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| 11:40.31 | qakhan | i am getting message ERROR[9768]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe |
| 11:40.56 | qakhan | when i call a php-agi script |
| 11:44.45 | WIMPy | That scpript seems to behave badly then. |
| 11:45.18 | WIMPy | It usually happens when you try to just send messages back to Asterisk without reading what Asterisk sends to you. |
| 11:50.28 | *** join/#asterisk qakhan (~qakhan@39.59.52.122) |
| 11:50.33 | qakhan | i am getting message ERROR[9768]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe |
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| 11:50.36 | qakhan | when i call a php-agi script |
| 11:52.06 | WIMPy | That scpript seems to behave badly then. |
| 11:52.07 | WIMPy | It usually happens when you try to just send messages back to Asterisk without reading what Asterisk sends to you. |
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| 12:20.37 | qakhan | WIMPy here is my script http://pastebin.com/GP7Hc9Cp |
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| 12:31.09 | WIMPy | I don't see it reading the opening sequence, but I have no idea about PHP so I might just not see it. |
| 12:31.25 | qakhan | ok |
| 12:31.33 | qakhan | thanks though |
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| 13:31.25 | boratynskikamil | Good afternoon. |
| 13:31.46 | boratynskikamil | Question to you: http://downloads.openvox.cn/pub/manuals/Release/English/chan_extra-2.0.5%20User%20Manual.pdf How would you implement SMS-sending? Via dialplan? Is it comfortable? |
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| 14:17.58 | DruidZ | Good morning. |
| 14:18.36 | DruidZ | I was looking for a public SIP address database. |
| 14:19.05 | DruidZ | I found one that requires you to own a particular brand of hardware but no fully open ones. |
| 14:19.27 | DruidZ | So, I was wondering how useful such a service would be. |
| 14:19.37 | WIMPy | What is a sip address database? |
| 14:20.32 | DruidZ | Like, I am looking for Joe Smith so I go to a web site and plug in his name. |
| 14:20.56 | DruidZ | It gives me a list of all the Joe Smiths and their SIP address that it finds. |
| 14:21.51 | DruidZ | And I have some ideas for other uses. |
| 14:22.11 | WIMPy | Maybe Joe Smiths should get Smiths.name so you can call joe@smiths.name. |
| 14:22.37 | DruidZ | Maybe. Doesn't scale well though. |
| 14:23.34 | DruidZ | An example other use - connect the entries to VoIP provider and real phone number if entered by trusted VoIP provider. |
| 14:23.52 | WIMPy | Look up ENUM |
| 14:24.15 | DruidZ | Then allow those VoIP providers to query the database for a phone number and redirect calls directly instead of going out the PSTN. |
| 14:25.45 | DruidZ | I know about ENUM. Not sure if it is a complete solution though. |
| 14:26.13 | WIMPy | It will give you an URL for a phone number, if available. |
| 14:26.44 | DruidZ | Can you look up a name? |
| 14:27.29 | WIMPy | no |
| 14:27.51 | WIMPy | NAMES ARE RARELY UNIQUE. |
| 14:27.54 | DruidZ | I also wonder how much acceptance it has outside the techie community. |
| 14:27.56 | WIMPy | oops |
| 14:28.23 | DruidZ | Exactly. That's why you need a web search like the various 411 pages but for SIP. |
| 14:28.39 | WIMPy | The telcos/itsps surely use it. Either the public ENUM or probably more often private solutions between them. |
| 14:29.59 | DruidZ | So how do I tap into that? Probably no way. |
| 14:30.23 | DruidZ | That's sort of what I am thinking but more open. |
| 14:30.36 | WIMPy | ENUM is the open ting. |
| 14:30.58 | WIMPy | There is also DUNDi, but that didn't take off as a public service. |
| 14:31.24 | boratynskikamil | WIMPy: Question to you, if I may. Where should I look for information abot text communication between SIPs? |
| 14:31.38 | DruidZ | I might investigate ENUM again. When I was running FS it would crash whenever I tried to use it. |
| 14:32.37 | WIMPy | Asterisks implementaion is only usefull for closed dial plans by now :-( |
| 14:33.26 | WIMPy | boratynskikamil: The good thing about SIP is that everyone has their own standard. So check the devides/applications you want to use. |
