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| 00:22.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 09:54.34 | kchehab | hi |
| 09:54.40 | kchehab | hi ppl, can i get the SPC and the SOC Values in The RTCP message from dialplan in order to use it also in the dialplan ?> |
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| 12:03.30 | defsdoor | How do I tell asterisk to use a specific IP address for one trunk - while leaving other trunk on another ? |
| 12:03.48 | defsdoor | I thought bind /bindaddr was it |
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| 13:05.23 | theodotos | Hi. We have sign up with a SIP trunk from a local provider. According to their support, their SIP trunk does not require authentication or even registration. In their ow words: |
| 13:05.23 | theodotos | "It seems when placing a call you are trying to register to our sip proxy. The service provided is sip trunk, no registration is used." |
| 13:05.23 | theodotos | Can anyone reference a guide where you can connect to a SIP trunk without registration? We are using the pjsip module. |
| 13:11.03 | [TK]D-Fender | You don't register. Te end. |
| 13:11.07 | [TK]D-Fender | The* |
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| 13:11.33 | [TK]D-Fender | Don't call it a "connection". They send you calls to a fixed IP you should have given them |
| 13:11.45 | [TK]D-Fender | You send them calls from your IP and they ID you that way. |
| 13:12.17 | [TK]D-Fender | Stop trying to register |
| 13:22.29 | [TK]D-Fender | defsdoor, chan_sip has no "bind" for a specific peer. That is a [general] only parameter. |
| 13:22.54 | [TK]D-Fender | defsdoor, PJSIP allows you to configure multiple transports. |
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| 13:26.53 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:27.05 | vader- | Any recommendations for settings or timing sources to use for running asterisk on a VMWare server? I am having some really bad audio issues with voicemail, system recording and MOH playback. |
| 13:27.15 | vader- | regular SIP calls and stuff seem to be fine |
| 13:29.16 | theodotos | [TK]D-Fender, I was looking for examples on how to do this with pjsip but I am constantly bumping on examples on how to do that with registration/authentication. Can you reference an example on how this is done without registration (yours truly is a complete newbie) |
| 13:30.09 | [TK]D-Fender | You are the one telling it to register |
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| 13:30.33 | [TK]D-Fender | There is nothing else to do except stopping what you're not supposed to be doing |
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| 13:40.45 | vader- | TK any thoughts on my choppy audio issue? |
| 13:41.46 | theodotos | Reading this: https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships |
| 13:41.46 | theodotos | [TK]D-Fender, what type should I use instead of Registration? |
| 13:41.57 | [TK]D-Fender | leave that part OUT |
| 13:42.31 | [TK]D-Fender | This isn't an "insteadl". this is one thing LESS that any other regular provider |
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| 13:48.40 | theodotos | If I leave that out, Asterisk does not even try to connect to the SIP proxy |
| 13:49.41 | [TK]D-Fender | There is no "connect". You don't seem to be grasping this |
| 13:49.44 | [TK]D-Fender | there is no CONNECTION. |
| 13:49.49 | [TK]D-Fender | they send you call. |
| 13:49.52 | [TK]D-Fender | You send them calls. |
| 13:49.54 | [TK]D-Fender | there is nothing else |
| 13:49.59 | [TK]D-Fender | there is no other communication. |
| 13:50.03 | [TK]D-Fender | there is no registration. |
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| 13:50.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.09 | [TK]D-Fender | Providers may choose to work this way if your IP is FIXED since they know where to send the calls to and expect them from. |
