |
| 17:36.59 | kaawee | sure |
| 17:39.38 | kaawee | suppose I have the theory that "From" is user-provided and "P-Asserted-Identity" is network provided, how can I actually test this? I need to make any of my telephone companies (either over ISDN or over SIP) to allow me to send a "user-provided, not screened" number in order to get, at the receiving end, a call with 2 callerids? |
| 17:41.11 | WIMPy | although I see a question mark, that was not a question. But as a statement it is true :-) |
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| 18:52.29 | j4jackj | flash memory is lame |
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| 19:20.26 | Marquel | morning. |
| 19:22.18 | Marquel | i have a small problem with my asterisk. trying to call a sip-client from another sip-client on the same asterisk instance gives a failing connection "because port number is zero", dialing out using my ISDN-trunk as well as dialing in from the isdn-trunk to SIP clients work quite well though. unfortunately google does not say where i can look here. since i do have that problem with two software sip clients as well as the sip-client in my android |
| 19:22.20 | Marquel | phone i suspect a sip.conf error on asterisk. |
| 19:22.37 | Marquel | anything i might have missed? |
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| 19:48.22 | Marquel | nvm. directmedia=no was the answer. |
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| 20:40.26 | twanny796 | just installed 11.8.1, and the sip module is *not* present!!! |
| 20:45.25 | Penguin | Not present, as in not loaded? Not present, as in not installed? |
| 20:46.27 | twanny796 | Penguin, not present |
| 20:46.51 | WIMPy | :-) |
| 20:46.52 | Penguin | You already said not present. I'm asking you to define not present. |
| 20:48.05 | twanny796 | Penguin: when I make reload in rasterisk there is no chan_sip |
| 20:48.25 | Penguin | So load it up. module load chan_sip.so |
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| 20:49.53 | twanny796 | <PROTECTED> |
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| 20:51.31 | twanny796 | Penguin, cannot understand how chan_sip did not get installed? |
| 20:51.45 | [TK]D-Fender | How did you install it? |
| 20:52.16 | twanny796 | Penguin, make all, make install, make config |
| 20:52.26 | Penguin | Maybe you deselected it in the menu. |
| 20:52.36 | WIMPy | Let me guess, you thought you don't need websockets an deselected that? |
| 20:52.44 | [TK]D-Fender | you skipped /configure |
| 20:52.55 | twanny796 | I did ./configure |
| 20:53.07 | [TK]D-Fender | Then answer completely |
| 20:53.12 | WIMPy | No menuselect? |
| 20:53.15 | [TK]D-Fender | go look in menconfig |
| 20:53.20 | twanny796 | I only did menuselect just to see, did not save |
| 20:53.40 | Penguin | I would start over. |
| 20:53.46 | WIMPy | Then you must be missing some dependency. |
| 20:54.11 | WIMPy | do menuselect and look what it says. |
| 20:54.18 | twanny796 | WIMPy, ok |
| 20:54.29 | WIMPy | Or config.log |
| 20:56.27 | twanny796 | ok, will continue tomorrow gtg |
| 20:56.38 | Penguin | WTF |
| 20:56.45 | Penguin | Now that is commitment! |
| 21:01.51 | Penguin | I sure wish I could figure out this TCP SIP thing. It is very annoying.` |
| 21:06.30 | WIMPy | At least you know a configuration that works. |
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| 21:50.13 | file | tacos! |
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| 22:28.08 | devlap | I'm looking at provisioning a few hard phones. What is the difference with sip user id's, authenticate Id's and passwords. I was just going to give every device an account with its mac as it userid and a random 16 character password. |
| 22:28.35 | devlap | And then route users via extensions to a device account. |
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| 22:56.11 | [TK]D-Fender | [18:28]devlapAnd then route users via extensions to a device account. <- not sure what you mean here exactly... |
| 22:56.51 | devlap | well if I had device 01:23:45:67:89 and thats johns desk, and johns ext is 200, then 200 rings up 01:23:45 |
| 22:57.16 | devlap | The device sip account, basically wouldn't care who actually had the device. |
| 22:59.50 | [TK]D-Fender | Your wording is a bit twisted |