| 14:34.30 | boratynskikamil | WIMPy: I have Jitsi App. And I see "Send message" option. I am even able to get message from SIP, but Jitsi tells me it is impossible to send message to SIP and grays the menu entry. |
| 14:34.45 | DruidZ | I am thinking about a FastAGI server. Quick check |
| 14:35.28 | DruidZ | Quick check to see if number is with a trusted ITSP and connect as appropriate. |
| 14:36.54 | DruidZ | Hmm. Can FastCGI work over UDP or only TCP? The latter I assume. |
| 14:38.19 | DruidZ | s/FastCGI/FastAGI/ |
| 14:39.54 | WIMPy | TCP off course. |
| 14:42.29 | DruidZ | Not sure why "of course" other than some minor programming complications. |
| 14:43.03 | WIMPy | It's a stream/connection. |
| 14:43.13 | DruidZ | Maybe that's a new feature for *. Simple call, simple response and no connection socket used up. |
| 14:43.37 | WIMPy | Like ENUM od DUNDi? |
| 14:43.56 | DruidZ | But generalized. |
| 14:44.47 | WIMPy | You can do in an AGI whatever you like. |
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| 14:45.15 | DruidZ | In fact, if both of those use UDP then the code is probably already there and just needs a public interface to it. |
| 14:45.46 | WIMPy | What extra interface do you need? |
| 14:45.51 | WIMPy | Have you looked at those? |
| 14:46.36 | DruidZ | Something to connect to a UDP socket from the dialplan, send a string and return a sting that can be parsed as needed. |
| 14:47.03 | DruidZ | So you are saying use AGI to connect to a program that makes the UDP connection then? |
| 14:47.56 | WIMPy | UDP or connection. |
| 14:49.41 | DruidZ | Right. I am saying UDP because I am imagining a server that handles thousands of queries a second and doesn't want to have the overhead of TCP. Receive a query, reply to it and forget about that connection. |
| 14:54.11 | WIMPy | There is no connection to forget about :-) |
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| 14:54.22 | boratynskikamil | WIMPy: Suggestion? :-) |
| 14:54.49 | WIMPy | The trouble is that you will have to implement retries. |
| 14:55.18 | WIMPy | boratynskikamil: No. |
| 14:55.43 | DruidZ | Exactly and yes or, depending on the applicatino, just treat a timeout as a negative reply. |
| 14:56.53 | DruidZ | IOW, in my example, ask about a number and if the reply is negative or no reply is received just go out the PSTN. |
| 14:57.32 | WIMPy | Yes, but that already exists. |
| 14:57.56 | DruidZ | UNUM or DUNDi, right? |
| 14:58.19 | WIMPy | ENUM, yes |
| 14:58.30 | WIMPy | ENUM or DUNDi, yes |
| 14:58.49 | DruidZ | Again, not the full solution I think. |
| 14:58.49 | WIMPy | (to avoid possible confusion) |
| 14:59.13 | WIMPy | I don't see how anything more makes sense on Asterisk. |
| 14:59.37 | WIMPy | If you want to find names you have to do that outside of Asterisk. |
| 14:59.58 | DruidZ | Well, no but I am thinking of the same database also being the address lookup database. |
| 15:00.23 | WIMPy | What address? |
| 15:00.50 | DruidZ | SIP address from name as I mentioned at the start of this conversation. |
| 15:01.34 | DruidZ | So the DB has name, PSTN phone number, SIP address and possibly more. |
| 15:01.57 | DruidZ | The * server only cares about the last two. |
| 15:01.57 | WIMPy | Use LDAP or something? |
| 15:02.13 | WIMPy | Many phones support LDAP already. |
| 15:03.26 | DruidZ | How does LDAP help when I want to look up Joe Blow in Chicoutimi, PQ and I am in Toronto, ON? |
| 15:03.57 | WIMPy | How does it matter where you are? |
| 15:04.19 | DruidZ | Well, more to the point, we use different ITSPs. |
| 15:04.45 | WIMPy | So what? |
| 15:04.53 | DruidZ | Or do you mean serve LDAP from the DB I am proposing? |
| 15:05.00 | WIMPy | But you obviousely have to agree to use the same database. |
| 15:05.30 | WIMPy | Maybe you should just ask Facebook. It's unlikely you find a more complete database. |
| 15:05.33 | DruidZ | Right. I thought you were arguing against the need for the common DB. |
| 15:06.00 | DruidZ | FB is not complete. For example, it doesn't include me. |