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| 14:26.30 | theodotos | [TK]D-Fender, I think I got it. I gave a call to the phone number they gave me and I see the request coming through. Now I need to figure out how to terminate it. |
| 14:26.35 | theodotos | Thanks for the help! |
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| 15:01.41 | igcewieling | Does anyone know what might cause this? Apr 3 11:00:52] NOTICE[26609]: acl.c:748 ast_apply_acl: SIP Peer ACL: Rejecting '50.74.52.26' due to a failure to pass ACL '(BASELINE)' [Apr 3 11:00:52] -- Registered SIP 'amayerhoff-desk' at 50.74.52.26:1076 |
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| 15:38.21 | Samot | You have an ACL called baseline somewhere. |
| 15:39.47 | Samot | Probably in acl.conf |
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| 15:42.54 | igcewieling | acl.conf is empty. |
| 15:43.25 | igcewieling | even it it was not empty it would be pretty useless if it denied once, then allowed in the reg. |
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| 15:44.03 | Samot | That's what that error means. You have an ACL that is blocking that IP. |
| 15:44.34 | igcewieling | then why is it letting in the IP> |
| 15:44.49 | Samot | I don't know. You didn't say it was. |
| 15:44.50 | igcewieling | ? You see the successful registation message right? |
| 15:45.01 | igcewieling | -- Registered SIP 'amayerhoff-desk' at 50.74.52.26:1076 |
| 15:45.02 | Samot | You showed a log entry of it being rejected. |
| 15:45.14 | Samot | I don't know. |
| 15:45.22 | Samot | You're showing a snippet of the actual transaction. |
| 15:45.25 | igcewieling | no, I showed a log entry being rejected AND a log entry showing it accepted. |
| 15:45.45 | defsdoor | [TK]D-Fender, I've switched my config over to using pjsip and have endpoints registering etc.. again, and my trunks are working, however a call delivered to asterisk IP2, from the same peer, is matching against the wrong trunk. I cannot find any documentation on the match syntax to specify the source and dest ip - firstly is this what I need to do ? |
| 15:46.20 | Samot | How is it matching against the wrong trunk? |
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| 15:46.44 | defsdoor | samot - well I'm very new to pjsip - so I could be flailing aimlessly ;) |
| 15:47.00 | defsdoor | I have 2 trunks connecting to the same peer ip |
| 15:47.04 | Samot | OK. |
| 15:47.12 | defsdoor | and I've bound each to 2 different IP addresses on the asterisk config |
| 15:47.25 | Samot | How? |
| 15:48.18 | defsdoor | using transports and bind |
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| 15:48.30 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:49.39 | igcewieling | I tried Asterisk 13 for a few hours the other day. |
| 15:49.41 | Samot | So then both IPs are listening on different ports for UDP? |
| 15:50.15 | Samot | Both IPs cannot listen on the same port for the same transport. |
| 15:50.33 | defsdoor | https://gist.github.com/anonymous/3df3f2c1b3298cdd3b0dbcf493da1620 |
| 15:50.41 | defsdoor | thats my transports and peers config |
| 15:50.55 | Samot | 11:50:16 AMÂ <Samot>Â Both IPs cannot listen on the same port for the same transport. |
| 15:51.07 | Samot | ; You can have more than one of any type of transport, as long as it doesn't use the same resources (bind address, port, etc) as the others. |
| 15:51.17 | Samot | ^^^ Straight from the pjsip.conf sample config. |
| 15:51.23 | defsdoor | oh really ? even if bound to completety different ip ? |
| 15:51.31 | Samot | Yes. |
| 15:51.37 | Samot | The documentation is clear. |
| 15:51.56 | Samot | .91 and .92 cannot both use 5060 for UDP |
| 15:51.57 | defsdoor | I read it as being the same ip AND port |
| 15:52.29 | Samot | Well... |
| 15:52.33 | igcewieling | That isn't anymore useful than chan_sip in that regard. |
| 15:52.59 | Samot | Yeah.. |
| 15:53.00 | file | same IP and port can't be bound, different IP same port can be bound |