| 22:59.54 | devlap | Ok. |
| 23:01.02 | [TK]D-Fender | [sip_device_name] has nothing to do with dialplan extensions implicitly. Calls FROM that device are sent to the context it specifies. Everything else is incidental |
| 23:01.20 | devlap | Yea I am well aware of that. |
| 23:01.46 | devlap | 200 => Dial(SIP/MACOFDEVICE) is what im going for. |
| 23:02.56 | [TK]D-Fender | You don't dial a MAC. you dial peer. |
| 23:03.13 | [TK]D-Fender | If you choose to name your peers to match... that's up to you, but nothing special. |
| 23:04.15 | [TK]D-Fender | It will make using an alternate UA to register to that peer awkward looking however and isn't intuitive at that point. |
| 23:04.33 | [TK]D-Fender | But OK for fixed phone layouts |
| 23:04.42 | devlap | I know I don't dial a mac. I don't have any clever name for the sip accounts. |
| 23:04.56 | devlap | So I just made the sip account for each device named its mac. |
| 23:05.18 | [TK]D-Fender | I might have made a clever basic name that also implies the extension # commonly used to dial it... |
| 23:05.52 | devlap | couldn't phones have different extensions based on whos sitting there currently and if they got swapped around? |
| 23:06.42 | [TK]D-Fender | That is implying a hot-desk scenario. that is a different concept as well. |
| 23:07.18 | devlap | I just thought and was told I think here that it was best to disconnect the phones sip account and its extension. |
| 23:07.26 | [TK]D-Fender | You should introduce your description with that because the logic should use a different approach than fixed phones would be |
| 23:07.56 | [TK]D-Fender | [19:07]devlapI just thought and was told I think here that it was best to disconnect the phones sip account and its extension. <- It's best not to over-associate. This has nothing to do with "hot-desking" necessarily |
| 23:08.19 | [TK]D-Fender | ie: don't a call a phone an "extension". |
| 23:08.54 | [TK]D-Fender | naming them different is on one side a mental trick. There are certain security reasons not to use simple numbers as SIP device names as well though |
| 23:09.42 | devlap | Beyond they are easy to guess the sip user name? |
| 23:09.51 | devlap | My original question however was: What is the difference with sip user id's, authenticate Id's and passwords? |
| 23:10.19 | [TK]D-Fender | Passwrod is obviously... a password |
| 23:10.32 | devlap | so userid vs auth id. |
| 23:11.45 | [TK]D-Fender | https://www.google.ca/#q=sip%20username%20vs%20authentication%20name <- take your pick |
| 23:12.01 | [TK]D-Fender | there is a 1st-run, then challenge response concept there |
| 23:12.06 | devlap | It isn't like I didn't already google that exact phrase. |
| 23:12.14 | [TK]D-Fender | I've never seen a reason for them not to be the same |
| 23:12.23 | devlap | That is what I've been doing. |
| 23:12.33 | [TK]D-Fender | Then just leave it at that |
| 23:12.49 | [TK]D-Fender | I certain don't think I can word it any better than any of the docs written for it already |
| 23:13.18 | [TK]D-Fender | So are you looking to run a "host-desk" user scenario? |
| 23:13.22 | [TK]D-Fender | hot* |
| 23:13.27 | devlap | Maybe. |
| 23:13.57 | devlap | I just need to get these phones working. Beyond that is a bit of a reach at this point in time. |
| 23:14.01 | [TK]D-Fender | I'drecommend settling that decision first |
| 23:14.18 | devlap | More than likely it will happen at some point. |
| 23:14.30 | [TK]D-Fender | knowing how you want things to feel should be an easy workflow decision. How to implement it is another matter. |
| 23:14.56 | [TK]D-Fender | If it looks likely, then you might just aim for that more complex scenario from the start |
| 23:17.03 | devlap | Yea I'd like to do that now, but time constraints make getting that done extremely risky. |
| 23:18.13 | devlap | Better to have a working simple system ( like exists now ) and then test a single phone later to get hot desking and fine tune the parameters. |
| 23:21.30 | [TK]D-Fender | Could be.... |
| 23:21.55 | [TK]D-Fender | Depends how and how many you'd need to adapt later and how complex your setup will be starting from. |
| 23:26.13 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 23:57.40 | *** join/#asterisk tuxx- (~tuxx@2a02:2308::1c1:7313:c475) |