| 15:06.17 | WIMPy | It doesn't include me, either. |
| 15:06.37 | WIMPy | But you still wont find a more complete database. |
| 15:08.54 | DruidZ | "More complete"? |
| 15:09.13 | DruidZ | pulls out Fowler's 3rd ed. |
| 15:09.29 | WIMPy | The only organization that hase more user records is the NSA and they won't share. |
| 15:10.38 | DruidZ | Will FB? Can I give them a name and get a SIP address? |
| 15:11.02 | DruidZ | Or, can I give them a PSTN number and get a SIP address? |
| 15:11.07 | WIMPy | You can mke calls via FB. |
| 15:11.29 | WIMPy | And anything else is just a matter of payment, I'm sure. |
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| 17:56.54 | df-1975 | Hi, I'm new to this channel, and a newbie asterisk user. |
| 18:00.20 | df-1975 | I was running version 11.4 and now made a reinstall with version 12.2. I copied the changes I had made to 11.4 over to 12.2, and found out that it works out of the box. I thought that sip.conf had become obsolete, in favor of pjsip.conf. I didn't modify pjsip.conf, but overwrote sip.conf with my ol values. Why is it still working? |
| 18:00.52 | df-1975 | Oh, I meant 12.1.1, not 12.2 |
| 18:00.55 | [TK]D-Fender | it is not "obsolete yet, and chan_sip is not GONE |
| 18:01.03 | [TK]D-Fender | Since it's still there... it works |
| 18:01.57 | df-1975 | Could I migrate sip.conf to pjsip.conf and remove sip.conf? |
| 18:02.59 | [TK]D-Fender | Sure |
| 18:03.38 | df-1975 | Would then chan_sip stopped being used, and chan_pjsip being used instead? |
| 18:05.45 | df-1975 | or is chan_sip now using pjsip under the hood? |
| 18:08.50 | [TK]D-Fender | it is not. |
| 18:09.13 | [TK]D-Fender | they are COMPLETELY separate. You could run them simultaneously, though you'd have to bind them to different ports |
| 18:12.49 | df-1975 | Ok, thanks. For new projects, is it recommended to use pjsip instead? |
| 18:13.23 | ChannelZ | It's probably something you should learn eventually |
| 18:13.36 | [TK]D-Fender | Compare the functionality you need and what each offers and pick whatever works best for you. |
| 18:13.44 | ChannelZ | and/or if you could benefit from some of it's features now |
| 18:14.23 | [TK]D-Fender | Going forward I suspect PJSIP will outstrip chan_sip in all/enough areas to give them reason to reexamine development of it. Time will tell |
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| 18:16.19 | df-1975 | Thanks to both of you. |
| 18:17.32 | file | VIVA PJSIP |
| 18:18.05 | BludSuckingFiend | Viva SCCP! |
| 18:19.02 | ChannelZ | speaking of I need to update my v12 |
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| 20:20.36 | gusto | hi folks |
| 20:20.39 | gusto | whats up? |
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| 20:34.30 | anaxagoras | Good evening everybody. |
| 20:35.54 | anaxagoras | After three sleepless nights i am seeking for your help: I have a nat-issue with my asterisk setup on ubuntu. I took the time to document the problem at http://pastebin.com/SBnK49pG I would be greatful for every input regarding the issue. Thanks in advance. |
| 20:38.25 | anaxagoras | should i explain it in chat or is it the right way to paste these information on pastebin? |
| 20:40.25 | pabelanger | anaxagoras: pb a SIP trace of a call that is not working |
| 20:41.18 | anaxagoras | pabelanger: there is a snipped of a call that is not working form the debug output of ther asterisk console |
| 20:41.35 | pabelanger | anaxagoras: that is not a SIP debug though |
| 20:41.40 | pabelanger | ~collectdebug |
| 20:41.40 | infobot | extra, extra, read all about it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 20:42.03 | pabelanger | the SIP trace will show actually IP address |
| 20:42.07 | pabelanger | and the issue with your NAT |
| 20:43.22 | gusto | why do you have that fritzbox in between? |
| 20:43.43 | gusto | aha, its a cable router |
| 20:43.44 | gusto | ok |
| 20:44.19 | gusto | who is your provider? |
| 20:44.24 | anaxagoras | gusto: unitymedia |
| 20:44.32 | gusto | so you do not have ipv6 |
| 20:44.34 | gusto | ha? |
| 20:44.43 | anaxagoras | gusto: no its ipv4 i suppose |