| 15:53.07 | Samot | But trunks don't care. |
| 15:53.13 | Samot | See.. |
| 15:53.27 | Samot | A call is being send from 55.55.55.128 (for example) |
| 15:53.27 | igcewieling | file: thank you for the clarification. |
| 15:53.40 | Samot | Both peers are accepting calls from the IP |
| 15:53.57 | Samot | The trunks don't care what IPs Asterisk is listening on. |
| 15:54.01 | defsdoor | regardless of the destiantion IP ? |
| 15:54.03 | defsdoor | hmm |
| 15:54.14 | Samot | The care about where the call is COMING FROM |
| 15:54.33 | Samot | Your trunks say "I accept and send calls to these hosts" |
| 15:55.31 | *** part/#asterisk igcewieling (~ewieling@ip98-170-215-112.pn.at.cox.net) |
| 15:55.47 | defsdoor | I can change my contexts to be the same and solve this |
| 15:56.08 | Samot | Well I haven't looked at how PJSIP handles domains.. |
| 15:56.16 | Samot | But with Chan_SIP... |
| 15:56.17 | defsdoor | the calls are presented to the right IP etc.. from the peer |
| 15:56.32 | Samot | I have only one incoming peer from my softswitch to Asterisk.. |
| 15:56.47 | Samot | And I push the call to the proper context based on the domain . |
| 15:56.49 | defsdoor | just asterisk thinks it's the other peer |
| 15:57.26 | [TK]D-Fender | <Samot> .91 and .92 cannot both use 5060 for UDP <- sure they can |
| 15:57.31 | [TK]D-Fender | it's different IP's |
| 15:57.34 | Samot | Yes, I was wrong. |
| 15:57.41 | Samot | That's been covered and corrected. |
| 15:57.46 | Samot | Regardless of that |
| 15:57.53 | [TK]D-Fender | Missed it. Was scrolling to when I got tagged |
| 15:58.04 | Samot | His trunk to the other PBX doesn't care what IPs Asterisk is listening. |
| 15:58.16 | Samot | His trunk cares what IP the other side is coming from. |
| 15:58.43 | defsdoor | I want it to care about the IP the call is presented to |
| 15:59.01 | [TK]D-Fender | Routing table <- |
| 15:59.28 | [TK]D-Fender | because your kernel is still picking the interface for initiated packets based on kernel routing rules |
| 16:00.17 | defsdoor | that all seems ok |
| 16:00.50 | defsdoor | the peer is giving the call to the correct outbound IP so it must be |
| 16:01.02 | defsdoor | corresponding outbound ip |
| 16:01.51 | defsdoor | I have to go for 20mins |
| 16:04.44 | *** join/#asterisk roycroft (~roycroft@dharmabums.kerouac.org) |
| 16:05.03 | roycroft | hi folks |
| 16:05.17 | roycroft | we have a trixbox that's been under attack lately |
| 16:05.38 | [TK]D-Fender | Trixbox has been dead for over half a decade |
| 16:05.40 | roycroft | a single ip will attempt to make thousands of simultaneous sip connections to the machine, pegging the cpu |
| 16:05.48 | [TK]D-Fender | So go firewall them out |
| 16:05.57 | roycroft | i block that ip on my edge router |
| 16:06.05 | roycroft | but then a different ip will start it later on |
| 16:06.17 | roycroft | is there a way on the switch to throttle simultaneous connections? |
| 16:06.40 | [TK]D-Fender | iptables. |
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| 18:10.31 | Maliuta_ | roycroft: have you heard of this thing call fail2ban |
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| 18:15.40 | [TK]D-Fender | * roycroft (~roycroft@dharmabums.kerouac.org) has left |
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| 18:24.58 | defsdoor | I switched back to chan_sip |
| 18:26.56 | defsdoor | wondering if I can make it work just with some iptables voodo |
| 18:26.58 | defsdoor | o |
| 18:28.22 | defsdoor | something like, tell asterisk that peer is 1.2.3.4, and SNAT and DNAT it to correct destination/port and from address |
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| 20:43.37 | defsdoor | is there a way to make pjsip verbose - I'm trying to create endpoints via wizard and if I add a syntax error it complains but without its silent and no endpoints created |
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