| 20:45.48 | gusto | your commedia and nat settings are useless, because you are the client, so one has to expect the other side (the side you are registering to) that they have set nat=yes ... which ... of course may not be the case |
| 20:46.43 | gusto | isnt it possible that the fritzbox is catching the VoIP connections away? |
| 20:47.15 | gusto | i ve seen such problems on DT speedports (technically also fritzboxes) |
| 20:48.27 | gusto | two public ip addresses? how is that? do you have some kind of business plan? |
| 20:49.03 | anaxagoras | gusto: its a business-account of unitymedia thats why we have the sip-account information and an own gateway / subnet and two public ips |
| 20:50.18 | gusto | hm, interesting |
| 20:50.37 | gusto | i am on kabeldeutschland ds_lite |
| 20:51.18 | anaxagoras | gusto: unfortunately they dont give away their sip-account credentials |
| 20:53.00 | gusto | yes, that is always a problem, but i managed always to find it out someway |
| 20:53.16 | gusto | you mean that credentials that are inside of your fritzbox? |
| 20:53.33 | gusto | maybe you should make a backup of that configuration and then search the file after the credentials |
| 20:53.41 | gusto | that would be my first try |
| 20:55.05 | gusto | localnet= 94.XX.190.20/255.255.255.252 ??? |
| 20:55.16 | gusto | what's that bullshit? |
| 20:55.49 | anaxagoras | gusto: that is not my issue, because unitymedia handed me the credentials |
| 20:56.03 | gusto | localnets are not translated into extern addresses |
| 20:56.07 | gusto | so put that line away |
| 20:56.09 | anaxagoras | gusto: that is the local subnet adress and the subnet-mask |
| 20:56.49 | gusto | no |
| 20:57.45 | gusto | put that line away and also those ridiculous extern ip and extern addr lines, one is sufficient and when you want to use DNS, then use externhost |
| 20:57.56 | anaxagoras | gusto: the fritzbox is not reachable from the wan, so it should be considered as a local net isnt it? |
| 20:58.20 | gusto | no |
| 20:58.35 | gusto | also those NAT settings are useless since you are the one behind nat |
| 20:58.59 | anaxagoras | gusto: should i loose all nat settings? |
| 21:00.01 | gusto | they may do more harm than good |
| 21:00.07 | gusto | especially those commedia settings |
| 21:00.23 | gusto | when he is waiting for the other side to send rtp packet first, it may never happen |
| 21:00.32 | anaxagoras | gusto: should i set nat=no |
| 21:00.36 | gusto | yes |
| 21:01.06 | gusto | nat=no in general and remove all the nat settings per peer |
| 21:02.14 | gusto | and you also seem to have SIP connection working, however, i can not understand why, because the way you configured it, not even that should work |
| 21:02.27 | gusto | but it maybe works because you enabled qualify |
| 21:02.52 | gusto | when you are qualifying with your SIP UDP connection, you do not need a portforwarding for that one |
| 21:03.06 | gusto | so - NAT/Portforward: WAN -> 6969 -> 5060 to IP of VM is also useless |
| 21:03.32 | anaxagoras | gusto: i have two external sip-clients |
| 21:04.05 | gusto | let us not care about them first |
| 21:04.20 | gusto | you need to make that setup working first, do care about the external bullshit later |
| 21:04.25 | anaxagoras | gusto: which are not that important i may switch to a vpn solution for them |
| 21:04.32 | gusto | no |
| 21:04.53 | anaxagoras | gusto: alright :) |
| 21:04.54 | gusto | that should work as well, but why did you choose 6969 port for them? |
| 21:05.13 | anaxagoras | gusto: because 5060 was spammed immediately |
| 21:05.22 | gusto | what do you care? |
| 21:05.28 | anaxagoras | gusto: i was afraid |
| 21:05.37 | gusto | hm |
| 21:05.58 | gusto | you can adjust your firewall to block away unknown ip address ranges |
| 21:06.11 | gusto | on that port |
| 21:06.55 | gusto | switching to another port is not really a more secure option, those bots who are just spamming into your standard port will not do you any harm, as long as you do not misconfigure your asterisk |
| 21:07.32 | gusto | and when there would come someone about who is able to break into your setup, you will not stop such a person with different port |
| 21:07.34 | gusto | :-D |
| 21:07.38 | anaxagoras | gusto: okay. like you said we will care about the external later |
| 21:08.24 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 21:09.06 | anaxagoras | gusto: i am restarting my asterisk and have a try if issue is still present |
| 21:09.13 | anaxagoras | gusto: just a sec |
| 21:11.02 | gusto | ssl36.telefon.unitymedia.de seems to resovle to an IPv4 so you should not need the SRV record as well, or is it needed for sipgate? it has nothing to do with your issue, but i would be careful enabling SRV on asterisk, you can get a loop very easily, i do not use it, i ll rather resovle the SRV record using DIG manually and put the domain manually there which resolves than to a A record |
| 21:12.51 | gusto | so, what is the status? |
| 21:14.27 | anaxagoras | gusto: the status is the same. Calls are routed to the phone, but there is no audio |
| 21:15.05 | gusto | well, but the adjustments helped, because it made the configuration a bit smaller (less likely for catching more errors) |
| 21:15.17 | anaxagoras | gusto: yes indeed |
| 21:15.21 | gusto | so in this sense it is good, even when there is no progress |
| 21:15.42 | gusto | so ... maybe we should now enable the sip debug using 'sip set debug on' |
| 21:15.45 | gusto | to see more |
| 21:16.14 | anaxagoras | gusto: its enabled |
| 21:16.23 | gusto | hm |
| 21:16.31 | anaxagoras | gusto: now another call and paste it to the bin? |
| 21:16.40 | gusto | yes |
| 21:16.47 | gusto | debug output helps always |
| 21:18.04 | gusto | listen up ... do you have directmedia enabled? |
| 21:18.32 | gusto | try to set directmedia = no to [general] section, maybe that would help |
| 21:19.22 | gusto | <PROTECTED> |
| 21:19.26 | gusto | that was the problem |
| 21:19.32 | gusto | he is trying to bridge it remotely |
| 21:19.38 | gusto | ah, i should have seen that from the start |
| 21:20.07 | gusto | a beginner mistake i made ... i expected the defaults not to bridge remotely, that is disgraceful |
| 21:20.10 | anaxagoras | gusto: with debuggin on there is no ringing. is that normal |
| 21:20.26 | gusto | eh, that is not normal |
| 21:20.35 | gusto | debugging should not make any changes |
| 21:20.51 | gusto | when it does not ring, we ran into a different problem |
| 21:21.04 | gusto | but first we have to disable the remote bridging, that is very important!!! |
| 21:21.17 | gusto | try to set directmedia = no to [general] section! |
| 21:21.18 | anaxagoras | gusto: alright. |
| 21:21.26 | anaxagoras | gusto: just a sec |
| 21:21.49 | gusto | he has to bridge locally, because when he does not, then all your portforwarding bullshit is useless |
| 21:23.08 | gusto | theoretically it would be possible to make it work with remote bridging as well, but i did not see such intelligent sip phones yet that would use externaddr only for cases when it is connecting to the outside net, but i may be mistaken, however, behind nat it is always a good idea to bridge locally |
| 21:24.34 | gusto | for that things to see you do not need sip debug 'sip set debug off' and you only need core verbosity 'core set verbose 4' and 'core set debug 4' |
| 21:27.31 | *** join/#asterisk anaxagoras (~jochen@b2b-94-79-190-22.unitymedia.biz) |
| 21:27.42 | anaxagoras | gusto: i am sorry. disconnect on my end |
| 21:27.59 | gusto | so |
| 21:28.02 | gusto | what is the status now? |
| 21:28.03 | anaxagoras | gusto: i made the changes |
| 21:29.32 | anaxagoras | gusto: just a sec please |
| 21:32.34 | anaxagoras | gusto: I can hear audio |
| 21:32.39 | anaxagoras | gusto: GREAT! |
| 21:32.58 | anaxagoras | gusto: now i am testing outbound calls, because they were cut after 30 seconds |
| 21:33.04 | gusto | when he is doing local bridging now, it is no surprise that you hear audio |
| 21:33.31 | gusto | that should not be a problem either, if it still exists, then we need to take a closer look at your NAT setup |
| 21:34.51 | anaxagoras | gusto: no breakups |
| 21:35.28 | anaxagoras | gusto: great |
| 21:35.53 | anaxagoras | gusto: thank you very much. now that i can understand the issue. what was the problem |
| 21:36.29 | gusto | yes, but removing remote bridging only would not help much, since you also had your externaddr configured as localnet |
| 21:37.12 | gusto | so there were 2 problems, but i first caught only the localnet one, and then i had to take a closer look to spot that you had no setting on directmedia |
| 21:38.08 | anaxagoras | gusto: ah okay. directmedia. I just read the explaination. which sounds logical - now that it works :) |
| 21:38.10 | gusto | and of course that nat settings like same port RTP (comedia) could make problems as well, since it makes asterisk wait for the other side |
| 21:38.59 | gusto | he was trying to bridge the telephone directly with your provider, which could not work, because you had no RTP redirection in NAT for your phones |
| 21:40.32 | gusto | directmedia is a nice thing, but you need to have all devices using direct (remote) bridging have reachable ip addresses (ipv6 for example) |
| 21:42.04 | anaxagoras | gusto: i think i get it. do you have any further advice on remote sip clients to connect |
| 21:43.14 | gusto | and you have also be careful about your external phones then, because when you do remote bridging with a phone outside your network trying to make a call over your provider, that could result in a ban, because in the contract you are only allowed to use that VoIP from your internet connection and not from other ones |
| 21:43.32 | gusto | even worse - giving others the access to your telephone plan ;-) |
| 21:44.22 | gusto | well, i see a problem |
| 21:44.30 | anaxagoras | gusto: it is a different provider in that case sipgate |
| 21:44.58 | gusto | for example ... you did say that you used a port 6969 because you was worried to use 5060 standard port |
| 21:45.19 | anaxagoras | gusto: yes |
| 21:45.38 | gusto | well, but did you also think about that the server also sends packets back? |
| 21:45.39 | anaxagoras | gusto: should i change the port back to 5060? |
| 21:45.52 | gusto | that would make a lot of things easier |
| 21:46.35 | anaxagoras | gusto: okay. i will do that |
| 21:46.40 | gusto | because the packets for 5060 are arriving on 6969 ... so the client is sending them to 6969, but the server responds from 5060 ... and that results in rejection |
| 21:47.06 | gusto | so you would need to have insecure=port enabled on those clients, and on phones, it is unlikely for them to do that |
| 21:47.45 | gusto | moving your SIP port to 6969 or another port would make more sense |
| 21:48.11 | gusto | but 5060 is the best solution |
| 21:48.48 | gusto | an alternative way would be to change port forwarding for 6969 -> 6969 and make asterisk on your ubuntu VM listen on 6969 |
| 21:49.53 | anaxagoras | gusto: insecure=port instead of insecure=invite? |
| 21:49.56 | gusto | for your providers it should be not important where the SIP messages come from, because you are registering to them anyway, so they should notice that you would send it from a different port, and since you are behind a nat anyway, it would not make much difference |
| 21:50.57 | gusto | i was talking about the external clients of yours (not sure if that are asterisk clients or what) |
| 21:51.24 | anaxagoras | gusto: no these are android phones - with zoiper or csimple installed |
| 21:51.47 | anaxagoras | gusto: which want to connect to the two seperate sipgate accounts |
| 21:52.09 | gusto | for the clients connecting to you, where you have type=friend, it is a dangerous method to use insecure=invite ... that could result in someone else taking over your connection! |
| 21:53.10 | gusto | for your peers on the other hand, you HAVE TO use insecure=invite, because they will not register to you, you only authenticate yourself to them, but not the other way (logically, they have no way to know YOUR credentials - if you had any) |
| 21:54.09 | [TK]D-Fender | [17:53]gustofor your peers on the other hand, you HAVE TO use insecure=invite, because they will not register to you, you only authenticate yourself to them, but not the other way (logically, they have no way to know YOUR credentials - if you had any) <- no |
| 21:54.12 | gusto | on the other hand, for VOIP providers like sipgate or your cable provider you can RELY on them using the right ports (because they are professionals) so you SHOULD remove insecure=port from their peer section |
| 21:54.55 | gusto | [TK]D-Fender: i meant the VOIP providers |
| 21:55.07 | gusto | as in type=peer |
| 21:55.16 | [TK]D-Fender | Use of insecure there has nothing to do with registration |
| 21:55.25 | gusto | the external connections (friends) he has as type=friend |
| 21:56.24 | gusto | [TK]D-Fender: to my understanding insecure=invite makes asterisk not require an authentification on every call being made |
| 21:56.27 | anaxagoras | gusto: yes external has type=freind |
| 21:58.46 | gusto | anaxagoras: type=friend means that you authenticate to them as well as they are authenticating to you, type=peer means that you have no credentials for that connection at all, but you can/should authenticate to them using registry => for example, so that part is right |
| 21:58.54 | gusto | but here just for clarification |
| 21:59.44 | [TK]D-Fender | [17:58]gustoanaxagoras: type=friend means that you authenticate to them as well as they are authenticating to you <- no. |
| 22:00.46 | gusto | [TK]D-Fender: why not? type=friend causes asterisk creating not only the connection (peer) but also a user (with a password given) |
| 22:00.46 | [TK]D-Fender | user = auth by username, peer = auth by IP (regsitration required if dynamic). friend = user + peer. (user matched first. |
| 22:01.10 | [TK]D-Fender | Both require authentication unless you use "insecure" |
| 22:01.13 | gusto | did i mix it up with IAX? |
| 22:01.21 | [TK]D-Fender | No, you're just wrong overall. |
| 22:01.31 | [TK]D-Fender | auth implies PROVING who you are |
| 22:02.00 | [TK]D-Fender | type change how you IDENTIFY them, INSECURE bypasses PASSWORD CHECKING |
| 22:02.30 | gusto | yes, but there is no password when you have type=peer |
| 22:02.34 | [TK]D-Fender | WRONG |
| 22:02.40 | gusto | NEW TO ME |
| 22:02.42 | gusto | :-D |
| 22:04.12 | [TK]D-Fender | There is a secret if you put one and bypassed on INCOMING if you use "insecure" appropriately. |
| 22:04.54 | gusto | well |
| 22:08.56 | anaxagoras | gusto: [TK]D-Fender, thanks very much for your help. I will change the other settings tomorrow, for today - my brain is too empty :9 |
| 22:09.21 | anaxagoras | gusto: have a good night. |
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| 23:50.05 | lvlinux | anybody setup SLA lately? i'm going through a nightmare just to get a super simple setup going. |
| 23:50.22 | lvlinux | or I should say a "Key System" |
| 23:50.53 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 23:51.33 | lvlinux | I'm using LG LIP-6812 phones |
| 23:51.44 | lvlinux | set to DSS/BLF on the lines |
| 23:53.39 | lvlinux | When I hit a line key and it speedials the sla line in the sla_stations context, asterisk gives a 603 declined |
| 23:55.18 | lvlinux | anyone here ever messed with that before? |
| 23:55.20 | [TK]D-Fender | Which * throws out if a call is accepted to the dialplan but nothing answers it or passes on any other status |
| 23:55.41 | [TK]D-Fender | You ran out of dialplan which means you should actually be paying attention to what is being called |
| 23:56.03 | lvlinux | [TK]D-Fender: I thought * was supposed to instantly connect the station to the trunk, which is a DAHDI line? |
| 23:56.30 | WIMPy | Do YOU have it configured like that? |
| 23:57.14 | lvlinux | I thought so---I've never set this up before, so I've been basing my config on the examples in the asterisk documentation and the asterisk book example |
| 23:57.34 | lvlinux | i have my sla.conf setup where line1 is DAHDI/1 line2 is DAHDI/2 etc |
| 23:57.37 | [TK]D-Fender | You aren't looking at your call |
| 23:58.24 | lvlinux | so I need to manually tell * to connect to DAHDI/1 in the context that the station hits? |
| 23:58.43 | [TK]D-Fender | DIALPLAN <--------- |
| 23:59.06 | lvlinux | yes that's what I mean---in the dialplan context |
| 23:59.12 | [TK]D-Fender | yes |